Warning: Permanently added '163.109.86.57' (ED25519) to the list of known hosts. You can reproduce this build on your computer by running: sudo dnf install copr-rpmbuild /usr/bin/copr-rpmbuild --verbose --drop-resultdir --task-url https://copr.fedorainfracloud.org/backend/get-build-task/10948227-fedora-rawhide-s390x --chroot fedora-rawhide-s390x Version: 1.9 PID: 8397 Logging PID: 8399 Task: {'allow_user_ssh': False, 'appstream': False, 'background': False, 'build_id': 10948227, 'buildroot_pkgs': [], 'chroot': 'fedora-rawhide-s390x', 'enable_net': False, 'fedora_review': False, 'git_hash': 'bae29d1374e87b5ce8878369fbd4b0060a5f46e7', 'git_repo': 'https://copr-dist-git.fedorainfracloud.org/git/rathann/ffmpeg/baresip', 'isolation': 'default', 'memory_reqs': 2048, 'package_name': 'baresip', 'package_version': '4.11.0-1', 'project_dirname': 'ffmpeg', 'project_name': 'ffmpeg', 'project_owner': 'rathann', 'repo_priority': None, 'repos': [{'baseurl': 'https://download.copr.fedorainfracloud.org/results/rathann/ffmpeg/fedora-rawhide-s390x/', 'id': 'copr_base', 'name': 'Copr repository', 'priority': None, 'repo_gpgcheck': False}], 'sandbox': 'rathann/ffmpeg--rathann', 'source_json': {}, 'source_type': None, 'ssh_public_keys': None, 'storage': 1, 'submitter': 'rathann', 'tags': [], 'task_id': '10948227-fedora-rawhide-s390x', 'timeout': 18000, 'uses_devel_repo': False, 'with_opts': [], 'without_opts': []} Running: git clone https://copr-dist-git.fedorainfracloud.org/git/rathann/ffmpeg/baresip /var/lib/copr-rpmbuild/workspace/workdir-n6mi2rvw/baresip --depth 500 --no-single-branch --recursive cmd: ['git', 'clone', 'https://copr-dist-git.fedorainfracloud.org/git/rathann/ffmpeg/baresip', '/var/lib/copr-rpmbuild/workspace/workdir-n6mi2rvw/baresip', '--depth', '500', '--no-single-branch', '--recursive'] cwd: . rc: 0 stdout: stderr: Cloning into '/var/lib/copr-rpmbuild/workspace/workdir-n6mi2rvw/baresip'... Running: git checkout bae29d1374e87b5ce8878369fbd4b0060a5f46e7 -- cmd: ['git', 'checkout', 'bae29d1374e87b5ce8878369fbd4b0060a5f46e7', '--'] cwd: /var/lib/copr-rpmbuild/workspace/workdir-n6mi2rvw/baresip rc: 0 stdout: stderr: Note: switching to 'bae29d1374e87b5ce8878369fbd4b0060a5f46e7'. You are in 'detached HEAD' state. You can look around, make experimental changes and commit them, and you can discard any commits you make in this state without impacting any branches by switching back to a branch. If you want to create a new branch to retain commits you create, you may do so (now or later) by using -c with the switch command. Example: git switch -c Or undo this operation with: git switch - Turn off this advice by setting config variable advice.detachedHead to false HEAD is now at bae29d1 automatic import of baresip Running: dist-git-client sources cmd: ['dist-git-client', 'sources'] cwd: /var/lib/copr-rpmbuild/workspace/workdir-n6mi2rvw/baresip rc: 0 stdout: stderr: INFO: Reading stdout from command: git rev-parse --abbrev-ref HEAD INFO: Reading stdout from command: git rev-parse HEAD INFO: Reading sources specification file: sources INFO: Downloading baresip-4.11.0.tar.gz INFO: Reading stdout from command: curl --help all INFO: Calling: curl -H Pragma: -H 'Accept-Encoding: identity' -o baresip-4.11.0.tar.gz --location --connect-timeout 60 --retry 3 --retry-delay 10 --remote-time --show-error --fail --retry-all-errors https://copr-dist-git.fedorainfracloud.org/repo/pkgs/rathann/ffmpeg/baresip/baresip-4.11.0.tar.gz/md5/2a61ec6a44412b9771274612c7eeac1a/baresip-4.11.0.tar.gz % Total % Received % Xferd Average Speed Time Time Time Current Dload Upload Total Spent Left Speed 100 1.15M 100 1.15M 0 0 1.11M 0 00:01 00:01 857.4k INFO: Reading stdout from command: md5sum baresip-4.11.0.tar.gz Running (timeout=18000): unbuffer mock --spec /var/lib/copr-rpmbuild/workspace/workdir-n6mi2rvw/baresip/baresip.spec --sources /var/lib/copr-rpmbuild/workspace/workdir-n6mi2rvw/baresip --resultdir /var/lib/copr-rpmbuild/results --uniqueext 1788464321.539005 -r /var/lib/copr-rpmbuild/results/configs/child.cfg tail: /var/lib/copr-rpmbuild/main.log: file truncated INFO: mock.py version 6.8 starting (python version = 3.14.5, NVR = mock-6.8-1.fc44), args: /usr/libexec/mock/mock --spec /var/lib/copr-rpmbuild/workspace/workdir-n6mi2rvw/baresip/baresip.spec --sources /var/lib/copr-rpmbuild/workspace/workdir-n6mi2rvw/baresip --resultdir /var/lib/copr-rpmbuild/results --uniqueext 1788464321.539005 -r /var/lib/copr-rpmbuild/results/configs/child.cfg Start(bootstrap): init plugins INFO: tmpfs initialized INFO: selinux enabled INFO: chroot_scan: initialized INFO: compress_logs: initialized Finish(bootstrap): init plugins Start: init plugins INFO: tmpfs initialized INFO: selinux enabled INFO: chroot_scan: initialized INFO: compress_logs: initialized Finish: init plugins INFO: Signal handler active Start: run INFO: Start(/var/lib/copr-rpmbuild/workspace/workdir-n6mi2rvw/baresip/baresip.spec) Config(fedora-rawhide-s390x) Start: clean chroot Finish: clean chroot Mock Version: 6.8 INFO: Mock Version: 6.8 Start(bootstrap): chroot init INFO: mounting tmpfs at /var/lib/mock/fedora-rawhide-s390x-bootstrap-1788464321.539005/root. INFO: calling preinit hooks INFO: enabled root cache INFO: enabled package manager cache Start(bootstrap): cleaning package manager metadata Finish(bootstrap): cleaning package manager metadata INFO: Guessed host environment type: unknown INFO: Using container image: registry.fedoraproject.org/fedora:rawhide INFO: Pulling image: registry.fedoraproject.org/fedora:rawhide INFO: Tagging container image as mock-bootstrap-d785116a-4757-4c47-8630-2254703aaa55 INFO: Checking that c77fdfa4af69084d046f613a6e4d8478d1923cfc783b89cbac5642f9e9c9b325 image matches host's architecture INFO: Copy content of container c77fdfa4af69084d046f613a6e4d8478d1923cfc783b89cbac5642f9e9c9b325 to /var/lib/mock/fedora-rawhide-s390x-bootstrap-1788464321.539005/root INFO: mounting c77fdfa4af69084d046f613a6e4d8478d1923cfc783b89cbac5642f9e9c9b325 with podman image mount INFO: image c77fdfa4af69084d046f613a6e4d8478d1923cfc783b89cbac5642f9e9c9b325 as /var/lib/containers/storage/overlay/912b98b0d668e001bf811731f5d6bfe27a050060b30f8e04d776241ee0132ad0/merged INFO: umounting image c77fdfa4af69084d046f613a6e4d8478d1923cfc783b89cbac5642f9e9c9b325 (/var/lib/containers/storage/overlay/912b98b0d668e001bf811731f5d6bfe27a050060b30f8e04d776241ee0132ad0/merged) with podman image umount INFO: Removing image mock-bootstrap-d785116a-4757-4c47-8630-2254703aaa55 INFO: Package manager dnf5 detected and used (fallback) INFO: Not updating bootstrap chroot, bootstrap_image_ready=True Start(bootstrap): creating root cache Finish(bootstrap): creating root cache Finish(bootstrap): chroot init Start: chroot init INFO: mounting tmpfs at /var/lib/mock/fedora-rawhide-s390x-1788464321.539005/root. INFO: calling preinit hooks INFO: enabled root cache INFO: enabled package manager cache Start: cleaning package manager metadata Finish: cleaning package manager metadata INFO: enabled HW Info plugin INFO: Package manager dnf5 detected and used (direct choice) INFO: Buildroot is handled by package management downloaded with a bootstrap image: rpm-6.1.0-1.fc46.s390x rpm-sequoia-1.10.2-5.fc45.s390x dnf5-5.4.3.0-2.fc46.s390x dnf5-plugins-5.4.3.0-2.fc46.s390x Start: installing minimal buildroot with dnf5 Updating and loading repositories: fedora 100% | 3.0 KiB/s | 4.1 KiB | 00m01s Copr repository 100% | 1.3 KiB/s | 1.5 KiB | 00m01s Repositories loaded. Package Arch Version Repository Size Installing group/module packages: bash s390x 0:5.3.15-2.fc45 fedora 8.6 MiB bzip2 s390x 0:1.0.8-24.fc45 fedora 86.6 KiB coreutils s390x 0:9.11-6.fc45 fedora 6.0 MiB cpio s390x 0:2.15-10.fc45 fedora 1.1 MiB diffutils s390x 0:3.12-6.fc45 fedora 1.6 MiB fedora-release-common noarch 0:46-0.1 fedora 4.6 KiB findutils s390x 1:4.11.0-2.fc45 fedora 2.0 MiB gawk s390x 0:5.4.1-1.fc45 fedora 2.2 MiB glibc-minimal-langpack s390x 0:2.44.9000-2.fc46 fedora 0.0 B grep s390x 0:3.12-4.fc45 fedora 1.0 MiB gzip s390x 0:1.14-4.fc45 fedora 397.2 KiB info s390x 0:7.3-3.fc45 fedora 403.7 KiB patch s390x 0:2.8-5.fc45 fedora 238.1 KiB redhat-rpm-config noarch 0:344-5.fc45 fedora 184.2 KiB rpm-build s390x 0:6.1.0-1.fc46 fedora 302.1 KiB sed s390x 0:4.10-2.fc45 fedora 870.1 KiB shadow-utils s390x 2:4.20.0-1.fc45 fedora 3.9 MiB tar s390x 2:1.35-9.fc45 fedora 3.0 MiB unzip s390x 0:6.0-71.fc45 fedora 440.9 KiB util-linux s390x 0:2.42.2-3.fc45 fedora 3.5 MiB which s390x 0:2.25-2.fc45 fedora 83.3 KiB xz s390x 1:5.8.3-2.fc45 fedora 1.4 MiB Installing dependencies: R-srpm-macros noarch 0:1.3.7-3.fc45 fedora 3.5 KiB add-determinism s390x 0:0.7.3-3.fc45 fedora 2.3 MiB alternatives s390x 0:1.33-6.fc45 fedora 61.8 KiB ansible-srpm-macros noarch 0:1-22.fc45 fedora 35.7 KiB audit-libs s390x 0:4.2.1-2.fc45 fedora 397.8 KiB binutils s390x 0:2.47.50-6.fc46 fedora 29.7 MiB build-reproducibility-srpm-macros noarch 0:0.7.3-3.fc45 fedora 1.2 KiB bzip2-libs s390x 0:1.0.8-24.fc45 fedora 84.3 KiB ca-certificates noarch 0:2026.2.90_v9.0.317-1.fc46 fedora 2.6 MiB cmake-srpm-macros noarch 0:4.3.0-5.fc45 fedora 524.0 B coreutils-common s390x 0:9.11-6.fc45 fedora 10.9 MiB crypto-policies noarch 0:20260713-2.gitc42da77.fc45 fedora 134.3 KiB curl s390x 0:8.22.0~rc3-1.fc46 fedora 510.4 KiB cyrus-sasl-lib s390x 0:2.1.28-37.fc45 fedora 2.4 MiB debugedit s390x 0:5.3-3.fc45 fedora 224.1 KiB dwz s390x 0:0.17-3.fc45 fedora 310.4 KiB ed s390x 0:1.22.6-1.fc46 fedora 150.6 KiB efi-srpm-macros noarch 0:6-9.fc45 fedora 40.3 KiB elfutils s390x 0:0.196-1.fc46 fedora 3.0 MiB elfutils-debuginfod-client s390x 0:0.196-1.fc46 fedora 75.4 KiB elfutils-libelf s390x 0:0.196-1.fc46 fedora 1.4 MiB elfutils-libs s390x 0:0.196-1.fc46 fedora 786.2 KiB erlang-srpm-macros noarch 0:0.3.11-2.fc45 fedora 1.9 KiB fedora-gpg-keys noarch 0:46-0.3 fedora 130.3 KiB fedora-release noarch 0:46-0.1 fedora 0.0 B fedora-release-identity-basic noarch 0:46-0.1 fedora 664.0 B fedora-repos noarch 0:46-0.3 fedora 4.9 KiB fedora-repos-rawhide noarch 0:46-0.3 fedora 2.2 KiB file s390x 0:5.48-3.fc46 fedora 101.6 KiB file-libs s390x 0:5.48-3.fc46 fedora 12.3 MiB filesystem s390x 0:3.18-59.fc45 fedora 112.0 B filesystem-srpm-macros noarch 0:3.18-59.fc45 fedora 38.2 KiB fips-provider s390x 0:1.5.2-3.fc45 fedora 4.0 MiB fonts-srpm-macros noarch 1:5.0.0-4.fc45 fedora 55.8 KiB forge-srpm-macros noarch 0:0.4.0-5.fc45 fedora 38.9 KiB fpc-srpm-macros noarch 0:1.3-17.fc45 fedora 144.0 B gap-srpm-macros noarch 0:2-3.fc45 fedora 2.1 KiB gdb-minimal s390x 0:17.2-4.fc45 fedora 15.9 MiB gdbm-libs s390x 1:1.23-12.fc45 fedora 125.2 KiB ghc-srpm-macros noarch 0:1.10-2.fc45 fedora 792.0 B glibc s390x 0:2.44.9000-2.fc46 fedora 5.5 MiB glibc-common s390x 0:2.44.9000-2.fc46 fedora 1.1 MiB glibc-gconv-extra s390x 0:2.44.9000-2.fc46 fedora 6.8 MiB gmp s390x 1:6.3.0-6.fc45 fedora 769.7 KiB gnat-srpm-macros noarch 0:7-3.fc45 fedora 1.0 KiB gnome-srpm-macros noarch 0:1.0-2.fc45 fedora 3.5 KiB gnulib-l10n noarch 0:20241231-4.fc45 fedora 655.0 KiB gnupg2 s390x 0:2.4.9-18.fc45 fedora 6.6 MiB gnupg2-dirmngr s390x 0:2.4.9-18.fc45 fedora 657.2 KiB gnupg2-gpg-agent s390x 0:2.4.9-18.fc45 fedora 717.4 KiB gnupg2-gpgconf s390x 0:2.4.9-18.fc45 fedora 269.1 KiB gnupg2-keyboxd s390x 0:2.4.9-18.fc45 fedora 213.0 KiB gnupg2-verify s390x 0:2.4.9-18.fc45 fedora 364.0 KiB gnutls s390x 0:3.8.13-3.fc45 fedora 3.7 MiB go-srpm-macros noarch 0:3.8.0-3.fc45 fedora 61.9 KiB gpgverify noarch 0:2.2-5.fc45 fedora 8.7 KiB ima-evm-utils-libs s390x 0:1.6.2-12.fc45 fedora 56.3 KiB jansson s390x 0:2.14-5.fc45 fedora 88.7 KiB java-srpm-macros noarch 0:1-9.fc45 fedora 870.0 B json-c s390x 0:0.18-9.fc45 fedora 82.3 KiB kernel-srpm-macros noarch 0:1.0-30.fc45 fedora 1.9 KiB keyutils-libs s390x 0:1.6.3-8.fc45 fedora 49.9 KiB krb5-libs s390x 0:1.22.2-9.fc45 fedora 2.5 MiB libacl s390x 0:2.4.0-2.fc45 fedora 43.6 KiB libarchive s390x 0:3.8.9-1.fc46 fedora 1.0 MiB libassuan s390x 0:2.5.7-6.fc45 fedora 163.6 KiB libattr s390x 0:2.6.0-2.fc45 fedora 24.1 KiB libblkid s390x 0:2.42.2-3.fc45 fedora 290.0 KiB libbrotli s390x 0:1.2.0-6.fc45 fedora 912.3 KiB libcap s390x 0:2.78-2.fc45 fedora 210.1 KiB libcap-ng s390x 0:0.9.5-1.fc46 fedora 64.3 KiB libcbor s390x 0:0.14.0-3.fc45 fedora 83.3 KiB libcom_err s390x 0:1.47.4-2.fc45 fedora 58.8 KiB libcurl s390x 0:8.22.0~rc3-1.fc46 fedora 1.1 MiB libeconf s390x 0:0.7.9-4.fc45 fedora 64.6 KiB libevent s390x 0:2.1.12-19.fc45 fedora 981.6 KiB libfdisk s390x 0:2.42.2-3.fc45 fedora 394.6 KiB libffi s390x 0:3.5.2-3.fc45 fedora 79.3 KiB libfido2 s390x 0:1.17.0-5.fc45 fedora 252.5 KiB libfsverity s390x 0:1.7-4.fc45 fedora 24.1 KiB libgcc s390x 0:16.2.1-2.fc46 fedora 213.6 KiB libgcrypt s390x 0:1.12.3-1.fc46 fedora 1.3 MiB libgomp s390x 0:16.2.1-2.fc46 fedora 576.9 KiB libgpg-error s390x 0:1.61-2.fc45 fedora 957.4 KiB libidn2 s390x 0:2.3.8-4.fc45 fedora 560.2 KiB libksba s390x 0:1.8.1-1.fc46 fedora 427.6 KiB liblastlog2 s390x 0:2.42.2-3.fc45 fedora 33.2 KiB libmount s390x 0:2.42.2-3.fc45 fedora 404.1 KiB libnghttp2 s390x 0:1.69.0-5.fc45 fedora 173.7 KiB libnghttp3 s390x 0:1.18.0-1.fc45 fedora 154.5 KiB libpkgconf s390x 0:2.5.1-2.fc45 fedora 93.7 KiB libpsl s390x 0:0.23.1-1.fc45 fedora 80.2 KiB libselinux s390x 0:3.11-4.fc45 fedora 217.0 KiB libselinux-utils s390x 0:3.11-4.fc45 fedora 269.5 KiB libsemanage s390x 0:3.11-3.fc45 fedora 311.8 KiB libsepol s390x 0:3.11-2.fc45 fedora 861.7 KiB libsmartcols s390x 0:2.42.2-3.fc45 fedora 196.2 KiB libssh s390x 0:0.12.2-1.fc45 fedora 709.1 KiB libssh-config noarch 0:0.12.2-1.fc45 fedora 277.0 B libstdc++ s390x 0:16.2.1-2.fc46 fedora 3.4 MiB libtasn1 s390x 0:4.21.0-2.fc45 fedora 184.4 KiB libtool-ltdl s390x 0:2.5.4-11.fc45 fedora 69.6 KiB libunistring s390x 0:1.1-12.fc45 fedora 1.8 MiB libusb1 s390x 0:1.0.30-2.fc45 fedora 175.4 KiB libuuid s390x 0:2.42.2-3.fc45 fedora 33.0 KiB libverto s390x 0:0.3.2-13.fc45 fedora 25.0 KiB libxcrypt s390x 0:4.5.2-4.fc45 fedora 289.0 KiB libxml2 s390x 0:2.13.9-4.fc45 fedora 1.8 MiB libzstd s390x 0:1.5.7-6.fc45 fedora 1.0 MiB linkdupes s390x 0:0.7.3-3.fc45 fedora 905.1 KiB lua-libs s390x 0:5.5.1-2.fc46 fedora 341.8 KiB lua-srpm-macros noarch 0:1-18.fc45 fedora 1.3 KiB lz4-libs s390x 0:1.10.0-5.fc45 fedora 197.1 KiB mpfr s390x 0:4.2.2-4.fc45 fedora 695.3 KiB ncurses-base noarch 0:6.6-3.fc45 fedora 329.7 KiB ncurses-libs s390x 0:6.6-3.fc45 fedora 1.0 MiB nettle s390x 0:4.0-5.fc45 fedora 849.1 KiB ngtcp2 s390x 0:1.22.1-3.fc45 fedora 316.9 KiB ngtcp2-crypto-ossl s390x 0:1.22.1-3.fc45 fedora 47.2 KiB npth s390x 0:1.8-5.fc45 fedora 45.1 KiB ocaml-srpm-macros noarch 0:11-4.fc45 fedora 1.9 KiB openblas-srpm-macros noarch 0:2-22.fc45 fedora 112.0 B openldap s390x 0:2.6.14-1.fc46 fedora 676.3 KiB openssl-libs s390x 1:4.0.2-1.fc46 fedora 6.0 MiB p11-kit s390x 0:0.26.5-1.fc45 fedora 3.0 MiB p11-kit-trust s390x 0:0.26.5-1.fc45 fedora 573.6 KiB package-notes-srpm-macros noarch 0:0.20-2.fc45 fedora 1.6 KiB pam-libs s390x 0:1.7.2-4.fc45 fedora 121.8 KiB pcre2 s390x 0:10.47-1.fc45.2 fedora 725.9 KiB pcre2-syntax noarch 0:10.47-1.fc45.2 fedora 281.9 KiB perl-srpm-macros noarch 0:1-65.fc45 fedora 861.0 B pkgconf s390x 0:2.5.1-2.fc45 fedora 92.5 KiB pkgconf-m4 noarch 0:2.5.1-2.fc45 fedora 14.3 KiB pkgconf-pkg-config s390x 0:2.5.1-2.fc45 fedora 988.0 B policycoreutils s390x 0:3.11-2.fc45 fedora 866.1 KiB popt s390x 0:1.19-11.fc45 fedora 140.4 KiB publicsuffix-list-dafsa noarch 0:20260624-2.fc45 fedora 71.3 KiB pyproject-srpm-macros noarch 0:1.23.2-1.fc45 fedora 3.4 KiB python-srpm-macros noarch 0:3.15-2.fc45 fedora 51.6 KiB qt5-srpm-macros noarch 0:5.15.18-3.fc45 fedora 500.0 B qt6-srpm-macros noarch 0:6.11.2-1.fc46 fedora 472.0 B readline s390x 0:8.3-6.fc45 fedora 571.1 KiB redhat-systemd-presets noarch 0:102-3.fc45 fedora 1.0 KiB redhat-systemd-presets-common noarch 0:102-3.fc45 fedora 16.9 KiB rpm s390x 0:6.1.0-1.fc46 fedora 3.1 MiB rpm-build-libs s390x 0:6.1.0-1.fc46 fedora 283.7 KiB rpm-libs s390x 0:6.1.0-1.fc46 fedora 1.1 MiB rpm-plugin-selinux s390x 0:6.1.0-1.fc46 fedora 11.6 KiB rpm-sequoia s390x 0:1.10.2-5.fc45 fedora 2.6 MiB rpm-sign-libs s390x 0:6.1.0-1.fc46 fedora 35.4 KiB rust-srpm-macros noarch 0:28.5-2.fc45 fedora 5.5 KiB selinux-policy noarch 0:45.15-2.fc46 fedora 32.1 KiB selinux-policy-targeted noarch 0:45.15-2.fc46 fedora 19.0 MiB setup noarch 0:2.15.1-2.fc45 fedora 725.6 KiB sqlite-libs s390x 0:3.53.4-1.fc45 fedora 1.7 MiB systemd-libs s390x 0:261.2-1.fc45 fedora 2.7 MiB systemd-standalone-sysusers s390x 0:261.2-1.fc45 fedora 938.4 KiB tpm2-tss s390x 0:4.2.0-1.fc45 fedora 1.6 MiB tree-sitter-srpm-macros noarch 0:0.4.6-2.fc45 fedora 8.6 KiB util-linux-core s390x 0:2.42.2-3.fc45 fedora 1.5 MiB xxhash-libs s390x 0:0.8.3-5.fc45 fedora 61.6 KiB xz-libs s390x 1:5.8.3-2.fc45 fedora 221.2 KiB zig-srpm-macros noarch 0:1-9.fc45 fedora 1.3 KiB zip s390x 0:3.0-46.fc45 fedora 729.1 KiB zlib-ng-compat s390x 0:2.3.3-6.fc45 fedora 125.1 KiB zstd s390x 0:1.5.7-6.fc45 fedora 509.6 KiB Installing groups: Buildsystem building group Transaction Summary: Installing: 188 packages Total size of inbound packages is 71 MiB. Need to download 0 B. After this operation, 232 MiB extra will be used (install 232 MiB, remove 0 B). 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publicsuffix-list-dafsa-0:202 100% | 0.0 B/s | 0.0 B | 00m00s >>> Already downloaded [181/188] libfido2-0:1.17.0-5.fc45.s390 100% | 0.0 B/s | 0.0 B | 00m00s >>> Already downloaded [182/188] libssh-config-0:0.12.2-1.fc45 100% | 0.0 B/s | 0.0 B | 00m00s >>> Already downloaded [183/188] libcbor-0:0.14.0-3.fc45.s390x 100% | 0.0 B/s | 0.0 B | 00m00s >>> Already downloaded [184/188] selinux-policy-targeted-0:45. 100% | 0.0 B/s | 0.0 B | 00m00s >>> Already downloaded [185/188] policycoreutils-0:3.11-2.fc45 100% | 0.0 B/s | 0.0 B | 00m00s >>> Already downloaded [186/188] selinux-policy-0:45.15-2.fc46 100% | 0.0 B/s | 0.0 B | 00m00s >>> Already downloaded [187/188] libselinux-utils-0:3.11-4.fc4 100% | 0.0 B/s | 0.0 B | 00m00s >>> Already downloaded [188/188] rpm-plugin-selinux-0:6.1.0-1. 100% | 0.0 B/s | 0.0 B | 00m00s >>> Already downloaded -------------------------------------------------------------------------------- [188/188] Total 100% | 0.0 B/s | 0.0 B | 00m00s Running transaction Importing OpenPGP key 0x91211FCE: UserID : "Fedora (46) " Fingerprint: D924B10D3E810DABDD8B56B596E7E91491211FCE From : file:///usr/share/distribution-gpg-keys/fedora/RPM-GPG-KEY-fedora-46-primary The key was successfully imported. Importing OpenPGP key 0x91211FCE: UserID : "Fedora (46) " Fingerprint: D924B10D3E810DABDD8B56B596E7E91491211FCE From : file:///usr/share/distribution-gpg-keys/fedora/RPM-GPG-KEY-fedora-46-primary The key was successfully imported. Importing OpenPGP key 0xF577861E: UserID : "Fedora (45) " Fingerprint: 4F50A6114CD5C6976A7F1179655A4B02F577861E From : file:///usr/share/distribution-gpg-keys/fedora/RPM-GPG-KEY-fedora-45-primary The key was successfully imported. Importing OpenPGP key 0x9488FB64: UserID : "Fedora (47) " Fingerprint: B2E766FA50CA6FA2D5ED06FC09D879E09488FB64 From : file:///usr/share/distribution-gpg-keys/fedora/RPM-GPG-KEY-fedora-47-primary The key was successfully imported. [ 1/190] Verify package files 100% | 1.0 KiB/s | 188.0 B | 00m00s >>> Running %pretrans scriptlet: filesystem-0:3.18-59.fc45.s390x >>> Finished %pretrans scriptlet: filesystem-0:3.18-59.fc45.s390x >>> [RPM] failed to open /var/lib/mock/fedora-rawhide-s390x-1788464321.539005/root/etc/group for id/name lookup: No such file or directory [ 2/190] Prepare transaction 100% | 2.3 KiB/s | 188.0 B | 00m00s [ 3/190] Installing libgcc-0:16.2.1-2. 100% | 210.3 MiB/s | 215.3 KiB | 00m00s [ 4/190] Installing redhat-systemd-pre 100% | 0.0 B/s | 1.5 KiB | 00m00s [ 5/190] Installing redhat-systemd-pre 100% | 0.0 B/s | 18.1 KiB | 00m00s [ 6/190] Installing libssh-config-0:0. 100% | 0.0 B/s | 816.0 B | 00m00s [ 7/190] Installing publicsuffix-list- 100% | 0.0 B/s | 72.0 KiB | 00m00s [ 8/190] Installing fedora-release-ide 100% | 0.0 B/s | 920.0 B | 00m00s [ 9/190] Installing fedora-repos-rawhi 100% | 0.0 B/s | 2.5 KiB | 00m00s [ 10/190] Installing fedora-gpg-keys-0: 100% | 35.5 MiB/s | 181.7 KiB | 00m00s [ 11/190] Installing fedora-repos-0:46- 100% | 0.0 B/s | 5.9 KiB | 00m00s [ 12/190] Installing fedora-release-com 100% | 7.8 MiB/s | 8.0 KiB | 00m00s [ 13/190] Installing fedora-release-0:4 100% | 20.2 KiB/s | 124.0 B | 00m00s >>> Running sysusers scriptlet: setup-0:2.15.1-2.fc45.noarch >>> Finished sysusers scriptlet: setup-0:2.15.1-2.fc45.noarch >>> Scriptlet output: >>> Creating group 'adm' with GID 4. >>> Creating group 'audio' with GID 63. >>> Creating group 'cdrom' with GID 11. >>> Creating group 'clock' with GID 103. >>> Creating group 'dialout' with GID 18. >>> Creating group 'disk' with GID 6. >>> Creating group 'floppy' with GID 19. >>> Creating group 'ftp' with GID 50. >>> Creating group 'games' with GID 20. >>> Creating group 'input' with GID 104. >>> Creating group 'kmem' with GID 9. >>> Creating group 'kvm' with GID 36. >>> Creating group 'lock' with GID 54. >>> Creating group 'lp' with GID 7. >>> Creating group 'mail' with GID 12. >>> Creating group 'man' with GID 15. >>> Creating group 'mem' with GID 8. >>> Creating group 'nobody' with GID 65534. >>> Creating group 'render' with GID 105. >>> Creating group 'root' with GID 0. >>> Creating group 'sgx' with GID 106. >>> Creating group 'sys' with GID 3. >>> Creating group 'tape' with GID 33. >>> Creating group 'tty' with GID 5. >>> Creating group 'users' with GID 100. >>> Creating group 'utmp' with GID 22. >>> Creating group 'video' with GID 39. >>> Creating group 'wheel' with GID 10. >>> Creating user 'adm' (adm) with UID 3 and GID 4. >>> Creating group 'bin' with GID 1. >>> Creating user 'bin' (bin) with UID 1 and GID 1. >>> Creating group 'daemon' with GID 2. >>> Creating user 'daemon' (daemon) with UID 2 and GID 2. >>> Creating user 'ftp' (FTP User) with UID 14 and GID 50. >>> Creating user 'games' (games) with UID 12 and GID 100. >>> Creating user 'halt' (halt) with UID 7 and GID 0. >>> Creating user 'lp' (lp) with UID 4 and GID 7. >>> Creating user 'mail' (mail) with UID 8 and GID 12. >>> Creating user 'nobody' (Kernel Overflow User) with UID 65534 and GID 65534. >>> Creating user 'operator' (operator) with UID 11 and GID 0. >>> Creating user 'root' (Super User) with UID 0 and GID 0. >>> Creating user 'shutdown' (shutdown) with UID 6 and GID 0. >>> Creating user 'sync' (sync) with UID 5 and GID 0. >>> [ 14/190] Installing setup-0:2.15.1-2.f 100% | 29.8 MiB/s | 731.3 KiB | 00m00s >>> [RPM] /etc/hosts created as /etc/hosts.rpmnew [ 15/190] Installing filesystem-0:3.18- 100% | 2.1 MiB/s | 253.6 KiB | 00m00s [ 16/190] Installing pcre2-syntax-0:10. 100% | 277.7 MiB/s | 284.3 KiB | 00m00s [ 17/190] Installing gnulib-l10n-0:2024 100% | 161.6 MiB/s | 661.9 KiB | 00m00s [ 18/190] Installing coreutils-common-0 100% | 322.3 MiB/s | 11.0 MiB | 00m00s [ 19/190] Installing ncurses-base-0:6.6 100% | 86.7 MiB/s | 355.3 KiB | 00m00s [ 20/190] Installing bash-0:5.3.15-2.fc 100% | 210.4 MiB/s | 8.6 MiB | 00m00s [ 21/190] Installing glibc-common-0:2.4 100% | 45.3 MiB/s | 1.1 MiB | 00m00s [ 22/190] Installing glibc-gconv-extra- 100% | 236.8 MiB/s | 6.9 MiB | 00m00s [ 23/190] Installing glibc-0:2.44.9000- 100% | 117.5 MiB/s | 5.5 MiB | 00m00s [ 24/190] Installing ncurses-libs-0:6.6 100% | 201.7 MiB/s | 1.0 MiB | 00m00s [ 25/190] Installing glibc-minimal-lang 100% | 0.0 B/s | 124.0 B | 00m00s [ 26/190] Installing zlib-ng-compat-0:2 100% | 123.0 MiB/s | 125.9 KiB | 00m00s [ 27/190] Installing bzip2-libs-0:1.0.8 100% | 83.4 MiB/s | 85.4 KiB | 00m00s [ 28/190] Installing libgpg-error-0:1.6 100% | 42.8 MiB/s | 963.3 KiB | 00m00s [ 29/190] Installing libstdc++-0:16.2.1 100% | 306.5 MiB/s | 3.4 MiB | 00m00s [ 30/190] Installing libzstd-0:1.5.7-6. 100% | 251.3 MiB/s | 1.0 MiB | 00m00s [ 31/190] Installing elfutils-libelf-0: 100% | 362.6 MiB/s | 1.5 MiB | 00m00s [ 32/190] Installing libassuan-0:2.5.7- 100% | 161.5 MiB/s | 165.4 KiB | 00m00s [ 33/190] Installing libgcrypt-0:1.12.3 100% | 260.8 MiB/s | 1.3 MiB | 00m00s [ 34/190] Installing readline-0:8.3-6.f 100% | 279.9 MiB/s | 573.2 KiB | 00m00s [ 35/190] Installing libacl-0:2.4.0-2.f 100% | 0.0 B/s | 44.5 KiB | 00m00s [ 36/190] Installing gmp-1:6.3.0-6.fc45 100% | 251.3 MiB/s | 772.0 KiB | 00m00s [ 37/190] Installing systemd-libs-0:261 100% | 300.5 MiB/s | 2.7 MiB | 00m00s [ 38/190] Installing libuuid-0:2.42.2-3 100% | 0.0 B/s | 34.0 KiB | 00m00s [ 39/190] Installing xz-libs-1:5.8.3-2. 100% | 217.1 MiB/s | 222.3 KiB | 00m00s [ 40/190] Installing popt-0:1.19-11.fc4 100% | 71.8 MiB/s | 147.0 KiB | 00m00s [ 41/190] Installing sqlite-libs-0:3.53 100% | 245.8 MiB/s | 1.7 MiB | 00m00s [ 42/190] Installing npth-0:1.8-5.fc45. 100% | 0.0 B/s | 46.2 KiB | 00m00s [ 43/190] Installing elfutils-libs-0:0. 100% | 256.5 MiB/s | 788.1 KiB | 00m00s [ 44/190] Installing libblkid-0:2.42.2- 100% | 284.4 MiB/s | 291.2 KiB | 00m00s [ 45/190] Installing libsepol-0:3.11-2. 100% | 280.8 MiB/s | 862.7 KiB | 00m00s [ 46/190] Installing gnupg2-gpgconf-0:2 100% | 13.2 MiB/s | 271.2 KiB | 00m00s [ 47/190] Installing pcre2-0:10.47-1.fc 100% | 236.8 MiB/s | 727.3 KiB | 00m00s [ 48/190] Installing libselinux-0:3.11- 100% | 213.1 MiB/s | 218.2 KiB | 00m00s [ 49/190] Installing grep-0:3.12-4.fc45 100% | 46.1 MiB/s | 1.0 MiB | 00m00s [ 50/190] Installing sed-0:4.10-2.fc45. 100% | 39.0 MiB/s | 878.6 KiB | 00m00s [ 51/190] Installing findutils-1:4.11.0 100% | 79.1 MiB/s | 2.0 MiB | 00m00s [ 52/190] Installing libxcrypt-0:4.5.2- 100% | 284.8 MiB/s | 291.7 KiB | 00m00s [ 53/190] Installing libtasn1-0:4.21.0- 100% | 181.9 MiB/s | 186.2 KiB | 00m00s [ 54/190] Installing libunistring-0:1.1 100% | 294.2 MiB/s | 1.8 MiB | 00m00s [ 55/190] Installing libidn2-0:2.3.8-4. 100% | 61.5 MiB/s | 566.3 KiB | 00m00s [ 56/190] Installing crypto-policies-0: 100% | 31.2 MiB/s | 159.5 KiB | 00m00s [ 57/190] Installing xz-1:5.8.3-2.fc45. 100% | 52.0 MiB/s | 1.4 MiB | 00m00s [ 58/190] Installing libmount-0:2.42.2- 100% | 197.8 MiB/s | 405.2 KiB | 00m00s [ 59/190] Installing gnupg2-verify-0:2. 100% | 17.8 MiB/s | 365.4 KiB | 00m00s [ 60/190] Installing mpfr-0:4.2.2-4.fc4 100% | 226.9 MiB/s | 697.0 KiB | 00m00s [ 61/190] Installing gawk-0:5.4.1-1.fc4 100% | 86.2 MiB/s | 2.2 MiB | 00m00s [ 62/190] Installing dwz-0:0.17-3.fc45. 100% | 16.0 MiB/s | 311.8 KiB | 00m00s [ 63/190] Installing libksba-0:1.8.1-1. 100% | 210.0 MiB/s | 430.2 KiB | 00m00s [ 64/190] Installing unzip-0:6.0-71.fc4 100% | 21.7 MiB/s | 444.5 KiB | 00m00s [ 65/190] Installing file-libs-0:5.48-3 100% | 536.7 MiB/s | 12.3 MiB | 00m00s [ 66/190] Installing file-0:5.48-3.fc46 100% | 5.3 MiB/s | 103.1 KiB | 00m00s [ 67/190] Installing diffutils-0:3.12-6 100% | 66.1 MiB/s | 1.6 MiB | 00m00s [ 68/190] Installing libattr-0:2.6.0-2. 100% | 0.0 B/s | 25.0 KiB | 00m00s [ 69/190] Installing libeconf-0:0.7.9-4 100% | 64.7 MiB/s | 66.2 KiB | 00m00s [ 70/190] Installing libcap-ng-0:0.9.5- 100% | 64.6 MiB/s | 66.2 KiB | 00m00s [ 71/190] Installing audit-libs-0:4.2.1 100% | 195.5 MiB/s | 400.4 KiB | 00m00s [ 72/190] Installing pam-libs-0:1.7.2-4 100% | 121.3 MiB/s | 124.2 KiB | 00m00s [ 73/190] Installing libcap-0:2.78-2.fc 100% | 11.1 MiB/s | 215.2 KiB | 00m00s [ 74/190] Installing libsemanage-0:3.11 100% | 153.1 MiB/s | 313.6 KiB | 00m00s [ 75/190] Installing libsmartcols-0:2.4 100% | 192.5 MiB/s | 197.2 KiB | 00m00s [ 76/190] Installing alternatives-0:1.3 100% | 3.4 MiB/s | 63.4 KiB | 00m00s [ 77/190] Installing lua-libs-0:5.5.1-2 100% | 167.6 MiB/s | 343.3 KiB | 00m00s [ 78/190] Installing json-c-0:0.18-9.fc 100% | 81.6 MiB/s | 83.5 KiB | 00m00s [ 79/190] Installing libffi-0:3.5.2-3.f 100% | 78.8 MiB/s | 80.7 KiB | 00m00s [ 80/190] Installing p11-kit-0:0.26.5-1 100% | 102.4 MiB/s | 3.0 MiB | 00m00s [ 81/190] Installing p11-kit-trust-0:0. 100% | 20.8 MiB/s | 575.3 KiB | 00m00s [ 82/190] Installing ngtcp2-0:1.22.1-3. 100% | 155.5 MiB/s | 318.4 KiB | 00m00s [ 83/190] Installing util-linux-core-0: 100% | 56.7 MiB/s | 1.5 MiB | 00m00s [ 84/190] Installing liblastlog2-0:2.42 100% | 8.6 MiB/s | 35.2 KiB | 00m00s [ 85/190] Installing zip-0:3.0-46.fc45. 100% | 31.1 MiB/s | 733.0 KiB | 00m00s [ 86/190] Installing gnupg2-keyboxd-0:2 100% | 52.3 MiB/s | 214.2 KiB | 00m00s [ 87/190] Installing libpsl-0:0.23.1-1. 100% | 79.4 MiB/s | 81.3 KiB | 00m00s [ 88/190] Installing tar-2:1.35-9.fc45. 100% | 100.7 MiB/s | 3.0 MiB | 00m00s [ 89/190] Installing linkdupes-0:0.7.3- 100% | 40.2 MiB/s | 906.5 KiB | 00m00s [ 90/190] Installing libselinux-utils-0 100% | 13.9 MiB/s | 283.9 KiB | 00m00s [ 91/190] Installing libfdisk-0:2.42.2- 100% | 193.1 MiB/s | 395.6 KiB | 00m00s [ 92/190] Installing fips-provider-0:1. 100% | 288.4 MiB/s | 4.0 MiB | 00m00s [ 93/190] Installing openssl-libs-1:4.0 100% | 284.1 MiB/s | 6.0 MiB | 00m00s [ 94/190] Installing coreutils-0:9.11-6 100% | 144.0 MiB/s | 6.0 MiB | 00m00s [ 95/190] Installing ca-certificates-0: 100% | 1.6 MiB/s | 2.5 MiB | 00m01s [ 96/190] Installing util-linux-0:2.42. 100% | 77.8 MiB/s | 3.6 MiB | 00m00s [ 97/190] Installing policycoreutils-0: 100% | 28.2 MiB/s | 894.4 KiB | 00m00s [ 98/190] Installing selinux-policy-0:4 100% | 2.0 MiB/s | 34.5 KiB | 00m00s [ 99/190] Installing selinux-policy-tar 100% | 187.0 MiB/s | 15.1 MiB | 00m00s [100/190] Installing gzip-0:1.14-4.fc45 100% | 17.1 MiB/s | 402.7 KiB | 00m00s [101/190] Installing rpm-sequoia-0:1.10 100% | 285.1 MiB/s | 2.6 MiB | 00m00s [102/190] Installing libfsverity-0:1.7- 100% | 0.0 B/s | 25.1 KiB | 00m00s [103/190] Installing libevent-0:2.1.12- 100% | 320.8 MiB/s | 985.4 KiB | 00m00s [104/190] Installing ngtcp2-crypto-ossl 100% | 0.0 B/s | 48.1 KiB | 00m00s [105/190] Installing libxml2-0:2.13.9-4 100% | 74.0 MiB/s | 1.8 MiB | 00m00s [106/190] Installing libusb1-0:1.0.30-2 100% | 28.8 MiB/s | 177.1 KiB | 00m00s >>> Running sysusers scriptlet: tpm2-tss-0:4.2.0-1.fc45.s390x >>> Finished sysusers scriptlet: tpm2-tss-0:4.2.0-1.fc45.s390x >>> Scriptlet output: >>> Creating group 'tss' with GID 59. >>> Creating user 'tss' (Account used for TPM access) with UID 59 and GID 59. >>> [107/190] Installing tpm2-tss-0:4.2.0-1 100% | 236.6 MiB/s | 1.7 MiB | 00m00s [108/190] Installing ima-evm-utils-libs 100% | 56.3 MiB/s | 57.6 KiB | 00m00s [109/190] Installing gnupg2-gpg-agent-0 100% | 25.2 MiB/s | 721.1 KiB | 00m00s [110/190] Installing nettle-0:4.0-5.fc4 100% | 208.1 MiB/s | 852.2 KiB | 00m00s [111/190] Installing gnutls-0:3.8.13-3. 100% | 283.5 MiB/s | 3.7 MiB | 00m00s [112/190] Installing zstd-0:1.5.7-6.fc4 100% | 23.9 MiB/s | 513.2 KiB | 00m00s [113/190] Installing bzip2-0:1.0.8-24.f 100% | 4.7 MiB/s | 91.1 KiB | 00m00s [114/190] Installing add-determinism-0: 100% | 90.1 MiB/s | 2.3 MiB | 00m00s [115/190] Installing build-reproducibil 100% | 0.0 B/s | 1.5 KiB | 00m00s [116/190] Installing cpio-0:2.15-10.fc4 100% | 48.5 MiB/s | 1.1 MiB | 00m00s [117/190] Installing ed-0:1.22.6-1.fc46 100% | 7.9 MiB/s | 152.9 KiB | 00m00s [118/190] Installing patch-0:2.8-5.fc45 100% | 12.3 MiB/s | 239.6 KiB | 00m00s [119/190] Installing jansson-0:2.14-5.f 100% | 88.0 MiB/s | 90.1 KiB | 00m00s [120/190] Installing lz4-libs-0:1.10.0- 100% | 193.5 MiB/s | 198.1 KiB | 00m00s [121/190] Installing libarchive-0:3.8.9 100% | 258.2 MiB/s | 1.0 MiB | 00m00s [122/190] Installing libgomp-0:16.2.1-2 100% | 282.4 MiB/s | 578.3 KiB | 00m00s [123/190] Installing libpkgconf-0:2.5.1 100% | 92.6 MiB/s | 94.9 KiB | 00m00s [124/190] Installing pkgconf-0:2.5.1-2. 100% | 4.9 MiB/s | 95.0 KiB | 00m00s [125/190] Installing libtool-ltdl-0:2.5 100% | 69.1 MiB/s | 70.7 KiB | 00m00s [126/190] Installing gdbm-libs-1:1.23-1 100% | 123.9 MiB/s | 126.9 KiB | 00m00s [127/190] Installing cyrus-sasl-lib-0:2 100% | 90.7 MiB/s | 2.4 MiB | 00m00s [128/190] Installing openldap-0:2.6.14- 100% | 221.4 MiB/s | 680.1 KiB | 00m00s [129/190] Installing gnupg2-dirmngr-0:2 100% | 23.9 MiB/s | 659.9 KiB | 00m00s [130/190] Installing gnupg2-0:2.4.9-18. 100% | 156.5 MiB/s | 6.6 MiB | 00m00s [131/190] Installing gpgverify-0:2.2-5. 100% | 0.0 B/s | 9.4 KiB | 00m00s [132/190] Installing systemd-standalone 100% | 41.7 MiB/s | 939.0 KiB | 00m00s [133/190] Installing rpm-libs-0:6.1.0-1 100% | 226.3 MiB/s | 1.1 MiB | 00m00s [134/190] Installing rpm-sign-libs-0:6. 100% | 0.0 B/s | 36.2 KiB | 00m00s [135/190] Installing rpm-build-libs-0:6 100% | 277.9 MiB/s | 284.6 KiB | 00m00s [136/190] Installing xxhash-libs-0:0.8. 100% | 61.5 MiB/s | 63.0 KiB | 00m00s [137/190] Installing libbrotli-0:1.2.0- 100% | 223.3 MiB/s | 914.6 KiB | 00m00s [138/190] Installing libnghttp2-0:1.69. 100% | 170.7 MiB/s | 174.8 KiB | 00m00s [139/190] Installing libnghttp3-0:1.18. 100% | 152.3 MiB/s | 155.9 KiB | 00m00s [140/190] Installing keyutils-libs-0:1. 100% | 0.0 B/s | 51.4 KiB | 00m00s [141/190] Installing libcom_err-0:1.47. 100% | 0.0 B/s | 59.9 KiB | 00m00s [142/190] Installing libverto-0:0.3.2-1 100% | 26.2 MiB/s | 26.8 KiB | 00m00s [143/190] Installing krb5-libs-0:1.22.2 100% | 248.1 MiB/s | 2.5 MiB | 00m00s [144/190] Installing libcbor-0:0.14.0-3 100% | 0.0 B/s | 84.7 KiB | 00m00s [145/190] Installing libfido2-0:1.17.0- 100% | 248.0 MiB/s | 254.0 KiB | 00m00s [146/190] Installing libssh-0:0.12.2-1. 100% | 231.5 MiB/s | 711.2 KiB | 00m00s [147/190] Installing libcurl-0:8.22.0~r 100% | 275.2 MiB/s | 1.1 MiB | 00m00s [148/190] Installing elfutils-debuginfo 100% | 3.8 MiB/s | 77.7 KiB | 00m00s [149/190] Installing binutils-0:2.47.50 100% | 275.1 MiB/s | 29.7 MiB | 00m00s [150/190] Installing elfutils-0:0.196-1 100% | 104.1 MiB/s | 3.0 MiB | 00m00s [151/190] Installing gdb-minimal-0:17.2 100% | 230.9 MiB/s | 15.9 MiB | 00m00s [152/190] Installing debugedit-0:5.3-3. 100% | 11.7 MiB/s | 227.4 KiB | 00m00s [153/190] Installing curl-0:8.22.0~rc3- 100% | 17.9 MiB/s | 513.1 KiB | 00m00s [154/190] Installing rpm-0:6.1.0-1.fc46 100% | 62.6 MiB/s | 2.5 MiB | 00m00s [155/190] Installing cmake-srpm-macros- 100% | 0.0 B/s | 804.0 B | 00m00s [156/190] Installing efi-srpm-macros-0: 100% | 0.0 B/s | 41.3 KiB | 00m00s [157/190] Installing java-srpm-macros-0 100% | 0.0 B/s | 1.1 KiB | 00m00s [158/190] Installing lua-srpm-macros-0: 100% | 0.0 B/s | 1.9 KiB | 00m00s [159/190] Installing tree-sitter-srpm-m 100% | 0.0 B/s | 9.6 KiB | 00m00s [160/190] Installing zig-srpm-macros-0: 100% | 0.0 B/s | 1.9 KiB | 00m00s [161/190] Installing filesystem-srpm-ma 100% | 0.0 B/s | 38.9 KiB | 00m00s [162/190] Installing pkgconf-m4-0:2.5.1 100% | 0.0 B/s | 14.7 KiB | 00m00s [163/190] Installing pkgconf-pkg-config 100% | 93.3 KiB/s | 1.8 KiB | 00m00s [164/190] Installing rust-srpm-macros-0 100% | 0.0 B/s | 6.4 KiB | 00m00s [165/190] Installing qt6-srpm-macros-0: 100% | 0.0 B/s | 748.0 B | 00m00s [166/190] Installing qt5-srpm-macros-0: 100% | 0.0 B/s | 776.0 B | 00m00s [167/190] Installing perl-srpm-macros-0 100% | 0.0 B/s | 1.1 KiB | 00m00s [168/190] Installing package-notes-srpm 100% | 343.8 KiB/s | 2.1 KiB | 00m00s [169/190] Installing openblas-srpm-macr 100% | 0.0 B/s | 392.0 B | 00m00s [170/190] Installing ocaml-srpm-macros- 100% | 0.0 B/s | 2.1 KiB | 00m00s [171/190] Installing kernel-srpm-macros 100% | 0.0 B/s | 2.3 KiB | 00m00s [172/190] Installing gnome-srpm-macros- 100% | 0.0 B/s | 4.1 KiB | 00m00s [173/190] Installing gnat-srpm-macros-0 100% | 0.0 B/s | 1.3 KiB | 00m00s [174/190] Installing ghc-srpm-macros-0: 100% | 0.0 B/s | 1.0 KiB | 00m00s [175/190] Installing gap-srpm-macros-0: 100% | 0.0 B/s | 2.7 KiB | 00m00s [176/190] Installing fpc-srpm-macros-0: 100% | 0.0 B/s | 420.0 B | 00m00s [177/190] Installing ansible-srpm-macro 100% | 35.4 MiB/s | 36.2 KiB | 00m00s [178/190] Installing rpm-build-0:6.1.0- 100% | 13.9 MiB/s | 312.1 KiB | 00m00s [179/190] Installing erlang-srpm-macros 100% | 0.0 B/s | 2.5 KiB | 00m00s [180/190] Installing pyproject-srpm-mac 100% | 0.0 B/s | 4.0 KiB | 00m00s [181/190] Installing redhat-rpm-config- 100% | 93.0 MiB/s | 190.4 KiB | 00m00s [182/190] Installing forge-srpm-macros- 100% | 0.0 B/s | 40.3 KiB | 00m00s [183/190] Installing fonts-srpm-macros- 100% | 0.0 B/s | 57.0 KiB | 00m00s [184/190] Installing go-srpm-macros-0:3 100% | 0.0 B/s | 63.0 KiB | 00m00s [185/190] Installing R-srpm-macros-0:1. 100% | 0.0 B/s | 4.4 KiB | 00m00s [186/190] Installing python-srpm-macros 100% | 0.0 B/s | 52.9 KiB | 00m00s [187/190] Installing rpm-plugin-selinux 100% | 0.0 B/s | 12.6 KiB | 00m00s [188/190] Installing which-0:2.25-2.fc4 100% | 4.2 MiB/s | 85.5 KiB | 00m00s [189/190] Installing shadow-utils-2:4.2 100% | 107.4 MiB/s | 4.0 MiB | 00m00s [190/190] Installing info-0:7.3-3.fc45. 100% | 51.9 KiB/s | 404.1 KiB | 00m08s Complete! Finish: installing minimal buildroot with dnf5 Start: creating root cache Finish: creating root cache Finish: chroot init INFO: Installed packages: INFO: R-srpm-macros-1.3.7-3.fc45.noarch add-determinism-0.7.3-3.fc45.s390x alternatives-1.33-6.fc45.s390x ansible-srpm-macros-1-22.fc45.noarch audit-libs-4.2.1-2.fc45.s390x bash-5.3.15-2.fc45.s390x binutils-2.47.50-6.fc46.s390x build-reproducibility-srpm-macros-0.7.3-3.fc45.noarch bzip2-1.0.8-24.fc45.s390x bzip2-libs-1.0.8-24.fc45.s390x ca-certificates-2026.2.90_v9.0.317-1.fc46.noarch cmake-srpm-macros-4.3.0-5.fc45.noarch coreutils-9.11-6.fc45.s390x coreutils-common-9.11-6.fc45.s390x cpio-2.15-10.fc45.s390x crypto-policies-20260713-2.gitc42da77.fc45.noarch curl-8.22.0~rc3-1.fc46.s390x cyrus-sasl-lib-2.1.28-37.fc45.s390x debugedit-5.3-3.fc45.s390x diffutils-3.12-6.fc45.s390x dwz-0.17-3.fc45.s390x ed-1.22.6-1.fc46.s390x efi-srpm-macros-6-9.fc45.noarch elfutils-0.196-1.fc46.s390x elfutils-debuginfod-client-0.196-1.fc46.s390x elfutils-libelf-0.196-1.fc46.s390x elfutils-libs-0.196-1.fc46.s390x erlang-srpm-macros-0.3.11-2.fc45.noarch fedora-gpg-keys-46-0.3.noarch fedora-release-46-0.1.noarch fedora-release-common-46-0.1.noarch fedora-release-identity-basic-46-0.1.noarch fedora-repos-46-0.3.noarch fedora-repos-rawhide-46-0.3.noarch file-5.48-3.fc46.s390x file-libs-5.48-3.fc46.s390x filesystem-3.18-59.fc45.s390x filesystem-srpm-macros-3.18-59.fc45.noarch findutils-4.11.0-2.fc45.s390x fips-provider-1.5.2-3.fc45.s390x fonts-srpm-macros-5.0.0-4.fc45.noarch forge-srpm-macros-0.4.0-5.fc45.noarch fpc-srpm-macros-1.3-17.fc45.noarch gap-srpm-macros-2-3.fc45.noarch gawk-5.4.1-1.fc45.s390x gdb-minimal-17.2-4.fc45.s390x gdbm-libs-1.23-12.fc45.s390x ghc-srpm-macros-1.10-2.fc45.noarch glibc-2.44.9000-2.fc46.s390x glibc-common-2.44.9000-2.fc46.s390x glibc-gconv-extra-2.44.9000-2.fc46.s390x glibc-minimal-langpack-2.44.9000-2.fc46.s390x gmp-6.3.0-6.fc45.s390x gnat-srpm-macros-7-3.fc45.noarch gnome-srpm-macros-1.0-2.fc45.noarch gnulib-l10n-20241231-4.fc45.noarch gnupg2-2.4.9-18.fc45.s390x gnupg2-dirmngr-2.4.9-18.fc45.s390x gnupg2-gpg-agent-2.4.9-18.fc45.s390x gnupg2-gpgconf-2.4.9-18.fc45.s390x gnupg2-keyboxd-2.4.9-18.fc45.s390x gnupg2-verify-2.4.9-18.fc45.s390x gnutls-3.8.13-3.fc45.s390x go-srpm-macros-3.8.0-3.fc45.noarch gpg-pubkey-4f50a6114cd5c6976a7f1179655a4b02f577861e-6888bc98 gpg-pubkey-b2e766fa50ca6fa2d5ed06fc09d879e09488fb64-6a6b46b4 gpg-pubkey-d924b10d3e810dabdd8b56b596e7e91491211fce-697c9899 gpgverify-2.2-5.fc45.noarch grep-3.12-4.fc45.s390x gzip-1.14-4.fc45.s390x ima-evm-utils-libs-1.6.2-12.fc45.s390x info-7.3-3.fc45.s390x jansson-2.14-5.fc45.s390x java-srpm-macros-1-9.fc45.noarch json-c-0.18-9.fc45.s390x kernel-srpm-macros-1.0-30.fc45.noarch keyutils-libs-1.6.3-8.fc45.s390x krb5-libs-1.22.2-9.fc45.s390x libacl-2.4.0-2.fc45.s390x libarchive-3.8.9-1.fc46.s390x libassuan-2.5.7-6.fc45.s390x libattr-2.6.0-2.fc45.s390x libblkid-2.42.2-3.fc45.s390x libbrotli-1.2.0-6.fc45.s390x libcap-2.78-2.fc45.s390x libcap-ng-0.9.5-1.fc46.s390x libcbor-0.14.0-3.fc45.s390x libcom_err-1.47.4-2.fc45.s390x libcurl-8.22.0~rc3-1.fc46.s390x libeconf-0.7.9-4.fc45.s390x libevent-2.1.12-19.fc45.s390x libfdisk-2.42.2-3.fc45.s390x libffi-3.5.2-3.fc45.s390x libfido2-1.17.0-5.fc45.s390x libfsverity-1.7-4.fc45.s390x libgcc-16.2.1-2.fc46.s390x libgcrypt-1.12.3-1.fc46.s390x libgomp-16.2.1-2.fc46.s390x libgpg-error-1.61-2.fc45.s390x libidn2-2.3.8-4.fc45.s390x libksba-1.8.1-1.fc46.s390x liblastlog2-2.42.2-3.fc45.s390x libmount-2.42.2-3.fc45.s390x libnghttp2-1.69.0-5.fc45.s390x libnghttp3-1.18.0-1.fc45.s390x libpkgconf-2.5.1-2.fc45.s390x libpsl-0.23.1-1.fc45.s390x libselinux-3.11-4.fc45.s390x libselinux-utils-3.11-4.fc45.s390x libsemanage-3.11-3.fc45.s390x libsepol-3.11-2.fc45.s390x libsmartcols-2.42.2-3.fc45.s390x libssh-0.12.2-1.fc45.s390x libssh-config-0.12.2-1.fc45.noarch libstdc++-16.2.1-2.fc46.s390x libtasn1-4.21.0-2.fc45.s390x libtool-ltdl-2.5.4-11.fc45.s390x libunistring-1.1-12.fc45.s390x libusb1-1.0.30-2.fc45.s390x libuuid-2.42.2-3.fc45.s390x libverto-0.3.2-13.fc45.s390x libxcrypt-4.5.2-4.fc45.s390x libxml2-2.13.9-4.fc45.s390x libzstd-1.5.7-6.fc45.s390x linkdupes-0.7.3-3.fc45.s390x lua-libs-5.5.1-2.fc46.s390x lua-srpm-macros-1-18.fc45.noarch lz4-libs-1.10.0-5.fc45.s390x mpfr-4.2.2-4.fc45.s390x ncurses-base-6.6-3.fc45.noarch ncurses-libs-6.6-3.fc45.s390x nettle-4.0-5.fc45.s390x ngtcp2-1.22.1-3.fc45.s390x ngtcp2-crypto-ossl-1.22.1-3.fc45.s390x npth-1.8-5.fc45.s390x ocaml-srpm-macros-11-4.fc45.noarch openblas-srpm-macros-2-22.fc45.noarch openldap-2.6.14-1.fc46.s390x openssl-libs-4.0.2-1.fc46.s390x p11-kit-0.26.5-1.fc45.s390x p11-kit-trust-0.26.5-1.fc45.s390x package-notes-srpm-macros-0.20-2.fc45.noarch pam-libs-1.7.2-4.fc45.s390x patch-2.8-5.fc45.s390x pcre2-10.47-1.fc45.2.s390x pcre2-syntax-10.47-1.fc45.2.noarch perl-srpm-macros-1-65.fc45.noarch pkgconf-2.5.1-2.fc45.s390x pkgconf-m4-2.5.1-2.fc45.noarch pkgconf-pkg-config-2.5.1-2.fc45.s390x policycoreutils-3.11-2.fc45.s390x popt-1.19-11.fc45.s390x publicsuffix-list-dafsa-20260624-2.fc45.noarch pyproject-srpm-macros-1.23.2-1.fc45.noarch python-srpm-macros-3.15-2.fc45.noarch qt5-srpm-macros-5.15.18-3.fc45.noarch qt6-srpm-macros-6.11.2-1.fc46.noarch readline-8.3-6.fc45.s390x redhat-rpm-config-344-5.fc45.noarch redhat-systemd-presets-102-3.fc45.noarch redhat-systemd-presets-common-102-3.fc45.noarch rpm-6.1.0-1.fc46.s390x rpm-build-6.1.0-1.fc46.s390x rpm-build-libs-6.1.0-1.fc46.s390x rpm-libs-6.1.0-1.fc46.s390x rpm-plugin-selinux-6.1.0-1.fc46.s390x rpm-sequoia-1.10.2-5.fc45.s390x rpm-sign-libs-6.1.0-1.fc46.s390x rust-srpm-macros-28.5-2.fc45.noarch sed-4.10-2.fc45.s390x selinux-policy-45.15-2.fc46.noarch selinux-policy-targeted-45.15-2.fc46.noarch setup-2.15.1-2.fc45.noarch shadow-utils-4.20.0-1.fc45.s390x sqlite-libs-3.53.4-1.fc45.s390x systemd-libs-261.2-1.fc45.s390x systemd-standalone-sysusers-261.2-1.fc45.s390x tar-1.35-9.fc45.s390x tpm2-tss-4.2.0-1.fc45.s390x tree-sitter-srpm-macros-0.4.6-2.fc45.noarch unzip-6.0-71.fc45.s390x util-linux-2.42.2-3.fc45.s390x util-linux-core-2.42.2-3.fc45.s390x which-2.25-2.fc45.s390x xxhash-libs-0.8.3-5.fc45.s390x xz-5.8.3-2.fc45.s390x xz-libs-5.8.3-2.fc45.s390x zig-srpm-macros-1-9.fc45.noarch zip-3.0-46.fc45.s390x zlib-ng-compat-2.3.3-6.fc45.s390x zstd-1.5.7-6.fc45.s390x Start: buildsrpm Start: rpmbuild -bs Building target platforms: s390x Building for target s390x setting SOURCE_DATE_EPOCH=1788134400 Wrote: /builddir/build/SRPMS/baresip-4.11.0-1.fc46.src.rpm Finish: rpmbuild -bs INFO: chroot_scan: 1 files copied to /var/lib/copr-rpmbuild/results/chroot_scan INFO: /var/lib/mock/fedora-rawhide-s390x-1788464321.539005/root/var/log/dnf5.log INFO: chroot_scan: creating tarball /var/lib/copr-rpmbuild/results/chroot_scan.tar.gz /bin/tar: Removing leading `/' from member names Finish: buildsrpm INFO: Done(/var/lib/copr-rpmbuild/workspace/workdir-n6mi2rvw/baresip/baresip.spec) Config(child) 0 minutes 25 seconds INFO: Results and/or logs in: /var/lib/copr-rpmbuild/results INFO: Cleaning up build root ('cleanup_on_success=True') Start: clean chroot INFO: unmounting tmpfs. Finish: clean chroot INFO: Start(/var/lib/copr-rpmbuild/results/baresip-4.11.0-1.fc46.src.rpm) Config(fedora-rawhide-s390x) Start(bootstrap): chroot init INFO: mounting tmpfs at /var/lib/mock/fedora-rawhide-s390x-bootstrap-1788464321.539005/root. INFO: reusing tmpfs at /var/lib/mock/fedora-rawhide-s390x-bootstrap-1788464321.539005/root. INFO: calling preinit hooks INFO: enabled root cache INFO: enabled package manager cache Start(bootstrap): cleaning package manager metadata Finish(bootstrap): cleaning package manager metadata Finish(bootstrap): chroot init Start: chroot init INFO: mounting tmpfs at /var/lib/mock/fedora-rawhide-s390x-1788464321.539005/root. INFO: calling preinit hooks INFO: enabled root cache Start: unpacking root cache Finish: unpacking root cache INFO: enabled package manager cache Start: cleaning package manager metadata Finish: cleaning package manager metadata INFO: enabled HW Info plugin INFO: Buildroot is handled by package management downloaded with a bootstrap image: rpm-6.1.0-1.fc46.s390x rpm-sequoia-1.10.2-5.fc45.s390x dnf5-5.4.3.0-2.fc46.s390x dnf5-plugins-5.4.3.0-2.fc46.s390x Finish: chroot init Start: build phase for baresip-4.11.0-1.fc46.src.rpm Start: build setup for baresip-4.11.0-1.fc46.src.rpm Building target platforms: s390x Building for target s390x setting SOURCE_DATE_EPOCH=1788134400 Wrote: /builddir/build/SRPMS/baresip-4.11.0-1.fc46.src.rpm Updating and loading repositories: fedora 100% | 3.7 KiB/s | 4.1 KiB | 00m01s Copr repository 100% | 1.7 KiB/s | 1.5 KiB | 00m01s Repositories loaded. Package Arch Version Repository Size Installing: alsa-lib-devel s390x 0:1.2.16.1-2.fc45 fedora 17.7 MiB cjson-devel s390x 0:1.7.19-2.fc45 fedora 51.1 KiB cmake s390x 0:4.3.0-5.fc45 fedora 37.2 MiB codec2-devel s390x 0:1.2.0-10.fc45 fedora 55.7 KiB desktop-file-utils s390x 0:0.28-8.fc45 fedora 241.4 KiB fdk-aac-free-devel s390x 0:2.0.3-3.fc45 fedora 1.1 MiB gcc s390x 0:16.2.1-2.fc46 fedora 88.6 MiB gcc-c++ s390x 0:16.2.1-2.fc46 fedora 34.8 MiB glib2-devel s390x 0:2.89.4-1.fc46 fedora 15.8 MiB gstreamer1-devel s390x 0:1.28.6-2.fc46 fedora 6.5 MiB gtk3-devel s390x 0:3.24.52-6.fc45 fedora 34.6 MiB libX11-devel s390x 0:1.8.13-2.fc45 fedora 1.0 MiB libXext-devel s390x 0:1.3.6-6.fc45 fedora 97.7 KiB libaom-devel s390x 0:3.13.3-3.fc45 fedora 224.5 KiB libavcodec-free-devel s390x 0:9.0.1-4.fc46 copr_base 265.3 KiB libavdevice-free-devel s390x 0:9.0.1-4.fc46 copr_base 17.9 KiB libavfilter-free-devel s390x 0:9.0.1-4.fc46 copr_base 63.3 KiB libavformat-free-devel s390x 0:9.0.1-4.fc46 copr_base 154.8 KiB libavutil-free-devel s390x 0:9.0.1-4.fc46 copr_base 740.3 KiB libpng-devel s390x 2:1.6.58-2.fc45 fedora 909.4 KiB libre-devel s390x 0:4.11.0-1.fc46 fedora 269.5 KiB libsndfile-devel s390x 0:1.2.2-12.fc45 fedora 212.9 KiB libswresample-free-devel s390x 0:9.0.1-4.fc46 copr_base 26.4 KiB libswscale-free-devel s390x 0:9.0.1-4.fc46 copr_base 32.7 KiB libv4l-devel s390x 0:1.32.0-4.fc45 fedora 715.9 KiB libvpx-devel s390x 0:1.17.0-1.fc46 fedora 3.8 MiB mosquitto-devel s390x 0:2.1.2-4.fc45 fedora 252.3 KiB openssl-devel s390x 1:4.0.2-1.fc46 fedora 4.6 MiB opus-devel s390x 0:1.6-3.fc45 fedora 2.6 MiB pipewire-devel s390x 0:1.6.8-3.fc45 fedora 994.1 KiB pipewire-jack-audio-connection-kit-devel s390x 0:1.6.8-3.fc45 fedora 187.4 KiB portaudio-devel s390x 0:19.7.0-4.fc45 fedora 2.5 MiB pulseaudio-libs-devel s390x 0:17.0-11.fc45 fedora 4.8 MiB python3-devel s390x 0:3.15.0~rc1-1.fc45 fedora 2.4 MiB sdl2-compat-devel s390x 0:2.32.70-2.fc45 fedora 3.3 MiB spandsp-devel s390x 0:0.0.6-23.fc45 fedora 843.7 KiB Installing dependencies: SDL3 s390x 0:3.4.14-1.fc45 fedora 3.4 MiB abattis-cantarell-vf-fonts noarch 0:0.301-18.fc45 fedora 192.7 KiB abseil-cpp s390x 0:20260526.0-3.fc45 fedora 2.5 MiB adwaita-cursor-theme noarch 0:51~beta-1.fc46 fedora 11.4 MiB adwaita-icon-theme noarch 0:51~beta-1.fc46 fedora 1.1 MiB adwaita-icon-theme-legacy noarch 0:46.2-12.fc45 fedora 2.1 MiB alsa-lib s390x 0:1.2.16.1-2.fc45 fedora 1.5 MiB annobin-docs noarch 0:13.31-1.fc46 fedora 99.7 KiB annobin-plugin-gcc s390x 0:13.31-1.fc46 fedora 715.8 KiB aom s390x 0:3.13.3-3.fc45 fedora 456.9 KiB aribb24 s390x 0:1.0.3^20160216git5e9be27-6.fc45 fedora 82.8 KiB at-spi2-atk s390x 0:2.61.1-1.fc45 fedora 295.2 KiB at-spi2-atk-devel s390x 0:2.61.1-1.fc45 fedora 1.6 KiB at-spi2-core s390x 0:2.61.1-1.fc45 fedora 1.6 MiB at-spi2-core-devel s390x 0:2.61.1-1.fc45 fedora 4.5 MiB atk s390x 0:2.61.1-1.fc45 fedora 272.5 KiB atk-devel s390x 0:2.61.1-1.fc45 fedora 6.0 MiB authselect s390x 0:1.8.0-1.fc46 fedora 231.6 KiB authselect-libs s390x 0:1.8.0-1.fc46 fedora 935.1 KiB autoconf noarch 0:2.73-4.fc45 fedora 2.9 MiB automake noarch 0:1.18.1-5.fc45 fedora 1.8 MiB avahi-glib s390x 0:0.9~rc2-9.fc45 fedora 15.2 KiB avahi-libs s390x 0:0.9~rc2-9.fc45 fedora 165.2 KiB bluez-libs s390x 0:5.87-7.fc46 fedora 222.1 KiB brotli s390x 0:1.2.0-6.fc45 fedora 33.2 KiB brotli-devel s390x 0:1.2.0-6.fc45 fedora 65.9 KiB bubblewrap s390x 0:0.12.0-1.fc46 fedora 139.5 KiB bzip2-devel s390x 0:1.0.8-24.fc45 fedora 309.8 KiB c-ares s390x 0:1.34.8-2.fc45 fedora 297.0 KiB cairo s390x 0:1.18.4-9.fc45 fedora 1.8 MiB cairo-devel s390x 0:1.18.4-9.fc45 fedora 2.3 MiB cairo-gobject s390x 0:1.18.4-9.fc45 fedora 34.8 KiB cairo-gobject-devel s390x 0:1.18.4-9.fc45 fedora 7.0 KiB check s390x 0:0.15.2-22.fc45 fedora 145.6 KiB check-devel s390x 0:0.15.2-22.fc45 fedora 120.1 KiB check-static s390x 0:0.15.2-22.fc45 fedora 378.1 KiB cjson s390x 0:1.7.19-2.fc45 fedora 63.6 KiB cmake-data noarch 0:4.3.0-5.fc45 fedora 9.4 MiB cmake-filesystem s390x 0:4.3.0-5.fc45 fedora 0.0 B cmake-rpm-macros noarch 0:4.3.0-5.fc45 fedora 8.4 KiB codec2 s390x 0:1.2.0-10.fc45 fedora 1.4 MiB colord-libs s390x 0:1.4.8-5.fc45 fedora 844.9 KiB cpp s390x 0:16.2.1-2.fc46 fedora 30.6 MiB cracklib s390x 0:2.10.3-2.fc45 fedora 237.7 KiB cups-filesystem noarch 1:2.4.19-4.fc45 fedora 0.0 B cups-libs s390x 1:2.4.19-4.fc45 fedora 718.3 KiB dbus s390x 1:1.16.2-6.fc45 fedora 0.0 B dbus-broker s390x 0:37-9.fc45 fedora 389.3 KiB dbus-common noarch 1:1.16.2-6.fc45 fedora 11.2 KiB dbus-devel s390x 1:1.16.2-6.fc45 fedora 131.7 KiB dbus-libs s390x 1:1.16.2-6.fc45 fedora 365.3 KiB default-fonts-core-sans noarch 0:4.3-3.fc45 fedora 11.9 KiB duktape s390x 0:2.7.0-12.fc45 fedora 727.4 KiB emacs-filesystem s390x 1:30.2-3.fc45 fedora 0.0 B expat s390x 0:2.8.3-1.fc46 fedora 353.8 KiB fdk-aac-free s390x 0:2.0.3-3.fc45 fedora 655.5 KiB fftw-libs-double s390x 0:3.3.10-18.fc45 fedora 1.7 MiB fftw-libs-single s390x 0:3.3.10-18.fc45 fedora 1.7 MiB flac s390x 0:1.5.0-9.fc45 fedora 343.2 KiB flac-devel s390x 0:1.5.0-9.fc45 fedora 5.2 MiB flac-libs s390x 0:1.5.0-9.fc45 fedora 578.5 KiB flite s390x 0:2.2-14.fc45 fedora 25.2 MiB fmt s390x 0:12.1.0-3.fc45 fedora 287.4 KiB fontconfig s390x 0:2.18.3-1.fc45 fedora 1.0 MiB fontconfig-devel s390x 0:2.18.3-1.fc45 fedora 127.7 KiB fonts-filesystem noarch 1:5.0.0-4.fc45 fedora 0.0 B freeglut s390x 0:3.8.0-3.fc45 fedora 532.6 KiB freetype s390x 0:2.14.3-2.fc45 fedora 966.6 KiB freetype-devel s390x 0:2.14.3-2.fc45 fedora 8.6 MiB fribidi s390x 0:1.0.16-5.fc45 fedora 193.5 KiB fribidi-devel s390x 0:1.0.16-5.fc45 fedora 78.0 KiB game-music-emu s390x 0:0.6.5-2.fc45 fedora 383.1 KiB gcc-plugin-annobin s390x 0:16.2.1-2.fc46 fedora 56.7 KiB gdbm s390x 1:1.23-12.fc45 fedora 470.9 KiB gdk-pixbuf2 s390x 0:2.44.8-3.fc46 fedora 2.6 MiB gdk-pixbuf2-devel s390x 0:2.44.8-3.fc46 fedora 2.2 MiB gettext s390x 0:1.0-3.fc45 fedora 17.8 MiB gettext-common-devel noarch 0:1.0-3.fc45 fedora 596.2 KiB gettext-devel s390x 0:1.0-3.fc45 fedora 1.4 MiB gettext-envsubst s390x 0:1.0-3.fc45 fedora 70.3 KiB gettext-libs s390x 0:1.0-3.fc45 fedora 2.4 MiB gettext-runtime s390x 0:1.0-3.fc45 fedora 710.9 KiB giflib s390x 0:6.1.3-3.fc45 fedora 125.9 KiB glfw s390x 1:3.4-8.fc45 fedora 425.0 KiB glib-networking s390x 0:2.90~alpha-2.fc45 fedora 795.8 KiB glib2 s390x 0:2.89.4-1.fc46 fedora 15.5 MiB glibc-devel s390x 0:2.44.9000-2.fc46 fedora 3.7 MiB glycin-devel s390x 0:2.2~beta-1.fc46 fedora 151.2 KiB glycin-libs s390x 0:2.2~beta-1.fc46 fedora 5.3 MiB glycin-loaders s390x 0:2.2~beta-1.fc46 fedora 17.2 MiB google-noto-fonts-common noarch 0:20260901-2.fc46 fedora 17.7 KiB google-noto-sans-vf-fonts noarch 0:20260901-2.fc46 fedora 1.4 MiB graphite2 s390x 0:1.3.14-22.fc45 fedora 194.8 KiB graphite2-devel s390x 0:1.3.14-22.fc45 fedora 49.3 KiB groff-base s390x 0:1.24.1-2.fc45 fedora 4.6 MiB gsettings-desktop-schemas s390x 0:51~beta-1.fc45 fedora 6.5 MiB gsm s390x 0:1.0.24-3.fc45 fedora 61.3 KiB gstreamer1 s390x 0:1.28.6-2.fc46 fedora 5.6 MiB gtk-update-icon-cache s390x 0:3.24.52-6.fc45 fedora 61.7 KiB gtk3 s390x 0:3.24.52-6.fc45 fedora 23.2 MiB harfbuzz s390x 0:14.4.0-1.fc46 fedora 3.0 MiB harfbuzz-cairo s390x 0:14.4.0-1.fc46 fedora 43.8 KiB harfbuzz-devel s390x 0:14.4.0-1.fc46 fedora 6.5 MiB harfbuzz-gpu s390x 0:14.4.0-1.fc46 fedora 167.9 KiB harfbuzz-icu s390x 0:14.4.0-1.fc46 fedora 11.0 KiB harfbuzz-raster s390x 0:14.4.0-1.fc46 fedora 104.1 KiB harfbuzz-vector s390x 0:14.4.0-1.fc46 fedora 108.8 KiB hicolor-icon-theme noarch 0:0.18-6.fc45 fedora 72.4 KiB hwdata noarch 0:0.410-1.fc45 fedora 10.0 MiB ilbc s390x 0:3.0.4-21.fc45 fedora 87.0 KiB jbigkit-libs s390x 0:2.1-34.fc45 fedora 116.7 KiB json-glib s390x 0:1.10.8-11.fc45 fedora 600.1 KiB jsoncpp s390x 0:1.9.6-3.fc44 fedora 263.5 KiB kernel-headers s390x 0:7.3.0-0.rc1.18.fc46 fedora 7.0 MiB lame-libs s390x 0:4.0-1.fc46 fedora 926.1 KiB lcms2 s390x 0:2.16-8.fc45 fedora 456.6 KiB leptonica s390x 0:1.87.0-5.fc45 fedora 3.3 MiB libGLEW s390x 0:2.2.0-13.fc45 fedora 760.1 KiB libX11 s390x 0:1.8.13-2.fc45 fedora 1.4 MiB libX11-common noarch 0:1.8.13-2.fc45 fedora 1.1 MiB libX11-xcb s390x 0:1.8.13-2.fc45 fedora 6.6 KiB libXau s390x 0:1.0.12-5.fc45 fedora 68.6 KiB libXau-devel s390x 0:1.0.12-5.fc45 fedora 7.3 KiB libXcomposite s390x 0:0.4.6-8.fc45 fedora 40.1 KiB libXcomposite-devel s390x 0:0.4.6-8.fc45 fedora 7.6 KiB libXcursor s390x 0:1.2.3-5.fc45 fedora 53.1 KiB libXcursor-devel s390x 0:1.2.3-5.fc45 fedora 20.5 KiB libXdamage s390x 0:1.1.6-8.fc45 fedora 39.3 KiB libXdamage-devel s390x 0:1.1.6-8.fc45 fedora 2.5 KiB libXext s390x 0:1.3.6-6.fc45 fedora 93.5 KiB libXfixes s390x 0:6.0.1-8.fc45 fedora 29.8 KiB libXfixes-devel s390x 0:6.0.1-8.fc45 fedora 9.2 KiB libXft s390x 0:2.3.8-11.fc45 fedora 168.0 KiB libXft-devel s390x 0:2.3.8-11.fc45 fedora 29.1 KiB libXi s390x 0:1.8.3-2.fc45 fedora 80.1 KiB libXi-devel s390x 0:1.8.3-2.fc45 fedora 132.2 KiB libXinerama s390x 0:1.1.5-11.fc45 fedora 14.6 KiB libXinerama-devel s390x 0:1.1.5-11.fc45 fedora 6.9 KiB libXrandr s390x 0:1.5.4-8.fc45 fedora 51.3 KiB libXrandr-devel s390x 0:1.5.4-8.fc45 fedora 21.3 KiB libXrender s390x 0:0.9.12-5.fc45 fedora 45.6 KiB libXrender-devel s390x 0:0.9.12-5.fc45 fedora 50.1 KiB libXtst s390x 0:1.2.5-5.fc45 fedora 33.1 KiB libXtst-devel s390x 0:1.2.5-5.fc45 fedora 11.2 KiB libXv s390x 0:1.0.13-5.fc45 fedora 25.6 KiB libXxf86vm s390x 0:1.1.6-5.fc45 fedora 24.8 KiB libaom s390x 0:3.13.3-3.fc45 fedora 3.4 MiB libaribcaption s390x 0:1.1.1-5.fc45 fedora 243.3 KiB libasan s390x 0:16.2.1-2.fc46 fedora 1.9 MiB libass s390x 0:0.17.4-3.fc45 fedora 257.7 KiB libasyncns s390x 0:0.8-35.fc45 fedora 55.0 KiB libatomic s390x 0:16.2.1-2.fc46 fedora 37.2 KiB libavcodec-free s390x 0:9.0.1-4.fc46 copr_base 10.2 MiB libavdevice-free s390x 0:9.0.1-4.fc46 copr_base 248.0 KiB libavfilter-free s390x 0:9.0.1-4.fc46 copr_base 4.9 MiB libavformat-free s390x 0:9.0.1-4.fc46 copr_base 3.1 MiB libavutil-free s390x 0:9.0.1-4.fc46 copr_base 861.9 KiB libblkid-devel s390x 0:2.42.2-3.fc45 fedora 45.0 KiB libbluray s390x 0:1.4.0-4.fc45 fedora 338.2 KiB libbs2b s390x 0:3.1.0-38.fc45 fedora 54.9 KiB libcaca s390x 0:0.99-0.86.beta20.fc45 fedora 881.3 KiB libcdio s390x 0:2.3.0-2.fc45 fedora 593.7 KiB libcdio-paranoia s390x 0:10.2+2.0.2-8.fc45 fedora 185.7 KiB libchromaprint s390x 0:1.6.1-1.fc46 copr_base 73.4 KiB libcloudproviders s390x 0:0.4.0-8.fc45 fedora 128.1 KiB libcloudproviders-devel s390x 0:0.4.0-8.fc45 fedora 1.4 MiB libdatrie s390x 0:0.2.14-3.fc45 fedora 57.5 KiB libdatrie-devel s390x 0:0.2.14-3.fc45 fedora 411.1 KiB libdav1d s390x 0:1.5.3-2.fc45 fedora 645.2 KiB libdecor s390x 0:0.2.5-3.fc45 fedora 172.0 KiB libdovi s390x 0:3.3.2-4.fc45 fedora 656.8 KiB libdrm s390x 0:2.4.134-2.fc45 fedora 420.7 KiB libdvdnav s390x 0:7.0.0-2.fc45 fedora 112.5 KiB libdvdread s390x 0:7.1.1-1.fc45 fedora 191.9 KiB libebur128 s390x 0:1.2.6-16.fc45 fedora 39.0 KiB libedit s390x 0:3.1-60.20260512cvs.fc45 fedora 276.0 KiB libepoxy s390x 0:1.5.10-13.fc45 fedora 1.3 MiB libepoxy-devel s390x 0:1.5.10-13.fc45 fedora 1.6 MiB libffi-devel s390x 0:3.5.2-3.fc45 fedora 30.1 KiB libglvnd s390x 1:1.7.0-10.fc45 fedora 874.6 KiB libglvnd-core-devel s390x 1:1.7.0-10.fc45 fedora 40.3 KiB libglvnd-devel s390x 1:1.7.0-10.fc45 fedora 2.1 MiB libglvnd-egl s390x 1:1.7.0-10.fc45 fedora 68.4 KiB libglvnd-gles s390x 1:1.7.0-10.fc45 fedora 113.3 KiB libglvnd-glx s390x 1:1.7.0-10.fc45 fedora 765.0 KiB libglvnd-opengl s390x 1:1.7.0-10.fc45 fedora 208.9 KiB libgusb s390x 0:0.4.9-6.fc45 fedora 157.6 KiB libheif s390x 0:1.23.1-11.fc46 copr_base 2.3 MiB libicu s390x 0:78.3-8.fc45 fedora 38.1 MiB libicu-devel s390x 0:78.3-8.fc45 fedora 5.2 MiB libjpeg-turbo s390x 0:3.1.4.1-2.fc45 fedora 872.5 KiB libjpeg-turbo-devel s390x 0:3.1.4.1-2.fc45 fedora 354.2 KiB libjxl s390x 1:0.11.2-2.fc45 fedora 2.8 MiB libklvanc s390x 0:1.6.0-5.fc45 fedora 151.7 KiB liblc3 s390x 0:1.1.3-9.fc45 fedora 182.5 KiB libldb s390x 2:4.24.5-1.fc45 fedora 473.4 KiB liblerc s390x 0:4.2.0-1.fc45 fedora 258.2 KiB liblerc-devel s390x 0:4.2.0-1.fc45 fedora 4.3 MiB libmodplug s390x 1:0.8.9.0-30.fc45 fedora 387.0 KiB libmount-devel s390x 0:2.42.2-3.fc45 fedora 66.5 KiB libmpc s390x 0:1.4.1-2.fc45 fedora 164.5 KiB libmysofa s390x 0:1.3.3-5.fc45 fedora 78.3 KiB libnsl2 s390x 0:2.0.1-6.fc45 fedora 57.5 KiB libogg s390x 2:1.3.6-4.fc45 fedora 49.3 KiB libogg-devel s390x 2:1.3.6-4.fc45 fedora 27.0 KiB libopenjph s390x 0:0.31.0-1.fc46 fedora 357.6 KiB libopenmpt s390x 0:0.8.9-1.fc46 fedora 1.8 MiB libpciaccess s390x 0:0.16-18.fc45 fedora 44.2 KiB libplacebo s390x 0:7.360.1-4.fc45 fedora 1.2 MiB libpng s390x 2:1.6.58-2.fc45 fedora 253.4 KiB libproxy s390x 0:0.5.12-3.fc45 fedora 101.5 KiB libpwquality s390x 0:1.4.5-18.fc45 fedora 399.8 KiB librabbitmq s390x 0:0.17.0-2.fc45 fedora 97.2 KiB libre s390x 0:4.11.0-1.fc46 fedora 958.5 KiB librist s390x 0:0.2.11-2.fc45 fedora 169.8 KiB librsvg2 s390x 0:2.62.3-3.fc45 fedora 6.1 MiB libsamplerate s390x 0:0.2.2-13.fc45 fedora 1.4 MiB libsbc s390x 0:2.1-3.fc45 fedora 76.9 KiB libseccomp s390x 0:2.6.1-3.fc45 fedora 238.0 KiB libseccomp-devel s390x 0:2.6.1-3.fc45 fedora 115.4 KiB libselinux-devel s390x 0:3.11-4.fc45 fedora 121.9 KiB libsepol-devel s390x 0:3.11-2.fc45 fedora 120.7 KiB libshaderc s390x 0:2026.2-3.fc45 fedora 3.9 MiB libsmbclient s390x 2:4.24.5-1.fc45 fedora 171.5 KiB libsndfile s390x 0:1.2.2-12.fc45 fedora 617.8 KiB libsodium s390x 0:1.0.22-2.fc45 fedora 416.2 KiB libsoup3 s390x 0:3.7.2-1.fc46 fedora 1.2 MiB libstdc++-devel s390x 0:16.2.1-2.fc46 fedora 40.7 MiB libstemmer s390x 0:3.1.1-2.fc45 fedora 957.3 KiB libswresample-free s390x 0:9.0.1-4.fc46 copr_base 130.2 KiB libswscale-free s390x 0:9.0.1-4.fc46 copr_base 873.6 KiB libtalloc s390x 0:2.4.4-4.fc45 fedora 59.5 KiB libtdb s390x 0:1.4.15-4.fc45 fedora 104.2 KiB libtevent s390x 0:0.17.1-7.fc45 fedora 105.1 KiB libtextstyle s390x 0:1.0-3.fc45 fedora 214.9 KiB libthai s390x 0:0.1.30-3.fc45 fedora 800.6 KiB libthai-devel s390x 0:0.1.30-3.fc45 fedora 510.5 KiB libtheora s390x 1:1.1.1-42.fc45 fedora 439.7 KiB libtiff s390x 0:4.7.2-2.fc45 fedora 670.9 KiB libtiff-devel s390x 0:4.7.2-2.fc45 fedora 885.0 KiB libtinysparql s390x 0:3.12~alpha-2.fc45 fedora 1.0 MiB libtirpc s390x 0:1.3.7-3.fc45 fedora 207.7 KiB libubsan s390x 0:16.2.1-2.fc46 fedora 499.8 KiB libudfread s390x 0:1.2.0-4.fc45 fedora 61.8 KiB libunibreak s390x 0:6.1-5.fc45 fedora 138.7 KiB libunwind s390x 0:1.8.3-4.fc46 fedora 166.1 KiB libuv s390x 1:1.52.1-2.fc45 fedora 588.2 KiB libv4l s390x 0:1.32.0-4.fc45 fedora 370.8 KiB libva s390x 0:2.24.1-2.fc45 fedora 336.7 KiB libvdpau s390x 0:1.5-12.fc45 fedora 16.4 KiB libvorbis s390x 1:1.3.7-21.fc45 fedora 892.4 KiB libvorbis-devel s390x 1:1.3.7-21.fc45 fedora 49.4 KiB libvpx s390x 0:1.17.0-1.fc46 fedora 7.6 MiB libwayland-client s390x 0:1.26.0-1.fc45 fedora 73.7 KiB libwayland-cursor s390x 0:1.26.0-1.fc45 fedora 36.9 KiB libwayland-egl s390x 0:1.26.0-1.fc45 fedora 8.0 KiB libwayland-server s390x 0:1.26.0-1.fc45 fedora 98.3 KiB libwbclient s390x 2:4.24.5-1.fc45 fedora 69.9 KiB libwebp s390x 0:1.6.0-4.fc45 fedora 711.1 KiB libwebp-devel s390x 0:1.6.0-4.fc45 fedora 121.7 KiB libxcb s390x 0:1.17.0-8.fc45 fedora 1.1 MiB libxcb-devel s390x 0:1.17.0-8.fc45 fedora 2.6 MiB libxcrypt-devel s390x 0:4.5.2-4.fc45 fedora 31.0 KiB libxkbcommon s390x 0:1.13.1-3.fc45 fedora 439.1 KiB libxkbcommon-devel s390x 0:1.13.1-3.fc45 fedora 393.4 KiB libxml2-devel s390x 0:2.13.9-4.fc45 fedora 3.2 MiB libxshmfence s390x 0:1.3.2-9.fc45 fedora 12.1 KiB libzstd-devel s390x 0:1.5.7-6.fc45 fedora 217.6 KiB lilv-libs s390x 0:0.28.0-3.fc45 fedora 133.9 KiB llvm-filesystem s390x 0:22.1.8-26.fc46 fedora 0.0 B llvm-libs s390x 0:22.1.8-26.fc46 fedora 155.0 MiB lm_sensors-libs s390x 0:3.6.0-25.fc45 fedora 85.4 KiB lmdb-libs s390x 0:0.9.34-3.fc45 fedora 109.6 KiB lpcnetfreedv s390x 0:0.5-11.fc45 fedora 14.8 MiB lzo s390x 0:2.10-17.fc45 fedora 198.6 KiB lzo-devel s390x 0:2.10-17.fc45 fedora 212.2 KiB lzo-minilzo s390x 0:2.10-17.fc45 fedora 40.4 KiB m4 s390x 0:1.4.21-3.fc45 fedora 900.3 KiB make s390x 1:4.4.1-13.fc45 fedora 1.9 MiB mesa-dri-drivers s390x 0:26.2.1-1.fc46 fedora 20.2 MiB mesa-filesystem s390x 0:26.2.1-1.fc46 fedora 3.6 KiB mesa-libEGL s390x 0:26.2.1-1.fc46 fedora 382.1 KiB mesa-libGL s390x 0:26.2.1-1.fc46 fedora 370.1 KiB mesa-libGLU s390x 0:9.0.3-9.fc45 fedora 405.3 KiB mesa-libGLU-devel s390x 0:9.0.3-9.fc45 fedora 17.0 KiB mesa-libgbm s390x 0:26.2.1-1.fc46 fedora 19.3 KiB mosquitto s390x 0:2.1.2-4.fc45 fedora 1.8 MiB mpdecimal s390x 0:4.0.1-4.fc45 fedora 236.8 KiB mpg123-devel s390x 0:1.33.6-3.fc45 fedora 3.1 MiB mpg123-libs s390x 0:1.33.6-3.fc45 fedora 849.4 KiB ncurses s390x 0:6.6-3.fc45 fedora 621.7 KiB nettle3.10 s390x 0:3.10.1-7.fc45 fedora 764.2 KiB ngtcp2-crypto-gnutls s390x 0:1.22.1-3.fc45 fedora 39.2 KiB ninja-build s390x 0:1.13.2-6.fc45 fedora 484.4 KiB noopenh264 s390x 0:2.6.0-5.fc45 fedora 34.8 KiB openal-soft s390x 0:1.24.2-9.fc46 copr_base 1.7 MiB openapv-libs s390x 0:0.3.0.0-1.fc46 fedora 120.7 KiB opencore-amr s390x 0:0.1.6-11.fc45 fedora 336.4 KiB openjpeg s390x 0:2.5.4-4.fc45 fedora 484.0 KiB openpgm s390x 0:5.3.128-7.fc45 fedora 304.0 KiB opus s390x 0:1.6-3.fc45 fedora 475.2 KiB pam s390x 0:1.7.2-4.fc45 fedora 1.5 MiB pango s390x 0:1.58.2-1.fc45 fedora 1.1 MiB pango-devel s390x 0:1.58.2-1.fc45 fedora 1.5 MiB pcre2-devel s390x 0:10.47-1.fc45.2 fedora 2.1 MiB pcre2-utf16 s390x 0:10.47-1.fc45.2 fedora 674.8 KiB pcre2-utf32 s390x 0:10.47-1.fc45.2 fedora 642.7 KiB perl-AutoLoader noarch 0:5.74-527.fc45 fedora 20.6 KiB perl-B s390x 0:1.92-527.fc45 fedora 529.3 KiB perl-Carp noarch 0:1.54-527.fc45 fedora 46.6 KiB perl-Class-Struct noarch 0:0.68-527.fc45 fedora 25.4 KiB perl-Data-Dumper s390x 0:2.192-528.fc45 fedora 111.2 KiB perl-Digest noarch 0:1.20-527.fc45 fedora 35.2 KiB perl-Digest-MD5 s390x 0:2.59-527.fc45 fedora 55.3 KiB perl-DynaLoader s390x 0:1.57-527.fc45 fedora 32.1 KiB perl-Encode s390x 4:3.24-527.fc45 fedora 9.6 MiB perl-Errno s390x 0:1.39-527.fc45 fedora 8.4 KiB perl-Exporter noarch 0:5.79-527.fc45 fedora 54.2 KiB perl-Fcntl s390x 0:1.20-527.fc45 fedora 48.4 KiB perl-File-Basename noarch 0:2.86-527.fc45 fedora 14.0 KiB perl-File-Compare noarch 0:1.100.800-527.fc45 fedora 5.6 KiB perl-File-Copy noarch 0:2.43-527.fc45 fedora 19.6 KiB perl-File-Find noarch 0:1.44-527.fc45 fedora 42.0 KiB perl-File-Path noarch 0:2.18-527.fc45 fedora 63.4 KiB perl-File-Temp noarch 1:0.231.200-527.fc45 fedora 163.7 KiB perl-File-stat noarch 0:1.15-527.fc45 fedora 13.7 KiB perl-FileHandle noarch 0:2.05-527.fc45 fedora 9.4 KiB perl-Getopt-Long noarch 1:2.58-527.fc45 fedora 144.5 KiB perl-Getopt-Std noarch 0:1.15-527.fc45 fedora 12.5 KiB perl-HTTP-Tiny noarch 0:0.096-528.fc45 fedora 162.4 KiB perl-IO s390x 0:1.56-527.fc45 fedora 147.0 KiB perl-IO-Socket-IP noarch 0:0.44-527.fc45 fedora 100.5 KiB perl-IO-Socket-SSL noarch 0:2.099-2.fc45 fedora 726.9 KiB perl-IPC-Open3 noarch 0:1.24-527.fc45 fedora 27.7 KiB perl-MIME-Base32 noarch 0:1.303-26.fc45 fedora 30.7 KiB perl-MIME-Base64 s390x 0:3.16-527.fc45 fedora 45.6 KiB perl-Net-SSLeay s390x 0:1.96-5.fc45 fedora 1.4 MiB perl-POSIX s390x 0:2.26-527.fc45 fedora 241.1 KiB perl-PathTools s390x 0:3.95-527.fc45 fedora 175.7 KiB perl-Pod-Escapes noarch 1:1.07-527.fc45 fedora 24.9 KiB perl-Pod-Perldoc noarch 0:3.28.01-527.fc45 fedora 163.8 KiB perl-Pod-Simple noarch 1:3.48-527.fc45 fedora 565.2 KiB perl-Pod-Usage noarch 4:2.05-527.fc45 fedora 86.3 KiB perl-Scalar-List-Utils s390x 5:1.70-527.fc45 fedora 144.5 KiB perl-SelectSaver noarch 0:1.03-527.fc45 fedora 2.1 KiB perl-Socket s390x 4:2.043-1.fc46 fedora 125.5 KiB perl-Storable s390x 1:3.41-527.fc45 fedora 226.6 KiB perl-Symbol noarch 0:1.09-527.fc45 fedora 6.8 KiB perl-Term-ANSIColor noarch 0:5.01-527.fc45 fedora 97.5 KiB perl-Term-Cap noarch 0:1.18-527.fc45 fedora 29.3 KiB perl-Text-ParseWords noarch 0:3.31-527.fc45 fedora 13.6 KiB perl-Text-Tabs+Wrap noarch 0:2024.001-527.fc45 fedora 22.6 KiB perl-Thread-Queue noarch 0:3.14-527.fc45 fedora 28.9 KiB perl-Time-HiRes s390x 4:1.9780-527.fc45 fedora 115.4 KiB perl-Time-Local noarch 2:1.350-527.fc45 fedora 69.0 KiB perl-URI noarch 0:5.36-1.fc46 fedora 275.3 KiB perl-Unicode-Normalize s390x 0:1.32-527.fc45 fedora 672.2 KiB perl-base noarch 0:2.27-527.fc45 fedora 12.6 KiB perl-constant noarch 0:1.33-528.fc45 fedora 26.3 KiB perl-if noarch 0:0.61.000-527.fc45 fedora 5.8 KiB perl-interpreter s390x 4:5.44.0-527.fc45 fedora 119.5 KiB perl-libnet noarch 0:3.15-528.fc45 fedora 288.4 KiB perl-libs s390x 4:5.44.0-527.fc45 fedora 12.3 MiB perl-locale noarch 0:1.13-527.fc45 fedora 6.1 KiB perl-mro s390x 0:1.30-527.fc45 fedora 41.2 KiB perl-overload noarch 0:1.40-527.fc45 fedora 71.6 KiB perl-overloading noarch 0:0.03-527.fc45 fedora 4.8 KiB perl-parent noarch 1:0.244-527.fc45 fedora 10.3 KiB perl-podlators noarch 1:6.1.0-2.fc45 fedora 324.7 KiB perl-threads s390x 1:2.45-527.fc45 fedora 114.6 KiB perl-threads-shared s390x 0:1.73-527.fc45 fedora 83.3 KiB perl-vars noarch 0:1.05-527.fc45 fedora 3.9 KiB pipewire-jack-audio-connection-kit s390x 0:1.6.8-3.fc45 fedora 30.0 B pipewire-jack-audio-connection-kit-libs s390x 0:1.6.8-3.fc45 fedora 519.7 KiB pipewire-libs s390x 0:1.6.8-3.fc45 fedora 9.8 MiB pixman s390x 0:0.46.2-4.fc45 fedora 560.2 KiB pixman-devel s390x 0:0.46.2-4.fc45 fedora 56.1 KiB portaudio s390x 0:19.7.0-4.fc45 fedora 255.8 KiB pulseaudio-libs s390x 0:17.0-11.fc45 fedora 3.4 MiB pulseaudio-libs-glib2 s390x 0:17.0-11.fc45 fedora 19.3 KiB pyproject-rpm-macros noarch 0:1.23.2-1.fc45 fedora 156.3 KiB python-pip-wheel noarch 0:26.1.2-5.fc46 fedora 1.2 MiB python-rpm-macros noarch 0:3.15-2.fc45 fedora 28.0 KiB python3 s390x 0:3.15.0~rc1-1.fc45 fedora 24.6 KiB python3-libs s390x 0:3.15.0~rc1-1.fc45 fedora 48.0 MiB python3-packaging noarch 0:26.3-1.fc46 fedora 1.3 MiB python3-rpm-generators noarch 0:14-35.fc46 fedora 81.9 KiB python3-rpm-macros noarch 0:3.15-2.fc45 fedora 6.5 KiB qrencode-libs s390x 0:4.1.1-13.fc45 fedora 164.4 KiB rav1e-libs s390x 0:0.8.1-4.fc45 fedora 2.2 MiB rhash s390x 0:1.4.5-5.fc45 fedora 354.0 KiB rubberband-libs s390x 0:4.0.0-6.fc45 fedora 461.2 KiB samba-client-libs s390x 2:4.24.5-1.fc45 fedora 11.7 MiB samba-common noarch 2:4.24.5-1.fc45 fedora 218.1 KiB samba-core-libs s390x 2:4.24.5-1.fc45 fedora 2.0 MiB samba-ndr-libs s390x 2:4.24.5-1.fc45 fedora 4.7 MiB sdl2-compat s390x 0:2.32.70-2.fc45 fedora 464.2 KiB serd s390x 0:0.32.10-2.fc45 fedora 131.7 KiB shared-mime-info s390x 0:2.5.1-2.fc45 fedora 6.6 MiB slang s390x 0:2.3.3-10.fc45 fedora 2.0 MiB snappy s390x 0:1.2.2-6.fc45 fedora 79.3 KiB sord s390x 0:0.16.22-2.fc45 fedora 76.5 KiB soxr s390x 0:0.1.3-22.fc45 fedora 166.2 KiB spandsp s390x 0:0.0.6-23.fc45 fedora 920.0 KiB speex s390x 0:1.2.0-22.fc45 fedora 144.2 KiB spirv-tools-libs s390x 0:2026.2-3.fc45 fedora 6.4 MiB sratom s390x 0:0.6.22-2.fc45 fedora 44.3 KiB srt-libs s390x 0:1.5.7-1.fc46 fedora 982.9 KiB subunit s390x 0:1.4.6-3.fc45 fedora 67.7 KiB subunit-devel s390x 0:1.4.6-3.fc45 fedora 5.3 KiB svt-av1-libs s390x 0:3.1.2-3.fc45 fedora 2.7 MiB sysprof-capture-devel s390x 0:51~alpha-2.fc45 fedora 742.1 KiB systemd s390x 0:261.2-1.fc45 fedora 13.8 MiB systemd-devel s390x 0:261.2-1.fc45 fedora 635.8 KiB systemd-rpm-macros noarch 0:261.2-1.fc45 fedora 8.5 KiB systemd-shared s390x 0:261.2-1.fc45 fedora 6.1 MiB tesseract-common noarch 0:5.5.3-1.fc45 fedora 14.7 KiB tesseract-langpack-eng noarch 0:4.1.0-13.fc45 fedora 3.9 MiB tesseract-libs s390x 0:5.5.3-1.fc45 fedora 3.3 MiB tesseract-tessdata-doc noarch 0:4.1.0-13.fc45 fedora 15.0 KiB twolame-libs s390x 0:0.4.0-10.fc45 fedora 165.2 KiB tzdata noarch 0:2026c-1.fc45 fedora 1.2 MiB vid.stab s390x 0:1.1.1-9.fc45 fedora 99.8 KiB vim-filesystem noarch 2:9.2.967-1.fc46 fedora 40.0 B vo-amrwbenc s390x 0:0.1.3-25.fc45 fedora 153.5 KiB vulkan-loader s390x 0:1.4.350.0-2.fc45 fedora 561.7 KiB wayland-devel s390x 0:1.26.0-1.fc45 fedora 720.4 KiB webrtc-audio-processing s390x 0:2.1-11.fc45 fedora 1.1 MiB xevd-libs s390x 0:0.7.0-2.fc46 fedora 303.2 KiB xeve-libs s390x 0:0.7.0-2.fc46 fedora 923.9 KiB xkeyboard-config noarch 0:2.48-4.fc45 fedora 6.4 MiB xml-common noarch 0:0.6.3-69.fc45 fedora 78.4 KiB xorg-x11-proto-devel noarch 0:2025.1-3.fc45 fedora 1.8 MiB xprop s390x 0:1.2.8-6.fc45 fedora 58.3 KiB xvidcore s390x 0:1.3.7-20.fc45 fedora 720.0 KiB xz-devel s390x 1:5.8.3-2.fc45 fedora 258.8 KiB zeromq s390x 0:4.3.5-24.fc45 fedora 959.9 KiB zimg s390x 0:3.0.6-6.fc46 fedora 249.9 KiB zix s390x 0:0.8.0-3.fc45 fedora 66.5 KiB zlib-ng-compat-devel s390x 0:2.3.3-6.fc45 fedora 107.0 KiB zvbi s390x 0:0.2.45-1.fc46 fedora 1.3 MiB Transaction Summary: Installing: 477 packages Total size of inbound packages is 321 MiB. 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Installing tesseract-tessdata 100% | 0.0 B/s | 15.7 KiB | 00m00s [210/479] Installing tesseract-common-0 100% | 20.4 MiB/s | 20.9 KiB | 00m00s [211/479] Installing tesseract-langpack 100% | 326.9 MiB/s | 3.9 MiB | 00m00s [212/479] Installing google-noto-fonts- 100% | 0.0 B/s | 18.5 KiB | 00m00s [213/479] Installing google-noto-sans-v 100% | 231.9 MiB/s | 1.4 MiB | 00m00s [214/479] Installing default-fonts-core 100% | 17.8 MiB/s | 18.2 KiB | 00m00s [215/479] Installing fontconfig-0:2.18. 100% | 1.0 MiB/s | 1.1 MiB | 00m01s [216/479] Installing libaribcaption-0:1 100% | 23.9 MiB/s | 244.7 KiB | 00m00s >>> Running sysusers scriptlet: dbus-common-1:1.16.2-6.fc45.noarch >>> Finished sysusers scriptlet: dbus-common-1:1.16.2-6.fc45.noarch >>> Scriptlet output: >>> Creating group 'dbus' with GID 81. >>> Creating user 'dbus' (System Message Bus) with UID 81 and GID 81. >>> [217/479] Installing dbus-common-1:1.16 100% | 1.3 MiB/s | 13.6 KiB | 00m00s [218/479] Installing dbus-broker-0:37-9 100% | 13.7 MiB/s | 391.9 KiB | 00m00s [219/479] Installing dbus-1:1.16.2-6.fc 100% | 0.0 B/s | 124.0 B | 00m00s [220/479] Installing gettext-common-dev 100% | 582.5 MiB/s | 596.5 KiB | 00m00s [221/479] Installing lm_sensors-libs-0: 100% | 84.5 MiB/s | 86.5 KiB | 00m00s [222/479] Installing libxshmfence-0:1.3 100% | 0.0 B/s | 13.2 KiB | 00m00s [223/479] Installing ngtcp2-crypto-gnut 100% | 0.0 B/s | 40.0 KiB | 00m00s [224/479] Installing lmdb-libs-0:0.9.34 100% | 108.4 MiB/s | 111.0 KiB | 00m00s [225/479] Installing libldb-2:4.24.5-1. 100% | 155.9 MiB/s | 479.0 KiB | 00m00s [226/479] Installing giflib-0:6.1.3-3.f 100% | 124.6 MiB/s | 127.6 KiB | 00m00s [227/479] Installing spirv-tools-libs-0 100% | 305.6 MiB/s | 6.4 MiB | 00m00s [228/479] Installing libshaderc-0:2026. 100% | 302.4 MiB/s | 3.9 MiB | 00m00s [229/479] Installing hwdata-0:0.410-1.f 100% | 436.7 MiB/s | 10.0 MiB | 00m00s [230/479] Installing libpciaccess-0:0.1 100% | 44.5 MiB/s | 45.6 KiB | 00m00s [231/479] Installing libdrm-0:2.4.134-2 100% | 207.3 MiB/s | 424.6 KiB | 00m00s [232/479] Installing mesa-libgbm-0:26.2 100% | 0.0 B/s | 20.1 KiB | 00m00s [233/479] Installing mesa-dri-drivers-0 100% | 396.6 MiB/s | 20.2 MiB | 00m00s [234/479] Installing bubblewrap-0:0.12. 100% | 6.9 MiB/s | 141.7 KiB | 00m00s [235/479] Installing gettext-envsubst-0 100% | 3.7 MiB/s | 71.7 KiB | 00m00s [236/479] Installing gettext-runtime-0: 100% | 32.0 MiB/s | 721.4 KiB | 00m00s [237/479] Installing brotli-0:1.2.0-6.f 100% | 1.8 MiB/s | 34.1 KiB | 00m00s [238/479] Installing brotli-devel-0:1.2 100% | 66.8 MiB/s | 68.4 KiB | 00m00s [239/479] Installing m4-0:1.4.21-3.fc45 100% | 40.3 MiB/s | 906.8 KiB | 00m00s [240/479] Installing libudfread-0:1.2.0 100% | 61.7 MiB/s | 63.2 KiB | 00m00s [241/479] Installing libbluray-0:1.4.0- 100% | 165.9 MiB/s | 339.8 KiB | 00m00s [242/479] Installing openpgm-0:5.3.128- 100% | 298.3 MiB/s | 305.4 KiB | 00m00s [243/479] Installing libsodium-0:1.0.22 100% | 203.8 MiB/s | 417.4 KiB | 00m00s [244/479] Installing zeromq-0:4.3.5-24. 100% | 41.3 MiB/s | 972.4 KiB | 00m00s [245/479] Installing libsamplerate-0:0. 100% | 287.0 MiB/s | 1.4 MiB | 00m00s [246/479] Installing fftw-libs-double-0 100% | 280.1 MiB/s | 1.7 MiB | 00m00s [247/479] Installing rubberband-libs-0: 100% | 225.7 MiB/s | 462.3 KiB | 00m00s [248/479] Installing vulkan-loader-0:1. 100% | 275.6 MiB/s | 564.4 KiB | 00m00s [249/479] Installing libdovi-0:3.3.2-4. 100% | 214.4 MiB/s | 658.5 KiB | 00m00s [250/479] Installing libplacebo-0:7.360 100% | 234.7 MiB/s | 1.2 MiB | 00m00s [251/479] Installing libunibreak-0:6.1- 100% | 137.1 MiB/s | 140.4 KiB | 00m00s [252/479] Installing libass-0:0.17.4-3. 100% | 126.5 MiB/s | 259.1 KiB | 00m00s [253/479] Installing libcdio-0:2.3.0-2. 100% | 28.0 MiB/s | 601.8 KiB | 00m00s [254/479] Installing libcdio-paranoia-0 100% | 9.7 MiB/s | 188.9 KiB | 00m00s [255/479] Installing slang-0:2.3.3-10.f 100% | 281.4 MiB/s | 2.0 MiB | 00m00s [256/479] Installing jbigkit-libs-0:2.1 100% | 115.9 MiB/s | 118.7 KiB | 00m00s [257/479] Installing libtiff-0:4.7.2-2. 100% | 219.1 MiB/s | 673.1 KiB | 00m00s [258/479] Installing spandsp-0:0.0.6-23 100% | 225.1 MiB/s | 922.1 KiB | 00m00s [259/479] Installing libtiff-devel-0:4. 100% | 218.7 MiB/s | 895.9 KiB | 00m00s [260/479] Installing leptonica-0:1.87.0 100% | 296.4 MiB/s | 3.3 MiB | 00m00s [261/479] Installing tesseract-libs-0:5 100% | 272.1 MiB/s | 3.3 MiB | 00m00s [262/479] Installing abseil-cpp-0:20260 100% | 194.8 MiB/s | 2.5 MiB | 00m00s [263/479] Installing webrtc-audio-proce 100% | 264.8 MiB/s | 1.1 MiB | 00m00s [264/479] Installing glfw-1:3.4-8.fc45. 100% | 138.8 MiB/s | 426.3 KiB | 00m00s [265/479] Installing xkeyboard-config-0 100% | 269.7 MiB/s | 6.5 MiB | 00m00s [266/479] Installing libxkbcommon-0:1.1 100% | 215.2 MiB/s | 440.8 KiB | 00m00s [267/479] Installing libstemmer-0:3.1.1 100% | 312.1 MiB/s | 958.7 KiB | 00m00s [268/479] Installing libtinysparql-0:3. 100% | 262.4 MiB/s | 1.0 MiB | 00m00s [269/479] Installing cups-filesystem-1: 100% | 0.0 B/s | 1.8 KiB | 00m00s [270/479] Installing cups-libs-1:2.4.19 100% | 87.9 MiB/s | 719.9 KiB | 00m00s [271/479] Installing adwaita-icon-theme 100% | 65.1 MiB/s | 2.4 MiB | 00m00s [272/479] Installing adwaita-cursor-the 100% | 601.5 MiB/s | 11.4 MiB | 00m00s [273/479] Installing adwaita-icon-theme 100% | 74.3 MiB/s | 1.3 MiB | 00m00s [274/479] Installing lzo-minilzo-0:2.10 100% | 40.7 MiB/s | 41.7 KiB | 00m00s [275/479] Installing lzo-devel-0:2.10-1 100% | 210.2 MiB/s | 215.3 KiB | 00m00s [276/479] Installing bzip2-devel-0:1.0. 100% | 75.9 MiB/s | 310.7 KiB | 00m00s [277/479] Installing systemd-devel-0:26 100% | 58.4 MiB/s | 777.7 KiB | 00m00s [278/479] Installing dbus-devel-1:1.16. 100% | 133.3 MiB/s | 136.5 KiB | 00m00s [279/479] Installing systemd-shared-0:2 100% | 137.6 MiB/s | 6.1 MiB | 00m00s >>> Running sysusers scriptlet: systemd-0:261.2-1.fc45.s390x >>> Finished sysusers scriptlet: systemd-0:261.2-1.fc45.s390x >>> Scriptlet output: >>> Creating group 'empower' with GID 999. >>> >>> Running sysusers scriptlet: systemd-0:261.2-1.fc45.s390x >>> Finished sysusers scriptlet: systemd-0:261.2-1.fc45.s390x >>> Scriptlet output: >>> Creating group 'systemd-journal' with GID 190. >>> >>> Running sysusers scriptlet: systemd-0:261.2-1.fc45.s390x >>> Finished sysusers scriptlet: systemd-0:261.2-1.fc45.s390x >>> Scriptlet output: >>> Creating group 'systemd-imds' with GID 998. >>> Creating user 'systemd-imds' (systemd Instance Metadata) with UID 998 and GID 998. >>> >>> Running sysusers scriptlet: systemd-0:261.2-1.fc45.s390x >>> Finished sysusers scriptlet: systemd-0:261.2-1.fc45.s390x >>> Scriptlet output: >>> Creating group 'systemd-oom' with GID 997. >>> Creating user 'systemd-oom' (systemd Userspace OOM Killer) with UID 997 and GID 997. >>> [280/479] Installing systemd-0:261.2-1. 100% | 84.8 MiB/s | 13.9 MiB | 00m00s >>> Running sysusers scriptlet: samba-common-2:4.24.5-1.fc45.noarch >>> Finished sysusers scriptlet: samba-common-2:4.24.5-1.fc45.noarch >>> Scriptlet output: >>> Creating group 'printadmin' with GID 996. >>> [281/479] Installing samba-common-2:4.2 100% | 27.1 MiB/s | 221.7 KiB | 00m00s [282/479] Installing samba-core-libs-2: 100% | 282.7 MiB/s | 2.0 MiB | 00m00s [283/479] Installing samba-ndr-libs-2:4 100% | 278.5 MiB/s | 4.7 MiB | 00m00s [284/479] Installing samba-client-libs- 100% | 292.2 MiB/s | 11.7 MiB | 00m00s [285/479] Installing libsmbclient-2:4.2 100% | 168.3 MiB/s | 172.3 KiB | 00m00s [286/479] Installing soxr-0:0.1.3-22.fc 100% | 164.3 MiB/s | 168.3 KiB | 00m00s [287/479] Installing librabbitmq-0:0.17 100% | 96.0 MiB/s | 98.3 KiB | 00m00s [288/479] Installing libmodplug-1:0.8.9 100% | 189.8 MiB/s | 388.8 KiB | 00m00s [289/479] Installing game-music-emu-0:0 100% | 187.8 MiB/s | 384.5 KiB | 00m00s [290/479] Installing zimg-0:3.0.6-6.fc4 100% | 245.5 MiB/s | 251.4 KiB | 00m00s [291/479] Installing vid.stab-0:1.1.1-9 100% | 98.7 MiB/s | 101.1 KiB | 00m00s [292/479] Installing qrencode-libs-0:4. 100% | 162.4 MiB/s | 166.3 KiB | 00m00s [293/479] Installing libmysofa-0:1.3.3- 100% | 77.9 MiB/s | 79.7 KiB | 00m00s [294/479] Installing libklvanc-0:1.6.0- 100% | 149.5 MiB/s | 153.1 KiB | 00m00s [295/479] Installing xvidcore-0:1.3.7-2 100% | 234.9 MiB/s | 721.6 KiB | 00m00s [296/479] Installing xeve-libs-0:0.7.0- 100% | 301.1 MiB/s | 925.0 KiB | 00m00s [297/479] Installing xevd-libs-0:0.7.0- 100% | 148.6 MiB/s | 304.3 KiB | 00m00s [298/479] Installing vo-amrwbenc-0:0.1. 100% | 151.4 MiB/s | 155.0 KiB | 00m00s [299/479] Installing twolame-libs-0:0.4 100% | 162.4 MiB/s | 166.3 KiB | 00m00s [300/479] Installing speex-0:1.2.0-22.f 100% | 142.7 MiB/s | 146.1 KiB | 00m00s [301/479] Installing snappy-0:1.2.2-6.f 100% | 79.0 MiB/s | 80.9 KiB | 00m00s [302/479] Installing opencore-amr-0:0.1 100% | 165.5 MiB/s | 338.9 KiB | 00m00s [303/479] Installing openapv-libs-0:0.3 100% | 23.9 MiB/s | 122.1 KiB | 00m00s [304/479] Installing ilbc-0:3.0.4-21.fc 100% | 14.4 MiB/s | 88.5 KiB | 00m00s [305/479] Installing tzdata-0:2026c-1.f 100% | 45.6 MiB/s | 1.5 MiB | 00m00s [306/479] Installing python-pip-wheel-0 100% | 623.3 MiB/s | 1.2 MiB | 00m00s [307/479] Installing mpdecimal-0:4.0.1- 100% | 29.1 MiB/s | 238.4 KiB | 00m00s [308/479] Installing python3-libs-0:3.1 100% | 295.0 MiB/s | 48.4 MiB | 00m00s [309/479] Installing python3-0:3.15.0~r 100% | 1.3 MiB/s | 26.3 KiB | 00m00s [310/479] Installing python3-packaging- 100% | 261.2 MiB/s | 1.3 MiB | 00m00s [311/479] Installing gettext-0:1.0-3.fc 100% | 318.9 MiB/s | 17.9 MiB | 00m00s [312/479] Installing gettext-devel-0:1. 100% | 62.3 MiB/s | 1.4 MiB | 00m00s [313/479] Installing autoconf-0:2.73-4. 100% | 110.8 MiB/s | 2.9 MiB | 00m00s [314/479] Installing automake-0:1.18.1- 100% | 73.5 MiB/s | 1.8 MiB | 00m00s [315/479] Installing libogg-devel-2:1.3 100% | 28.1 MiB/s | 28.8 KiB | 00m00s [316/479] Installing opus-devel-0:1.6-3 100% | 292.3 MiB/s | 2.6 MiB | 00m00s [317/479] Installing flac-devel-0:1.5.0 100% | 406.1 MiB/s | 5.3 MiB | 00m00s [318/479] Installing libvorbis-devel-1: 100% | 50.6 MiB/s | 51.9 KiB | 00m00s [319/479] Installing python3-rpm-genera 100% | 81.1 MiB/s | 83.1 KiB | 00m00s [320/479] Installing gstreamer1-0:1.28. 100% | 145.0 MiB/s | 5.7 MiB | 00m00s [321/479] Installing ninja-build-0:1.13 100% | 23.8 MiB/s | 487.5 KiB | 00m00s [322/479] Installing cmake-rpm-macros-0 100% | 0.0 B/s | 9.0 KiB | 00m00s [323/479] Installing libasyncns-0:0.8-3 100% | 55.1 MiB/s | 56.4 KiB | 00m00s [324/479] Installing pulseaudio-libs-0: 100% | 286.7 MiB/s | 3.4 MiB | 00m00s [325/479] Installing pulseaudio-libs-gl 100% | 19.7 MiB/s | 20.2 KiB | 00m00s [326/479] Installing flite-0:2.2-14.fc4 100% | 280.3 MiB/s | 25.2 MiB | 00m00s [327/479] Installing libsbc-0:2.1-3.fc4 100% | 76.2 MiB/s | 78.0 KiB | 00m00s [328/479] Installing libebur128-0:1.2.6 100% | 39.4 MiB/s | 40.4 KiB | 00m00s [329/479] Installing fftw-libs-single-0 100% | 282.7 MiB/s | 1.7 MiB | 00m00s [330/479] Installing bluez-libs-0:5.87- 100% | 109.0 MiB/s | 223.2 KiB | 00m00s [331/479] Installing pipewire-libs-0:1. 100% | 265.7 MiB/s | 9.8 MiB | 00m00s [332/479] Installing pipewire-jack-audi 100% | 25.5 MiB/s | 522.3 KiB | 00m00s [333/479] Installing pipewire-jack-audi 100% | 0.0 B/s | 312.0 B | 00m00s [334/479] Installing portaudio-0:19.7.0 100% | 251.8 MiB/s | 257.8 KiB | 00m00s [335/479] Installing libwayland-server- 100% | 97.1 MiB/s | 99.4 KiB | 00m00s [336/479] Installing wayland-devel-0:1. 100% | 33.9 MiB/s | 729.1 KiB | 00m00s [337/479] Installing hicolor-icon-theme 100% | 19.5 MiB/s | 179.7 KiB | 00m00s [338/479] Installing xz-devel-1:5.8.3-2 100% | 128.2 MiB/s | 262.6 KiB | 00m00s [339/479] Installing libxml2-devel-0:2. 100% | 127.2 MiB/s | 3.2 MiB | 00m00s [340/479] Installing libxkbcommon-devel 100% | 385.8 MiB/s | 395.1 KiB | 00m00s [341/479] Installing check-static-0:0.1 100% | 369.7 MiB/s | 378.6 KiB | 00m00s [342/479] Installing check-devel-0:0.15 100% | 123.8 MiB/s | 126.7 KiB | 00m00s [343/479] Installing c-ares-0:1.34.8-2. 100% | 24.3 MiB/s | 298.5 KiB | 00m00s >>> Running sysusers scriptlet: mosquitto-0:2.1.2-4.fc45.s390x >>> Finished sysusers scriptlet: mosquitto-0:2.1.2-4.fc45.s390x >>> Scriptlet output: >>> Creating group 'mosquitto' with GID 995. >>> Creating user 'mosquitto' (Mosquitto Broker) with UID 995 and GID 995. >>> [344/479] Installing mosquitto-0:2.1.2- 100% | 64.5 MiB/s | 1.8 MiB | 00m00s [345/479] Installing libX11-common-0:1. 100% | 148.2 MiB/s | 1.2 MiB | 00m00s [346/479] Installing libX11-0:1.8.13-2. 100% | 175.2 MiB/s | 1.4 MiB | 00m00s [347/479] Installing libX11-devel-0:1.8 100% | 73.7 MiB/s | 1.1 MiB | 00m00s [348/479] Installing libXext-0:1.3.6-6. 100% | 92.5 MiB/s | 94.7 KiB | 00m00s [349/479] Installing libXext-devel-0:1. 100% | 53.6 MiB/s | 109.8 KiB | 00m00s [350/479] Installing libXrender-0:0.9.1 100% | 45.8 MiB/s | 46.9 KiB | 00m00s [351/479] Installing cairo-0:1.18.4-9.f 100% | 254.2 MiB/s | 1.8 MiB | 00m00s [352/479] Installing libXrender-devel-0 100% | 0.0 B/s | 51.0 KiB | 00m00s [353/479] Installing libXi-0:1.8.3-2.fc 100% | 79.3 MiB/s | 81.2 KiB | 00m00s [354/479] Installing libXfixes-0:6.0.1- 100% | 0.0 B/s | 31.2 KiB | 00m00s [355/479] Installing libXfixes-devel-0: 100% | 0.0 B/s | 9.9 KiB | 00m00s [356/479] Installing cairo-gobject-0:1. 100% | 34.8 MiB/s | 35.6 KiB | 00m00s [357/479] Installing libXi-devel-0:1.8. 100% | 70.4 MiB/s | 144.3 KiB | 00m00s [358/479] Installing libXcursor-0:1.2.3 100% | 53.5 MiB/s | 54.8 KiB | 00m00s [359/479] Installing libXdamage-0:1.1.6 100% | 0.0 B/s | 40.8 KiB | 00m00s [360/479] Installing libXtst-0:1.2.5-5. 100% | 0.0 B/s | 34.2 KiB | 00m00s [361/479] Installing libXrandr-0:1.5.4- 100% | 0.0 B/s | 52.5 KiB | 00m00s [362/479] Installing libXft-0:2.3.8-11. 100% | 165.6 MiB/s | 169.5 KiB | 00m00s [363/479] Installing pango-0:1.58.2-1.f 100% | 45.9 MiB/s | 1.1 MiB | 00m00s [364/479] Installing libXinerama-0:1.1. 100% | 0.0 B/s | 15.7 KiB | 00m00s [365/479] Installing libXcomposite-0:0. 100% | 0.0 B/s | 41.7 KiB | 00m00s [366/479] Installing mesa-libEGL-0:26.2 100% | 187.1 MiB/s | 383.1 KiB | 00m00s [367/479] Installing libglvnd-egl-1:1.7 100% | 68.4 MiB/s | 70.0 KiB | 00m00s [368/479] Installing libglvnd-gles-1:1. 100% | 112.0 MiB/s | 114.7 KiB | 00m00s [369/479] Installing libXcomposite-deve 100% | 0.0 B/s | 10.1 KiB | 00m00s [370/479] Installing libXinerama-devel- 100% | 0.0 B/s | 8.4 KiB | 00m00s [371/479] Installing libXrandr-devel-0: 100% | 0.0 B/s | 24.1 KiB | 00m00s [372/479] Installing libXtst-devel-0:1. 100% | 0.0 B/s | 13.6 KiB | 00m00s [373/479] Installing libXdamage-devel-0 100% | 0.0 B/s | 3.1 KiB | 00m00s [374/479] Installing libXcursor-devel-0 100% | 29.8 MiB/s | 30.5 KiB | 00m00s [375/479] Installing harfbuzz-cairo-0:1 100% | 0.0 B/s | 44.6 KiB | 00m00s [376/479] Installing libXv-0:1.0.13-5.f 100% | 26.2 MiB/s | 26.8 KiB | 00m00s [377/479] Installing libvdpau-0:1.5-12. 100% | 0.0 B/s | 18.0 KiB | 00m00s [378/479] Installing libXxf86vm-0:1.1.6 100% | 0.0 B/s | 26.1 KiB | 00m00s [379/479] Installing mesa-libGL-0:26.2. 100% | 181.2 MiB/s | 371.0 KiB | 00m00s [380/479] Installing libglvnd-glx-1:1.7 100% | 249.5 MiB/s | 766.4 KiB | 00m00s [381/479] Installing libva-0:2.24.1-2.f 100% | 166.3 MiB/s | 340.7 KiB | 00m00s [382/479] Installing libavutil-free-0:9 100% | 210.7 MiB/s | 863.2 KiB | 00m00s [383/479] Installing libavutil-free-dev 100% | 184.7 MiB/s | 756.5 KiB | 00m00s [384/479] Installing libswresample-free 100% | 128.4 MiB/s | 131.5 KiB | 00m00s [385/479] Installing libswresample-free 100% | 0.0 B/s | 27.6 KiB | 00m00s [386/479] Installing libswscale-free-0: 100% | 284.8 MiB/s | 874.9 KiB | 00m00s [387/479] Installing libswscale-free-de 100% | 0.0 B/s | 33.9 KiB | 00m00s [388/479] Installing libchromaprint-0:1 100% | 73.2 MiB/s | 75.0 KiB | 00m00s [389/479] Installing libGLEW-0:2.2.0-13 100% | 247.8 MiB/s | 761.2 KiB | 00m00s [390/479] Installing freeglut-0:3.8.0-3 100% | 260.9 MiB/s | 534.3 KiB | 00m00s [391/479] Installing libcaca-0:0.99-0.8 100% | 287.4 MiB/s | 883.0 KiB | 00m00s [392/479] Installing xprop-0:1.2.8-6.fc 100% | 3.1 MiB/s | 59.6 KiB | 00m00s [393/479] Installing at-spi2-core-0:2.6 100% | 178.9 MiB/s | 1.6 MiB | 00m00s [394/479] Installing atk-0:2.61.1-1.fc4 100% | 267.5 MiB/s | 273.9 KiB | 00m00s [395/479] Installing at-spi2-atk-0:2.61 100% | 72.6 MiB/s | 297.2 KiB | 00m00s [396/479] Installing kernel-headers-0:7 100% | 174.7 MiB/s | 7.2 MiB | 00m00s [397/479] Installing glibc-devel-0:2.44 100% | 219.3 MiB/s | 3.7 MiB | 00m00s [398/479] Installing libxcrypt-devel-0: 100% | 32.5 MiB/s | 33.3 KiB | 00m00s [399/479] Installing lpcnetfreedv-0:0.5 100% | 494.2 MiB/s | 14.8 MiB | 00m00s [400/479] Installing codec2-0:1.2.0-10. 100% | 279.2 MiB/s | 1.4 MiB | 00m00s [401/479] Installing glycin-loaders-0:2 100% | 390.3 MiB/s | 17.2 MiB | 00m00s [402/479] Installing glycin-libs-0:2.2~ 100% | 292.2 MiB/s | 5.3 MiB | 00m00s [403/479] Installing gdk-pixbuf2-0:2.44 100% | 90.1 MiB/s | 2.6 MiB | 00m00s [404/479] Installing librsvg2-0:2.62.3- 100% | 305.6 MiB/s | 6.1 MiB | 00m00s [405/479] Installing libavcodec-free-0: 100% | 282.5 MiB/s | 10.2 MiB | 00m00s [406/479] Installing libheif-0:1.23.1-1 100% | 260.9 MiB/s | 2.3 MiB | 00m00s [407/479] Installing libavcodec-free-de 100% | 131.7 MiB/s | 269.7 KiB | 00m00s [408/479] Installing libavformat-free-0 100% | 284.4 MiB/s | 3.1 MiB | 00m00s [409/479] Installing libavformat-free-d 100% | 152.5 MiB/s | 156.1 KiB | 00m00s [410/479] Installing libavfilter-free-0 100% | 289.1 MiB/s | 4.9 MiB | 00m00s [411/479] Installing libavfilter-free-d 100% | 63.2 MiB/s | 64.7 KiB | 00m00s [412/479] Installing libavdevice-free-0 100% | 121.7 MiB/s | 249.3 KiB | 00m00s [413/479] Installing gtk-update-icon-ca 100% | 3.2 MiB/s | 62.9 KiB | 00m00s [414/479] Installing gtk3-0:3.24.52-6.f 100% | 252.3 MiB/s | 23.2 MiB | 00m00s [415/479] Installing libdecor-0:0.2.5-3 100% | 170.5 MiB/s | 174.6 KiB | 00m00s [416/479] Installing SDL3-0:3.4.14-1.fc 100% | 286.1 MiB/s | 3.4 MiB | 00m00s [417/479] Installing sdl2-compat-0:2.32 100% | 227.5 MiB/s | 466.0 KiB | 00m00s [418/479] Installing libglvnd-core-deve 100% | 0.0 B/s | 41.1 KiB | 00m00s [419/479] Installing libglvnd-devel-1:1 100% | 530.1 MiB/s | 2.1 MiB | 00m00s [420/479] Installing mesa-libGLU-devel- 100% | 0.0 B/s | 17.5 KiB | 00m00s [421/479] Installing libepoxy-devel-0:1 100% | 529.8 MiB/s | 1.6 MiB | 00m00s [422/479] Installing pcre2-utf32-0:10.4 100% | 209.5 MiB/s | 643.5 KiB | 00m00s [423/479] Installing pcre2-utf16-0:10.4 100% | 219.9 MiB/s | 675.6 KiB | 00m00s [424/479] Installing pcre2-devel-0:10.4 100% | 78.5 MiB/s | 2.1 MiB | 00m00s [425/479] Installing libsepol-devel-0:3 100% | 31.3 MiB/s | 128.2 KiB | 00m00s [426/479] Installing libselinux-devel-0 100% | 38.2 MiB/s | 156.6 KiB | 00m00s [427/479] Installing libblkid-devel-0:2 100% | 0.0 B/s | 46.2 KiB | 00m00s [428/479] Installing libmount-devel-0:2 100% | 0.0 B/s | 67.4 KiB | 00m00s [429/479] Installing libre-0:4.11.0-1.f 100% | 156.3 MiB/s | 960.0 KiB | 00m00s [430/479] Installing libstdc++-devel-0: 100% | 388.8 MiB/s | 40.8 MiB | 00m00s [431/479] Installing libubsan-0:16.2.1- 100% | 244.4 MiB/s | 500.6 KiB | 00m00s [432/479] Installing libatomic-0:16.2.1 100% | 37.4 MiB/s | 38.3 KiB | 00m00s [433/479] Installing libasan-0:16.2.1-2 100% | 267.2 MiB/s | 1.9 MiB | 00m00s [434/479] Installing gcc-0:16.2.1-2.fc4 100% | 302.6 MiB/s | 88.7 MiB | 00m00s [435/479] Installing rhash-0:1.4.5-5.fc 100% | 15.9 MiB/s | 359.3 KiB | 00m00s [436/479] Installing libuv-1:1.52.1-2.f 100% | 192.4 MiB/s | 591.0 KiB | 00m00s [437/479] Installing jsoncpp-0:1.9.6-3. 100% | 129.4 MiB/s | 265.0 KiB | 00m00s [438/479] Installing cmake-0:4.3.0-5.fc 100% | 244.8 MiB/s | 37.2 MiB | 00m00s [439/479] Installing cmake-data-0:4.3.0 100% | 104.9 MiB/s | 10.1 MiB | 00m00s [440/479] Installing sysprof-capture-de 100% | 181.9 MiB/s | 744.9 KiB | 00m00s [441/479] Installing glib2-devel-0:2.89 100% | 283.9 MiB/s | 15.9 MiB | 00m00s [442/479] Installing atk-devel-0:2.61.1 100% | 303.8 MiB/s | 6.1 MiB | 00m00s [443/479] Installing harfbuzz-devel-0:1 100% | 185.8 MiB/s | 6.5 MiB | 00m00s [444/479] Installing freetype-devel-0:2 100% | 214.9 MiB/s | 8.6 MiB | 00m00s [445/479] Installing fontconfig-devel-0 100% | 32.1 MiB/s | 164.2 KiB | 00m00s [446/479] Installing cairo-devel-0:1.18 100% | 381.8 MiB/s | 2.3 MiB | 00m00s [447/479] Installing cairo-gobject-deve 100% | 0.0 B/s | 7.6 KiB | 00m00s [448/479] Installing glycin-devel-0:2.2 100% | 148.8 MiB/s | 152.3 KiB | 00m00s [449/479] Installing gdk-pixbuf2-devel- 100% | 83.6 MiB/s | 2.3 MiB | 00m00s [450/479] Installing libXft-devel-0:2.3 100% | 40.6 MiB/s | 41.6 KiB | 00m00s [451/479] Installing pango-devel-0:1.58 100% | 386.1 MiB/s | 1.5 MiB | 00m00s [452/479] Installing libcloudproviders- 100% | 198.3 MiB/s | 1.4 MiB | 00m00s [453/479] Installing at-spi2-core-devel 100% | 287.0 MiB/s | 4.6 MiB | 00m00s [454/479] Installing at-spi2-atk-devel- 100% | 1.0 MiB/s | 2.1 KiB | 00m00s [455/479] Installing gtk3-devel-0:3.24. 100% | 340.1 MiB/s | 34.7 MiB | 00m00s [456/479] Installing gstreamer1-devel-0 100% | 435.8 MiB/s | 6.5 MiB | 00m00s [457/479] Installing pulseaudio-libs-de 100% | 348.2 MiB/s | 4.9 MiB | 00m00s [458/479] Installing gcc-c++-0:16.2.1-2 100% | 259.8 MiB/s | 34.8 MiB | 00m00s [459/479] Installing annobin-plugin-gcc 100% | 53.9 MiB/s | 717.4 KiB | 00m00s [460/479] Installing gcc-plugin-annobin 100% | 4.7 MiB/s | 58.4 KiB | 00m00s [461/479] Installing libre-devel-0:4.11 100% | 91.9 MiB/s | 282.4 KiB | 00m00s [462/479] Installing sdl2-compat-devel- 100% | 112.8 MiB/s | 3.3 MiB | 00m00s [463/479] Installing libavdevice-free-d 100% | 18.6 MiB/s | 19.0 KiB | 00m00s [464/479] Installing codec2-devel-0:1.2 100% | 57.2 MiB/s | 58.6 KiB | 00m00s [465/479] Installing mosquitto-devel-0: 100% | 126.5 MiB/s | 259.0 KiB | 00m00s [466/479] Installing portaudio-devel-0: 100% | 360.4 MiB/s | 2.5 MiB | 00m00s [467/479] Installing pipewire-jack-audi 100% | 93.2 MiB/s | 190.8 KiB | 00m00s [468/479] Installing pipewire-devel-0:1 100% | 126.3 MiB/s | 1.0 MiB | 00m00s [469/479] Installing python3-devel-0:3. 100% | 86.2 MiB/s | 2.5 MiB | 00m00s [470/479] Installing libsndfile-devel-0 100% | 105.8 MiB/s | 216.8 KiB | 00m00s [471/479] Installing spandsp-devel-0:0. 100% | 169.1 MiB/s | 865.8 KiB | 00m00s [472/479] Installing authselect-0:1.8.0 100% | 11.1 MiB/s | 238.1 KiB | 00m00s [473/479] Installing pam-0:1.7.2-4.fc45 100% | 51.4 MiB/s | 1.6 MiB | 00m00s [474/479] Installing libaom-devel-0:3.1 100% | 110.9 MiB/s | 227.1 KiB | 00m00s [475/479] Installing libvpx-devel-0:1.1 100% | 387.5 MiB/s | 3.9 MiB | 00m00s [476/479] Installing libv4l-devel-0:1.3 100% | 234.4 MiB/s | 720.1 KiB | 00m00s [477/479] Installing desktop-file-utils 100% | 12.6 MiB/s | 245.2 KiB | 00m00s [478/479] Installing fdk-aac-free-devel 100% | 554.7 MiB/s | 1.1 MiB | 00m00s [479/479] Installing systemd-rpm-macros 100% | 10.3 KiB/s | 8.9 KiB | 00m01s Warning: skipped OpenPGP checks for 17 packages from repository: copr_base Complete! Building target platforms: s390x Building for target s390x setting SOURCE_DATE_EPOCH=1788134400 Wrote: /builddir/build/SRPMS/baresip-4.11.0-1.fc46.src.rpm Updating and loading repositories: fedora 100% | 3.4 KiB/s | 4.1 KiB | 00m01s Copr repository 100% | 1.5 KiB/s | 1.5 KiB | 00m01s Repositories loaded. Package "glib2-devel-2.89.4-1.fc46.s390x" is already installed. Package "sdl2-compat-devel-2.32.70-2.fc45.s390x" is already installed. Package "alsa-lib-devel-1.2.16.1-2.fc45.s390x" is already installed. Package "cjson-devel-1.7.19-2.fc45.s390x" is already installed. Package "cmake-4.3.0-5.fc45.s390x" is already installed. Package "codec2-devel-1.2.0-10.fc45.s390x" is already installed. Package "desktop-file-utils-0.28-8.fc45.s390x" is already installed. Package "fdk-aac-free-devel-2.0.3-3.fc45.s390x" is already installed. Package "gcc-16.2.1-2.fc46.s390x" is already installed. Package "gcc-c++-16.2.1-2.fc46.s390x" is already installed. Package "libX11-devel-1.8.13-2.fc45.s390x" is already installed. Package "libXext-devel-1.3.6-6.fc45.s390x" is already installed. Package "libaom-devel-3.13.3-3.fc45.s390x" is already installed. Package "libpng-devel-2:1.6.58-2.fc45.s390x" is already installed. Package "libre-devel-4.11.0-1.fc46.s390x" is already installed. Package "libsndfile-devel-1.2.2-12.fc45.s390x" is already installed. Package "libv4l-devel-1.32.0-4.fc45.s390x" is already installed. Package "libvpx-devel-1.17.0-1.fc46.s390x" is already installed. Package "mosquitto-devel-2.1.2-4.fc45.s390x" is already installed. Package "openssl-devel-1:4.0.2-1.fc46.s390x" is already installed. Package "opus-devel-1.6-3.fc45.s390x" is already installed. Package "glib2-devel-2.89.4-1.fc46.s390x" is already installed. Package "glib2-devel-2.89.4-1.fc46.s390x" is already installed. Package "glib2-devel-2.89.4-1.fc46.s390x" is already installed. Package "gstreamer1-devel-1.28.6-2.fc46.s390x" is already installed. Package "gtk3-devel-3.24.52-6.fc45.s390x" is already installed. Package "pipewire-jack-audio-connection-kit-devel-1.6.8-3.fc45.s390x" is already installed. Package "libavcodec-free-devel-9.0.1-4.fc46.s390x" is already installed. Package "libavdevice-free-devel-9.0.1-4.fc46.s390x" is already installed. Package "libavfilter-free-devel-9.0.1-4.fc46.s390x" is already installed. Package "libavformat-free-devel-9.0.1-4.fc46.s390x" is already installed. Package "libavutil-free-devel-9.0.1-4.fc46.s390x" is already installed. Package "pipewire-devel-1.6.8-3.fc45.s390x" is already installed. Package "pulseaudio-libs-devel-17.0-11.fc45.s390x" is already installed. Package "libswresample-free-devel-9.0.1-4.fc46.s390x" is already installed. Package "libswscale-free-devel-9.0.1-4.fc46.s390x" is already installed. Package "libvpx-devel-1.17.0-1.fc46.s390x" is already installed. Package "portaudio-devel-19.7.0-4.fc45.s390x" is already installed. Package "python3-devel-3.15.0~rc1-1.fc45.s390x" is already installed. Package "spandsp-devel-0.0.6-23.fc45.s390x" is already installed. Nothing to do. Finish: build setup for baresip-4.11.0-1.fc46.src.rpm Start: rpmbuild baresip-4.11.0-1.fc46.src.rpm Building target platforms: s390x Building for target s390x setting SOURCE_DATE_EPOCH=1788134400 Executing(mkbuilddir): /bin/sh -e /var/tmp/rpm-tmp.3pkMeX Creating(rpmbuild.env): /builddir/build/BUILD/baresip-4.11.0-build/rpmbuild.env RPM_SOURCE_DIR="/builddir/build/SOURCES" RPM_BUILD_DIR="/builddir/build/BUILD/baresip-4.11.0-build" RPM_OPT_FLAGS="-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Wno-complain-wrong-lang -Werror=format-security -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection" RPM_LD_FLAGS="-Wl,-z,relro -Wl,--as-needed -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -specs=/usr/lib/rpm/redhat/redhat-hardened-ld-errors -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -Wl,--build-id=sha1 -specs=/usr/lib/rpm/redhat/redhat-package-notes " RPM_ARCH="s390x" RPM_OS="linux" RPM_BUILD_NCPUS="2" RPM_SPECPARTS_DIR="/builddir/build/BUILD/baresip-4.11.0-build/SPECPARTS" export RPM_SOURCE_DIR RPM_BUILD_DIR RPM_OPT_FLAGS RPM_ARCH RPM_OS RPM_BUILD_NCPUS RPM_SPECPARTS_DIR RPM_LD_FLAGS RPM_DOC_DIR="/usr/share/doc" export RPM_DOC_DIR RPM_PACKAGE_NAME="baresip" RPM_PACKAGE_VERSION="4.11.0" RPM_PACKAGE_RELEASE="1.fc46" export RPM_PACKAGE_NAME RPM_PACKAGE_VERSION RPM_PACKAGE_RELEASE LANG=C.UTF-8 export LANG unset CDPATH DISPLAY ||: unset DEBUGINFOD_URLS ||: unset RPM_CONFIG_DIR ||: RPM_BUILD_ROOT="/builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT" export RPM_BUILD_ROOT PKG_CONFIG_PATH="${PKG_CONFIG_PATH}:/usr/lib64/pkgconfig:/usr/share/pkgconfig" export PKG_CONFIG_PATH CONFIG_SITE=${CONFIG_SITE:-NONE} export CONFIG_SITE Executing(%prep): /bin/sh -e /var/tmp/rpm-tmp.Vv4wV2 + umask 022 + cd /builddir/build/BUILD/baresip-4.11.0-build + cd /builddir/build/BUILD/baresip-4.11.0-build + rm -rf baresip-4.11.0 + /usr/bin/rpmuncompress -x /builddir/build/SOURCES/baresip-4.11.0.tar.gz + STATUS=0 + '[' 0 -ne 0 ']' + cd baresip-4.11.0 + /usr/bin/chmod -Rf a+rX,u+w,g-w,o-w . + RPM_EC=0 ++ jobs -p + exit 0 Executing(%build): /bin/sh -e /var/tmp/rpm-tmp.ItJY4D + umask 022 + cd /builddir/build/BUILD/baresip-4.11.0-build + CFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection ' + export CFLAGS + CXXFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection ' + export CXXFLAGS + FFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection -I/usr/lib64/gfortran/modules ' + export FFLAGS + FCFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection -I/usr/lib64/gfortran/modules ' + export FCFLAGS + VALAFLAGS=-g + export VALAFLAGS + RUSTFLAGS='-Copt-level=3 -Cdebuginfo=2 -Ccodegen-units=1 -Cstrip=none -Clink-arg=-specs=/usr/lib/rpm/redhat/redhat-package-notes --cap-lints=warn' + export RUSTFLAGS + LDFLAGS='-Wl,-z,relro -Wl,--as-needed -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -specs=/usr/lib/rpm/redhat/redhat-hardened-ld-errors -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -Wl,--build-id=sha1 -specs=/usr/lib/rpm/redhat/redhat-package-notes ' + export LDFLAGS + LT_SYS_LIBRARY_PATH=/usr/lib64: + export LT_SYS_LIBRARY_PATH + CC=gcc + export CC + CXX=g++ + export CXX + cd baresip-4.11.0 + CFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection ' + export CFLAGS + CXXFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection ' + export CXXFLAGS + FFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection -I/usr/lib64/gfortran/modules ' + export FFLAGS + FCFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection -I/usr/lib64/gfortran/modules ' + export FCFLAGS + VALAFLAGS=-g + export VALAFLAGS + RUSTFLAGS='-Copt-level=3 -Cdebuginfo=2 -Ccodegen-units=1 -Cstrip=none -Clink-arg=-specs=/usr/lib/rpm/redhat/redhat-package-notes --cap-lints=warn' + export RUSTFLAGS + LDFLAGS='-Wl,-z,relro -Wl,--as-needed -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -specs=/usr/lib/rpm/redhat/redhat-hardened-ld-errors -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -Wl,--build-id=sha1 -specs=/usr/lib/rpm/redhat/redhat-package-notes ' + export LDFLAGS + LT_SYS_LIBRARY_PATH=/usr/lib64: + export LT_SYS_LIBRARY_PATH + CC=gcc + export CC + CXX=g++ + export CXX + /usr/bin/cmake -S . -B redhat-linux-build -G Ninja -DCMAKE_C_FLAGS_RELEASE:STRING=-DNDEBUG -DCMAKE_CXX_FLAGS_RELEASE:STRING=-DNDEBUG -DCMAKE_Fortran_FLAGS_RELEASE:STRING=-DNDEBUG -DCMAKE_VERBOSE_MAKEFILE:BOOL=ON -DCMAKE_INSTALL_DO_STRIP:BOOL=OFF -DCMAKE_INSTALL_PREFIX:PATH=/usr -DCMAKE_INSTALL_FULL_SBINDIR:PATH=/usr/bin -DCMAKE_INSTALL_SBINDIR:PATH=bin -DINCLUDE_INSTALL_DIR:PATH=/usr/include -DLIB_INSTALL_DIR:PATH=/usr/lib64 -DSYSCONF_INSTALL_DIR:PATH=/etc -DSHARE_INSTALL_PREFIX:PATH=/usr/share -DLIB_SUFFIX=64 -DBUILD_SHARED_LIBS:BOOL=ON -DDEFAULT_CAFILE:PATH=/etc/pki/tls/certs/ca-bundle.crt -DDEFAULT_CAPATH:PATH=/etc/pki/tls/certs -DDEFAULT_AUDIO_DEVICE:STRING=pipewire -- The C compiler identification is GNU 16.2.1 -- The CXX compiler identification is GNU 16.2.1 -- Detecting C compiler ABI info -- Detecting C compiler ABI info - done -- Check for working C compiler: /usr/bin/gcc - skipped -- Detecting C compile features -- Detecting C compile features - done -- Detecting CXX compiler ABI info -- Detecting CXX compiler ABI info - done -- Check for working CXX compiler: /usr/bin/g++ - skipped -- Detecting CXX compile features -- Detecting CXX compile features - done -- Found RE: /usr/lib64/libre.so -- Looking for backtrace -- Looking for backtrace - found -- backtrace facility detected in default set of libraries -- Found Backtrace: /usr/include -- Performing Test CMAKE_HAVE_LIBC_PTHREAD -- Performing Test CMAKE_HAVE_LIBC_PTHREAD - Success -- Found Threads: TRUE -- Found ZLIB: /usr/lib64/libz.so (found version "1.3.1") -- Found OpenSSL: /usr/lib64/libcrypto.so (found suitable version "4.0.2", minimum required is "1.1.1") -- Setting build type to 'Debug' as none was specified. -- Looking for LIBRESSL_VERSION_NUMBER -- Looking for LIBRESSL_VERSION_NUMBER - not found -- Looking for arc4random -- Looking for arc4random - found -- Looking for syslog.h -- Looking for syslog.h - found -- Looking for getopt.h -- Looking for getopt.h - found -- Looking for unistd.h -- Looking for unistd.h - found -- Looking for res_ninit -- Looking for res_ninit - found -- Looking for thrd_create -- Looking for thrd_create - found -- Looking for threads.h -- Looking for threads.h - found -- Looking for accept4 -- Looking for accept4 - found -- Looking for epoll_create -- Looking for epoll_create - found -- Looking for kqueue -- Looking for kqueue - not found -- Looking for sys/prctl.h -- Looking for sys/prctl.h - found -- Looking for ifaddrs.h -- Looking for ifaddrs.h - found -- Performing Test atomic_test -- Performing Test atomic_test - Success -- Checking for one of the modules 'fdk-aac' -- Found AAC: /usr/lib64/libfdk-aac.so -- Could NOT find AAUDIO (missing: AAUDIO_LIBRARY AAUDIO_INCLUDE_DIR) -- Found ALSA: /usr/lib64/libasound.so (found version "1.2.16.1") -- Could NOT find AMR (missing: AMR_FOUND) -- Found PkgConfig: /usr/bin/pkg-config (found version "2.5.1") -- Checking for one of the modules 'aptx' -- Could NOT find APTX (missing: APTX_LIBRARY APTX_INCLUDE_DIR) -- Found AOM: /usr/lib64/libaom.so (found version "3.13.3") -- Found FFMPEG: /usr/lib64/libavcodec.so (found version "#define LIBAVCODEC_VERSION_MINOR 1;#define LIBAVCODEC_VERSION_MICRO 101.1.101") found components: avcodec avfilter avformat swscale avdevice avutil -- Found FFMPEG: /usr/lib64/libavcodec.so (found version "#define LIBAVCODEC_VERSION_MINOR 1;#define LIBAVCODEC_VERSION_MICRO 101.1.101") found components: avcodec avfilter avformat swscale swresample avdevice avutil -- Checking for one of the modules 'codec2' -- Found CODEC2: /usr/lib64/libcodec2.so -- Checking for module 'gio-unix-2.0' -- Found gio-unix-2.0, version 2.89.4 -- Found GIO: /usr/lib64/libglib-2.0.so -- Could NOT find LIBG722 (missing: LIBG722_LIBRARY LIBG722_INCLUDE_DIR) -- Found SPANDSP: /usr/lib64/libspandsp.so -- Checking for one of the modules 'g7221' -- Could NOT find G7221 (missing: G7221_LIBRARY G7221_INCLUDE_DIR) -- Found GST: TRUE -- Checking for module 'gtk+-3.0' -- Found gtk+-3.0, version 3.24.52 -- Found GTK3: gtk-3;gdk-3;pangocairo-1.0;cairo-gobject;gdk_pixbuf-2.0;atk-1.0;pango-1.0;cairo;harfbuzz;gio-2.0;gobject-2.0;glib-2.0;z -- Could NOT find GZRTP (missing: GZRTP_LIBRARY GZRTP_INCLUDE_DIR) -- Checking for one of the modules 'jack' -- Found JACK: /usr/lib64/pipewire-0.3/jack/libjack.so -- Found MOSQUITTO: /usr/lib64/libmosquitto.so -- libmosquitto include dir: /usr/include -- libmosquitto: /usr/lib64/libmosquitto.so -- Looking for linux/netlink.h -- Looking for linux/netlink.h - found -- Could NOT find OPENSLES (missing: OPENSLES_LIBRARY OPENSLES_INCLUDE_DIR) -- Checking for one of the modules 'opus' -- Found OPUS: /usr/lib64/libopus.so -- Checking for one of the modules 'libpipewire-0.3' -- Found PIPEWIRE: /usr/include/pipewire-0.3;/usr/include/spa-0.2 -- Checking for one of the modules 'portaudio' -- Found PORTAUDIO: /lib64/libportaudio.so -- Checking for one of the modules 'pulse' -- Found PULSE: /lib64/libpulse.so -- Checking for one of the modules 'sdl2' -- Found SDL: /usr/lib64/libSDL2.so -- Found PNG: /lib64/libpng.so -- Checking for module 'sndfile' -- Found sndfile, version 1.2.2 -- Found SNDFILE: /usr/include -- Checking for module 'libv4l2' -- Found libv4l2, version 1.32.0 -- Found V4L2: /usr/lib64/libv4l2.so -- Found VPX: /lib64/libvpx.so -- Could NOT find WEBRTC_AEC (missing: WEBRTC_AEC_LIBRARY WEBRTC_AEC_INCLUDE_DIR) -- Found X11: /usr/include -- Looking for XOpenDisplay in /usr/lib64/libX11.so;/usr/lib64/libXext.so -- Looking for XOpenDisplay in /usr/lib64/libX11.so;/usr/lib64/libXext.so - found -- Looking for gethostbyname -- Looking for gethostbyname - found -- Looking for connect -- Looking for connect - found -- Looking for remove -- Looking for remove - found -- Looking for shmat -- Looking for shmat - found MODULES_DETECTED=aac;account;alsa;aubridge;auconv;aufile;augain;auresamp;ausine;av1;avcodec;avfilter;avformat;codec2;cons;contact;ctrl_dbus;ctrl_tcp;debug_cmd;dtls_srtp;echo;evdev;fakevideo;g711;g722;gst;gtk;httpd;httpreq;ice;in_band_dtmf;jack;l16;menu;mixausrc;mixminus;mqtt;mwi;natpmp;netroam;opus;opus_multistream;pcp;pipewire;plc;portaudio;presence;pulse;rtcpsummary;sdl;selfview;serreg;snapshot;sndfile;srtp;stdio;stun;swscale;syslog;turn;uuid;v4l2;vidbridge;vidinfo;vp8;vp9;vumeter;x11 -- Configuring done (7.1s) -- Generating done (0.1s) CMake Warning: Manually-specified variables were not used by the project: CMAKE_CXX_FLAGS_RELEASE CMAKE_C_FLAGS_RELEASE CMAKE_Fortran_FLAGS_RELEASE CMAKE_INSTALL_DO_STRIP CMAKE_VERBOSE_MAKEFILE INCLUDE_INSTALL_DIR LIB_INSTALL_DIR LIB_SUFFIX SHARE_INSTALL_PREFIX SYSCONF_INSTALL_DIR -- Build files have been written to: /builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/redhat-linux-build + /usr/bin/cmake --build redhat-linux-build -j2 --verbose Change Dir: '/builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/redhat-linux-build' Run Build Command(s): /usr/bin/ninja-build -v -j 2 [1/293] /usr/bin/gcc -DARCH=\"s390x\" -DDEFAULT_AUDIO_DEVICE=\"pipewire\" -DDEFAULT_CAFILE=\"/etc/pki/tls/certs/ca-bundle.crt\" -DDEFAULT_CAPATH=\"/etc/pki/tls/certs\" -DHAVE_ACCEPT4 -DHAVE_ARC4RANDOM -DHAVE_ATOMIC -DHAVE_EPOLL -DHAVE_EXECINFO -DHAVE_FORK -DHAVE_GETIFADDRS -DHAVE_GETOPT -DHAVE_PRCTL -DHAVE_PTHREAD -DHAVE_PWD_H -DHAVE_RESOLV -DHAVE_ROUTE_LIST -DHAVE_SELECT -DHAVE_SELECT_H -DHAVE_SETRLIMIT -DHAVE_SIGNAL -DHAVE_STRERROR_R -DHAVE_STRINGS_H -DHAVE_SYSLOG -DHAVE_SYS_TIME_H -DHAVE_THREADS -DHAVE_TLS1_3_POST_HANDSHAKE_AUTH -DHAVE_UNAME -DHAVE_UNISTD_H -DHAVE_UNIXSOCK=1 -DLINUX -DMOD_PATH=\"/usr/lib64/baresip/modules\" -DOS=\"Linux\" -DSHARE_PATH=\"/usr/share/baresip\" -DUSE_DTLS -DUSE_OPENSSL -DUSE_OPENSSL_AES -DUSE_OPENSSL_HMAC -DUSE_TLS -DUSE_ZLIB -DVERSION=\"4.11.0\" -DVER_MAJOR=4 -DVER_MINOR=11 -DVER_PATCH=0 -D_GNU_SOURCE -Dbaresip_EXPORTS -I/builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/include -I/builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/src -I/usr/include/re -O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection -g -std=gnu11 -fPIC -Wall -Wextra -pedantic -Wcast-align -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wnested-externs -Wno-strict-aliasing -Wold-style-definition -Wshadow -Waggregate-return -Wstrict-prototypes -Wuninitialized -Wvla -MD -MT CMakeFiles/baresip.dir/src/aucodec.c.o -MF CMakeFiles/baresip.dir/src/aucodec.c.o.d -o CMakeFiles/baresip.dir/src/aucodec.c.o -c /builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/src/aucodec.c [2/293] /usr/bin/gcc -DARCH=\"s390x\" -DDEFAULT_AUDIO_DEVICE=\"pipewire\" -DDEFAULT_CAFILE=\"/etc/pki/tls/certs/ca-bundle.crt\" -DDEFAULT_CAPATH=\"/etc/pki/tls/certs\" -DHAVE_ACCEPT4 -DHAVE_ARC4RANDOM -DHAVE_ATOMIC -DHAVE_EPOLL -DHAVE_EXECINFO -DHAVE_FORK -DHAVE_GETIFADDRS -DHAVE_GETOPT -DHAVE_PRCTL -DHAVE_PTHREAD -DHAVE_PWD_H -DHAVE_RESOLV -DHAVE_ROUTE_LIST -DHAVE_SELECT -DHAVE_SELECT_H -DHAVE_SETRLIMIT -DHAVE_SIGNAL -DHAVE_STRERROR_R -DHAVE_STRINGS_H -DHAVE_SYSLOG -DHAVE_SYS_TIME_H -DHAVE_THREADS -DHAVE_TLS1_3_POST_HANDSHAKE_AUTH -DHAVE_UNAME -DHAVE_UNISTD_H -DHAVE_UNIXSOCK=1 -DLINUX -DMOD_PATH=\"/usr/lib64/baresip/modules\" -DOS=\"Linux\" -DSHARE_PATH=\"/usr/share/baresip\" -DUSE_DTLS -DUSE_OPENSSL -DUSE_OPENSSL_AES -DUSE_OPENSSL_HMAC -DUSE_TLS -DUSE_ZLIB -DVERSION=\"4.11.0\" -DVER_MAJOR=4 -DVER_MINOR=11 -DVER_PATCH=0 -D_GNU_SOURCE -Dbaresip_EXPORTS -I/builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/include -I/builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/src -I/usr/include/re -O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection -g -std=gnu11 -fPIC -Wall -Wextra -pedantic -Wcast-align -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wnested-externs -Wno-strict-aliasing -Wold-style-definition -Wshadow -Waggregate-return -Wstrict-prototypes -Wuninitialized -Wvla -MD -MT CMakeFiles/baresip.dir/src/account.c.o -MF CMakeFiles/baresip.dir/src/account.c.o.d -o CMakeFiles/baresip.dir/src/account.c.o -c /builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/src/account.c [3/293] /usr/bin/gcc -DARCH=\"s390x\" -DDEFAULT_AUDIO_DEVICE=\"pipewire\" -DDEFAULT_CAFILE=\"/etc/pki/tls/certs/ca-bundle.crt\" -DDEFAULT_CAPATH=\"/etc/pki/tls/certs\" -DHAVE_ACCEPT4 -DHAVE_ARC4RANDOM -DHAVE_ATOMIC -DHAVE_EPOLL -DHAVE_EXECINFO -DHAVE_FORK -DHAVE_GETIFADDRS -DHAVE_GETOPT -DHAVE_PRCTL -DHAVE_PTHREAD -DHAVE_PWD_H -DHAVE_RESOLV -DHAVE_ROUTE_LIST -DHAVE_SELECT -DHAVE_SELECT_H -DHAVE_SETRLIMIT -DHAVE_SIGNAL -DHAVE_STRERROR_R -DHAVE_STRINGS_H -DHAVE_SYSLOG -DHAVE_SYS_TIME_H -DHAVE_THREADS -DHAVE_TLS1_3_POST_HANDSHAKE_AUTH -DHAVE_UNAME -DHAVE_UNISTD_H -DHAVE_UNIXSOCK=1 -DLINUX -DMOD_PATH=\"/usr/lib64/baresip/modules\" -DOS=\"Linux\" -DSHARE_PATH=\"/usr/share/baresip\" -DUSE_DTLS -DUSE_OPENSSL -DUSE_OPENSSL_AES -DUSE_OPENSSL_HMAC -DUSE_TLS -DUSE_ZLIB -DVERSION=\"4.11.0\" -DVER_MAJOR=4 -DVER_MINOR=11 -DVER_PATCH=0 -D_GNU_SOURCE -Dbaresip_EXPORTS -I/builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/include -I/builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/src -I/usr/include/re -O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection -g -std=gnu11 -fPIC -Wall -Wextra -pedantic -Wcast-align -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wnested-externs -Wno-strict-aliasing -Wold-style-definition -Wshadow -Waggregate-return -Wstrict-prototypes -Wuninitialized -Wvla -MD -MT CMakeFiles/baresip.dir/src/aufilt.c.o -MF CMakeFiles/baresip.dir/src/aufilt.c.o.d -o CMakeFiles/baresip.dir/src/aufilt.c.o -c /builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/src/aufilt.c [4/293] /usr/bin/gcc -DARCH=\"s390x\" -DDEFAULT_AUDIO_DEVICE=\"pipewire\" -DDEFAULT_CAFILE=\"/etc/pki/tls/certs/ca-bundle.crt\" 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-Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection -g -std=gnu11 -fPIC -Wall -Wextra -pedantic -Wcast-align -Wbad-function-cast -Wmissing-declarations -Wmissing-prototypes -Wnested-externs -Wno-strict-aliasing -Wold-style-definition -Wshadow -Waggregate-return -Wstrict-prototypes -Wuninitialized -Wvla -MD -MT CMakeFiles/baresip.dir/src/audio.c.o -MF CMakeFiles/baresip.dir/src/audio.c.o.d -o CMakeFiles/baresip.dir/src/audio.c.o -c /builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/src/audio.c [5/293] /usr/bin/gcc -DARCH=\"s390x\" -DDEFAULT_AUDIO_DEVICE=\"pipewire\" -DDEFAULT_CAFILE=\"/etc/pki/tls/certs/ca-bundle.crt\" -DDEFAULT_CAPATH=\"/etc/pki/tls/certs\" -DHAVE_ACCEPT4 -DHAVE_ARC4RANDOM -DHAVE_ATOMIC -DHAVE_EPOLL -DHAVE_EXECINFO -DHAVE_FORK -DHAVE_GETIFADDRS 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-pipe -Wall -Werror=format-security -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection -g -Wl,-z,relro -Wl,--as-needed -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -specs=/usr/lib/rpm/redhat/redhat-hardened-ld-errors -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -Wl,--build-id=sha1 -specs=/usr/lib/rpm/redhat/redhat-package-notes -Wl,--export-dynamic -rdynamic -Wl,--dependency-file=webrtc/CMakeFiles/baresip-webrtc.dir/link.d webrtc/CMakeFiles/baresip-webrtc.dir/src/demo.c.o webrtc/CMakeFiles/baresip-webrtc.dir/src/main.c.o webrtc/CMakeFiles/baresip-webrtc.dir/src/sess.c.o -o webrtc/baresip-webrtc -Wl,-rpath,/builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/redhat-linux-build libbaresip.so.28.11.0 /usr/lib64/libre.so && : [293/293] : && /usr/bin/gcc -O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection -g -Wl,-z,relro -Wl,--as-needed -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -specs=/usr/lib/rpm/redhat/redhat-hardened-ld-errors -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -Wl,--build-id=sha1 -specs=/usr/lib/rpm/redhat/redhat-package-notes -Wl,--export-dynamic -rdynamic -Wl,--dependency-file=test/CMakeFiles/selftest.dir/link.d test/CMakeFiles/selftest.dir/account.c.o test/CMakeFiles/selftest.dir/ausrc.c.o test/CMakeFiles/selftest.dir/bevent.c.o test/CMakeFiles/selftest.dir/call.c.o test/CMakeFiles/selftest.dir/call_cancelrule.c.o test/CMakeFiles/selftest.dir/call_fixture.c.o test/CMakeFiles/selftest.dir/cmd.c.o test/CMakeFiles/selftest.dir/contact.c.o test/CMakeFiles/selftest.dir/cparam.c.o test/CMakeFiles/selftest.dir/dial_number.c.o test/CMakeFiles/selftest.dir/jbuf.c.o test/CMakeFiles/selftest.dir/jbuf_gnack.c.o test/CMakeFiles/selftest.dir/message.c.o test/CMakeFiles/selftest.dir/net.c.o test/CMakeFiles/selftest.dir/peerconn.c.o test/CMakeFiles/selftest.dir/play.c.o test/CMakeFiles/selftest.dir/stunuri.c.o test/CMakeFiles/selftest.dir/ua.c.o test/CMakeFiles/selftest.dir/video.c.o test/CMakeFiles/selftest.dir/mock/dnssrv.c.o test/CMakeFiles/selftest.dir/sip/aor.c.o test/CMakeFiles/selftest.dir/sip/auth.c.o test/CMakeFiles/selftest.dir/sip/domain.c.o test/CMakeFiles/selftest.dir/sip/location.c.o test/CMakeFiles/selftest.dir/sip/sipsrv.c.o test/CMakeFiles/selftest.dir/sip/user.c.o test/CMakeFiles/selftest.dir/mock/cert.c.o test/CMakeFiles/selftest.dir/mock/mock_aucodec.c.o test/CMakeFiles/selftest.dir/mock/mock_auplay.c.o test/CMakeFiles/selftest.dir/mock/mock_mnat.c.o test/CMakeFiles/selftest.dir/mock/mock_vidcodec.c.o test/CMakeFiles/selftest.dir/mock/mock_vidisp.c.o test/CMakeFiles/selftest.dir/test.c.o test/CMakeFiles/selftest.dir/main.c.o -o test/selftest -Wl,-rpath,/builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/redhat-linux-build libbaresip.so.28.11.0 /usr/lib64/libre.so && : + RPM_EC=0 ++ jobs -p + exit 0 Executing(%install): /bin/sh -e /var/tmp/rpm-tmp.AFDmFu + umask 022 + cd /builddir/build/BUILD/baresip-4.11.0-build + '[' /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT '!=' / ']' + rm -rf /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT ++ dirname /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT + mkdir -p /builddir/build/BUILD/baresip-4.11.0-build + mkdir /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT + CFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection ' + export CFLAGS + CXXFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection ' + export CXXFLAGS + FFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection -I/usr/lib64/gfortran/modules ' + export FFLAGS + FCFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection -I/usr/lib64/gfortran/modules ' + export FCFLAGS + VALAFLAGS=-g + export VALAFLAGS + RUSTFLAGS='-Copt-level=3 -Cdebuginfo=2 -Ccodegen-units=1 -Cstrip=none -Clink-arg=-specs=/usr/lib/rpm/redhat/redhat-package-notes --cap-lints=warn' + export RUSTFLAGS + LDFLAGS='-Wl,-z,relro -Wl,--as-needed -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -specs=/usr/lib/rpm/redhat/redhat-hardened-ld-errors -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -Wl,--build-id=sha1 -specs=/usr/lib/rpm/redhat/redhat-package-notes ' + export LDFLAGS + LT_SYS_LIBRARY_PATH=/usr/lib64: + export LT_SYS_LIBRARY_PATH + CC=gcc + export CC + CXX=g++ + export CXX + cd baresip-4.11.0 + DESTDIR=/builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT + /usr/bin/cmake --install redhat-linux-build -- Install configuration: "Debug" -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/pkgconfig/libbaresip.pc -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/bin/baresip -- Set non-toolchain portion of runtime path of "/builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/bin/baresip" to "" -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/libbaresip.so.28.11.0 -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/libbaresip.so.28 -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/include/baresip.h -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/aac.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/account.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/alsa.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/aubridge.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/auconv.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/aufile.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/augain.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/auresamp.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/ausine.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/av1.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/avcodec.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/avfilter.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/avformat.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/codec2.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/cons.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/contact.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/ctrl_dbus.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/ctrl_tcp.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/debug_cmd.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/dtls_srtp.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/echo.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/evdev.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/fakevideo.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/g711.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/g722.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/gst.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/gtk.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/httpd.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/httpreq.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/ice.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/in_band_dtmf.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/jack.so -- Set non-toolchain portion of runtime path of "/builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/jack.so" to "" -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/l16.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/menu.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/mixausrc.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/mixminus.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/mqtt.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/mwi.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/natpmp.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/netroam.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/opus.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/opus_multistream.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/pcp.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/pipewire.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/plc.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/portaudio.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/presence.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/pulse.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/rtcpsummary.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/sdl.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/selfview.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/serreg.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/snapshot.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/sndfile.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/srtp.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/stdio.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/stun.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/swscale.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/syslog.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/turn.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/uuid.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/v4l2.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/vidbridge.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/vidinfo.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/vp8.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/vp9.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/vumeter.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/baresip/modules/x11.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/lib64/libbaresip.so -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/autoanswer.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/busy.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/callwaiting.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/error.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/logo.png -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/message.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/notfound.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/ring.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/ringback.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/sound0.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/sound1.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/sound2.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/sound3.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/sound4.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/sound5.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/sound6.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/sound7.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/sound8.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/sound9.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/sounda.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/soundb.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/soundc.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/soundd.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/soundroute.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/baresip/soundstar.wav -- Installing: /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/applications/com.github.baresip.desktop + install -p -m 0755 tools/fritzbox2baresip /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/bin/fritzbox2baresip + /usr/bin/find-debuginfo -j2 --strict-build-id -m -i --build-id-seed 4.11.0-1.fc46 --unique-debug-suffix -4.11.0-1.fc46.s390x --unique-debug-src-base baresip-4.11.0-1.fc46.s390x --run-dwz --dwz-low-mem-die-limit 10000000 --dwz-max-die-limit 50000000 --remove-section .gnu.build.attributes -S debugsourcefiles.list /builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0 find-debuginfo: starting Extracting debug info from 70 files DWARF-compressing 70 files sepdebugcrcfix: Updated 70 CRC32s, 0 CRC32s did match. Creating .debug symlinks for symlinks to ELF files Copying sources found by 'debugedit -l' to /usr/src/debug/baresip-4.11.0-1.fc46.s390x find-debuginfo: done + /usr/lib/rpm/check-buildroot + /usr/lib/rpm/redhat/brp-ldconfig + COMPRESS='gzip -9 -n' + COMPRESS_EXT=.gz + /usr/lib/rpm/brp-compress + /usr/lib/rpm/redhat/brp-strip-lto /usr/bin/strip + /usr/lib/rpm/check-rpaths + /usr/lib/rpm/redhat/brp-mangle-shebangs + /usr/lib/rpm/brp-remove-la-files + /usr/lib/rpm/redhat/brp-python-rpm-in-distinfo + env /usr/lib/rpm/redhat/brp-python-bytecompile '' 1 0 -j2 + /usr/lib/rpm/redhat/brp-python-hardlink + /usr/bin/add-det --brp -j2 /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT Scanned 97 directories and 389 files, processed 0 inodes, 0 modified (0 replaced + 0 rewritten), 0 unsupported format, 0 errors + /usr/bin/linkdupes --brp /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr Scanned 96 directories and 389 files, considered 387 files, read 44 files, linked 1 files, 0 errors sum of sizes of linked files: 65328 bytes Reading /builddir/build/BUILD/baresip-4.11.0-build/SPECPARTS/rpm-debuginfo.specpart Executing(%check): /bin/sh -e /var/tmp/rpm-tmp.YND7bq + umask 022 + cd /builddir/build/BUILD/baresip-4.11.0-build + CFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection ' + export CFLAGS + CXXFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Werror=format-security -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection ' + export CXXFLAGS + FFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection -I/usr/lib64/gfortran/modules ' + export FFLAGS + FCFLAGS='-O2 -flto=auto -ffat-lto-objects -fexceptions -g -grecord-gcc-switches -pipe -Wall -Wp,-U_FORTIFY_SOURCE,-D_FORTIFY_SOURCE=3 -Wp,-D_GLIBCXX_ASSERTIONS -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -fstack-protector-strong -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -m64 -march=z15 -mtune=z17 -fasynchronous-unwind-tables -fstack-clash-protection -I/usr/lib64/gfortran/modules ' + export FCFLAGS + VALAFLAGS=-g + export VALAFLAGS + RUSTFLAGS='-Copt-level=3 -Cdebuginfo=2 -Ccodegen-units=1 -Cstrip=none -Clink-arg=-specs=/usr/lib/rpm/redhat/redhat-package-notes --cap-lints=warn' + export RUSTFLAGS + LDFLAGS='-Wl,-z,relro -Wl,--as-needed -Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld -specs=/usr/lib/rpm/redhat/redhat-hardened-ld-errors -specs=/usr/lib/rpm/redhat/redhat-annobin-cc1 -Wl,--build-id=sha1 -specs=/usr/lib/rpm/redhat/redhat-package-notes ' + export LDFLAGS + LT_SYS_LIBRARY_PATH=/usr/lib64: + export LT_SYS_LIBRARY_PATH + CC=gcc + export CC + CXX=g++ + export CXX + cd baresip-4.11.0 + desktop-file-validate /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/applications/com.github.baresip.desktop + cd redhat-linux-build + ./test/selftest -d ../test/data/ -v running baresip selftest version 4.11.0 with 67 tests Local network addresses: None available for net_interface: - [ RUN ] test_account (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 dtls_srtp: use prime256v1 for elliptic curve cryptography mediaenc: dtls_srtp medianat: ice vidcodec: H266 using stunserver: 'stun:stunserver.org' unloading module: ice.so unloading module: dtls_srtp.so unloading module: g711.so [ OK ] [ RUN ] test_account_uri_complete (rx main) [ OK ] [ RUN ] test_ausrc (rx main) ausrc: aufile auplay: aufile unloading module: aufile.so [ OK ] [ RUN ] test_call_answer (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:43919'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:43919) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:43919) a@127.0.0.1: Call answered: sip:b@127.0.0.1:43919 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:43919) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:43919 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:43919) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:54550'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:54550) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:54550) a@127.0.0.1: Call answered: sip:b@127.0.0.1:54550 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:54550) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:54550 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:54550) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_answer_hangup_a (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:38374'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:38374) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:38374) a@127.0.0.1: Call answered: sip:b@127.0.0.1:38374 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:38374) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:38374 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:38374) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) call: terminate call '2a204a5977d768c6' with sip:b@127.0.0.1:38374 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_answer_hangup_b (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to ''.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:43878) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:43878) a@127.0.0.1: Call answered: sip:b@127.0.0.1:43878 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:43878) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:43878 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:43878) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) call: terminate call '5202a829f4642b38' with sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) sip:b@127.0.0.1:43878: session closed: Connection reset by peer [104] test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_aulevel (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: ausine auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:36248'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:36248) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 audio: client-to-mixer audio levels enabled stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:36248) a@127.0.0.1: Call answered: sip:b@127.0.0.1:36248 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:36248) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) audio: client-to-mixer audio levels enabled stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:36248 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:36248) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) canceled by --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so [ OK ] [ RUN ] test_call_custom_headers (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:58792'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:58792) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:58792) a@127.0.0.1: Call answered: sip:b@127.0.0.1:58792 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:58792) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:58792 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:58792) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_dtmf (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 ausrc: ausine call: connecting to 'sip:b@127.0.0.1:51992'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:51992) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:51992) a@127.0.0.1: Call answered: sip:b@127.0.0.1:51992 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:51992) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:51992 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:51992) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) audio: send DTMF digit: '1' audio: send DTMF digit: '2' audio: send DTMF digit: '3' audio: send DTMF digit: '4' audio: send DTMF digit end: '4' test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) received in-band DTMF event: '1' (end=0) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_START (1) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) received in-band DTMF event: '1' (end=1) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_END () received in-band DTMF event: '2' (end=0) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_START (2) received in-band DTMF event: '2' (end=1) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_END () received in-band DTMF event: '3' (end=0) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_START (3) received in-band DTMF event: '3' (end=1) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_END () received in-band DTMF event: '4' (end=0) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_START (4) received in-band DTMF event: '4' (end=1) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_END () test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so [ OK ] [ RUN ] test_call_format_float (rx main) aufilt: auconv aufilt: auresamp aucodec: aucmock/48000/2 aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: ausine auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:51491'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:51491) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:51491) a@127.0.0.1: Call answered: sip:b@127.0.0.1:51491 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:51491) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format S16LE audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock a@127.0.0.1: Call established: sip:b@127.0.0.1:51491 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:51491) audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format S16LE audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_auframe: 3 met: yes test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 26 26 avg. bitrate: 1468.2 1479.1 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 27 26 avg. bitrate: 1513.5 1446.9 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: ausine auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:36455'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:36455) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:36455) a@127.0.0.1: Call answered: sip:b@127.0.0.1:36455 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:36455) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format S16LE audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock a@127.0.0.1: Call established: sip:b@127.0.0.1:36455 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:36455) audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format S16LE audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 n_auframe: 3 met: yes AND --> --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_auframe: 3 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 25 25 avg. bitrate: 3000.0 2909.0 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 26 25 avg. bitrate: 2980.2 2953.8 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: ausine auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:39387'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:39387) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:39387) a@127.0.0.1: Call answered: sip:b@127.0.0.1:39387 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:39387) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format FLOAT audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format FLOAT audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock a@127.0.0.1: Call established: sip:b@127.0.0.1:39387 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:39387) audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format FLOAT audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format FLOAT audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 n_auframe: 3 met: yes AND --> --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_auframe: 3 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 31 31 avg. bitrate: 1478.7 1469.6 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 33 31 avg. bitrate: 1517.6 1434.2 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: ausine auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:52403'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:52403) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:52403) a@127.0.0.1: Call answered: sip:b@127.0.0.1:52403 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:52403) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format FLOAT audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format FLOAT audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock a@127.0.0.1: Call established: sip:b@127.0.0.1:52403 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:52403) audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format FLOAT audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format FLOAT audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 n_auframe: 3 met: yes AND --> --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_auframe: 3 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:b@127.0.0.1:52403: Call with sip:a@127.0.0.1 terminated (duration: 1 sec) audio Transmit: Receive: packets: 26 26 avg. bitrate: 2980.2 2915.0 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:a@127.0.0.1: Call with sip:b@127.0.0.1:52403 terminated (duration: 1 sec) audio Transmit: Receive: packets: 27 26 avg. bitrate: 3005.2 2958.2 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so unloading module: auresamp.so unloading module: auconv.so [ OK ] [ RUN ] test_call_max (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:56884'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:56884) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) call: connecting to 'sip:b@127.0.0.1:56884'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:56884) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) ua: rejected call from sip:a@127.0.0.1 (maximum 3 calls) test: [ ] event: SIPSESS_FAILED (486 Max Calls) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:56884) a@127.0.0.1: Call answered: sip:b@127.0.0.1:56884 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:56884) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:56884 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:56884) sip:b@127.0.0.1:56884: session closed: 486 Max Calls test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (486 Max Calls) test: re_cancel on call failed test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: re_cancel on call failed test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: re_cancel on call failed test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: re_cancel on call failed unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_mediaenc (rx main) mediaenc: srtp mediaenc: srtp-mand mediaenc: srtp-mandf aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using media encryption 'srtp' b@127.0.0.1: Using media encryption 'srtp' ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:51047'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:51047) stream: audio: starting mediaenc 'srtp' (wait_secure=0) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer stream: audio: starting mediaenc 'srtp' (wait_secure=0) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' stream: audio: starting mediaenc 'srtp' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:51047) a@127.0.0.1: Call answered: sip:b@127.0.0.1:51047 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:51047) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: audio: starting mediaenc 'srtp' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch a@127.0.0.1: Call established: sip:b@127.0.0.1:51047 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:51047) audio_recv: Set audio decoder: PCMU 8000Hz 1ch b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 1 1 avg. bitrate: 5.8 6.4 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 2 1 avg. bitrate: 7.1 3.7 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so unloading module: srtp.so mediaenc: srtp mediaenc: srtp-mand mediaenc: srtp-mandf aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using media encryption 'srtp' b@127.0.0.1: Using media encryption 'srtp' ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:44439'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:44439) stream: audio: starting mediaenc 'srtp' (wait_secure=0) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer stream: audio: starting mediaenc 'srtp' (wait_secure=0) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' audio: RTP/RTCP multiplexing enabled stream: audio: starting mediaenc 'srtp' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:44439) a@127.0.0.1: Call answered: sip:b@127.0.0.1:44439 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:44439) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio: RTP/RTCP multiplexing enabled stream: audio: starting mediaenc 'srtp' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch a@127.0.0.1: Call established: sip:b@127.0.0.1:44439 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:44439) audio_recv: Set audio decoder: PCMU 8000Hz 1ch b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes AND --> --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 2 1 avg. bitrate: 11.6 9.1 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 2 1 avg. bitrate: 11.6 4.2 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so unloading module: srtp.so dtls_srtp: use prime256v1 for elliptic curve cryptography mediaenc: dtls_srtp aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using media encryption 'dtls_srtp' b@127.0.0.1: Using media encryption 'dtls_srtp' ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:36228'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:36228) stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) dtls_srtp: media=audio -- start DTLS client dtls_srtp: 'audio,RTP' dtls connect to 127.0.0.1:44540 dtls_srtp: 'audio,RTCP' dtls connect to 127.0.0.1:44541 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:36228) a@127.0.0.1: Call answered: sip:b@127.0.0.1:36228 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:36228) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) dtls_srtp: media=audio -- start DTLS server a@127.0.0.1: Call established: sip:b@127.0.0.1:36228 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:36228) b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) dtls_srtp: audio: incoming DTLS connect from 127.0.0.1:9508 dtls_srtp: audio: incoming DTLS connect from 127.0.0.1:9509 dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTP) Profile=AES_CM_128_HMAC_SHA1_80 audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,RTP) dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTCP) Profile=AES_CM_128_HMAC_SHA1_80 audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,RTCP) dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTP) Profile=AES_CM_128_HMAC_SHA1_80 audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,RTP) dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTCP) Profile=AES_CM_128_HMAC_SHA1_80 audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,RTCP) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes AND --> --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:b@127.0.0.1:36228: Call with sip:a@127.0.0.1 terminated (duration: 1 sec) audio Transmit: Receive: packets: 1 1 avg. bitrate: 5.8 7.1 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:a@127.0.0.1: Call with sip:b@127.0.0.1:36228 terminated (duration: 1 sec) audio Transmit: Receive: packets: 3 1 avg. bitrate: 13.7 4.2 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so unloading module: dtls_srtp.so dtls_srtp: use prime256v1 for elliptic curve cryptography mediaenc: dtls_srtp aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using media encryption 'dtls_srtp' b@127.0.0.1: Using media encryption 'dtls_srtp' ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:52922'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:52922) stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' audio: RTP/RTCP multiplexing enabled stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) dtls_srtp: media=audio -- start DTLS client dtls_srtp: 'audio,RTP' dtls connect to 127.0.0.1:17528 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:52922) a@127.0.0.1: Call answered: sip:b@127.0.0.1:52922 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:52922) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio: RTP/RTCP multiplexing enabled stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) dtls_srtp: media=audio -- start DTLS server a@127.0.0.1: Call established: sip:b@127.0.0.1:52922 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:52922) b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) dtls_srtp: audio: incoming DTLS connect from 127.0.0.1:33318 dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTP) Profile=AES_CM_128_HMAC_SHA1_80 audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,RTP) dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTP) Profile=AES_CM_128_HMAC_SHA1_80 audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,RTP) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:b@127.0.0.1:52922: Call with sip:a@127.0.0.1 terminated (duration: 1 sec) audio Transmit: Receive: packets: 1 1 avg. bitrate: 5.8 5.3 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:a@127.0.0.1: Call with sip:b@127.0.0.1:52922 terminated (duration: 1 sec) audio Transmit: Receive: packets: 3 1 avg. bitrate: 10.6 4.0 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so unloading module: dtls_srtp.so [ OK ] [ RUN ] test_call_medianat (rx main) medianat: XNAT aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using medianat 'XNAT' b@127.0.0.1: Using medianat 'XNAT' ausrc: ausine call: connecting to 'sip:b@127.0.0.1:42154'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:42154) call: media-nat 'XNAT' established/gathered test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) call: media-nat 'XNAT' established/gathered test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:42154) stream: 'audio' mnat 'XNAT' connected: raddr 127.0.0.1:30674 127.0.0.1:30675 audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call answered: sip:b@127.0.0.1:42154 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:42154) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' a@127.0.0.1: Call established: sip:b@127.0.0.1:42154 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:42154) stream: 'audio' mnat 'XNAT' connected: raddr 127.0.0.1:30516 127.0.0.1:30517 audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU audio_recv: Set audio decoder: PCMU 8000Hz 1ch b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes AND --> --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 1 1 avg. bitrate: 8.0 8.0 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 1 1 avg. bitrate: 8.0 8.0 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so [ OK ] [ RUN ] test_call_mixausrc (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: aufile auplay: aufile aufilt: mixausrc auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:35634'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:35634) audio: using account specific source: (aufile,../test/data//wav/square_500Hz_0.8.wav) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific source: (aufile,../test/data//wav/square_500Hz_0.8.wav) audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:35634) a@127.0.0.1: Call answered: sip:b@127.0.0.1:35634 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:35634) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch aufile: loading input file '../test/data//wav/square_500Hz_0.8.wav' aufile: ../test/data//wav/square_500Hz_0.8.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=10 sampc=80 aufile: read end of file aufile: loaded 16000 bytes audio: source started with sample format S16LE audio tx pipeline: aufile ---> aubuf ---> mixausrc ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- mixausrc <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:35634 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:35634) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch aufile: loading input file '../test/data//wav/square_500Hz_0.8.wav' aufile: ../test/data//wav/square_500Hz_0.8.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=10 sampc=80 aufile: read end of file aufile: loaded 16000 bytes audio: source started with sample format S16LE audio tx pipeline: aufile ---> aubuf ---> mixausrc ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- mixausrc <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [auframe]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [auframe]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_auframe: 3 met: yes test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [auframe]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM prm=[auframe 4] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM prm=[auframe 4] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [mixed]) aufile: loading input file '../test/data//wav/square_1kHz_0.4.wav' aufile: ../test/data//wav/square_1kHz_0.4.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=40 sampc=320 aufile: read end of file aufile: loaded 16000 bytes test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 5) test: event CUSTOM prm=[auframe 5] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 5) test: event CUSTOM prm=[auframe 5] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 6) test: event CUSTOM prm=[auframe 6] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 6) test: event CUSTOM prm=[auframe 6] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 7) test: event CUSTOM prm=[auframe 7] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 7) test: event CUSTOM prm=[auframe 7] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 8) test: event CUSTOM prm=[auframe 8] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 8) test: event CUSTOM prm=[auframe 8] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 9) test: event CUSTOM prm=[auframe 9] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 9) test: event CUSTOM prm=[auframe 9] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 10) test: event CUSTOM prm=[auframe 10] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 10) test: event CUSTOM prm=[auframe 10] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 11) test: event CUSTOM prm=[auframe 11] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 11) test: event CUSTOM prm=[auframe 11] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 12) test: event CUSTOM prm=[auframe 12] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 12) test: event CUSTOM prm=[auframe 12] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (mixed) canceled by --- CUSTOM --- prm: mixed ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 met: yes test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 13) test: event CUSTOM prm=[auframe 13] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 13) test: event CUSTOM prm=[auframe 13] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (mixed) test: event CUSTOM prm=[mixed] (expected [original]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 14) test: event CUSTOM prm=[auframe 14] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 14) test: event CUSTOM prm=[auframe 14] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (mixed) test: event CUSTOM prm=[mixed] (expected [original]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 15) test: event CUSTOM prm=[auframe 15] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 15) test: event CUSTOM prm=[auframe 15] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [original]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 16) test: event CUSTOM prm=[auframe 16] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 16) test: event CUSTOM prm=[auframe 16] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) canceled by --- CUSTOM --- prm: original ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 18 18 avg. bitrate: 62.2 62.9 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 19 18 avg. bitrate: 63.6 59.6 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: mixausrc.so unloading module: aufile.so aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: aufile auplay: aufile aufilt: mixausrc auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:35376'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:35376) audio: using account specific source: (aufile,../test/data//wav/square_500Hz_0.8.wav) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific source: (aufile,../test/data//wav/square_500Hz_0.8.wav) audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:35376) a@127.0.0.1: Call answered: sip:b@127.0.0.1:35376 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:35376) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch aufile: loading input file '../test/data//wav/square_500Hz_0.8.wav' aufile: ../test/data//wav/square_500Hz_0.8.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=10 sampc=80 aufile: read end of file aufile: loaded 16000 bytes audio: source started with sample format S16LE audio tx pipeline: aufile ---> aubuf ---> mixausrc ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- mixausrc <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:35376 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:35376) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch aufile: loading input file '../test/data//wav/square_500Hz_0.8.wav' aufile: ../test/data//wav/square_500Hz_0.8.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=10 sampc=80 aufile: read end of file aufile: loaded 16000 bytes audio: source started with sample format S16LE audio tx pipeline: aufile ---> aubuf ---> mixausrc ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- mixausrc <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [auframe]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [auframe]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_auframe: 3 met: yes test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [auframe]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM prm=[auframe 4] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM prm=[auframe 4] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [mixed]) aufile: loading input file '../test/data//wav/square_1kHz_0.4.wav' aufile: ../test/data//wav/square_1kHz_0.4.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=40 sampc=320 aufile: read end of file aufile: loaded 16000 bytes test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 5) test: event CUSTOM prm=[auframe 5] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 5) test: event CUSTOM prm=[auframe 5] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 6) test: event CUSTOM prm=[auframe 6] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 6) test: event CUSTOM prm=[auframe 6] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 7) test: event CUSTOM prm=[auframe 7] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 7) test: event CUSTOM prm=[auframe 7] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 8) test: event CUSTOM prm=[auframe 8] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 8) test: event CUSTOM prm=[auframe 8] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 9) test: event CUSTOM prm=[auframe 9] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 9) test: event CUSTOM prm=[auframe 9] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 10) test: event CUSTOM prm=[auframe 10] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 10) test: event CUSTOM prm=[auframe 10] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 11) test: event CUSTOM prm=[auframe 11] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 11) test: event CUSTOM prm=[auframe 11] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (mixed) canceled by --- CUSTOM --- prm: mixed ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 met: yes test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 12) test: event CUSTOM prm=[auframe 12] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 12) test: event CUSTOM prm=[auframe 12] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (mixed) test: event CUSTOM prm=[mixed] (expected [original]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 13) test: event CUSTOM prm=[auframe 13] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 13) test: event CUSTOM prm=[auframe 13] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (mixed) test: event CUSTOM prm=[mixed] (expected [original]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 14) test: event CUSTOM prm=[auframe 14] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 14) test: event CUSTOM prm=[auframe 14] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [original]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 15) test: event CUSTOM prm=[auframe 15] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 15) test: event CUSTOM prm=[auframe 15] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) canceled by --- CUSTOM --- prm: original ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 17 18 avg. bitrate: 63.2 66.5 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 18 17 avg. bitrate: 62.6 59.4 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: mixausrc.so unloading module: aufile.so [ OK ] [ RUN ] test_call_multiple (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:54783'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:54783) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) call: connecting to 'sip:b@127.0.0.1:54783'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:54783) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) call: connecting to 'sip:b@127.0.0.1:54783'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:54783) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) call: connecting to 'sip:b@127.0.0.1:54783'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:54783) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 2 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 3 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 4 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:54783) a@127.0.0.1: Call answered: sip:b@127.0.0.1:54783 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:54783) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:54783 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:54783) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:54783) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) a@127.0.0.1: Call answered: sip:b@127.0.0.1:54783 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:54783) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:54783 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:54783) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:54783) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) a@127.0.0.1: Call answered: sip:b@127.0.0.1:54783 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:54783) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:54783 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:54783) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:54783) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) a@127.0.0.1: Call answered: sip:b@127.0.0.1:54783 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:54783) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:54783 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:54783) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) call: terminate call '09636ca9caa0e0df' with sip:b@127.0.0.1:54783 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) call: terminate call '214843b15083efdd' with sip:b@127.0.0.1:54783 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) call: connecting to 'sip:b@127.0.0.1:54783'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:54783) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) call: connecting to 'sip:b@127.0.0.1:54783'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:54783) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 2 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 4 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:54783) a@127.0.0.1: Call answered: sip:b@127.0.0.1:54783 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:54783) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:54783 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:54783) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:54783) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) a@127.0.0.1: Call answered: sip:b@127.0.0.1:54783 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:54783) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:54783 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:54783) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_progress (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:53979'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:53979) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:53979) call: SIP Progress: 183 Session Progress (application/sdp) a@127.0.0.1: Call in-progress: sip:b@127.0.0.1:53979 test: [ sip:a@127.0.0.1 ] event: CALL_PROGRESS (sip:b@127.0.0.1:53979) canceled by --- CALL_PROGRESS --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_reject (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:44980'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:44980) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: rejecting incoming call from sip:a@127.0.0.1 (486 Busy Here) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:44980) sip:b@127.0.0.1:44980: session closed: 486 Busy Here test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (486 Busy Here) test: re_cancel on call failed unloading module: g711.so ua: stop all (forced=1) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:56599'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:56599) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: rejecting incoming call from sip:a@127.0.0.1 (302 Moved Temporarily) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Moved Temporarily) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:56599) call: redirect to sip:c@127.0.0.1 test: [ sip:a@127.0.0.1 ] event: CALL_REDIRECT (302,sip:c@127.0.0.1) sip:b@127.0.0.1:56599: session closed: 302 Moved Temporarily test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (302 Moved Temporarily) test: re_cancel on call failed unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_cancel (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:52555'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:52555) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) call: terminate call 'c8a32cf64c977b4a' with sip:b@127.0.0.1:52555 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) canceled by --- CALL_CLOSED --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 0 n_closed: 1 met: yes test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: rejecting incoming call from sip:a@127.0.0.1 (486 Busy Here) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) test: event CALL_CLOSED ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] unloading module: g711.so ua: stop all (forced=1) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:35132'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:35132) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:35132) call: SIP Progress: 183 Session Progress (application/sdp) a@127.0.0.1: Call in-progress: sip:b@127.0.0.1:35132 test: [ sip:a@127.0.0.1 ] event: CALL_PROGRESS (sip:b@127.0.0.1:35132) canceled by --- CALL_PROGRESS --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU call: terminate call 'a88f199b4d2dfb3a' with sip:b@127.0.0.1:35132 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) unloading module: g711.so ua: stop all (forced=1) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:38678'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:38678) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:38678) call: SIP Progress: 183 Session Progress (application/sdp) a@127.0.0.1: Call in-progress: sip:b@127.0.0.1:38678 test: [ sip:a@127.0.0.1 ] event: CALL_PROGRESS (sip:b@127.0.0.1:38678) canceled by --- CALL_PROGRESS --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: terminate call '360030c50e5746bb' with sip:b@127.0.0.1:38678 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) audio_recv: Set audio decoder: PCMU 8000Hz 1ch b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_rtcp (rx main) ausrc: ausine aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:55429'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:55429) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:55429) a@127.0.0.1: Call answered: sip:b@127.0.0.1:55429 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:55429) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:55429 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:55429) test: event CALL_ESTABLISHED ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) canceled by --- CALL_ESTABLISHED --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 4) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 4) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 5) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 5) test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 6) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 6) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) canceled by --- CALL_RTCP --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_rtcp: 5 met: yes AND --> --- CALL_RTCP --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_rtcp: 5 met: yes AND --> --- CUSTOM --- prm: audebug ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_audebug: 5 met: yes AND --> --- CUSTOM --- prm: audebug ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_audebug: 5 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 7) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 7) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 3 3 avg. bitrate: 11.2 14.7 (kbit/s) errors: 0 0 pkt.report: 3 2 lost: 0 0 jitter: 0.6 0.3 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 4 2 avg. bitrate: 15.0 6.4 (kbit/s) errors: 0 0 pkt.report: 3 2 lost: 0 0 jitter: 0.3 0.6 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so ausrc: ausine aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:57386'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:57386) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' audio: RTP/RTCP multiplexing enabled test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:57386) a@127.0.0.1: Call answered: sip:b@127.0.0.1:57386 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:57386) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio: RTP/RTCP multiplexing enabled audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:57386 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:57386) test: event CALL_ESTABLISHED ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) canceled by --- CALL_ESTABLISHED --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 4) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 4) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 5) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 5) test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 6) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) canceled by --- CALL_RTCP --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_rtcp: 5 met: yes AND --> --- CUSTOM --- prm: audebug ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_audebug: 5 met: yes AND --> --- CUSTOM --- prm: audebug ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_audebug: 5 met: yes test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 6) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 7) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 3 2 avg. bitrate: 16.0 7.5 (kbit/s) errors: 0 0 pkt.report: 2 2 lost: 0 0 jitter: 0.0 0.3 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 4 2 avg. bitrate: 12.1 8.5 (kbit/s) errors: 0 0 pkt.report: 2 2 lost: 0 0 jitter: 0.3 0.1 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so [ OK ] [ RUN ] test_call_rtp_timeout (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:36567'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:36567) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:36567) a@127.0.0.1: Call answered: sip:b@127.0.0.1:36567 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:36567) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU stream: Enable RTP timeout (1 milliseconds) a@127.0.0.1: Call established: sip:b@127.0.0.1:36567 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:36567) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) stream: no audio RTP packets received for 10 milliseconds call: error in "audio" rtp stream (Connection timed out [110]) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (rtp stream error) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_tcp (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:36029;transport=tcp'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:36029;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:36029;transport=tcp) a@127.0.0.1: Call answered: sip:b@127.0.0.1:36029;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:36029;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:36029;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:36029;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_deny_udp (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:a@127.0.0.1 ] event: CREATE (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) call: connecting to 'sip:b@127.0.0.1:40413;transport=udp'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:40413;transport=udp) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) ua: sip:a@127.0.0.1: UA not found: sip:b@127.0.0.1:40413;transport=udp test: [ ] event: SIPSESS_FAILED (404 Not Found) sip:b@127.0.0.1:40413;transport=udp: session closed: 404 Not Found test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (404 Not Found) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_transfer (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:c@127.0.0.1 ] event: CREATE (sip:c@127.0.0.1) call: connecting to 'sip:b@127.0.0.1:59078'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:59078) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:59078) a@127.0.0.1: Call answered: sip:b@127.0.0.1:59078 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:59078) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:59078 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:59078) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) call: hold sip:b@127.0.0.1:59078 test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' transferring call to sip:c@127.0.0.1:59078 test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_HOLD () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: TRANSFER (sip:c@127.0.0.1:59078) call: connecting to 'sip:c@127.0.0.1:59078'.. test: [ sip:b@127.0.0.1 ] event: CALL_OUTGOING (sip:c@127.0.0.1:59078) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:c@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:c@127.0.0.1 ] event: CALL_INCOMING (sip:b@127.0.0.1:59078) call: answering call on line 1 from sip:b@127.0.0.1:59078 with 200 stream: update 'audio' test: [ sip:c@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:b@127.0.0.1 ] event: CALL_RINGING (sip:c@127.0.0.1:59078) b@127.0.0.1: Call answered: sip:c@127.0.0.1:59078 test: [ sip:b@127.0.0.1 ] event: CALL_ANSWERED (sip:c@127.0.0.1:59078) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:c@127.0.0.1:59078 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:c@127.0.0.1:59078) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU c@127.0.0.1: Call established: sip:b@127.0.0.1:59078 test: [ sip:c@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:59078) test: [ sip:c@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Call transfered) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) test: [ sip:c@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_transfer_fail (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:c@127.0.0.1 ] event: CREATE (sip:c@127.0.0.1) call: connecting to 'sip:b@127.0.0.1:41676'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:41676) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:41676) a@127.0.0.1: Call answered: sip:b@127.0.0.1:41676 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:41676) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:41676 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:41676) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) call: hold sip:b@127.0.0.1:41676 test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' transferring call to sip:c@127.0.0.1:41676 test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_HOLD () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: TRANSFER (sip:c@127.0.0.1:41676) call: connecting to 'sip:c@127.0.0.1:41676'.. test: [ sip:b@127.0.0.1 ] event: CALL_OUTGOING (sip:c@127.0.0.1:41676) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:c@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:c@127.0.0.1 ] event: CALL_INCOMING (sip:b@127.0.0.1:41676) call: rejecting incoming call from sip:b@127.0.0.1:41676 (486 Busy Here) test: [ sip:c@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) call: SIP Progress: 180 Ringing (/) test: [ sip:b@127.0.0.1 ] event: CALL_RINGING (sip:c@127.0.0.1:41676) sip:c@127.0.0.1:41676: session closed: 486 Busy Here test: [ sip:b@127.0.0.1 ] event: TRANSFER_FAILED (486 Busy Here) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (486 Busy Here) call: transfer failed: 486 Busy Here test: [ sip:a@127.0.0.1 ] event: TRANSFER_FAILED (486 Busy Here) call: resume sip:b@127.0.0.1:41676 test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_attended_transfer (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:c@127.0.0.1 ] event: CREATE (sip:c@127.0.0.1) call: connecting to 'sip:b@127.0.0.1:49147'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:49147) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:49147) a@127.0.0.1: Call answered: sip:b@127.0.0.1:49147 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:49147) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:49147 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:49147) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) call: hold sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' call: connecting to 'sip:c@127.0.0.1:49147'.. test: [ sip:b@127.0.0.1 ] event: CALL_OUTGOING (sip:c@127.0.0.1:49147) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ sip:a@127.0.0.1 ] event: CALL_HOLD () test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:c@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:c@127.0.0.1 ] event: CALL_INCOMING (sip:b@127.0.0.1) call: answering call on line 1 from sip:b@127.0.0.1 with 200 stream: update 'audio' test: [ sip:c@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' call: SIP Progress: 180 Ringing (/) test: [ sip:b@127.0.0.1 ] event: CALL_RINGING (sip:c@127.0.0.1:49147) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) b@127.0.0.1: Call answered: sip:c@127.0.0.1:49147 test: [ sip:b@127.0.0.1 ] event: CALL_ANSWERED (sip:c@127.0.0.1:49147) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:c@127.0.0.1:49147 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:c@127.0.0.1:49147) call: hold sip:c@127.0.0.1:49147 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' transferring call to sip:c@127.0.0.1:49147 test: [ sip:c@127.0.0.1 ] event: CALL_RTCP (audio) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU c@127.0.0.1: Call established: sip:b@127.0.0.1 test: [ sip:c@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1) test: [ sip:c@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:c@127.0.0.1 ] event: CALL_HOLD () test: [ sip:c@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:c@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: TRANSFER () call: connecting to ''.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:c@127.0.0.1:49147) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' test: [ sip:c@127.0.0.1 ] event: CALL_RTCP (audio) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:c@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:c@127.0.0.1 ] event: CALL_CLOSED (c9e49c6d80401e93;to-tag=a3f9620f2577e0cb;from-tag=8eba9779f4a73012 replaced) test: [ sip:c@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 2 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:c@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:c@127.0.0.1:49147) a@127.0.0.1: Call answered: sip:c@127.0.0.1:49147 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:c@127.0.0.1:49147) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:c@127.0.0.1:49147 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:c@127.0.0.1:49147) sip:c@127.0.0.1:49147: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) test: [ sip:c@127.0.0.1 ] event: CALL_RTCP (audio) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU c@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:c@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Call transfered) sip:b@127.0.0.1:49147: session closed: Connection reset by peer [104] test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) test: [ sip:c@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_video (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 vidcodec: H266 vidisp: mock-vidisp vidsrc: fakevideo vidisp: fakevideo call: connecting to 'sip:b@127.0.0.1:34279'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:34279) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' stream: update 'video' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:34279) a@127.0.0.1: Call answered: sip:b@127.0.0.1:34279 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:34279) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 a@127.0.0.1: Call established: sip:b@127.0.0.1:34279 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:34279) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (video) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (video) video: receiving with resolution 640 x 480 and format 'yuv420p' video: receiving with resolution 640 x 480 and format 'yuv420p' mock_vidisp: got 10 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 10 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 3) canceled by --- CUSTOM --- prm: vidframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 n_vidframe: 3 met: yes test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:b@127.0.0.1:34279: Call with sip:a@127.0.0.1 terminated (duration: 1 sec) video Transmit: Receive: packets: 22 21 avg. bitrate: 11.3 11.2 (kbit/s) errors: 0 0 pkt.report: 1 0 lost: 0 0 jitter: 0.0 0.0 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:a@127.0.0.1: Call with sip:b@127.0.0.1:34279 terminated (duration: 1 sec) video Transmit: Receive: packets: 21 22 avg. bitrate: 11.1 11.5 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: fakevideo.so [ OK ] [ RUN ] test_call_change_videodir (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 vidcodec: H266 vidisp: mock-vidisp vidsrc: fakevideo vidisp: fakevideo call: connecting to 'sip:b@127.0.0.1:49121'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:49121) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:49121) call: SIP Progress: 183 Session Progress (application/sdp) a@127.0.0.1: Call in-progress: sip:b@127.0.0.1:49121 test: [ sip:a@127.0.0.1 ] event: CALL_PROGRESS (sip:b@127.0.0.1:49121) canceled by --- CALL_PROGRESS --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' stream: update 'video' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) a@127.0.0.1: Call answered: sip:b@127.0.0.1:49121 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:49121) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch a@127.0.0.1: Call established: sip:b@127.0.0.1:49121 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:49121) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (video) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (video) audio_recv: Set audio decoder: PCMU 8000Hz 1ch b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (video) video: receiving with resolution 640 x 480 and format 'yuv420p' video: receiving with resolution 640 x 480 and format 'yuv420p' mock_vidisp: got 10 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 10 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 3) canceled by --- CUSTOM --- prm: vidframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_vidframe: 3 met: yes AND --> --- CUSTOM --- prm: vidframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 1 n_vidframe: 3 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 call: got INVITE (SDP Offer) audio-video: sendrecv-sendrecv test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) mock_vidisp: got 10 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) mock_vidisp: got 11 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 13 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 4) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 14 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 5) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 15 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 6) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 16 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 7) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 17 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 8) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 18 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 9) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 10 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 1) mock_vidisp: got 19 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 10) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 2) mock_vidisp: got 20 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 11) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 3) canceled by --- CUSTOM --- prm: vidframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_vidframe: 3 met: yes AND --> --- CUSTOM --- prm: vidframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 1 n_vidframe: 3 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 44 43 avg. bitrate: 10.8 10.8 (kbit/s) errors: 0 0 pkt.report: 35 33 lost: 0 0 jitter: 2.0 1.8 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 43 42 avg. bitrate: 10.7 10.6 (kbit/s) errors: 0 0 pkt.report: 23 23 lost: 0 0 jitter: 1.7 1.9 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: fakevideo.so [ OK ] [ RUN ] test_call_webrtc (rx main) medianat: XNAT dtls_srtp: use prime256v1 for elliptic curve cryptography mediaenc: dtls_srtp ausrc: ausine vidcodec: H266 vidsrc: fakevideo vidisp: fakevideo aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using medianat 'XNAT' a@127.0.0.1: Using media encryption 'dtls_srtp' b@127.0.0.1: Using medianat 'XNAT' b@127.0.0.1: Using media encryption 'dtls_srtp' call: connecting to 'sip:b@127.0.0.1:45652'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:45652) stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) stream: video: starting mediaenc 'dtls_srtp' (wait_secure=1) call: media-nat 'XNAT' established/gathered test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) stream: video: starting mediaenc 'dtls_srtp' (wait_secure=1) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) call: media-nat 'XNAT' established/gathered test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' audio: RTP/RTCP multiplexing enabled stream: update 'video' video: RTP/RTCP multiplexing enabled test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:45652) stream: 'audio' mnat 'XNAT' connected: raddr 127.0.0.1:17714 127.0.0.1:17715 stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) dtls_srtp: media=audio -- start DTLS client dtls_srtp: 'audio,RTP' dtls connect to 127.0.0.1:17714 stream: 'video' mnat 'XNAT' connected: raddr 127.0.0.1:7710 127.0.0.1:7711 stream: video: starting mediaenc 'dtls_srtp' (wait_secure=1) dtls_srtp: media=video -- start DTLS client dtls_srtp: 'video,RTP' dtls connect to 127.0.0.1:7710 a@127.0.0.1: Call answered: sip:b@127.0.0.1:45652 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:45652) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio: RTP/RTCP multiplexing enabled stream: update 'video' video: RTP/RTCP multiplexing enabled a@127.0.0.1: Call established: sip:b@127.0.0.1:45652 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:45652) stream: 'audio' mnat 'XNAT' connected: raddr 127.0.0.1:12804 127.0.0.1:12805 stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) dtls_srtp: media=audio -- start DTLS server stream: 'video' mnat 'XNAT' connected: raddr 127.0.0.1:2288 127.0.0.1:2289 stream: video: starting mediaenc 'dtls_srtp' (wait_secure=1) dtls_srtp: media=video -- start DTLS server b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) dtls_srtp: audio: incoming DTLS connect from 127.0.0.1:12804 dtls_srtp: video: incoming DTLS connect from 127.0.0.1:2288 dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTP) Profile=AES_CM_128_HMAC_SHA1_80 audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,RTP) dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (video/RTP) Profile=AES_CM_128_HMAC_SHA1_80 Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: fakevideo <--- H266 test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,video,RTP) dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTP) Profile=AES_CM_128_HMAC_SHA1_80 audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,RTP) dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (video/RTP) Profile=AES_CM_128_HMAC_SHA1_80 Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: fakevideo <--- H266 test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,video,RTP) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (video) test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: event CALL_RTPESTAB n_audio_estab=0 (expected 1) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB n_video_estab=0 (expected 1) test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (video) test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_audio_estab: 1 n_video_estab: 1 met: yes AND --> --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 n_audio_estab: 1 n_video_estab: 1 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:b@127.0.0.1:45652: Call with sip:a@127.0.0.1 terminated (duration: 1 sec) audio Transmit: Receive: packets: 1 1 avg. bitrate: 75.2 256.0 (kbit/s) errors: 0 0 video Transmit: Receive: packets: 2 2 avg. bitrate: 13.1 13.1 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:a@127.0.0.1: Call with sip:b@127.0.0.1:45652 terminated (duration: 1 sec) audio Transmit: Receive: packets: 2 1 avg. bitrate: 73.1 49.2 (kbit/s) errors: 0 0 video Transmit: Receive: packets: 3 2 avg. bitrate: 10.1 8.6 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: fakevideo.so unloading module: ausine.so unloading module: dtls_srtp.so [ OK ] [ RUN ] test_call_bundle (rx main) vidcodec: H266 vidisp: mock-vidisp vidsrc: fakevideo vidisp: fakevideo aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:33290'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:33290) bundle: alloc stream: bundle init offerer: generate id=1 bundle: alloc stream: bundle init offerer: generate id=1 test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer bundle: alloc bundle: alloc stream: 'audio' enable bundle (Base) bundle: start socket <0x2aa0db6a560> stream: 'video' enable bundle (Mux) bundle: start socket <0x2aa0dbb4f20> stream: parse mid: '0' -> '0' stream: parse mid: '1' -> '1' test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' stream: update 'video' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:33290) a@127.0.0.1: Call answered: sip:b@127.0.0.1:33290 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:33290) stream: parse mid: '0' -> '0' stream: parse mid: '1' -> '1' stream: 'audio' enable bundle (Base) bundle: start socket <0x2aa0db9e380> stream: 'video' enable bundle (Mux) bundle: start socket <0x2aa0db9d930> stream: parse mid: '0' -> '0' stream: parse mid: '1' -> '1' test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 a@127.0.0.1: Call established: sip:b@127.0.0.1:33290 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:33290) test: event CALL_ESTABLISHED ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (video) test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (video) canceled by --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 met: yes AND --> --- CALL_ESTABLISHED --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes AND --> --- CALL_ESTABLISHED --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 2 1 avg. bitrate: 22.0 14.0 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 1 1 avg. bitrate: 17.6 18.6 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: fakevideo.so ERROR: Module srtp is not currently loaded medianat: XNAT vidcodec: H266 vidisp: mock-vidisp vidsrc: fakevideo vidisp: fakevideo aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using medianat 'XNAT' b@127.0.0.1: Using medianat 'XNAT' call: connecting to 'sip:b@127.0.0.1:43733'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:43733) bundle: alloc stream: bundle init offerer: generate id=1 bundle: alloc stream: bundle init offerer: generate id=1 call: media-nat 'XNAT' established/gathered test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer bundle: alloc bundle: alloc stream: 'audio' enable bundle (Base) bundle: start socket <0x2aa0db2b3d0> stream: 'video' enable bundle (Mux) stream: disable MNAT (video) bundle: start socket <0x2aa0db319a0> stream: parse mid: '0' -> '0' stream: parse mid: '1' -> '1' test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) call: media-nat 'XNAT' established/gathered test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' stream: update 'video' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 stream: 'audio' mnat 'XNAT' connected: raddr 127.0.0.1:4494 127.0.0.1:4495 audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:43733) a@127.0.0.1: Call answered: sip:b@127.0.0.1:43733 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:43733) stream: parse mid: '0' -> '0' stream: parse mid: '1' -> '1' stream: 'audio' enable bundle (Base) bundle: start socket <0x2aa0db6bb50> stream: 'video' enable bundle (Mux) stream: disable MNAT (video) bundle: start socket <0x2aa0dbadba0> stream: parse mid: '0' -> '0' stream: parse mid: '1' -> '1' test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: update 'video' Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 a@127.0.0.1: Call established: sip:b@127.0.0.1:43733 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:43733) test: event CALL_ESTABLISHED ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (video) test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] stream: 'audio' mnat 'XNAT' connected: raddr 127.0.0.1:22052 127.0.0.1:22053 audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU audio_recv: Set audio decoder: PCMU 8000Hz 1ch b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (video) canceled by --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 met: yes AND --> --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 met: yes AND --> --- CALL_ESTABLISHED --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes AND --> --- CALL_ESTABLISHED --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 1 1 avg. bitrate: 14.6 22.3 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 1 1 avg. bitrate: 22.0 14.0 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: fakevideo.so ERROR: Module srtp is not currently loaded mediaenc: srtp mediaenc: srtp-mand mediaenc: srtp-mandf vidcodec: H266 vidisp: mock-vidisp vidsrc: fakevideo vidisp: fakevideo aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using media encryption 'srtp' b@127.0.0.1: Using media encryption 'srtp' call: connecting to 'sip:b@127.0.0.1:37914'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:37914) stream: audio: starting mediaenc 'srtp' (wait_secure=0) stream: video: starting mediaenc 'srtp' (wait_secure=0) bundle: alloc stream: bundle init offerer: generate id=1 bundle: alloc stream: bundle init offerer: generate id=1 test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer stream: audio: starting mediaenc 'srtp' (wait_secure=0) stream: video: starting mediaenc 'srtp' (wait_secure=0) bundle: alloc bundle: alloc stream: 'audio' enable bundle (Base) bundle: start socket <0x2aa0db3d820> stream: 'video' enable bundle (Mux) stream: disable MENC (video) bundle: start socket <0x2aa0db319a0> stream: parse mid: '0' -> '0' stream: parse mid: '1' -> '1' test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' stream: audio: starting mediaenc 'srtp' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) stream: update 'video' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:37914) a@127.0.0.1: Call answered: sip:b@127.0.0.1:37914 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:37914) stream: parse mid: '0' -> '0' stream: parse mid: '1' -> '1' stream: 'audio' enable bundle (Base) bundle: start socket <0x2aa0db90de0> stream: 'video' enable bundle (Mux) stream: disable MENC (video) bundle: start socket <0x2aa0db0cc60> stream: parse mid: '0' -> '0' stream: parse mid: '1' -> '1' test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: audio: starting mediaenc 'srtp' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 a@127.0.0.1: Call established: sip:b@127.0.0.1:37914 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:37914) test: event CALL_ESTABLISHED ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] audio_recv: Set audio decoder: PCMU 8000Hz 1ch b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) srtp: failed to decrypt RTCP packet with 52 bytes (Unknown error 217 [217]) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (video) test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (video) canceled by --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 met: yes AND --> --- CALL_ESTABLISHED --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes AND --> --- CALL_ESTABLISHED --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 2 1 avg. bitrate: 20.7 14.0 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 1 1 avg. bitrate: 13.5 16.0 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: fakevideo.so unloading module: srtp.so medianat: XNAT mediaenc: srtp mediaenc: srtp-mand mediaenc: srtp-mandf vidcodec: H266 vidisp: mock-vidisp vidsrc: fakevideo vidisp: fakevideo aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using medianat 'XNAT' a@127.0.0.1: Using media encryption 'srtp' b@127.0.0.1: Using medianat 'XNAT' b@127.0.0.1: Using media encryption 'srtp' call: connecting to 'sip:b@127.0.0.1:56815'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:56815) stream: audio: starting mediaenc 'srtp' (wait_secure=0) stream: video: starting mediaenc 'srtp' (wait_secure=0) bundle: alloc stream: bundle init offerer: generate id=1 bundle: alloc stream: bundle init offerer: generate id=1 call: media-nat 'XNAT' established/gathered test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer stream: audio: starting mediaenc 'srtp' (wait_secure=0) stream: video: starting mediaenc 'srtp' (wait_secure=0) bundle: alloc bundle: alloc stream: 'audio' enable bundle (Base) bundle: start socket <0x2aa0db6a760> stream: 'video' enable bundle (Mux) stream: disable MNAT (video) stream: disable MENC (video) bundle: start socket <0x2aa0db319a0> stream: parse mid: '0' -> '0' stream: parse mid: '1' -> '1' test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) call: media-nat 'XNAT' established/gathered test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' stream: update 'video' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:56815) stream: 'audio' mnat 'XNAT' connected: raddr 127.0.0.1:32608 127.0.0.1:32609 stream: audio: starting mediaenc 'srtp' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) a@127.0.0.1: Call answered: sip:b@127.0.0.1:56815 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:56815) stream: parse mid: '0' -> '0' stream: parse mid: '1' -> '1' stream: 'audio' enable bundle (Base) bundle: start socket <0x2aa0db4a590> stream: 'video' enable bundle (Mux) stream: disable MNAT (video) stream: disable MENC (video) bundle: start socket <0x2aa0db67380> stream: parse mid: '0' -> '0' stream: parse mid: '1' -> '1' test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: update 'video' Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 a@127.0.0.1: Call established: sip:b@127.0.0.1:56815 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:56815) test: event CALL_ESTABLISHED ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] stream: 'audio' mnat 'XNAT' connected: raddr 127.0.0.1:27840 127.0.0.1:27841 stream: audio: starting mediaenc 'srtp' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) srtp: failed to decrypt RTCP packet with 52 bytes (Unknown error 217 [217]) srtp: failed to decrypt RTP packet with 34 bytes (Unknown error 217 [217]) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (video) test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (video) canceled by --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 met: yes AND --> --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 met: yes AND --> --- CALL_ESTABLISHED --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes AND --> --- CALL_ESTABLISHED --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 2 1 avg. bitrate: 16.7 56.0 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 2 1 avg. bitrate: 20.7 10.1 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: fakevideo.so unloading module: srtp.so [ OK ] [ RUN ] test_call_ipv6ll (rx main) no IPv6 address -- skipping test test_call_ipv6ll [ OK ] [ RUN ] test_call_100rel_audio (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: ausine auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:41135'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:41135) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:41135) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU call: SIP Progress: 183 Session Progress (application/sdp) a@127.0.0.1: Call in-progress: sip:b@127.0.0.1:41135 test: [ sip:a@127.0.0.1 ] event: CALL_PROGRESS (sip:b@127.0.0.1:41135) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 n_auframe: 3 met: yes test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' call: got UPDATE (SDP Offer) audio-video: inactive-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU call: got UPDATE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 n_auframe: 3 met: yes test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 32 31 avg. bitrate: 14.8 14.1 (kbit/s) errors: 0 0 pkt.report: 31 30 lost: 0 0 jitter: 2.6 2.6 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 31 32 avg. bitrate: 13.8 14.5 (kbit/s) errors: 0 0 pkt.report: 30 31 lost: 0 0 jitter: 2.6 2.6 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so [ OK ] [ RUN ] test_call_100rel_video (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 vidcodec: H266 vidisp: mock-vidisp vidsrc: fakevideo vidisp: fakevideo call: connecting to 'sip:b@127.0.0.1:33356'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:33356) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:33356) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 call: SIP Progress: 183 Session Progress (application/sdp) a@127.0.0.1: Call in-progress: sip:b@127.0.0.1:33356 test: [ sip:a@127.0.0.1 ] event: CALL_PROGRESS (sip:b@127.0.0.1:33356) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (video) video: receiving with resolution 640 x 480 and format 'yuv420p' video: receiving with resolution 640 x 480 and format 'yuv420p' mock_vidisp: got 10 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 10 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 3) canceled by --- CUSTOM --- prm: vidframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 n_vidframe: 3 met: yes test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got UPDATE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 call: got UPDATE (SDP Offer) audio-video: sendrecv-sendrecv test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) mock_vidisp: got 10 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 13 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 4) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 14 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 5) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 15 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 6) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 16 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 7) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 17 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 8) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 10 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 1) mock_vidisp: got 18 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 9) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 2) mock_vidisp: got 19 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 10) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 3) canceled by --- CUSTOM --- prm: vidframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 0 n_vidframe: 3 met: yes AND --> --- CUSTOM --- prm: vidframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 n_vidframe: 3 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 43 42 avg. bitrate: 10.7 10.7 (kbit/s) errors: 0 0 pkt.report: 34 32 lost: 0 0 jitter: 2.2 1.7 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 42 43 avg. bitrate: 10.6 10.9 (kbit/s) errors: 0 0 pkt.report: 22 22 lost: 0 0 jitter: 1.3 1.4 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: fakevideo.so [ OK ] [ RUN ] test_call_100rel_required_reject (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) call: connecting to 'sip:b@127.0.0.1:55440'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:55440) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) ua: call from sip:a@127.0.0.1 rejected with 421 -- option-tag '100rel' not supported by remote sip:b@127.0.0.1:55440: session closed: 421 Extension required test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (421 Extension required) test: re_cancel on call failed unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_hold_resume (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:46113'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:46113) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:46113) a@127.0.0.1: Call answered: sip:b@127.0.0.1:46113 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:46113) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:46113 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:46113) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 n_audio_estab: 1 met: yes AND --> --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_audio_estab: 1 met: yes call: hold sip:b@127.0.0.1:46113 test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_HOLD () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 1 met: yes call: resume sip:b@127.0.0.1:46113 test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RESUME () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE call: got INVITE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 1 met: yes call: terminate call '9d0df5c361124dbc' with sip:b@127.0.0.1:46113 sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) canceled by --- CALL_CLOSED --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes audio Transmit: Receive: packets: 2 1 avg. bitrate: 45.7 22.4 (kbit/s) errors: 0 0 pkt.report: 2 1 lost: 0 0 jitter: 0.0 0.0 (ms) call: connecting to 'sip:b@127.0.0.1:37845;transport=tcp'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:37845;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:37845;transport=tcp) a@127.0.0.1: Call answered: sip:b@127.0.0.1:37845;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:37845;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE a@127.0.0.1: Call established: sip:b@127.0.0.1:37845;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:37845;transport=tcp) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 n_audio_estab: 2 met: yes call: hold sip:b@127.0.0.1:37845;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 2 met: yes call: resume sip:b@127.0.0.1:37845;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 2 met: yes call: hold sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_HOLD () test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:a@127.0.0.1 n_established: 2 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' call: got INVITE (SDP Offer) audio-video: inactive-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:a@127.0.0.1 n_established: 2 met: yes call: resume sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_RESUME () test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE call: got INVITE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE test: [ sip:a@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:a@127.0.0.1 n_established: 2 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 0 4 avg. bitrate: 0.0 23.2 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 1 2 avg. bitrate: 4.3 8.8 (kbit/s) errors: 0 0 pkt.report: 1 1 lost: 0 0 jitter: 0.0 0.0 (ms) audio Transmit: Receive: packets: 4 0 avg. bitrate: 22.5 0.0 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:34235;transport=tcp'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:34235;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:34235;transport=tcp) a@127.0.0.1: Call answered: sip:b@127.0.0.1:34235;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:34235;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:34235;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:34235;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 n_audio_estab: 1 met: yes AND --> --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_audio_estab: 1 met: yes call: hold sip:b@127.0.0.1:34235;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_HOLD () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 1 met: yes call: resume sip:b@127.0.0.1:34235;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RESUME () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE call: got INVITE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 1 met: yes call: terminate call '485d4497c693a1cb' with sip:b@127.0.0.1:34235;transport=tcp sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) canceled by --- CALL_CLOSED --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes sip:b@127.0.0.1:34235;transport=tcp: Call with sip:a@127.0.0.1 terminated (duration: 1 sec) audio Transmit: Receive: packets: 5 4 avg. bitrate: 37.6 28.7 (kbit/s) errors: 0 0 pkt.report: 2 4 lost: 0 0 jitter: 0.0 0.0 (ms) call: connecting to 'sip:b@127.0.0.1:34235;transport=tcp'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:34235;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:34235;transport=tcp) a@127.0.0.1: Call answered: sip:b@127.0.0.1:34235;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:34235;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE a@127.0.0.1: Call established: sip:b@127.0.0.1:34235;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:34235;transport=tcp) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 n_audio_estab: 2 met: yes call: hold sip:b@127.0.0.1:34235;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 2 met: yes call: resume sip:b@127.0.0.1:34235;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 2 met: yes call: hold sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_HOLD () test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:a@127.0.0.1 n_established: 2 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' call: got INVITE (SDP Offer) audio-video: inactive-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:a@127.0.0.1 n_established: 2 met: yes call: resume sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_RESUME () test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE call: got INVITE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE test: [ sip:a@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:a@127.0.0.1 n_established: 2 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 0 3 avg. bitrate: 0.0 15.0 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:a@127.0.0.1: Call with sip:b@127.0.0.1:34235;transport=tcp terminated (duration: 1 sec) audio Transmit: Receive: packets: 4 1 avg. bitrate: 10.2 2.6 (kbit/s) errors: 0 0 audio Transmit: Receive: packets: 5 0 avg. bitrate: 24.7 0.0 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:48705'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:48705) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:48705) a@127.0.0.1: Call answered: sip:b@127.0.0.1:48705 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:48705) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:48705 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:48705) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 n_audio_estab: 1 met: yes AND --> --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_audio_estab: 1 met: yes call: hold sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_HOLD () test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:a@127.0.0.1 n_established: 1 met: yes call: got INVITE audio-video: recvonly-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) test: event CALL_REMOTE_SDP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE call: got (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_RESUME () test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) test: event CALL_REMOTE_SDP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 1 1 avg. bitrate: 28.4 27.2 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 1 1 avg. bitrate: 17.7 18.2 (kbit/s) errors: 0 0 pkt.report: 1 1 lost: 0 0 jitter: 0.0 0.0 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:35090'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:35090) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:35090) a@127.0.0.1: Call answered: sip:b@127.0.0.1:35090 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:35090) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:35090 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:35090) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 n_audio_estab: 1 met: yes AND --> --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_audio_estab: 1 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_HOLD () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' call: got INVITE (SDP Offer) audio-video: inactive-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 1 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RESUME () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU call: got INVITE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 1 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 2 1 avg. bitrate: 41.9 20.6 (kbit/s) errors: 0 0 pkt.report: 1 1 lost: 0 0 jitter: 0.0 0.0 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 2 1 avg. bitrate: 30.4 15.4 (kbit/s) errors: 0 0 pkt.report: 1 1 lost: 0 0 jitter: 0.0 0.0 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so [ OK ] [ RUN ] test_call_srtp_tx_rekey (rx main) mediaenc: srtp mediaenc: srtp-mand mediaenc: srtp-mandf ausrc: ausine auplay: mock-auplay aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using media encryption 'srtp-mand' b@127.0.0.1: Using media encryption 'srtp-mand' b@127.0.0.1: Using media encryption 'srtp-mand' test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) call: connecting to 'sip:b@127.0.0.1:47491'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:47491) stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:47491) a@127.0.0.1: Call answered: sip:b@127.0.0.1:47491 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:47491) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch a@127.0.0.1: Call established: sip:b@127.0.0.1:47491 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:47491) test: event CALL_ESTABLISHED ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] audio_recv: Set audio decoder: PCMU 8000Hz 1ch b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) canceled by --- CALL_ESTABLISHED --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: event CALL_ESTABLISHED ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) stream: update 'audio' stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) stream: update 'audio' stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) srtp: audio: re-keying in progress srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 5) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 6) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 7) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 5) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 8) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 6) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 9) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 7) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 10) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 8) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 11) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 9) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 12) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 10) canceled by --- CUSTOM --- prm: auframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 n_auframe: 10 met: yes AND --> --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_auframe: 10 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 34 33 avg. bitrate: 15.1 14.9 (kbit/s) errors: 0 0 pkt.report: 3 1 lost: 0 0 jitter: 0.0 0.0 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 35 34 avg. bitrate: 15.1 15.4 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so unloading module: srtp.so [ OK ] [ RUN ] test_call_uag_find_msg (rx main) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:alice@127.0.0.1:54402 ] event: CREATE (sip:alice@127.0.0.1:54402) test: [ sip:alice@127.0.0.1:55564 ] event: CREATE (sip:alice@127.0.0.1:55564) test: [ sip:bob@127.0.0.1:55564 ] event: CREATE (sip:bob@127.0.0.1:55564) test: [ sip:alice@127.0.0.1:54402 ] event: REGISTERING () test: [ sip:alice@127.0.0.1:55564 ] event: REGISTERING () test: [ sip:bob@127.0.0.1:55564 ] event: REGISTERING () alice@127.0.0.1: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] test: [ sip:alice@127.0.0.1:54402 ] event: REGISTER_OK (200 OK) test: event REGISTER_OK ua=[sip:alice@127.0.0.1:54402] (expected [sip:bob@127.0.0.1:55564] test: event REGISTER_OK ua=[sip:alice@127.0.0.1:54402] (expected [sip:alice@127.0.0.1:55564] alice@127.0.0.1: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] test: [ sip:alice@127.0.0.1:55564 ] event: REGISTER_OK (200 OK) test: event REGISTER_OK ua=[sip:alice@127.0.0.1:55564] (expected [sip:bob@127.0.0.1:55564] bob@127.0.0.1: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] test: [ sip:bob@127.0.0.1:55564 ] event: REGISTER_OK (200 OK) canceled by --- REGISTER_OK --- prm: ua: sip:bob@127.0.0.1:55564 n_incoming: 0 n_progress: 0 n_established: 0 met: yes test: event REGISTER_OK ua=[sip:bob@127.0.0.1:55564] (expected [sip:alice@127.0.0.1:55564] test: event REGISTER_OK ua=[sip:bob@127.0.0.1:55564] (expected [sip:alice@127.0.0.1:54402] call: connecting to 'sip:alice@127.0.0.1:55564'.. test: [ sip:bob@127.0.0.1:55564 ] event: CALL_OUTGOING (sip:alice@127.0.0.1:55564) test: [ sip:bob@127.0.0.1:55564 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:alice@127.0.0.1:55564 ] event: CALL_REMOTE_SDP (offer) test: [ sip:alice@127.0.0.1:55564 ] event: CALL_INCOMING (sip:bob@127.0.0.1:55564) call: answering call on line 1 from sip:bob@127.0.0.1:55564 with 200 stream: update 'audio' test: [ sip:alice@127.0.0.1:55564 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:bob@127.0.0.1:55564 ] event: CALL_RINGING (sip:alice@127.0.0.1:55564) bob@127.0.0.1: Call answered: sip:alice@127.0.0.1:55564 test: [ sip:bob@127.0.0.1:55564 ] event: CALL_ANSWERED (sip:alice@127.0.0.1:55564) test: [ sip:bob@127.0.0.1:55564 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU bob@127.0.0.1: Call established: sip:alice@127.0.0.1:55564 test: [ sip:bob@127.0.0.1:55564 ] event: CALL_ESTABLISHED (sip:alice@127.0.0.1:55564) test: event CALL_ESTABLISHED ua=[sip:bob@127.0.0.1:55564] (expected [sip:alice@127.0.0.1:55564] audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU alice@127.0.0.1: Call established: sip:bob@127.0.0.1:55564 test: [ sip:alice@127.0.0.1:55564 ] event: CALL_ESTABLISHED (sip:bob@127.0.0.1:55564) canceled by --- CALL_ESTABLISHED --- prm: ua: sip:alice@127.0.0.1:55564 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: event CALL_ESTABLISHED ua=[sip:alice@127.0.0.1:55564] (expected [sip:bob@127.0.0.1:55564] test: [ sip:alice@127.0.0.1:55564 ] event: CALL_RTCP (audio) test: [ sip:alice@127.0.0.1:54402 ] event: UNREGISTERING () test: [ sip:alice@127.0.0.1:54402 ] event: CREATE (sip:alice@127.0.0.1:54402) test: [ sip:alice@127.0.0.1:54402 ] event: REGISTERING () alice@127.0.0.1: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] test: [ sip:alice@127.0.0.1:54402 ] event: REGISTER_OK (200 OK) canceled by --- REGISTER_OK --- prm: ua: sip:alice@127.0.0.1:54402 n_incoming: 0 n_progress: 0 n_established: 0 met: yes call: connecting to 'sip:alice@127.0.0.1:54401'.. test: [ sip:bob@127.0.0.1:55564 ] event: CALL_OUTGOING (sip:alice@127.0.0.1:54401) test: [ sip:bob@127.0.0.1:55564 ] event: CALL_LOCAL_SDP (offer) ua: sip:bob@127.0.0.1:55564: UA not found: sip:alice@127.0.0.1:54401 test: [ ] event: SIPSESS_FAILED (404 Not Found) test: could not find agent/ua sip:alice@127.0.0.1:54401: session closed: 404 Not Found test: [ sip:bob@127.0.0.1:55564 ] event: CALL_CLOSED (404 Not Found) canceled by --- CALL_CLOSED --- prm: ua: sip:bob@127.0.0.1:55564 n_incoming: 0 n_progress: 0 n_established: 0 met: yes test: [ sip:bob@127.0.0.1:55564 ] event: UNREGISTERING () test: [ sip:bob@127.0.0.1:55564 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:alice@127.0.0.1:55564 ] event: UNREGISTERING () test: [ sip:alice@127.0.0.1:55564 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:alice@127.0.0.1:54402 ] event: UNREGISTERING () unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_samplerate_switch (rx main) aucodec: L16/48000/2 aucodec: L16/44100/2 aucodec: L16/32000/2 aucodec: L16/16000/2 aucodec: L16/8000/2 aucodec: L16/48000/1 aucodec: L16/44100/1 aucodec: L16/32000/1 aucodec: L16/16000/1 aucodec: L16/8000/1 aufilt: auresamp aufilt: auconv ausrc: aufile auplay: aufile aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:33562'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:33562) audio: using account specific source: (aufile,../test/data//wav/square_500Hz_0.8.wav) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific source: (aufile,../test/data//wav/square_500Hz_0.8.wav) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:33562) a@127.0.0.1: Call answered: sip:b@127.0.0.1:33562 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:33562) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: L16 48000Hz 2ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: L16 48000Hz 2ch aufile: loading input file '../test/data//wav/square_500Hz_0.8.wav' aufile: ../test/data//wav/square_500Hz_0.8.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=4 sampc=32 aufile: read end of file aufile: loaded 16000 bytes audio: source started with sample format S16LE audio tx pipeline: aufile ---> aubuf ---> auresamp ---> auconv ---> L16 audio rx pipeline: aufile <--- aubuf <--- auresamp <--- auconv <--- L16 a@127.0.0.1: Call established: sip:b@127.0.0.1:33562 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:33562) audio_recv: Set audio decoder: L16 48000Hz 2ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: L16 48000Hz 2ch aufile: loading input file '../test/data//wav/square_500Hz_0.8.wav' aufile: ../test/data//wav/square_500Hz_0.8.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=4 sampc=32 aufile: read end of file aufile: loaded 16000 bytes audio: source started with sample format S16LE audio tx pipeline: aufile ---> aubuf ---> auresamp ---> auconv ---> L16 audio rx pipeline: aufile <--- aubuf <--- auresamp <--- auconv <--- L16 b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) audio_recv: Set audio decoder: PCMA 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 5 0 avg. bitrate: 853.3 0.0 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 10 0 avg. bitrate: 1059.3 0.0 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: l16.so unloading module: aufile.so unloading module: auconv.so unloading module: auresamp.so [ OK ] [ RUN ] test_call_sni (rx main) SIP Certificate: ../test/data//sni/other-cert.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:a@retest.client.org ] event: CREATE (sip:a@retest.client.org) test: [ sip:b@retest.server.org ] event: CREATE (sip:b@retest.server.org) test: [ sip:c@retest.unknown.org ] event: CREATE (sip:c@retest.unknown.org) call: connecting to 'sip:b@retest.server.org:33867;transport=tls'.. test: [ sip:a@retest.client.org ] event: CALL_OUTGOING (sip:b@retest.server.org:33867;transport=tls) test: [ sip:a@retest.client.org ] event: CALL_LOCAL_SDP (offer) sip:b@retest.server.org:33867;transport=tls: session closed: Protocol error [71] test: [ sip:a@retest.client.org ] event: CALL_CLOSED (Protocol error [71]) call: connecting to 'sip:c@retest.unknown.org:33867;transport=tls'.. test: [ sip:a@retest.client.org ] event: CALL_OUTGOING (sip:c@retest.unknown.org:33867;transport=tls) test: [ sip:a@retest.client.org ] event: CALL_LOCAL_SDP (offer) sip:c@retest.unknown.org:33867;transport=tls: session closed: Protocol error [71] test: [ sip:a@retest.client.org ] event: CALL_CLOSED (Protocol error [71]) call: connecting to 'sip:b@retest.server.org:33867;transport=tls'.. test: [ sip:a@retest.client.org ] event: CALL_OUTGOING (sip:b@retest.server.org:33867;transport=tls) test: [ sip:a@retest.client.org ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@retest.server.org ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@retest.server.org ] event: CALL_INCOMING (sip:a@retest.client.org) call: answering call on line 1 from sip:a@retest.client.org with 200 stream: update 'audio' test: [ sip:b@retest.server.org ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@retest.client.org ] event: CALL_RINGING (sip:b@retest.server.org:33867;transport=tls) a@retest.client.org: Call answered: sip:b@retest.server.org:33867;transport=tls test: [ sip:a@retest.client.org ] event: CALL_ANSWERED (sip:b@retest.server.org:33867;transport=tls) test: [ sip:a@retest.client.org ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@retest.client.org: Call established: sip:b@retest.server.org:33867;transport=tls test: [ sip:a@retest.client.org ] event: CALL_ESTABLISHED (sip:b@retest.server.org:33867;transport=tls) test: [ sip:b@retest.server.org ] event: CALL_RTCP (audio) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@retest.server.org: Call established: sip:a@retest.client.org test: [ sip:b@retest.server.org ] event: CALL_ESTABLISHED (sip:a@retest.client.org) call: terminate call '863f82cb6a87a479' with sip:b@retest.server.org:33867;transport=tls test: [ sip:a@retest.client.org ] event: CALL_CLOSED (Rejected by user) test: [ sip:b@retest.server.org ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_cert_select (rx main) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:a@127.0.0.1 ] event: CREATE (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) call: connecting to 'sip:b@127.0.0.1:38157;transport=tls'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:38157;transport=tls) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) sip:b@127.0.0.1:38157;transport=tls: session closed: Protocol error [71] test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Protocol error [71]) call: connecting to 'sip:a@127.0.0.1:38157;transport=tls'.. test: [ sip:b@127.0.0.1 ] event: CALL_OUTGOING (sip:a@127.0.0.1:38157;transport=tls) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) sip:a@127.0.0.1:38157;transport=tls: session closed: Protocol error [71] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Protocol error [71]) call: connecting to 'sip:b@127.0.0.1:38157;transport=tls'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:38157;transport=tls) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:38157;transport=tls) a@127.0.0.1: Call answered: sip:b@127.0.0.1:38157;transport=tls test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:38157;transport=tls) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:38157;transport=tls test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:38157;transport=tls) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) call: terminate call 'fd3899b286483b3e' with sip:b@127.0.0.1:38157;transport=tls test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_cmd (rx main) [ OK ] [ RUN ] test_cmd_long (rx main) [ OK ] [ RUN ] test_cparam_call_decode (rx main) test 0: audio=sendonly video=inactive callid=123 test 1: video=recvonly callid=234 audio=sendonly test 2: video=recvonly callid=234 audio=fail test 3: video=sendonly test 4: video=fail2 test 5: recvonly test 6: callid2 test 7: recvonly 345 test 8: fail3 456 test 9: video=inactive callid=234 audio=inactive [ OK ] [ RUN ] test_cparam_ua_decode (rx main) test 0: "display name" audio=sendonly video=inactive userdata=mydata test 1: displayname userdata=mydata test 2: dn userdata=mydata2 audio=recvonly test 3: dn userdata=mydata2 video=fail4 test 4: sip:user2@domain2 audio=recvonly video=sendonly test 5: sip:user3@domain3 audio=fail video=sendonly test 6: sip:user4@domain4 video=inactive test 7: sip:user5@domain5 video=fail2 test 8: user6 sendonly test 9: audio=sendonly test 10: sip:user4@domain4;transport=tcp video=recvonly [ OK ] [ RUN ] test_contact (rx main) [ OK ] [ RUN ] test_contact_find_call (rx main) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:55324'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:55324) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:55324) a@127.0.0.1: Call answered: sip:b@127.0.0.1:55324 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:55324) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:55324 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:55324) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) call: terminate call 'afbdb16f025ed432' with sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@localhost ] event: CREATE (sip:a@localhost) call: connecting to 'sip:b@127.0.0.1:55324'.. test: [ sip:a@localhost ] event: CALL_OUTGOING (sip:b@127.0.0.1:55324) test: [ sip:a@localhost ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@localhost) call: answering call on line 1 from sip:a@localhost with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@localhost ] event: CALL_RINGING (sip:b@127.0.0.1:55324) a@localhost: Call answered: sip:b@127.0.0.1:55324 test: [ sip:a@localhost ] event: CALL_ANSWERED (sip:b@127.0.0.1:55324) test: [ sip:a@localhost ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@localhost: Call established: sip:b@127.0.0.1:55324 test: [ sip:a@localhost ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:55324) call: abort call '0d0054c24631681f' with sip:a@localhost test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) test: [ sip:a@localhost ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_bevent_register (rx main) [ OK ] [ RUN ] test_jbuf (rx main) [ OK ] [ RUN ] test_jbuf_adaptive (rx main) [ OK ] [ RUN ] test_jbuf_video (rx main) [ OK ] [ RUN ] test_jbuf_gnack (rx main) jbuf: RTCP_GNACK too big blp_pkts 20 [ OK ] [ RUN ] test_message (rx main) [ sip:b@127.0.0.1:57958;transport=UDP ] recv msg from sip:a@127.0.0.1: "hei paa deg" [ sip:a@127.0.0.1:57958;transport=UDP ] message sent OK [ sip:b@127.0.0.1:37561;transport=TCP ] recv msg from sip:a@127.0.0.1: "hei paa deg" [ sip:a@127.0.0.1:37561;transport=TCP ] message sent OK [ sip:a@127.0.0.1:59775;transport=UDP ] message sent OK [ sip:a@127.0.0.1:46435;transport=TCP ] message sent OK [ OK ] [ RUN ] test_network (rx main) Local network addresses: None available for net_interface: - [ OK ] [ RUN ] test_peerconn (rx main) [ OK ] [ RUN ] test_play (rx main) auplay: mock-auplay [ OK ] [ RUN ] test_stunuri (rx main) [ OK ] [ RUN ] test_ua_alloc (rx main) [ OK ] [ RUN ] test_ua_cuser (rx main) [ OK ] [ RUN ] test_ua_options (rx main) ua: stop all (forced=1) ua: stop all (forced=1) [ OK ] [ RUN ] test_ua_refer (rx main) ua: REFER forbidden for user@127.0.0.1 ua: stop all (forced=1) ua: REFER forbidden for user@127.0.0.1 ua: stop all (forced=1) [ OK ] [ RUN ] test_ua_register (rx main) SIP Certificate: ../test/data//sni/server-interm.pem x@127.0.0.1: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! x@127.0.0.1: (prio 0) {0/TCP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! x@127.0.0.1: (prio 0) {0/TLS/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) [ OK ] [ RUN ] test_ua_register_auth (rx main) SIP Certificate: ../test/data//sni/server-interm.pem alfredh@localhost: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! alfredh@localhost: (prio 0) {0/TCP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! alfredh@localhost: (prio 0) {0/TLS/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) [ OK ] [ RUN ] test_ua_register_auth_dns (rx main) | DNS-server on 127.0.0.1:58688 | SIP-server on 127.0.0.1:37213 | SIP-server on 127.0.0.1:35099 SIP Certificate: ../test/data//sni/server-interm.pem alfredh@test.invalid: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) | DNS-server on 127.0.0.1:49004 | SIP-server on 127.0.0.1:36053 | SIP-server on 127.0.0.1:44141 SIP Certificate: ../test/data//sni/server-interm.pem alfredh@test.invalid: (prio 0) {0/TCP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) | DNS-server on 127.0.0.1:34924 | SIP-server on 127.0.0.1:33979 | SIP-server on 127.0.0.1:35061 SIP Certificate: ../test/data//sni/server-interm.pem alfredh@test.invalid: (prio 0) {0/TLS/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) [ OK ] [ RUN ] test_ua_register_dns (rx main) | DNS-server on 127.0.0.1:51829 | SIP-server on 127.0.0.1:33884 SIP Certificate: ../test/data//sni/server-interm.pem x@test.invalid: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) | DNS-server on 127.0.0.1:36100 | SIP-server on 127.0.0.1:43711 SIP Certificate: ../test/data//sni/server-interm.pem x@test.invalid: (prio 0) {0/TCP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) | DNS-server on 127.0.0.1:59444 | SIP-server on 127.0.0.1:40123 SIP Certificate: ../test/data//sni/server-interm.pem x@test.invalid: (prio 0) {0/TLS/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) [ OK ] [ RUN ] test_uag_find_param (rx main) [ OK ] [ RUN ] test_video (rx main) [ OK ] [ RUN ] test_clean_number (rx main) [ OK ] [ RUN ] test_clean_number_only_numeric (rx main) [ OK ] [ RUN ] test_account (rx thread) aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 dtls_srtp: use prime256v1 for elliptic curve cryptography mediaenc: dtls_srtp medianat: ice vidcodec: H266 using stunserver: 'stun:stunserver.org' unloading module: ice.so unloading module: dtls_srtp.so unloading module: g711.so [ OK ] [ RUN ] test_account_uri_complete (rx thread) [ OK ] [ RUN ] test_ausrc (rx thread) ausrc: aufile auplay: aufile unloading module: aufile.so [ OK ] [ RUN ] test_call_answer (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:33714'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:33714) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:33714) a@127.0.0.1: Call answered: sip:b@127.0.0.1:33714 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:33714) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:33714 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:33714) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) rtp_receiver: RTP RX thread started test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:45434'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:45434) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:45434) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:45434 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:45434) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:45434 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:45434) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_answer_hangup_a (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:47994'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:47994) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:47994) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:47994 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:47994) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:47994 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:47994) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) call: terminate call 'bcd47a9fd0d380d7' with sip:b@127.0.0.1:47994 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_answer_hangup_b (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to ''.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:39158) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:39158) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:39158 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:39158) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:39158 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:39158) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) call: terminate call '033f00d2fee48f7e' with sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) rtp_receiver: RTP RX thread started sip:b@127.0.0.1:39158: session closed: Connection reset by peer [104] test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_aulevel (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: ausine auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:46391'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:46391) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 audio: client-to-mixer audio levels enabled stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:46391) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:46391 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:46391) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) audio: client-to-mixer audio levels enabled stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:46391 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:46391) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) canceled by --- CUSTOM --- prm: auframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes AND --> --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 29 31 avg. bitrate: 30.1 16.0 (kbit/s) errors: 0 0 pkt.report: 1 2 lost: 0 0 jitter: 0.0 0.1 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 32 29 avg. bitrate: 30.3 13.8 (kbit/s) errors: 0 0 pkt.report: 1 0 lost: 0 0 jitter: 0.0 0.0 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so [ OK ] [ RUN ] test_call_custom_headers (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:59449'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:59449) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:59449) a@127.0.0.1: Call answered: sip:b@127.0.0.1:59449 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:59449) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:59449 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:59449) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_dtmf (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 ausrc: ausine call: connecting to 'sip:b@127.0.0.1:58544'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:58544) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:58544) a@127.0.0.1: Call answered: sip:b@127.0.0.1:58544 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:58544) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:58544 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:58544) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) audio: send DTMF digit: '1' audio: send DTMF digit: '2' audio: send DTMF digit: '3' audio: send DTMF digit: '4' audio: send DTMF digit end: '4' test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) received in-band DTMF event: '1' (end=0) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_START (1) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) received in-band DTMF event: '1' (end=1) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_END () received in-band DTMF event: '2' (end=0) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_START (2) received in-band DTMF event: '2' (end=1) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_END () received in-band DTMF event: '3' (end=0) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_START (3) received in-band DTMF event: '3' (end=1) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_END () received in-band DTMF event: '4' (end=0) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_START (4) received in-band DTMF event: '4' (end=1) test: [ sip:b@127.0.0.1 ] event: CALL_DTMF_END () test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 47 96 avg. bitrate: 15.2 23.2 (kbit/s) errors: 0 0 pkt.report: 1 3 lost: 0 0 jitter: 0.0 0.0 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 98 47 avg. bitrate: 22.6 14.4 (kbit/s) errors: 0 0 pkt.report: 2 1 lost: 0 0 jitter: 0.0 0.0 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so [ OK ] [ RUN ] test_call_format_float (rx thread) aufilt: auconv aufilt: auresamp aucodec: aucmock/48000/2 SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: ausine auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:60847'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:60847) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:60847) a@127.0.0.1: Call answered: sip:b@127.0.0.1:60847 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:60847) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format S16LE audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock a@127.0.0.1: Call established: sip:b@127.0.0.1:60847 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:60847) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format S16LE audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_auframe: 3 met: yes test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 26 27 avg. bitrate: 1479.1 1513.5 (kbit/s) errors: 0 0 pkt.report: 1 2 lost: 0 0 jitter: 0.0 0.0 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 30 26 avg. bitrate: 1505.8 1322.3 (kbit/s) errors: 0 0 pkt.report: 2 1 lost: 0 0 jitter: 0.0 0.0 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: ausine auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:58890'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:58890) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:58890) a@127.0.0.1: Call answered: sip:b@127.0.0.1:58890 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:58890) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format S16LE audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock a@127.0.0.1: Call established: sip:b@127.0.0.1:58890 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:58890) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format S16LE audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_auframe: 3 met: yes test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:b@127.0.0.1:58890: Call with sip:a@127.0.0.1 terminated (duration: 1 sec) audio Transmit: Receive: packets: 25 25 avg. bitrate: 3023.6 3000.0 (kbit/s) errors: 0 0 pkt.report: 1 1 lost: 0 0 jitter: 0.0 0.0 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:a@127.0.0.1: Call with sip:b@127.0.0.1:58890 terminated (duration: 1 sec) audio Transmit: Receive: packets: 27 25 avg. bitrate: 2983.5 2782.6 (kbit/s) errors: 0 0 pkt.report: 1 1 lost: 0 0 jitter: 0.0 0.0 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: ausine auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:55432'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:55432) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:55432) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:55432 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:55432) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format FLOAT audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format FLOAT audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock a@127.0.0.1: Call established: sip:b@127.0.0.1:55432 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:55432) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format FLOAT audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format FLOAT audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_auframe: 3 met: yes test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 31 32 avg. bitrate: 1526.1 1575.3 (kbit/s) errors: 0 0 pkt.report: 2 1 lost: 0 0 jitter: 0.0 0.0 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 33 31 avg. bitrate: 1473.4 1392.2 (kbit/s) errors: 0 0 pkt.report: 1 1 lost: 0 0 jitter: 0.0 0.0 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: ausine auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:40661'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:40661) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:40661) a@127.0.0.1: Call answered: sip:b@127.0.0.1:40661 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:40661) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format FLOAT audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format FLOAT audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock a@127.0.0.1: Call established: sip:b@127.0.0.1:40661 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:40661) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: aucmock 48000Hz 2ch audio_recv: player started with sample format FLOAT audio: Set audio encoder: aucmock 48000Hz 2ch ausine: 16000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=5 sampc=80 audio: source started with sample format FLOAT audio tx pipeline: ausine ---> aubuf ---> auconv ---> auresamp ---> aucmock audio rx pipeline: mock-auplay <--- aubuf <--- auconv <--- auresamp <--- aucmock b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_auframe: 3 met: yes test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 26 26 avg. bitrate: 3025.4 3002.7 (kbit/s) errors: 0 0 pkt.report: 1 1 lost: 0 0 jitter: 0.0 0.0 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 27 26 avg. bitrate: 2983.5 2893.9 (kbit/s) errors: 0 0 pkt.report: 1 1 lost: 0 0 jitter: 0.0 0.0 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so unloading module: auresamp.so unloading module: auconv.so [ OK ] [ RUN ] test_call_max (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:35528'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:35528) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) call: connecting to 'sip:b@127.0.0.1:35528'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:35528) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) ua: rejected call from sip:a@127.0.0.1 (maximum 3 calls) test: [ ] event: SIPSESS_FAILED (486 Max Calls) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:35528) a@127.0.0.1: Call answered: sip:b@127.0.0.1:35528 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:35528) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:35528 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:35528) sip:b@127.0.0.1:35528: session closed: 486 Max Calls test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (486 Max Calls) test: re_cancel on call failed rtp_receiver: RTP RX thread started test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: re_cancel on call failed test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: re_cancel on call failed unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_mediaenc (rx thread) mediaenc: srtp mediaenc: srtp-mand mediaenc: srtp-mandf SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using media encryption 'srtp' b@127.0.0.1: Using media encryption 'srtp' ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:51409'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:51409) stream: audio: starting mediaenc 'srtp' (wait_secure=0) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer stream: audio: starting mediaenc 'srtp' (wait_secure=0) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' stream: audio: starting mediaenc 'srtp' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:51409) a@127.0.0.1: Call answered: sip:b@127.0.0.1:51409 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:51409) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: audio: starting mediaenc 'srtp' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch a@127.0.0.1: Call established: sip:b@127.0.0.1:51409 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:51409) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 1 2 avg. bitrate: 9.1 18.2 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 4 1 avg. bitrate: 11.1 2.7 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so unloading module: srtp.so mediaenc: srtp mediaenc: srtp-mand mediaenc: srtp-mandf SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using media encryption 'srtp' b@127.0.0.1: Using media encryption 'srtp' ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:46645'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:46645) stream: audio: starting mediaenc 'srtp' (wait_secure=0) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer stream: audio: starting mediaenc 'srtp' (wait_secure=0) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' audio: RTP/RTCP multiplexing enabled stream: audio: starting mediaenc 'srtp' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:46645) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:46645 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:46645) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio: RTP/RTCP multiplexing enabled stream: audio: starting mediaenc 'srtp' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch a@127.0.0.1: Call established: sip:b@127.0.0.1:46645 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:46645) audio_recv: Set audio decoder: PCMU 8000Hz 1ch b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) rtp_receiver: RTP RX thread started test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 1 2 avg. bitrate: 9.1 18.2 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 4 1 avg. bitrate: 11.1 2.7 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so unloading module: srtp.so dtls_srtp: use prime256v1 for elliptic curve cryptography mediaenc: dtls_srtp SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using media encryption 'dtls_srtp' b@127.0.0.1: Using media encryption 'dtls_srtp' ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:44419'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:44419) stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) dtls_srtp: media=audio -- start DTLS client dtls_srtp: 'audio,RTP' dtls connect to 127.0.0.1:11280 dtls_srtp: 'audio,RTCP' dtls connect to 127.0.0.1:11281 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:44419) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:44419 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:44419) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) dtls_srtp: media=audio -- start DTLS server a@127.0.0.1: Call established: sip:b@127.0.0.1:44419 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:44419) b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) rtp_receiver: RTP RX thread started dtls_srtp: audio: incoming DTLS connect from 127.0.0.1:42556 dtls_srtp: audio: incoming DTLS connect from 127.0.0.1:42557 dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTP) Profile=AES_CM_128_HMAC_SHA1_80 audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,RTP) dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTP) Profile=AES_CM_128_HMAC_SHA1_80 audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,RTP) dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTCP) Profile=AES_CM_128_HMAC_SHA1_80 audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,RTCP) dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTCP) Profile=AES_CM_128_HMAC_SHA1_80 audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,RTCP) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:b@127.0.0.1:44419: Call with sip:a@127.0.0.1 terminated (duration: 1 sec) audio Transmit: Receive: packets: 1 2 avg. bitrate: 5.8 9.8 (kbit/s) errors: 0 0 dtls_srtp: dtls-connection closed (Connection reset by peer [104]) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) dtls_srtp: dtls-connection closed (Connection reset by peer [104]) sip:a@127.0.0.1: Call with sip:b@127.0.0.1:44419 terminated (duration: 1 sec) audio Transmit: Receive: packets: 4 1 avg. bitrate: 9.8 2.6 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so unloading module: dtls_srtp.so dtls_srtp: use prime256v1 for elliptic curve cryptography mediaenc: dtls_srtp SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using media encryption 'dtls_srtp' b@127.0.0.1: Using media encryption 'dtls_srtp' ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:57522'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:57522) stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' audio: RTP/RTCP multiplexing enabled stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) dtls_srtp: media=audio -- start DTLS client dtls_srtp: 'audio,RTP' dtls connect to 127.0.0.1:4036 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:57522) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:57522 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:57522) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio: RTP/RTCP multiplexing enabled stream: audio: starting mediaenc 'dtls_srtp' (wait_secure=1) dtls_srtp: media=audio -- start DTLS server a@127.0.0.1: Call established: sip:b@127.0.0.1:57522 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:57522) b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) rtp_receiver: RTP RX thread started dtls_srtp: audio: incoming DTLS connect from 127.0.0.1:4362 dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTP) Profile=AES_CM_128_HMAC_SHA1_80 audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch dtls_srtp: verified SHA-256 fingerprint OK dtls_srtp: ---> DTLS-SRTP complete (audio/RTP) Profile=AES_CM_128_HMAC_SHA1_80 audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,RTP) ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,RTP) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes AND --> --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:b@127.0.0.1:57522: Call with sip:a@127.0.0.1 terminated (duration: 1 sec) audio Transmit: Receive: packets: 1 2 avg. bitrate: 5.8 16.0 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) dtls_srtp: dtls-connection closed (Connection reset by peer [104]) sip:a@127.0.0.1: Call with sip:b@127.0.0.1:57522 terminated (duration: 1 sec) audio Transmit: Receive: packets: 4 1 avg. bitrate: 13.4 2.9 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so unloading module: dtls_srtp.so [ OK ] [ RUN ] test_call_medianat (rx thread) medianat: XNAT SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using medianat 'XNAT' b@127.0.0.1: Using medianat 'XNAT' ausrc: ausine call: connecting to 'sip:b@127.0.0.1:50522'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:50522) call: media-nat 'XNAT' established/gathered test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) call: media-nat 'XNAT' established/gathered test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:50522) stream: 'audio' mnat 'XNAT' connected: raddr 127.0.0.1:33420 127.0.0.1:33421 audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call answered: sip:b@127.0.0.1:50522 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:50522) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' a@127.0.0.1: Call established: sip:b@127.0.0.1:50522 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:50522) stream: 'audio' mnat 'XNAT' connected: raddr 127.0.0.1:24510 127.0.0.1:24511 audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch rtp_receiver: RTP RX thread started ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU audio_recv: Set audio decoder: PCMU 8000Hz 1ch b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) rtp_receiver: RTP RX thread started test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 1 1 avg. bitrate: 16.0 10.6 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 1 1 avg. bitrate: 7.1 9.1 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so [ OK ] [ RUN ] test_call_mixausrc (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: aufile auplay: aufile aufilt: mixausrc auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:54488'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:54488) audio: using account specific source: (aufile,../test/data//wav/square_500Hz_0.8.wav) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific source: (aufile,../test/data//wav/square_500Hz_0.8.wav) audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:54488) a@127.0.0.1: Call answered: sip:b@127.0.0.1:54488 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:54488) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch aufile: loading input file '../test/data//wav/square_500Hz_0.8.wav' aufile: ../test/data//wav/square_500Hz_0.8.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=10 sampc=80 aufile: read end of file aufile: loaded 16000 bytes audio: source started with sample format S16LE audio tx pipeline: aufile ---> aubuf ---> mixausrc ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- mixausrc <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:54488 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:54488) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch aufile: loading input file '../test/data//wav/square_500Hz_0.8.wav' aufile: ../test/data//wav/square_500Hz_0.8.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=10 sampc=80 aufile: read end of file aufile: loaded 16000 bytes audio: source started with sample format S16LE audio tx pipeline: aufile ---> aubuf ---> mixausrc ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- mixausrc <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [auframe]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [auframe]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_auframe: 3 met: yes test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [auframe]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM prm=[auframe 4] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM prm=[auframe 4] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [mixed]) aufile: loading input file '../test/data//wav/square_1kHz_0.4.wav' aufile: ../test/data//wav/square_1kHz_0.4.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=40 sampc=320 aufile: read end of file aufile: loaded 16000 bytes test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 5) test: event CUSTOM prm=[auframe 5] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 5) test: event CUSTOM prm=[auframe 5] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 6) test: event CUSTOM prm=[auframe 6] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 6) test: event CUSTOM prm=[auframe 6] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 7) test: event CUSTOM prm=[auframe 7] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 7) test: event CUSTOM prm=[auframe 7] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 8) test: event CUSTOM prm=[auframe 8] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 8) test: event CUSTOM prm=[auframe 8] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 9) test: event CUSTOM prm=[auframe 9] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 9) test: event CUSTOM prm=[auframe 9] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 10) test: event CUSTOM prm=[auframe 10] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 10) test: event CUSTOM prm=[auframe 10] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 11) test: event CUSTOM prm=[auframe 11] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 11) test: event CUSTOM prm=[auframe 11] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 12) test: event CUSTOM prm=[auframe 12] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 12) test: event CUSTOM prm=[auframe 12] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (mixed) canceled by --- CUSTOM --- prm: mixed ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 met: yes test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 13) test: event CUSTOM prm=[auframe 13] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 13) test: event CUSTOM prm=[auframe 13] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (mixed) test: event CUSTOM prm=[mixed] (expected [original]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 14) test: event CUSTOM prm=[auframe 14] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 14) test: event CUSTOM prm=[auframe 14] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (mixed) test: event CUSTOM prm=[mixed] (expected [original]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 15) test: event CUSTOM prm=[auframe 15] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 15) test: event CUSTOM prm=[auframe 15] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (mixed) test: event CUSTOM prm=[mixed] (expected [original]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 16) test: event CUSTOM prm=[auframe 16] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 16) test: event CUSTOM prm=[auframe 16] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [original]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 17) test: event CUSTOM prm=[auframe 17] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 17) test: event CUSTOM prm=[auframe 17] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) canceled by --- CUSTOM --- prm: original ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 19 20 avg. bitrate: 61.7 64.3 (kbit/s) errors: 0 0 pkt.report: 1 1 lost: 0 0 jitter: 0.0 0.0 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 21 19 avg. bitrate: 64.0 58.4 (kbit/s) errors: 0 0 pkt.report: 1 1 lost: 0 0 jitter: 0.0 0.0 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: mixausrc.so unloading module: aufile.so SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: aufile auplay: aufile aufilt: mixausrc auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:36739'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:36739) audio: using account specific source: (aufile,../test/data//wav/square_500Hz_0.8.wav) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific source: (aufile,../test/data//wav/square_500Hz_0.8.wav) audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:36739) a@127.0.0.1: Call answered: sip:b@127.0.0.1:36739 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:36739) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch aufile: loading input file '../test/data//wav/square_500Hz_0.8.wav' aufile: ../test/data//wav/square_500Hz_0.8.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=10 sampc=80 aufile: read end of file aufile: loaded 16000 bytes audio: source started with sample format S16LE audio tx pipeline: aufile ---> aubuf ---> mixausrc ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- mixausrc <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:36739 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:36739) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch aufile: loading input file '../test/data//wav/square_500Hz_0.8.wav' aufile: ../test/data//wav/square_500Hz_0.8.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=10 sampc=80 aufile: read end of file aufile: loaded 16000 bytes audio: source started with sample format S16LE audio tx pipeline: aufile ---> aubuf ---> mixausrc ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- mixausrc <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [auframe]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [auframe]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_auframe: 3 met: yes test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [auframe]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM prm=[auframe 4] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM prm=[auframe 4] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) test: event CUSTOM prm=[original] (expected [mixed]) aufile: loading input file '../test/data//wav/square_1kHz_0.4.wav' aufile: ../test/data//wav/square_1kHz_0.4.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=40 sampc=320 aufile: read end of file aufile: loaded 16000 bytes test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 5) test: event CUSTOM prm=[auframe 5] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 5) test: event CUSTOM prm=[auframe 5] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 6) test: event CUSTOM prm=[auframe 6] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 6) test: event CUSTOM prm=[auframe 6] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 7) test: event CUSTOM prm=[auframe 7] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 7) test: event CUSTOM prm=[auframe 7] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 8) test: event CUSTOM prm=[auframe 8] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 8) test: event CUSTOM prm=[auframe 8] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 9) test: event CUSTOM prm=[auframe 9] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 9) test: event CUSTOM prm=[auframe 9] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 10) test: event CUSTOM prm=[auframe 10] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 10) test: event CUSTOM prm=[auframe 10] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [mixed]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 11) test: event CUSTOM prm=[auframe 11] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 11) test: event CUSTOM prm=[auframe 11] (expected [mixed]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (mixed) canceled by --- CUSTOM --- prm: mixed ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 met: yes test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 12) test: event CUSTOM prm=[auframe 12] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 12) test: event CUSTOM prm=[auframe 12] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (mixed) test: event CUSTOM prm=[mixed] (expected [original]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 13) test: event CUSTOM prm=[auframe 13] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 13) test: event CUSTOM prm=[auframe 13] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (mixed) test: event CUSTOM prm=[mixed] (expected [original]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 14) test: event CUSTOM prm=[auframe 14] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 14) test: event CUSTOM prm=[auframe 14] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (low) test: event CUSTOM prm=[low] (expected [original]) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 15) test: event CUSTOM prm=[auframe 15] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 15) test: event CUSTOM prm=[auframe 15] (expected [original]) test: [ sip:b@127.0.0.1 ] event: CUSTOM (original) canceled by --- CUSTOM --- prm: original ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:b@127.0.0.1:36739: Call with sip:a@127.0.0.1 terminated (duration: 1 sec) audio Transmit: Receive: packets: 16 18 avg. bitrate: 59.5 66.2 (kbit/s) errors: 0 0 pkt.report: 1 1 lost: 0 0 jitter: 0.0 0.0 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:a@127.0.0.1: Call with sip:b@127.0.0.1:36739 terminated (duration: 1 sec) audio Transmit: Receive: packets: 18 16 avg. bitrate: 62.2 55.9 (kbit/s) errors: 0 0 pkt.report: 1 1 lost: 0 0 jitter: 0.0 0.0 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: mixausrc.so unloading module: aufile.so [ OK ] [ RUN ] test_call_multiple (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:57154'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:57154) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) call: connecting to 'sip:b@127.0.0.1:57154'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:57154) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) call: connecting to 'sip:b@127.0.0.1:57154'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:57154) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) call: connecting to 'sip:b@127.0.0.1:57154'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:57154) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 2 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started rtp_receiver: RTP RX thread started test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 3 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 4 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:57154) a@127.0.0.1: Call answered: sip:b@127.0.0.1:57154 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:57154) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:57154 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:57154) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:57154) a@127.0.0.1: Call answered: sip:b@127.0.0.1:57154 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:57154) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:57154 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:57154) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:57154) a@127.0.0.1: Call answered: sip:b@127.0.0.1:57154 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:57154) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:57154 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:57154) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:57154) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:57154 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:57154) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:57154 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:57154) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) call: terminate call '1c50ab5648a8e572' with sip:b@127.0.0.1:57154 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) call: terminate call '60b7cbcd03771ed9' with sip:b@127.0.0.1:57154 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) call: connecting to 'sip:b@127.0.0.1:57154'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:57154) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) call: connecting to 'sip:b@127.0.0.1:57154'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:57154) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 2 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) rtp_receiver: RTP RX thread started test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 4 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:57154) a@127.0.0.1: Call answered: sip:b@127.0.0.1:57154 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:57154) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:57154 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:57154) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:57154) a@127.0.0.1: Call answered: sip:b@127.0.0.1:57154 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:57154) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:57154 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:57154) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_progress (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:44206'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:44206) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:44206) rtp_receiver: RTP RX thread started call: SIP Progress: 183 Session Progress (application/sdp) a@127.0.0.1: Call in-progress: sip:b@127.0.0.1:44206 test: [ sip:a@127.0.0.1 ] event: CALL_PROGRESS (sip:b@127.0.0.1:44206) canceled by --- CALL_PROGRESS --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_reject (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:35703'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:35703) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: rejecting incoming call from sip:a@127.0.0.1 (486 Busy Here) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:35703) sip:b@127.0.0.1:35703: session closed: 486 Busy Here test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (486 Busy Here) test: re_cancel on call failed unloading module: g711.so ua: stop all (forced=1) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:53212'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:53212) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: rejecting incoming call from sip:a@127.0.0.1 (302 Moved Temporarily) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Moved Temporarily) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:53212) call: redirect to sip:c@127.0.0.1 test: [ sip:a@127.0.0.1 ] event: CALL_REDIRECT (302,sip:c@127.0.0.1) sip:b@127.0.0.1:53212: session closed: 302 Moved Temporarily test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (302 Moved Temporarily) test: re_cancel on call failed unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_cancel (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:38442'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:38442) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) call: terminate call 'f28ae414cb17711a' with sip:b@127.0.0.1:38442 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) canceled by --- CALL_CLOSED --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 0 n_closed: 1 met: yes test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: rejecting incoming call from sip:a@127.0.0.1 (486 Busy Here) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) test: event CALL_CLOSED ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] unloading module: g711.so ua: stop all (forced=1) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:53936'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:53936) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:53936) rtp_receiver: RTP RX thread started call: SIP Progress: 183 Session Progress (application/sdp) a@127.0.0.1: Call in-progress: sip:b@127.0.0.1:53936 test: [ sip:a@127.0.0.1 ] event: CALL_PROGRESS (sip:b@127.0.0.1:53936) canceled by --- CALL_PROGRESS --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU call: terminate call '2ac164faed82d4a0' with sip:b@127.0.0.1:53936 rtp_receiver: RTP RX thread started test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) unloading module: g711.so ua: stop all (forced=1) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:59029'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:59029) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:59029) call: SIP Progress: 183 Session Progress (application/sdp) a@127.0.0.1: Call in-progress: sip:b@127.0.0.1:59029 test: [ sip:a@127.0.0.1 ] event: CALL_PROGRESS (sip:b@127.0.0.1:59029) canceled by --- CALL_PROGRESS --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: terminate call '1c86843f3859b736' with sip:b@127.0.0.1:59029 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) audio_recv: Set audio decoder: PCMU 8000Hz 1ch b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) rtp_receiver: RTP RX thread started unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_rtcp (rx thread) ausrc: ausine SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:40573'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:40573) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:40573) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:40573 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:40573) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:40573 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:40573) test: event CALL_ESTABLISHED ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) canceled by --- CALL_ESTABLISHED --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 4) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 4) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 5) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 5) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 6) test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 6) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) canceled by --- CALL_RTCP --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_rtcp: 5 met: yes AND --> --- CALL_RTCP --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_rtcp: 5 met: yes AND --> --- CUSTOM --- prm: audebug ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_audebug: 5 met: yes AND --> --- CUSTOM --- prm: audebug ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_audebug: 5 met: yes test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 7) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 3 4 avg. bitrate: 10.6 13.4 (kbit/s) errors: 0 0 pkt.report: 3 3 lost: 0 0 jitter: 0.5 0.5 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 6 3 avg. bitrate: 11.6 6.0 (kbit/s) errors: 0 0 pkt.report: 3 3 lost: 0 0 jitter: 0.3 0.5 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so ausrc: ausine SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:35828'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:35828) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' audio: RTP/RTCP multiplexing enabled test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:35828) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:35828 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:35828) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio: RTP/RTCP multiplexing enabled audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:35828 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:35828) test: event CALL_ESTABLISHED ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) canceled by --- CALL_ESTABLISHED --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 4) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 4) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 5) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 5) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 6) test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 6) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: event CALL_RTCP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (audebug 7) test: [ sip:b@127.0.0.1 ] event: CUSTOM (audebug 7) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) canceled by --- CALL_RTCP --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_rtcp: 5 met: yes AND --> --- CALL_RTCP --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_rtcp: 5 met: yes AND --> --- CUSTOM --- prm: audebug ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_audebug: 5 met: yes AND --> --- CUSTOM --- prm: audebug ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_audebug: 5 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 4 7 avg. bitrate: 10.2 17.2 (kbit/s) errors: 0 0 pkt.report: 3 4 lost: 0 0 jitter: 0.5 0.5 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 9 4 avg. bitrate: 14.0 6.4 (kbit/s) errors: 0 0 pkt.report: 4 3 lost: 0 0 jitter: 0.5 0.5 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so [ OK ] [ RUN ] test_call_rtp_timeout (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:39827'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:39827) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:39827) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:39827 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:39827) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU stream: Enable RTP timeout (1 milliseconds) a@127.0.0.1: Call established: sip:b@127.0.0.1:39827 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:39827) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) stream: no audio RTP packets received for 2 milliseconds call: error in "audio" rtp stream (Connection timed out [110]) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (rtp stream error) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_tcp (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:39615;transport=tcp'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:39615;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:39615;transport=tcp) a@127.0.0.1: Call answered: sip:b@127.0.0.1:39615;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:39615;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:39615;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:39615;transport=tcp) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) rtp_receiver: RTP RX thread started test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_deny_udp (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:a@127.0.0.1 ] event: CREATE (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) call: connecting to 'sip:b@127.0.0.1:59787;transport=udp'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:59787;transport=udp) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) ua: sip:a@127.0.0.1: UA not found: sip:b@127.0.0.1:59787;transport=udp test: [ ] event: SIPSESS_FAILED (404 Not Found) sip:b@127.0.0.1:59787;transport=udp: session closed: 404 Not Found test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (404 Not Found) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_transfer (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:c@127.0.0.1 ] event: CREATE (sip:c@127.0.0.1) call: connecting to 'sip:b@127.0.0.1:48696'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:48696) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:48696) a@127.0.0.1: Call answered: sip:b@127.0.0.1:48696 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:48696) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:48696 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:48696) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) call: hold sip:b@127.0.0.1:48696 test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' transferring call to sip:c@127.0.0.1:48696 test: [ sip:b@127.0.0.1 ] event: CALL_HOLD () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: TRANSFER (sip:c@127.0.0.1:48696) call: connecting to 'sip:c@127.0.0.1:48696'.. test: [ sip:b@127.0.0.1 ] event: CALL_OUTGOING (sip:c@127.0.0.1:48696) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:c@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:c@127.0.0.1 ] event: CALL_INCOMING (sip:b@127.0.0.1:48696) call: answering call on line 1 from sip:b@127.0.0.1:48696 with 200 stream: update 'audio' test: [ sip:c@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:b@127.0.0.1 ] event: CALL_RINGING (sip:c@127.0.0.1:48696) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) b@127.0.0.1: Call answered: sip:c@127.0.0.1:48696 test: [ sip:b@127.0.0.1 ] event: CALL_ANSWERED (sip:c@127.0.0.1:48696) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:c@127.0.0.1:48696 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:c@127.0.0.1:48696) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU c@127.0.0.1: Call established: sip:b@127.0.0.1:48696 test: [ sip:c@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:48696) rtp_receiver: RTP RX thread started test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Call transfered) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) test: [ sip:c@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_transfer_fail (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:c@127.0.0.1 ] event: CREATE (sip:c@127.0.0.1) call: connecting to 'sip:b@127.0.0.1:46747'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:46747) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:46747) a@127.0.0.1: Call answered: sip:b@127.0.0.1:46747 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:46747) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:46747 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:46747) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) call: hold sip:b@127.0.0.1:46747 test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' transferring call to sip:c@127.0.0.1:46747 test: [ sip:b@127.0.0.1 ] event: CALL_HOLD () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: TRANSFER (sip:c@127.0.0.1:46747) call: connecting to 'sip:c@127.0.0.1:46747'.. test: [ sip:b@127.0.0.1 ] event: CALL_OUTGOING (sip:c@127.0.0.1:46747) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:c@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:c@127.0.0.1 ] event: CALL_INCOMING (sip:b@127.0.0.1:46747) call: rejecting incoming call from sip:b@127.0.0.1:46747 (486 Busy Here) test: [ sip:c@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) call: SIP Progress: 180 Ringing (/) test: [ sip:b@127.0.0.1 ] event: CALL_RINGING (sip:c@127.0.0.1:46747) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) sip:c@127.0.0.1:46747: session closed: 486 Busy Here test: [ sip:b@127.0.0.1 ] event: TRANSFER_FAILED (486 Busy Here) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (486 Busy Here) call: transfer failed: 486 Busy Here test: [ sip:a@127.0.0.1 ] event: TRANSFER_FAILED (486 Busy Here) call: resume sip:b@127.0.0.1:46747 test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_attended_transfer (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:c@127.0.0.1 ] event: CREATE (sip:c@127.0.0.1) call: connecting to 'sip:b@127.0.0.1:44930'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:44930) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:44930) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:44930 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:44930) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:44930 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:44930) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) call: hold sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' call: connecting to 'sip:c@127.0.0.1:44930'.. test: [ sip:b@127.0.0.1 ] event: CALL_OUTGOING (sip:c@127.0.0.1:44930) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ sip:a@127.0.0.1 ] event: CALL_HOLD () test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:c@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:c@127.0.0.1 ] event: CALL_INCOMING (sip:b@127.0.0.1) call: answering call on line 1 from sip:b@127.0.0.1 with 200 stream: update 'audio' test: [ sip:c@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' call: SIP Progress: 180 Ringing (/) test: [ sip:b@127.0.0.1 ] event: CALL_RINGING (sip:c@127.0.0.1:44930) b@127.0.0.1: Call answered: sip:c@127.0.0.1:44930 test: [ sip:b@127.0.0.1 ] event: CALL_ANSWERED (sip:c@127.0.0.1:44930) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:c@127.0.0.1:44930 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:c@127.0.0.1:44930) call: hold sip:c@127.0.0.1:44930 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' transferring call to sip:c@127.0.0.1:44930 test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU c@127.0.0.1: Call established: sip:b@127.0.0.1 test: [ sip:c@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1) test: [ sip:c@127.0.0.1 ] event: CALL_HOLD () test: [ sip:c@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:c@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: TRANSFER () call: connecting to ''.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:c@127.0.0.1:44930) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:c@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:c@127.0.0.1 ] event: CALL_CLOSED (395e7cb44086d730;to-tag=6f3ff543e910a295;from-tag=9118f0082f502787 replaced) test: [ sip:c@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 2 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:c@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:c@127.0.0.1:44930) a@127.0.0.1: Call answered: sip:c@127.0.0.1:44930 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:c@127.0.0.1:44930) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:c@127.0.0.1:44930 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:c@127.0.0.1:44930) rtp_receiver: RTP RX thread started sip:c@127.0.0.1:44930: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU c@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:c@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Call transfered) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) sip:b@127.0.0.1:44930: session closed: Connection reset by peer [104] test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) test: [ sip:c@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_video (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 vidcodec: H266 vidisp: mock-vidisp vidsrc: fakevideo vidisp: fakevideo call: connecting to 'sip:b@127.0.0.1:49191'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:49191) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' stream: update 'video' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:49191) a@127.0.0.1: Call answered: sip:b@127.0.0.1:49191 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:49191) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 a@127.0.0.1: Call established: sip:b@127.0.0.1:49191 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:49191) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (video) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (video) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) video: receiving with resolution 640 x 480 and format 'yuv420p' video: receiving with resolution 640 x 480 and format 'yuv420p' mock_vidisp: got 10 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 10 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 3) canceled by --- CUSTOM --- prm: vidframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 n_vidframe: 3 met: yes test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 22 21 avg. bitrate: 11.0 10.9 (kbit/s) errors: 0 0 pkt.report: 1 0 lost: 0 0 jitter: 0.0 0.0 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 22 22 avg. bitrate: 11.0 11.0 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: fakevideo.so [ OK ] [ RUN ] test_call_change_videodir (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 vidcodec: H266 vidisp: mock-vidisp vidsrc: fakevideo vidisp: fakevideo call: connecting to 'sip:b@127.0.0.1:45420'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:45420) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:45420) call: SIP Progress: 183 Session Progress (application/sdp) a@127.0.0.1: Call in-progress: sip:b@127.0.0.1:45420 test: [ sip:a@127.0.0.1 ] event: CALL_PROGRESS (sip:b@127.0.0.1:45420) canceled by --- CALL_PROGRESS --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' rtp_receiver: RTP RX thread started stream: update 'video' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) a@127.0.0.1: Call answered: sip:b@127.0.0.1:45420 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:45420) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch a@127.0.0.1: Call established: sip:b@127.0.0.1:45420 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:45420) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (video) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (video) audio_recv: Set audio decoder: PCMU 8000Hz 1ch b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) video: receiving with resolution 640 x 480 and format 'yuv420p' video: receiving with resolution 640 x 480 and format 'yuv420p' mock_vidisp: got 10 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 10 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 3) canceled by --- CUSTOM --- prm: vidframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_vidframe: 3 met: yes AND --> --- CUSTOM --- prm: vidframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 1 n_vidframe: 3 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 call: got INVITE (SDP Offer) audio-video: sendrecv-sendrecv test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) mock_vidisp: got 10 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) mock_vidisp: got 11 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 13 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 4) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 14 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 5) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 15 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 6) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 16 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 7) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 17 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 8) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 18 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 9) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 10 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 1) mock_vidisp: got 19 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 10) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 2) mock_vidisp: got 20 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 11) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 3) canceled by --- CUSTOM --- prm: vidframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_vidframe: 3 met: yes AND --> --- CUSTOM --- prm: vidframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 1 n_vidframe: 3 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:b@127.0.0.1:45420: Call with sip:a@127.0.0.1 terminated (duration: 1 sec) video Transmit: Receive: packets: 44 43 avg. bitrate: 10.8 10.7 (kbit/s) errors: 0 0 pkt.report: 34 33 lost: 0 0 jitter: 1.8 1.5 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) sip:a@127.0.0.1: Call with sip:b@127.0.0.1:45420 terminated (duration: 1 sec) video Transmit: Receive: packets: 44 42 avg. bitrate: 10.8 10.5 (kbit/s) errors: 0 0 pkt.report: 23 23 lost: 0 0 jitter: 1.4 1.7 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: fakevideo.so [ OK ] [ RUN ] test_call_webrtc (rx thread) [ OK ] [ RUN ] test_call_bundle (rx thread) [ OK ] [ RUN ] test_call_ipv6ll (rx thread) no IPv6 address -- skipping test test_call_ipv6ll [ OK ] [ RUN ] test_call_100rel_audio (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) ausrc: ausine auplay: mock-auplay call: connecting to 'sip:b@127.0.0.1:47094'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:47094) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:47094) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU rtp_receiver: RTP RX thread started call: SIP Progress: 183 Session Progress (application/sdp) a@127.0.0.1: Call in-progress: sip:b@127.0.0.1:47094 test: [ sip:a@127.0.0.1 ] event: CALL_PROGRESS (sip:b@127.0.0.1:47094) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU rtp_receiver: RTP RX thread started test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 5) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_established: 0 n_auframe: 3 met: yes AND --> --- CUSTOM --- prm: auframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 n_auframe: 3 met: yes test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 6) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' call: got UPDATE (SDP Offer) audio-video: inactive-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU call: got UPDATE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) canceled by --- CUSTOM --- prm: auframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 n_auframe: 3 met: yes test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 34 32 avg. bitrate: 14.4 13.9 (kbit/s) errors: 0 0 pkt.report: 2 1 lost: 0 0 jitter: 0.1 0.0 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 34 33 avg. bitrate: 13.7 13.0 (kbit/s) errors: 0 0 pkt.report: 30 33 lost: 0 0 jitter: 2.5 3.1 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so [ OK ] [ RUN ] test_call_100rel_video (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 vidcodec: H266 vidisp: mock-vidisp vidsrc: fakevideo vidisp: fakevideo call: connecting to 'sip:b@127.0.0.1:40498'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:40498) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:40498) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 rtp_receiver: RTP RX thread started call: SIP Progress: 183 Session Progress (application/sdp) a@127.0.0.1: Call in-progress: sip:b@127.0.0.1:40498 test: [ sip:a@127.0.0.1 ] event: CALL_PROGRESS (sip:b@127.0.0.1:40498) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU Set video encoder: H266 (1000000 bit/s, 100.00 fps) Set video decoder: H266 video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (video) rtp_receiver: RTP RX thread started test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (video) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) video: receiving with resolution 640 x 480 and format 'yuv420p' video: receiving with resolution 640 x 480 and format 'yuv420p' mock_vidisp: got 10 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 10 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 3) canceled by --- CUSTOM --- prm: vidframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 n_vidframe: 3 met: yes test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got UPDATE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 call: got UPDATE (SDP Offer) audio-video: sendrecv-sendrecv test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: update 'video' audio_recv: Set audio decoder: PCMU 8000Hz 1ch video tx pipeline: fakevideo ---> H266 video rx pipeline: mock-vidisp <--- H266 test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) mock_vidisp: got 10 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (video) mock_vidisp: got 11 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 13 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 4) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 14 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 5) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 15 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 6) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 16 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 7) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 17 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 8) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 18 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 9) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 10 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 1) mock_vidisp: got 19 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 10) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 11 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 2) mock_vidisp: got 20 frames test: [ sip:a@127.0.0.1 ] event: CUSTOM (vidframe 11) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] mock_vidisp: got 12 frames test: [ sip:b@127.0.0.1 ] event: CUSTOM (vidframe 3) canceled by --- CUSTOM --- prm: vidframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 0 n_vidframe: 3 met: yes AND --> --- CUSTOM --- prm: vidframe ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 1 n_established: 0 n_vidframe: 3 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 43 42 avg. bitrate: 10.8 10.7 (kbit/s) errors: 0 0 pkt.report: 33 31 lost: 0 0 jitter: 1.9 1.7 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) video Transmit: Receive: packets: 42 43 avg. bitrate: 10.5 10.7 (kbit/s) errors: 0 0 pkt.report: 22 22 lost: 0 0 jitter: 1.4 1.5 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: fakevideo.so [ OK ] [ RUN ] test_call_100rel_required_reject (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) call: connecting to 'sip:b@127.0.0.1:55811'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:55811) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) ua: call from sip:a@127.0.0.1 rejected with 421 -- option-tag '100rel' not supported by remote sip:b@127.0.0.1:55811: session closed: 421 Extension required test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (421 Extension required) test: re_cancel on call failed unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_hold_resume (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:56825'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:56825) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:56825) a@127.0.0.1: Call answered: sip:b@127.0.0.1:56825 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:56825) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:56825 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:56825) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) rtp_receiver: RTP RX thread started test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_audio_estab: 1 met: yes test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] call: hold sip:b@127.0.0.1:56825 test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_HOLD () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 1 met: yes call: resume sip:b@127.0.0.1:56825 test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RESUME () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE call: got INVITE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 1 met: yes call: terminate call '8758e4d2cd25d9cb' with sip:b@127.0.0.1:56825 test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) canceled by --- CALL_CLOSED --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes audio Transmit: Receive: packets: 2 1 avg. bitrate: 43.3 22.4 (kbit/s) errors: 0 0 pkt.report: 2 1 lost: 0 0 jitter: 0.0 0.0 (ms) call: connecting to 'sip:b@127.0.0.1:34295;transport=tcp'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:34295;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:34295;transport=tcp) a@127.0.0.1: Call answered: sip:b@127.0.0.1:34295;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:34295;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE a@127.0.0.1: Call established: sip:b@127.0.0.1:34295;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:34295;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 n_audio_estab: 2 met: yes call: hold sip:b@127.0.0.1:34295;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 2 met: yes call: resume sip:b@127.0.0.1:34295;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 2 met: yes call: hold sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_HOLD () test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:a@127.0.0.1 n_established: 2 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' call: got INVITE (SDP Offer) audio-video: inactive-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:a@127.0.0.1 n_established: 2 met: yes call: resume sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_RESUME () test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE call: got INVITE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE test: [ sip:a@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:a@127.0.0.1 n_established: 2 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 0 4 avg. bitrate: 0.0 21.1 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 1 2 avg. bitrate: 4.0 7.9 (kbit/s) errors: 0 0 pkt.report: 1 2 lost: 0 0 jitter: 0.0 0.6 (ms) audio Transmit: Receive: packets: 6 0 avg. bitrate: 30.3 0.0 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:45099;transport=tcp'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:45099;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:45099;transport=tcp) a@127.0.0.1: Call answered: sip:b@127.0.0.1:45099;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:45099;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:45099;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:45099;transport=tcp) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 n_audio_estab: 1 met: yes AND --> --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_audio_estab: 1 met: yes call: hold sip:b@127.0.0.1:45099;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_HOLD () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 1 met: yes call: resume sip:b@127.0.0.1:45099;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RESUME () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE call: got INVITE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 1 met: yes call: terminate call 'e292cc929a12b928' with sip:b@127.0.0.1:45099;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) sip:a@127.0.0.1: session closed: Connection reset by peer [104] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Connection reset by peer [104]) canceled by --- CALL_CLOSED --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes audio Transmit: Receive: packets: 4 3 avg. bitrate: 33.0 24.4 (kbit/s) errors: 0 0 pkt.report: 3 3 lost: 0 0 jitter: 0.0 0.0 (ms) call: connecting to 'sip:b@127.0.0.1:45099;transport=tcp'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:45099;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:45099;transport=tcp) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:45099;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:45099;transport=tcp) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE a@127.0.0.1: Call established: sip:b@127.0.0.1:45099;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:45099;transport=tcp) audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 n_audio_estab: 2 met: yes call: hold sip:b@127.0.0.1:45099;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 2 met: yes call: resume sip:b@127.0.0.1:45099;transport=tcp test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 2 met: yes call: hold sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_HOLD () test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:a@127.0.0.1 n_established: 2 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' call: got INVITE (SDP Offer) audio-video: inactive-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:a@127.0.0.1 n_established: 2 met: yes call: resume sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_RESUME () test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE call: got INVITE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:b@127.0.0.1 n_incoming: 2 n_progress: 0 n_established: 2 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE test: [ sip:a@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:a@127.0.0.1 n_established: 2 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 0 3 avg. bitrate: 0.0 16.1 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 3 1 avg. bitrate: 8.2 2.7 (kbit/s) errors: 0 0 pkt.report: 3 2 lost: 0 0 jitter: 0.0 0.0 (ms) audio Transmit: Receive: packets: 5 0 avg. bitrate: 25.4 0.0 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:53709'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:53709) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:53709) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:53709 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:53709) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:53709 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:53709) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 n_audio_estab: 1 met: yes AND --> --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_audio_estab: 1 met: yes call: hold sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CALL_HOLD () test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: recvonly-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:a@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:a@127.0.0.1 n_established: 1 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) call: got INVITE audio-video: recvonly-inactive test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) test: event CALL_REMOTE_SDP ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE call: got (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_RESUME () test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) test: event CALL_REMOTE_SDP ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 1 2 avg. bitrate: 27.8 53.3 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 3 1 avg. bitrate: 50.5 17.2 (kbit/s) errors: 0 0 pkt.report: 1 1 lost: 0 0 jitter: 0.0 0.0 (ms) unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 ausrc: ausine ausrc: aufile auplay: aufile call: connecting to 'sip:b@127.0.0.1:34380'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:34380) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:34380) a@127.0.0.1: Call answered: sip:b@127.0.0.1:34380 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:34380) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:34380 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:34380) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: event CALL_RTPESTAB ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) canceled by --- CALL_RTPESTAB --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_audio_estab: 1 met: yes test: event CALL_RTPESTAB ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_HOLD () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' call: got INVITE (SDP Offer) audio-video: inactive-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 1 met: yes test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_RESUME () test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: event CALL_REMOTE_SDP prm=[offer] (expected [answer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU call: got INVITE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) canceled by --- CALL_REMOTE_SDP --- prm: answer ua: sip:a@127.0.0.1 n_incoming: 0 n_progress: 0 n_established: 1 met: yes test: event CALL_REMOTE_SDP prm=[answer] (expected [offer]) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=20 sampc=160 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: aufile <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CUSTOM (gotack) canceled by --- CUSTOM --- prm: gotack ua: sip:b@127.0.0.1 n_established: 1 met: yes test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 1 2 avg. bitrate: 20.9 43.3 (kbit/s) errors: 0 0 pkt.report: 1 1 lost: 0 0 jitter: 0.0 0.0 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 3 1 avg. bitrate: 47.4 15.4 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: aufile.so unloading module: ausine.so [ OK ] [ RUN ] test_call_srtp_tx_rekey (rx thread) mediaenc: srtp mediaenc: srtp-mand mediaenc: srtp-mandf ausrc: ausine auplay: mock-auplay SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 a@127.0.0.1: Using media encryption 'srtp-mand' b@127.0.0.1: Using media encryption 'srtp-mand' b@127.0.0.1: Using media encryption 'srtp-mand' test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) call: connecting to 'sip:b@127.0.0.1:39182'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:39182) stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) audio: using account specific player: (mock-auplay,a) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) audio: using account specific player: (mock-auplay,b) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:39182) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:39182 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:39182) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio_recv: player started with sample format S16LE audio: Set audio encoder: PCMU 8000Hz 1ch ausine: 8000 Hz, 1 channels, frequency 440 Hz CH_MODE: 3 ausine: audio ptime=1 sampc=8 audio: source started with sample format S16LE audio tx pipeline: ausine ---> aubuf ---> PCMU audio rx pipeline: mock-auplay <--- aubuf <--- PCMU test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch a@127.0.0.1: Call established: sip:b@127.0.0.1:39182 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:39182) test: event CALL_ESTABLISHED ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] audio_recv: Set audio decoder: PCMU 8000Hz 1ch b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) canceled by --- CALL_ESTABLISHED --- prm: ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: event CALL_ESTABLISHED ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] rtp_receiver: RTP RX thread started srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) stream: update 'audio' stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) stream: update 'audio' stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) srtp: failed to decrypt RTP packet with 30 bytes (Unknown error 217 [217]) stream: update 'audio' stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) srtp: audio: re-keying in progress srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:b@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch call: got INVITE (SDP Offer) audio-video: sendrecv-inactive test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' stream: audio: starting mediaenc 'srtp-mand' (wait_secure=0) srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_MENC (0,audio,AES_CM_128_HMAC_SHA1_80) audio_recv: Set audio decoder: PCMU 8000Hz 1ch test: [ sip:a@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTPESTAB (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:b@127.0.0.1 ] event: CALL_RTCP (audio) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 1) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 5) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 2) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 6) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 3) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 7) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 4) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 8) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 5) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 9) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 6) test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 10) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 7) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 11) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 8) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 12) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 9) test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 13) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CUSTOM (auframe 10) canceled by --- CUSTOM --- prm: auframe ua: sip:b@127.0.0.1 n_incoming: 1 n_progress: 0 n_established: 1 n_auframe: 10 met: yes test: event CUSTOM ua=[sip:b@127.0.0.1] (expected [sip:a@127.0.0.1] test: [ sip:a@127.0.0.1 ] event: CUSTOM (auframe 14) test: event CUSTOM ua=[sip:a@127.0.0.1] (expected [sip:b@127.0.0.1] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 34 34 avg. bitrate: 15.2 15.6 (kbit/s) errors: 0 0 pkt.report: 2 1 lost: 0 0 jitter: 0.1 0.0 (ms) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 37 34 avg. bitrate: 15.2 14.0 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: ausine.so unloading module: srtp.so [ OK ] [ RUN ] test_call_uag_find_msg (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:alice@127.0.0.1:59942 ] event: CREATE (sip:alice@127.0.0.1:59942) test: [ sip:alice@127.0.0.1:38513 ] event: CREATE (sip:alice@127.0.0.1:38513) test: [ sip:bob@127.0.0.1:38513 ] event: CREATE (sip:bob@127.0.0.1:38513) test: [ sip:alice@127.0.0.1:59942 ] event: REGISTERING () test: [ sip:alice@127.0.0.1:38513 ] event: REGISTERING () test: [ sip:bob@127.0.0.1:38513 ] event: REGISTERING () alice@127.0.0.1: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] test: [ sip:alice@127.0.0.1:59942 ] event: REGISTER_OK (200 OK) test: event REGISTER_OK ua=[sip:alice@127.0.0.1:59942] (expected [sip:bob@127.0.0.1:38513] test: event REGISTER_OK ua=[sip:alice@127.0.0.1:59942] (expected [sip:alice@127.0.0.1:38513] alice@127.0.0.1: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] test: [ sip:alice@127.0.0.1:38513 ] event: REGISTER_OK (200 OK) test: event REGISTER_OK ua=[sip:alice@127.0.0.1:38513] (expected [sip:bob@127.0.0.1:38513] bob@127.0.0.1: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] test: [ sip:bob@127.0.0.1:38513 ] event: REGISTER_OK (200 OK) canceled by --- REGISTER_OK --- prm: ua: sip:bob@127.0.0.1:38513 n_incoming: 0 n_progress: 0 n_established: 0 met: yes test: event REGISTER_OK ua=[sip:bob@127.0.0.1:38513] (expected [sip:alice@127.0.0.1:38513] test: event REGISTER_OK ua=[sip:bob@127.0.0.1:38513] (expected [sip:alice@127.0.0.1:59942] call: connecting to 'sip:alice@127.0.0.1:38513'.. test: [ sip:bob@127.0.0.1:38513 ] event: CALL_OUTGOING (sip:alice@127.0.0.1:38513) test: [ sip:bob@127.0.0.1:38513 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:alice@127.0.0.1:38513 ] event: CALL_REMOTE_SDP (offer) test: [ sip:alice@127.0.0.1:38513 ] event: CALL_INCOMING (sip:bob@127.0.0.1:38513) call: answering call on line 1 from sip:bob@127.0.0.1:38513 with 200 stream: update 'audio' test: [ sip:alice@127.0.0.1:38513 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:bob@127.0.0.1:38513 ] event: CALL_RINGING (sip:alice@127.0.0.1:38513) bob@127.0.0.1: Call answered: sip:alice@127.0.0.1:38513 test: [ sip:bob@127.0.0.1:38513 ] event: CALL_ANSWERED (sip:alice@127.0.0.1:38513) test: [ sip:bob@127.0.0.1:38513 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU bob@127.0.0.1: Call established: sip:alice@127.0.0.1:38513 test: [ sip:bob@127.0.0.1:38513 ] event: CALL_ESTABLISHED (sip:alice@127.0.0.1:38513) test: event CALL_ESTABLISHED ua=[sip:bob@127.0.0.1:38513] (expected [sip:alice@127.0.0.1:38513] audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU alice@127.0.0.1: Call established: sip:bob@127.0.0.1:38513 test: [ sip:alice@127.0.0.1:38513 ] event: CALL_ESTABLISHED (sip:bob@127.0.0.1:38513) canceled by --- CALL_ESTABLISHED --- prm: ua: sip:alice@127.0.0.1:38513 n_incoming: 1 n_progress: 0 n_established: 1 met: yes test: event CALL_ESTABLISHED ua=[sip:alice@127.0.0.1:38513] (expected [sip:bob@127.0.0.1:38513] test: [ sip:alice@127.0.0.1:59942 ] event: UNREGISTERING () rtp_receiver: RTP RX thread started test: [ sip:alice@127.0.0.1:59942 ] event: CREATE (sip:alice@127.0.0.1:59942) test: [ sip:alice@127.0.0.1:59942 ] event: REGISTERING () alice@127.0.0.1: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] test: [ sip:alice@127.0.0.1:59942 ] event: REGISTER_OK (200 OK) canceled by --- REGISTER_OK --- prm: ua: sip:alice@127.0.0.1:59942 n_incoming: 0 n_progress: 0 n_established: 0 met: yes call: connecting to 'sip:alice@127.0.0.1:59918'.. test: [ sip:bob@127.0.0.1:38513 ] event: CALL_OUTGOING (sip:alice@127.0.0.1:59918) test: [ sip:bob@127.0.0.1:38513 ] event: CALL_LOCAL_SDP (offer) ua: sip:bob@127.0.0.1:38513: UA not found: sip:alice@127.0.0.1:59918 test: [ ] event: SIPSESS_FAILED (404 Not Found) test: could not find agent/ua sip:alice@127.0.0.1:59918: session closed: 404 Not Found test: [ sip:bob@127.0.0.1:38513 ] event: CALL_CLOSED (404 Not Found) canceled by --- CALL_CLOSED --- prm: ua: sip:bob@127.0.0.1:38513 n_incoming: 0 n_progress: 0 n_established: 0 met: yes test: [ sip:bob@127.0.0.1:38513 ] event: UNREGISTERING () test: [ sip:bob@127.0.0.1:38513 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:alice@127.0.0.1:38513 ] event: UNREGISTERING () test: [ sip:alice@127.0.0.1:38513 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:alice@127.0.0.1:59942 ] event: UNREGISTERING () unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_samplerate_switch (rx thread) aucodec: L16/48000/2 aucodec: L16/44100/2 aucodec: L16/32000/2 aucodec: L16/16000/2 aucodec: L16/8000/2 aucodec: L16/48000/1 aucodec: L16/44100/1 aucodec: L16/32000/1 aucodec: L16/16000/1 aucodec: L16/8000/1 aufilt: auresamp aufilt: auconv ausrc: aufile auplay: aufile SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:54326'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:54326) audio: using account specific source: (aufile,../test/data//wav/square_500Hz_0.8.wav) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer audio: using account specific source: (aufile,../test/data//wav/square_500Hz_0.8.wav) test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:54326) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:54326 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:54326) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: L16 48000Hz 2ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: L16 48000Hz 2ch aufile: loading input file '../test/data//wav/square_500Hz_0.8.wav' aufile: ../test/data//wav/square_500Hz_0.8.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=4 sampc=32 aufile: read end of file aufile: loaded 16000 bytes audio: source started with sample format S16LE audio tx pipeline: aufile ---> aubuf ---> auresamp ---> auconv ---> L16 audio rx pipeline: aufile <--- aubuf <--- auresamp <--- auconv <--- L16 a@127.0.0.1: Call established: sip:b@127.0.0.1:54326 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:54326) audio_recv: Set audio decoder: L16 48000Hz 2ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE audio: Set audio encoder: L16 48000Hz 2ch aufile: loading input file '../test/data//wav/square_500Hz_0.8.wav' aufile: ../test/data//wav/square_500Hz_0.8.wav: 8000 Hz, 1 channels, S16LE aufile: audio ptime=4 sampc=32 aufile: read end of file aufile: loaded 16000 bytes audio: source started with sample format S16LE audio tx pipeline: aufile ---> aubuf ---> auresamp ---> auconv ---> L16 audio rx pipeline: aufile <--- aubuf <--- auresamp <--- auconv <--- L16 b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMA 8000Hz 1ch aufile: writing speaker audio to speaker.wav audio_recv: player started with sample format S16LE test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 3 5 avg. bitrate: 614.3 960.0 (kbit/s) errors: 0 0 test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) audio Transmit: Receive: packets: 10 3 avg. bitrate: 1159.2 361.4 (kbit/s) errors: 0 0 unloading module: g711.so ua: stop all (forced=1) unloading module: l16.so unloading module: aufile.so unloading module: auconv.so unloading module: auresamp.so [ OK ] [ RUN ] test_call_sni (rx thread) SIP Certificate: ../test/data//sni/other-cert.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:a@retest.client.org ] event: CREATE (sip:a@retest.client.org) test: [ sip:b@retest.server.org ] event: CREATE (sip:b@retest.server.org) test: [ sip:c@retest.unknown.org ] event: CREATE (sip:c@retest.unknown.org) call: connecting to 'sip:b@retest.server.org:43163;transport=tls'.. test: [ sip:a@retest.client.org ] event: CALL_OUTGOING (sip:b@retest.server.org:43163;transport=tls) test: [ sip:a@retest.client.org ] event: CALL_LOCAL_SDP (offer) sip:b@retest.server.org:43163;transport=tls: session closed: Protocol error [71] test: [ sip:a@retest.client.org ] event: CALL_CLOSED (Protocol error [71]) call: connecting to 'sip:c@retest.unknown.org:43163;transport=tls'.. test: [ sip:a@retest.client.org ] event: CALL_OUTGOING (sip:c@retest.unknown.org:43163;transport=tls) test: [ sip:a@retest.client.org ] event: CALL_LOCAL_SDP (offer) sip:c@retest.unknown.org:43163;transport=tls: session closed: Protocol error [71] test: [ sip:a@retest.client.org ] event: CALL_CLOSED (Protocol error [71]) call: connecting to 'sip:b@retest.server.org:43163;transport=tls'.. test: [ sip:a@retest.client.org ] event: CALL_OUTGOING (sip:b@retest.server.org:43163;transport=tls) test: [ sip:a@retest.client.org ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@retest.server.org ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@retest.server.org ] event: CALL_INCOMING (sip:a@retest.client.org) call: answering call on line 1 from sip:a@retest.client.org with 200 stream: update 'audio' test: [ sip:b@retest.server.org ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@retest.client.org ] event: CALL_RINGING (sip:b@retest.server.org:43163;transport=tls) a@retest.client.org: Call answered: sip:b@retest.server.org:43163;transport=tls test: [ sip:a@retest.client.org ] event: CALL_ANSWERED (sip:b@retest.server.org:43163;transport=tls) test: [ sip:a@retest.client.org ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@retest.client.org: Call established: sip:b@retest.server.org:43163;transport=tls test: [ sip:a@retest.client.org ] event: CALL_ESTABLISHED (sip:b@retest.server.org:43163;transport=tls) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@retest.server.org: Call established: sip:a@retest.client.org test: [ sip:b@retest.server.org ] event: CALL_ESTABLISHED (sip:a@retest.client.org) call: terminate call '65d1f75eaf9b86e6' with sip:b@retest.server.org:43163;transport=tls test: [ sip:a@retest.client.org ] event: CALL_CLOSED (Rejected by user) test: [ sip:b@retest.server.org ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_call_cert_select (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 test: [ sip:a@127.0.0.1 ] event: CREATE (sip:a@127.0.0.1) test: [ sip:b@127.0.0.1 ] event: CREATE (sip:b@127.0.0.1) call: connecting to 'sip:b@127.0.0.1:36637;transport=tls'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:36637;transport=tls) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) sip:b@127.0.0.1:36637;transport=tls: session closed: Protocol error [71] test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Protocol error [71]) call: connecting to 'sip:a@127.0.0.1:36637;transport=tls'.. test: [ sip:b@127.0.0.1 ] event: CALL_OUTGOING (sip:a@127.0.0.1:36637;transport=tls) test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) sip:a@127.0.0.1:36637;transport=tls: session closed: Protocol error [71] test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Protocol error [71]) call: connecting to 'sip:b@127.0.0.1:36637;transport=tls'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:36637;transport=tls) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) rtp_receiver: RTP RX thread started call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:36637;transport=tls) a@127.0.0.1: Call answered: sip:b@127.0.0.1:36637;transport=tls test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:36637;transport=tls) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:36637;transport=tls test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:36637;transport=tls) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) call: terminate call 'f251cb531bec4c30' with sip:b@127.0.0.1:36637;transport=tls test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_cmd (rx thread) [ OK ] [ RUN ] test_cmd_long (rx thread) [ OK ] [ RUN ] test_cparam_call_decode (rx thread) test 0: audio=sendonly video=inactive callid=123 test 1: video=recvonly callid=234 audio=sendonly test 2: video=recvonly callid=234 audio=fail test 3: video=sendonly test 4: video=fail2 test 5: recvonly test 6: callid2 test 7: recvonly 345 test 8: fail3 456 test 9: video=inactive callid=234 audio=inactive [ OK ] [ RUN ] test_cparam_ua_decode (rx thread) test 0: "display name" audio=sendonly video=inactive userdata=mydata test 1: displayname userdata=mydata test 2: dn userdata=mydata2 audio=recvonly test 3: dn userdata=mydata2 video=fail4 test 4: sip:user2@domain2 audio=recvonly video=sendonly test 5: sip:user3@domain3 audio=fail video=sendonly test 6: sip:user4@domain4 video=inactive test 7: sip:user5@domain5 video=fail2 test 8: user6 sendonly test 9: audio=sendonly test 10: sip:user4@domain4;transport=tcp video=recvonly [ OK ] [ RUN ] test_contact (rx thread) [ OK ] [ RUN ] test_contact_find_call (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem aucodec: PCMU/8000/1 aucodec: PCMA/8000/1 call: connecting to 'sip:b@127.0.0.1:51975'.. test: [ sip:a@127.0.0.1 ] event: CALL_OUTGOING (sip:b@127.0.0.1:51975) test: [ sip:a@127.0.0.1 ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@127.0.0.1) call: answering call on line 1 from sip:a@127.0.0.1 with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@127.0.0.1 ] event: CALL_RINGING (sip:b@127.0.0.1:51975) rtp_receiver: RTP RX thread started a@127.0.0.1: Call answered: sip:b@127.0.0.1:51975 test: [ sip:a@127.0.0.1 ] event: CALL_ANSWERED (sip:b@127.0.0.1:51975) test: [ sip:a@127.0.0.1 ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@127.0.0.1: Call established: sip:b@127.0.0.1:51975 test: [ sip:a@127.0.0.1 ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:51975) rtp_receiver: RTP RX thread started audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU b@127.0.0.1: Call established: sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_ESTABLISHED (sip:a@127.0.0.1) call: terminate call '3b800a8b5707d58e' with sip:a@127.0.0.1 test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) test: [ sip:a@127.0.0.1 ] event: CALL_CLOSED (User-Agent deleted) test: [ sip:a@localhost ] event: CREATE (sip:a@localhost) call: connecting to 'sip:b@127.0.0.1:51975'.. test: [ sip:a@localhost ] event: CALL_OUTGOING (sip:b@127.0.0.1:51975) test: [ sip:a@localhost ] event: CALL_LOCAL_SDP (offer) test: [ ] event: SIPSESS_CONN (incoming call) ua: using connection-address 127.0.0.1 of SDP offer test: [ sip:b@127.0.0.1 ] event: CALL_REMOTE_SDP (offer) test: [ sip:b@127.0.0.1 ] event: CALL_INCOMING (sip:a@localhost) call: answering call on line 1 from sip:a@localhost with 200 stream: update 'audio' test: [ sip:b@127.0.0.1 ] event: CALL_LOCAL_SDP (answer) call: SIP Progress: 180 Ringing (/) test: [ sip:a@localhost ] event: CALL_RINGING (sip:b@127.0.0.1:51975) rtp_receiver: RTP RX thread started a@localhost: Call answered: sip:b@127.0.0.1:51975 test: [ sip:a@localhost ] event: CALL_ANSWERED (sip:b@127.0.0.1:51975) test: [ sip:a@localhost ] event: CALL_REMOTE_SDP (answer) stream: update 'audio' audio_recv: Set audio decoder: PCMU 8000Hz 1ch audio: Set audio encoder: PCMU 8000Hz 1ch audio tx pipeline: (src) ---> aubuf ---> PCMU audio rx pipeline: (play) <--- aubuf <--- PCMU a@localhost: Call established: sip:b@127.0.0.1:51975 test: [ sip:a@localhost ] event: CALL_ESTABLISHED (sip:b@127.0.0.1:51975) rtp_receiver: RTP RX thread started call: abort call '18ea1c9d8922e2ae' with sip:a@localhost test: [ sip:b@127.0.0.1 ] event: CALL_CLOSED (Rejected by user) test: [ sip:a@localhost ] event: CALL_CLOSED (User-Agent deleted) unloading module: g711.so ua: stop all (forced=1) [ OK ] [ RUN ] test_bevent_register (rx thread) [ OK ] [ RUN ] test_jbuf (rx thread) [ OK ] [ RUN ] test_jbuf_adaptive (rx thread) [ OK ] [ RUN ] test_jbuf_video (rx thread) [ OK ] [ RUN ] test_jbuf_gnack (rx thread) jbuf: RTCP_GNACK too big blp_pkts 20 [ OK ] [ RUN ] test_message (rx thread) [ sip:b@127.0.0.1:35501;transport=UDP ] recv msg from sip:a@127.0.0.1: "hei paa deg" [ sip:a@127.0.0.1:35501;transport=UDP ] message sent OK [ sip:b@127.0.0.1:34109;transport=TCP ] recv msg from sip:a@127.0.0.1: "hei paa deg" [ sip:a@127.0.0.1:34109;transport=TCP ] message sent OK [ sip:a@127.0.0.1:40939;transport=UDP ] message sent OK [ sip:a@127.0.0.1:37733;transport=TCP ] message sent OK [ OK ] [ RUN ] test_network (rx thread) Local network addresses: None available for net_interface: - [ OK ] [ RUN ] test_peerconn (rx thread) [ OK ] [ RUN ] test_play (rx thread) auplay: mock-auplay [ OK ] [ RUN ] test_stunuri (rx thread) [ OK ] [ RUN ] test_ua_alloc (rx thread) [ OK ] [ RUN ] test_ua_cuser (rx thread) [ OK ] [ RUN ] test_ua_options (rx thread) ua: stop all (forced=1) ua: stop all (forced=1) [ OK ] [ RUN ] test_ua_refer (rx thread) ua: REFER forbidden for user@127.0.0.1 ua: stop all (forced=1) ua: REFER forbidden for user@127.0.0.1 ua: stop all (forced=1) [ OK ] [ RUN ] test_ua_register (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem x@127.0.0.1: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! x@127.0.0.1: (prio 0) {0/TCP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! x@127.0.0.1: (prio 0) {0/TLS/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) [ OK ] [ RUN ] test_ua_register_auth (rx thread) SIP Certificate: ../test/data//sni/server-interm.pem alfredh@localhost: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! alfredh@localhost: (prio 0) {0/TCP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! alfredh@localhost: (prio 0) {0/TLS/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) [ OK ] [ RUN ] test_ua_register_auth_dns (rx thread) | DNS-server on 127.0.0.1:46472 | SIP-server on 127.0.0.1:41222 | SIP-server on 127.0.0.1:45226 SIP Certificate: ../test/data//sni/server-interm.pem alfredh@test.invalid: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) | DNS-server on 127.0.0.1:51806 | SIP-server on 127.0.0.1:38849 | SIP-server on 127.0.0.1:40697 SIP Certificate: ../test/data//sni/server-interm.pem alfredh@test.invalid: (prio 0) {0/TCP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) | DNS-server on 127.0.0.1:48016 | SIP-server on 127.0.0.1:33353 | SIP-server on 127.0.0.1:38439 SIP Certificate: ../test/data//sni/server-interm.pem alfredh@test.invalid: (prio 0) {0/TLS/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) [ OK ] [ RUN ] test_ua_register_dns (rx thread) | DNS-server on 127.0.0.1:50028 | SIP-server on 127.0.0.1:58167 SIP Certificate: ../test/data//sni/server-interm.pem x@test.invalid: (prio 0) {0/UDP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) | DNS-server on 127.0.0.1:52042 | SIP-server on 127.0.0.1:36313 SIP Certificate: ../test/data//sni/server-interm.pem x@test.invalid: (prio 0) {0/TCP/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) | DNS-server on 127.0.0.1:46950 | SIP-server on 127.0.0.1:40761 SIP Certificate: ../test/data//sni/server-interm.pem x@test.invalid: (prio 0) {0/TLS/v4} 200 OK (mock SIP server) [1 binding] event: Register OK! ua: stop all (forced=1) [ OK ] [ RUN ] test_uag_find_param (rx thread) [ OK ] [ RUN ] test_video (rx thread) [ OK ] [ RUN ] test_clean_number (rx thread) [ OK ] [ RUN ] test_clean_number_only_numeric (rx thread) [ OK ] ua: stop all (forced=1) OK. 67 tests passed successfully ua: stop all (forced=1) + RPM_EC=0 ++ jobs -p + exit 0 Processing files: baresip-4.11.0-1.fc46.s390x Executing(%doc): /bin/sh -e /var/tmp/rpm-tmp.QgE2md + umask 022 + cd /builddir/build/BUILD/baresip-4.11.0-build + cd baresip-4.11.0 + DOCDIR=/builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/doc/baresip + export LC_ALL=C.UTF-8 + LC_ALL=C.UTF-8 + export DOCDIR + /usr/bin/mkdir -p /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/doc/baresip + cp -pr /builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/CHANGELOG.md /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/doc/baresip + cp -pr /builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/docs/THANKS /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/doc/baresip + cp -pr /builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/docs/examples /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/doc/baresip + RPM_EC=0 ++ jobs -p + exit 0 Executing(%license): /bin/sh -e /var/tmp/rpm-tmp.zjZ9PG + umask 022 + cd /builddir/build/BUILD/baresip-4.11.0-build + cd baresip-4.11.0 + LICENSEDIR=/builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/licenses/baresip + export LC_ALL=C.UTF-8 + LC_ALL=C.UTF-8 + export LICENSEDIR + /usr/bin/mkdir -p /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/licenses/baresip + cp -pr /builddir/build/BUILD/baresip-4.11.0-build/baresip-4.11.0/LICENSE /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT/usr/share/licenses/baresip + RPM_EC=0 ++ jobs -p + exit 0 Provides: baresip = 4.11.0-1.fc46 baresip(s390-64) = 4.11.0-1.fc46 libbaresip.so.28()(64bit) Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libbaresip.so.28()(64bit) libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.3)(64bit) libc.so.6(GLIBC_2.3.4)(64bit) libc.so.6(GLIBC_2.33)(64bit) libc.so.6(GLIBC_2.34)(64bit) libc.so.6(GLIBC_2.38)(64bit) libc.so.6(GLIBC_2.4)(64bit) libre.so.45()(64bit) rtld(GNU_HASH) Obsoletes: baresip-cairo < 1.1.0-1 baresip-g726 < 4.8.0-1 baresip-gsm < 2.6.0-1 baresip-gst_video < 2.6.0-1 baresip-mpa < 4.3.0-1 baresip-omx < 2.7.0-1 baresip-rst < 2.0.0-1 baresip-speex_pp < 2.0.0-1 baresip-x11grab < 2.0.0-1 Recommends: baresip-pipewire(s390-64) = 4.11.0-1.fc46 Processing files: baresip-devel-4.11.0-1.fc46.s390x Provides: baresip-devel = 4.11.0-1.fc46 baresip-devel(s390-64) = 4.11.0-1.fc46 pkgconfig(libbaresip) = 4.11.0 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: /usr/bin/pkg-config libbaresip.so.28()(64bit) Processing files: baresip-aac-4.11.0-1.fc46.s390x Provides: baresip-aac = 4.11.0-1.fc46 baresip-aac(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) libfdk-aac.so.2()(64bit) rtld(GNU_HASH) Processing files: baresip-alsa-4.11.0-1.fc46.s390x Provides: baresip-alsa = 4.11.0-1.fc46 baresip-alsa(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libasound.so.2()(64bit) libasound.so.2(ALSA_0.9)(64bit) libasound.so.2(ALSA_0.9.0rc4)(64bit) libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.34)(64bit) libc.so.6(GLIBC_2.4)(64bit) rtld(GNU_HASH) Processing files: baresip-av1-4.11.0-1.fc46.s390x Provides: baresip-av1 = 4.11.0-1.fc46 baresip-av1(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libaom.so.3()(64bit) libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) rtld(GNU_HASH) Processing files: baresip-avcodec-4.11.0-1.fc46.s390x Provides: baresip-avcodec = 4.11.0-1.fc46 baresip-avcodec(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libavcodec.so.63()(64bit) libavcodec.so.63(LIBAVCODEC_63)(64bit) libavutil.so.61()(64bit) libavutil.so.61(LIBAVUTIL_61)(64bit) libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) rtld(GNU_HASH) Processing files: baresip-avfilter-4.11.0-1.fc46.s390x Provides: baresip-avfilter = 4.11.0-1.fc46 baresip-avfilter(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libavfilter.so.12()(64bit) libavfilter.so.12(LIBAVFILTER_12)(64bit) libavutil.so.61()(64bit) libavutil.so.61(LIBAVUTIL_61)(64bit) libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.34)(64bit) libc.so.6(GLIBC_2.4)(64bit) rtld(GNU_HASH) Processing files: baresip-avformat-4.11.0-1.fc46.s390x Provides: baresip-avformat = 4.11.0-1.fc46 baresip-avformat(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libavcodec.so.63()(64bit) libavcodec.so.63(LIBAVCODEC_63)(64bit) libavdevice.so.63()(64bit) libavdevice.so.63(LIBAVDEVICE_63)(64bit) libavformat.so.63()(64bit) libavformat.so.63(LIBAVFORMAT_63)(64bit) libavutil.so.61()(64bit) libavutil.so.61(LIBAVUTIL_61)(64bit) libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.34)(64bit) libc.so.6(GLIBC_2.4)(64bit) libswresample.so.7()(64bit) libswresample.so.7(LIBSWRESAMPLE_7)(64bit) rtld(GNU_HASH) Processing files: baresip-codec2-4.11.0-1.fc46.s390x Provides: baresip-codec2 = 4.11.0-1.fc46 baresip-codec2(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libcodec2.so.1.2()(64bit) rtld(GNU_HASH) Processing files: baresip-ctrl_dbus-4.11.0-1.fc46.s390x Provides: baresip-ctrl_dbus = 4.11.0-1.fc46 baresip-ctrl_dbus(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.34)(64bit) libc.so.6(GLIBC_2.4)(64bit) libgio-2.0.so.0()(64bit) libglib-2.0.so.0()(64bit) libgobject-2.0.so.0()(64bit) rtld(GNU_HASH) Processing files: baresip-g722-4.11.0-1.fc46.s390x Provides: baresip-g722 = 4.11.0-1.fc46 baresip-g722(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libspandsp.so.2()(64bit) rtld(GNU_HASH) Processing files: baresip-gst-4.11.0-1.fc46.s390x Provides: baresip-gst = 4.11.0-1.fc46 baresip-gst(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) libglib-2.0.so.0()(64bit) libgstreamer-1.0.so.0()(64bit) rtld(GNU_HASH) Processing files: baresip-gtk-4.11.0-1.fc46.s390x Provides: application() application(com.github.baresip.desktop) baresip-gtk = 4.11.0-1.fc46 baresip-gtk(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.34)(64bit) libc.so.6(GLIBC_2.4)(64bit) libgdk-3.so.0()(64bit) libgio-2.0.so.0()(64bit) libglib-2.0.so.0()(64bit) libgobject-2.0.so.0()(64bit) libgtk-3.so.0()(64bit) rtld(GNU_HASH) Recommends: libcanberra-gtk3 Processing files: baresip-jack-4.11.0-1.fc46.s390x Provides: baresip-jack = 4.11.0-1.fc46 baresip-jack(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) libjack.so.0()(64bit) rtld(GNU_HASH) Processing files: baresip-mqtt-4.11.0-1.fc46.s390x Provides: baresip-mqtt = 4.11.0-1.fc46 baresip-mqtt(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) libmosquitto.so.1()(64bit) libmosquitto.so.1(MOSQ_1.0)(64bit) rtld(GNU_HASH) Processing files: baresip-opus-4.11.0-1.fc46.s390x Provides: baresip-opus = 4.11.0-1.fc46 baresip-opus(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) libopus.so.0()(64bit) rtld(GNU_HASH) Processing files: baresip-pipewire-4.11.0-1.fc46.s390x Provides: baresip-pipewire = 4.11.0-1.fc46 baresip-pipewire(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) libpipewire-0.3.so.0()(64bit) rtld(GNU_HASH) Processing files: baresip-plc-4.11.0-1.fc46.s390x Provides: baresip-plc = 4.11.0-1.fc46 baresip-plc(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libspandsp.so.2()(64bit) rtld(GNU_HASH) Processing files: baresip-portaudio-4.11.0-1.fc46.s390x Provides: baresip-portaudio = 4.11.0-1.fc46 baresip-portaudio(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.38)(64bit) libc.so.6(GLIBC_2.4)(64bit) libportaudio.so.2()(64bit) rtld(GNU_HASH) Processing files: baresip-pulse-4.11.0-1.fc46.s390x Provides: baresip-pulse = 4.11.0-1.fc46 baresip-pulse(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) libpulse.so.0()(64bit) libpulse.so.0(PULSE_0)(64bit) rtld(GNU_HASH) Processing files: baresip-sdl-4.11.0-1.fc46.s390x Provides: baresip-sdl = 4.11.0-1.fc46 baresip-sdl(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libSDL2-2.0.so.0()(64bit) libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) rtld(GNU_HASH) Processing files: baresip-snapshot-4.11.0-1.fc46.s390x Provides: baresip-snapshot = 4.11.0-1.fc46 baresip-snapshot(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.11)(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) libpng16.so.16()(64bit) libpng16.so.16(PNG16_0)(64bit) rtld(GNU_HASH) Processing files: baresip-sndfile-4.11.0-1.fc46.s390x Provides: baresip-sndfile = 4.11.0-1.fc46 baresip-sndfile(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) libsndfile.so.1()(64bit) libsndfile.so.1(libsndfile.so.1.0)(64bit) rtld(GNU_HASH) Processing files: baresip-swscale-4.11.0-1.fc46.s390x Provides: baresip-swscale = 4.11.0-1.fc46 baresip-swscale(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) libswscale.so.10()(64bit) libswscale.so.10(LIBSWSCALE_10)(64bit) rtld(GNU_HASH) Processing files: baresip-tools-4.11.0-1.fc46.s390x Provides: baresip-tools = 4.11.0-1.fc46 baresip-tools(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: /usr/bin/python3 Processing files: baresip-vp8-4.11.0-1.fc46.s390x Provides: baresip-vp8 = 4.11.0-1.fc46 baresip-vp8(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) libvpx.so.12()(64bit) rtld(GNU_HASH) Processing files: baresip-vp9-4.11.0-1.fc46.s390x Provides: baresip-vp9 = 4.11.0-1.fc46 baresip-vp9(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) libvpx.so.12()(64bit) rtld(GNU_HASH) Processing files: baresip-v4l2-4.11.0-1.fc46.s390x Provides: baresip-v4l2 = 4.11.0-1.fc46 baresip-v4l2(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.34)(64bit) libc.so.6(GLIBC_2.4)(64bit) rtld(GNU_HASH) Processing files: baresip-x11-4.11.0-1.fc46.s390x Provides: baresip-x11 = 4.11.0-1.fc46 baresip-x11(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Requires: libX11.so.6()(64bit) libXext.so.6()(64bit) libc.so.6()(64bit) libc.so.6(GLIBC_2.2)(64bit) libc.so.6(GLIBC_2.4)(64bit) rtld(GNU_HASH) Processing files: baresip-debugsource-4.11.0-1.fc46.s390x Provides: baresip-debugsource = 4.11.0-1.fc46 baresip-debugsource(s390-64) = 4.11.0-1.fc46 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Processing files: baresip-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-debuginfo = 4.11.0-1.fc46 baresip-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 00b2429f5a4124270cf9f33a5f4a2f8c4b576c97 debuginfo(build-id) = 044d46dcbcbbbc0c44887fb9edd3e0aef2cbfdb2 debuginfo(build-id) = 0d33ca53388f60843243d6da24bf029be70b7ce0 debuginfo(build-id) = 1e69a604ad3d3616d8146db7ef4c171b656f906b debuginfo(build-id) = 20186a0847de10b4cc584cb60d1aa5b2e8c7c6bc debuginfo(build-id) = 2381ac05fd5f27eed96e9df5ef7883bb97f5d093 debuginfo(build-id) = 32951252103c8b95c9b5d33f0a1f5b1db27e6ff7 debuginfo(build-id) = 3a8a70a6fece4db57442265edaa9dd900c02bd8a debuginfo(build-id) = 409a9439247c4433f4290461d8790d68fb5b027b debuginfo(build-id) = 471ba90b486b4c0c2ab97ce45d16bfa40cd4b1e0 debuginfo(build-id) = 49661e379b5c3b5c46d4e311dba86c5454ba8e7a debuginfo(build-id) = 4c56a9fbed0d6af65a2fa29ebbec314531d4ac92 debuginfo(build-id) = 50000f1f81df23448f4e74cae3d183f11f4047e2 debuginfo(build-id) = 57c805b1cc6d3cd553a58d0c62fc91736d9ea558 debuginfo(build-id) = 6892f0a21f155562a96ae30c6ce9cedf742e23ad debuginfo(build-id) = 6c2816454c32360430699a43a053d0a10f0b95a9 debuginfo(build-id) = 6e735c0093f67d8dfae34f4c9ec31fedf1860598 debuginfo(build-id) = 759e023805d6c54a02d00ad4da005ce430675cd2 debuginfo(build-id) = 7a7f5b49b2525a5e6574c48447871bb337691d71 debuginfo(build-id) = 7c724ee70162b41b714e95b56c42a9bdeb6d916b debuginfo(build-id) = 8d736a4b10b8d4635f5b0c5d3b92c0cd5b9a6bea debuginfo(build-id) = 9e92da0fd01b048c4213d043f24622146f7f93bc debuginfo(build-id) = a1a8654ac5b96df32dab02fb82c8a8e0cb365d8a debuginfo(build-id) = a35714430efb0e80d8a0a730a39ebcc7f57964c8 debuginfo(build-id) = a6b4f4cdaa06d9d2d1b2d7608c9153935796f639 debuginfo(build-id) = adc1748a3e2b26f87e2446aae679915f3699c97c debuginfo(build-id) = af1500202dc35b60722e00b22ac66258ba71a5d9 debuginfo(build-id) = b107cd848fffa9d4e7d366a9fbc19c1cbd912642 debuginfo(build-id) = c74f3972f84c87b596d1d4d24425db815d33db07 debuginfo(build-id) = c891d3ba94687596dbd96f4187af7327668ac17a debuginfo(build-id) = cdbc072f7c29c485a6802a219c33c9786cad398b debuginfo(build-id) = d118537f89ed826ff7262b0b048d65bb8d6ed908 debuginfo(build-id) = d39a18759c537afbb743560531df6c81934c5191 debuginfo(build-id) = d788bcad389e7017f3f3a22607111031c690eec2 debuginfo(build-id) = dd566b24b25b879e9c64fd9e079b022c6002d49f debuginfo(build-id) = dda726ea5739cac4bd0a8b3a3bd988ab07094ccc debuginfo(build-id) = df7c3a598d94a5d29396a6616c5c68e6b3153f46 debuginfo(build-id) = e019ced9bb0bd5c9971a71c5e170981ac1b2ca96 debuginfo(build-id) = e3442441c76304eccbc2e148976ad46d7a8ca341 debuginfo(build-id) = e8c87d2414558f29fb0ff30382051a107987a4e9 debuginfo(build-id) = ef316201ea16228647acb80ba5dadd132d596989 debuginfo(build-id) = f1fc171b49f23828bc4fd73780ca2e21f40ed777 debuginfo(build-id) = f9ef14dee85f5021ba215bab3b6705f2609a19f8 libbaresip.so.28.11.0-4.11.0-1.fc46.s390x.debug()(64bit) Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-aac-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-aac-debuginfo = 4.11.0-1.fc46 baresip-aac-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 30cf9bf151c489eb9ce39c6fcea51d2b5b3f50a8 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-alsa-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-alsa-debuginfo = 4.11.0-1.fc46 baresip-alsa-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 1e66261bb0adb4739a9829602f74329334ce1b44 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-av1-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-av1-debuginfo = 4.11.0-1.fc46 baresip-av1-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 408683551529dc2d93a5437544fb8056f55b99ad Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-avcodec-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-avcodec-debuginfo = 4.11.0-1.fc46 baresip-avcodec-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = aadb8c5a98b6badba0c96fded96e38c003f4f673 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-avfilter-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-avfilter-debuginfo = 4.11.0-1.fc46 baresip-avfilter-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 4a8ac92162f1cce88f91d47eb8a8e0a4ba3c6be0 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-avformat-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-avformat-debuginfo = 4.11.0-1.fc46 baresip-avformat-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 6fab967f372c37445124376295be0a70c6c8a01e Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-codec2-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-codec2-debuginfo = 4.11.0-1.fc46 baresip-codec2-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = c3ae6c1d75d6b695f735ab57ef38e192381de815 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-ctrl_dbus-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-ctrl_dbus-debuginfo = 4.11.0-1.fc46 baresip-ctrl_dbus-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 7ef6053dd59f29aff0a0095f296aecad3dfc8a49 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-g722-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-g722-debuginfo = 4.11.0-1.fc46 baresip-g722-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 684e38d3217e47583c8b9a9e983b10cd3b7fa6a8 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-gst-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-gst-debuginfo = 4.11.0-1.fc46 baresip-gst-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = adc95c5864d09939ab85b9fd82f25f329f2bfc3a Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-gtk-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-gtk-debuginfo = 4.11.0-1.fc46 baresip-gtk-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = e0dc6a6b456c971f267c3d32a568155e8ff8fe7a Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-jack-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-jack-debuginfo = 4.11.0-1.fc46 baresip-jack-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = f298ac899a27587c79a58b2a4de40171557800da Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-mqtt-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-mqtt-debuginfo = 4.11.0-1.fc46 baresip-mqtt-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = d1ba152bf8611da606ed9ecc0b8a770ab87f86bb Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-opus-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-opus-debuginfo = 4.11.0-1.fc46 baresip-opus-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 6ee7705c6787b99374a53d444f25d55d798cad84 debuginfo(build-id) = e337c950006e1b14b06058f0c7c711355aa9caa8 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-pipewire-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-pipewire-debuginfo = 4.11.0-1.fc46 baresip-pipewire-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 43152efaffb53369ad00a5310a0393038e7572a8 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-plc-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-plc-debuginfo = 4.11.0-1.fc46 baresip-plc-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 6e1eba7e3a82c2fe8a7882258d8325e2f426177a Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-portaudio-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-portaudio-debuginfo = 4.11.0-1.fc46 baresip-portaudio-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 338c6bf4365c44aed4342eadae6f3804c64fed8c Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-pulse-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-pulse-debuginfo = 4.11.0-1.fc46 baresip-pulse-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 0fdcb751287d96a7a390b2279698e85ce64ccc13 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-sdl-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-sdl-debuginfo = 4.11.0-1.fc46 baresip-sdl-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 3f851446789b2a58a6922065aec2ddc36387cd3b Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-snapshot-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-snapshot-debuginfo = 4.11.0-1.fc46 baresip-snapshot-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = be8f9dcef34f93ecda1e88d16854030572627c4e Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-sndfile-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-sndfile-debuginfo = 4.11.0-1.fc46 baresip-sndfile-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = c8c20cc04643d4660f8ca87d40d37faa244411ff Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-swscale-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-swscale-debuginfo = 4.11.0-1.fc46 baresip-swscale-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 9f97198960d8c528b0d3f565ea932bd6359b8172 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-vp8-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-vp8-debuginfo = 4.11.0-1.fc46 baresip-vp8-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 45b263f80b241e99a5144e454c7706ecb2c8d629 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-vp9-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-vp9-debuginfo = 4.11.0-1.fc46 baresip-vp9-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 34123ab66abcff9e54c9f69fedb3f07b2de53ab0 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-v4l2-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-v4l2-debuginfo = 4.11.0-1.fc46 baresip-v4l2-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 2b45783562394be8247075b35840b47f5c67a325 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Processing files: baresip-x11-debuginfo-4.11.0-1.fc46.s390x Provides: baresip-x11-debuginfo = 4.11.0-1.fc46 baresip-x11-debuginfo(s390-64) = 4.11.0-1.fc46 debuginfo(build-id) = 53acaf4d7161a011f4079023c7cdd3ad46a61587 Requires(rpmlib): rpmlib(CompressedFileNames) <= 3.0.4-1 rpmlib(FileDigests) <= 4.6.0-1 rpmlib(PayloadFilesHavePrefix) <= 4.0-1 Recommends: baresip-debugsource(s390-64) = 4.11.0-1.fc46 Checking for unpackaged file(s): /usr/lib/rpm/check-files /builddir/build/BUILD/baresip-4.11.0-build/BUILDROOT Wrote: /builddir/build/RPMS/baresip-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-pipewire-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-gtk-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-ctrl_dbus-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-debugsource-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-avcodec-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-avformat-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-opus-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-sdl-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-pulse-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-av1-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-devel-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-aac-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-gst-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-avfilter-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-gtk-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-pipewire-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-vp9-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-vp8-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-mqtt-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-alsa-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-v4l2-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-x11-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-jack-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-ctrl_dbus-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-opus-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-swscale-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-portaudio-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-avcodec-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-sndfile-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-avformat-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-snapshot-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-codec2-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-g722-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-aac-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-pulse-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-mqtt-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-plc-debuginfo-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-vp9-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-alsa-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-jack-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-gst-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-sndfile-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-av1-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-vp8-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-avfilter-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-snapshot-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-portaudio-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-x11-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-sdl-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-swscale-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-v4l2-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-codec2-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-g722-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-tools-4.11.0-1.fc46.s390x.rpm Wrote: /builddir/build/RPMS/baresip-plc-4.11.0-1.fc46.s390x.rpm Executing(rmbuild): /bin/sh -e /var/tmp/rpm-tmp.XYvz0s + umask 022 + cd /builddir/build/BUILD/baresip-4.11.0-build + test -d /builddir/build/BUILD/baresip-4.11.0-build + /usr/bin/chmod -Rf a+rX,u+w,g-w,o-w /builddir/build/BUILD/baresip-4.11.0-build + rm -rf /builddir/build/BUILD/baresip-4.11.0-build + RPM_EC=0 ++ jobs -p + exit 0 Finish: rpmbuild baresip-4.11.0-1.fc46.src.rpm Finish: build phase for baresip-4.11.0-1.fc46.src.rpm INFO: chroot_scan: 1 files copied to /var/lib/copr-rpmbuild/results/chroot_scan INFO: /var/lib/mock/fedora-rawhide-s390x-1788464321.539005/root/var/log/dnf5.log INFO: chroot_scan: creating tarball /var/lib/copr-rpmbuild/results/chroot_scan.tar.gz /bin/tar: Removing leading `/' from member names INFO: Done(/var/lib/copr-rpmbuild/results/baresip-4.11.0-1.fc46.src.rpm) Config(child) 1 minutes 29 seconds INFO: Results and/or logs in: /var/lib/copr-rpmbuild/results INFO: Cleaning up build root ('cleanup_on_success=True') Start: clean chroot INFO: unmounting tmpfs. Finish: clean chroot Finish: run Running RPMResults tool Package info: { "packages": [ { "name": "baresip-x11-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-g722", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-av1", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "src" }, { "name": "baresip-x11", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-devel", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-gtk-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-plc", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-avfilter-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-gst-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-sdl-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-vp9-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-jack", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-aac", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-avformat", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-pulse-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-avcodec-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-codec2-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-sdl", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-avfilter", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-alsa-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-vp8", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-vp8-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-aac-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-pipewire-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-v4l2", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-opus-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-opus", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-g722-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-pulse", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-snapshot-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-codec2", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-snapshot", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-gtk", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-ctrl_dbus", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-gst", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-avformat-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-portaudio", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-tools", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-alsa", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-jack-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-sndfile", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-portaudio-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-swscale", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-mqtt-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-mqtt", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-avcodec", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-ctrl_dbus-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-swscale-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-vp9", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-av1-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-v4l2-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-debugsource", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-pipewire", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-sndfile-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip-plc-debuginfo", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" }, { "name": "baresip", "epoch": null, "version": "4.11.0", "release": "1.fc46", "arch": "s390x" } ] } RPMResults finished