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| Name: baresip-video | Distribution: openSUSE Tumbleweed |
| Version: 4.9.0 | Vendor: openSUSE |
| Release: 1.4 | Build date: Wed Jun 17 14:31:50 2026 |
| Group: Productivity/Telephony/SIP/Clients | Build host: reproducible |
| Size: 299210 | Source RPM: baresip-4.9.0-1.4.src.rpm |
| Packager: https://bugs.opensuse.org | |
| Url: https://github.com/baresip/baresip | |
| Summary: Video support for the Baresip useragent | |
A modular SIP user-agent with support for audio and video, and many IETF standards such as SIP, SDP, RTP/RTCP, STUN, TURN, and ICE. Supports both IPv4 and IPv6, and the following features. * Audio codecs: AMR, G.711, G.722, G.726, GSM, L16, MPA, OPUS. * Video codecs: H.263, H.264, H.265, MPEG4, VP8, VP9. * Audio drivers: Alsa, JACK, OSS, Portaudio, sndio. * Video sources: FFmpeg avformat, Video4Linux2, X11 Grabber. * Video output: SDL2, X11, DirectFB. * NAT Traversal: STUN, TURN, ICE, NATBD, NAT-PMP, PCP. * Media encryption: SRTP, DTLS-SRTP. * DNS Service Discovery: Avahi. * Telemetry messaging: MQTT. * Control interfaces: JSON-over-TCP. This subpackage provides the modules that are needed for video support.
BSD-3-Clause
* Wed Jun 17 2026 Martin Hauke <mardnh@gmx.de>
- Update to versoin 4.9.0
* tls: change to non-deprecated TLS_METHOD_XXX.
* ua: emit event for failed incoming INVITE.
* ua: avoid fallback 500 after unsupported address family.
* ua: stop processing INVITE after 421 response.
* test: add 100rel required reject testcase.
* docs: minor cleanup of example config.
* contact: add command to clear all contacts.
* uag: check registration state of fallback UA.
* call: make call_supported() input argument const.
* Thu May 14 2026 Martin Hauke <mardnh@gmx.de>
- Update to version 4.8.0
* contact: null pointer check.
* misc: auresamp lost after dynamic codec switch.
* menu: delayed answer fixes.
* contact: avoid duplicate contacts.
* contact: intelligent matching for incoming calls.
* call: support rejecting re-INVITE.
* jbuf/gnack: improve RTCP_GNACK handling (fixes UB).
* srtp: use rtp_is_rtcp_packet() from libre.
* g726: remove deprecated module.
* menu: multiple call auto answer.
* test: expand mediaenc to include dtls_srtp and mux.
* misc/modules: fix redefinition once_flag errors.
* ua: ua_regladdr LIST_FOREACH cleanup.
* test: update doxygen header for dial_number.c .
* call: Use CRLF in REFER NOTIFY sipfrag body.
* Wed Apr 29 2026 Luigi Baldoni <aloisio@gmx.com>
- Raise minimum version of libre.
* Tue Apr 07 2026 Martin Hauke <mardnh@gmx.de>
- Update to version 4.7.0
* https://github.com/baresip/baresip/releases/tag/v4.7.0
* https://github.com/baresip/baresip/releases/tag/v4.6.0
* https://github.com/baresip/baresip/releases/tag/v4.5.0
* Tue Jan 06 2026 Martin Hauke <mardnh@gmx.de>
- Update to version 4.4.0
* https://github.com/baresip/baresip/releases/tag/v4.4.0
* https://github.com/baresip/baresip/releases/tag/v4.3.0
* Thu Oct 16 2025 Martin Hauke <mardnh@gmx.de>
- Update to version 4.2.0
* https://github.com/baresip/baresip/releases/tag/v4.2.0
- Build with support for pipwire
* Sat Sep 13 2025 Martin Hauke <mardnh@gmx.de>
- Update to version 4.1.0
* This major release drops backward wrappers for the old
UA-events API. Applications now have to be ported to the new
baresip events API.
* Release v4.0.0 further includes breaking API renaming,
code/API cleanup and drops obsolete API functions.
Detailed changes since 3.21.0:
* https://github.com/baresip/baresip/releases/tag/v4.1.0
* https://github.com/baresip/baresip/releases/tag/v4.0.0
* https://github.com/baresip/baresip/releases/tag/v3.24.0
* https://github.com/baresip/baresip/releases/tag/v3.23.0
* https://github.com/baresip/baresip/releases/tag/v3.22.0
* Wed Apr 16 2025 Andreas Stieger <andreas.stieger@gmx.de>
- Update to 3.21.0:
* GTK added standalone main window
* wasapi: Add WASAPI Audio module
* add contact commands
* many bug fixes and functionality improvements
- fixes build, and build with cmake 4 (boo#1239788)
* Fri Oct 25 2024 Pedro Monreal <pmonreal@suse.com>
- Do not hardcode the openssl version [bsc#1219879]
* Replace libopenssl-1_1-devel with libopenssl-devel
* Fix in_band_dtmf.so lib directory for x86
* Thu Oct 03 2024 Martin Hauke <mardnh@gmx.de>
- Update to version 3.16.0
* https://github.com/baresip/baresip/releases/tag/v3.16.0
* https://github.com/baresip/baresip/releases/tag/v3.15.0
* https://github.com/baresip/baresip/releases/tag/v3.14.0
* https://github.com/baresip/baresip/releases/tag/v3.13.0
* https://github.com/baresip/baresip/releases/tag/v3.12.0
* Sun Apr 14 2024 Martin Hauke <mardnh@gmx.de>
- Update to version 3.11.0
* https://github.com/baresip/baresip/releases/tag/v3.11.0
* https://github.com/baresip/baresip/releases/tag/v3.10.1
* https://github.com/baresip/baresip/releases/tag/v3.10.0
* https://github.com/baresip/baresip/releases/tag/v3.9.0
* Wed Jan 03 2024 Martin Hauke <mardnh@gmx.de>
- Update to version 3.8.1
* aur: set audio format correctly.
* cmake: add only non-system link paths to rpath.
* Wed Dec 27 2023 Martin Hauke <mardnh@gmx.de>
- Update to version 3.8.0
Changes:
* https://github.com/baresip/baresip/releases/tag/v3.8.0
* https://github.com/baresip/baresip/releases/tag/v3.7.0
* https://github.com/baresip/baresip/releases/tag/v3.5.1
* https://github.com/baresip/baresip/releases/tag/v3.5.0
* https://github.com/baresip/baresip/releases/tag/v3.4.0
* https://github.com/baresip/baresip/releases/tag/v3.3.0
* https://github.com/baresip/baresip/releases/tag/v3.2.0
* https://github.com/baresip/baresip/releases/tag/v3.1.0
* https://github.com/baresip/baresip/releases/tag/v3.0.0
* https://github.com/baresip/baresip/releases/tag/v2.12.0
* https://github.com/baresip/baresip/releases/tag/v2.11.0
* Thu Jan 05 2023 Matthias Eliasson <elimat@opensuse.org>
- Update to release 2.10.0
* sdl: small improvements
* vidinfo: allow all pixel formats
* vid: add support for YUV422P pixel format
* avformat: fix hwaccel vaapi
* mk: add deprecate notice
* mingw: upgrade to OpenSSL 3.0.7
* dshow: fix some warnings
* dshow: fix pragma warning
* ci: install libsdl2 development package
* sdl: work in progress fixes for multi-threading
* Stop segfaulting when no URI is passed to dial command
* ice: local candidate policy config
* auresamp: check handler arguments
* fixes 2315 and GTK errors on quit
* auresamp: avoid division
* cmake: check for XShm.h
* pulse_async: avoid integer overrun for timestamps in recorder
* ua: use sdp connection data instead origin
* rtpext: move from baresip to re
* acc,stream: add rtcp_mux account param
* video: video_update cleanup
* aufile/src: add auframe support
* ice/tmr_async_handler: fix possible segfault
* webrtc: fix browser offer handling
* Space at the beginning of sip: creates errors
* opus_multistream: update mimetype to ad-hoc standard
* webrtc: add offerer and recvonly options
* test: replace RSA cert with EC cert
* Add OPTIONS handling for webrtc demo
* mk: remove rtpext.c from srcs.mk
* ua: change refer log to info()
- switch to Cmake builds
- add devel and libbaresip2 sub-packages
- add pcp module
- remove sndio and v4l2 modules
/usr/lib64/baresip/modules/av1.so /usr/lib64/baresip/modules/avcodec.so /usr/lib64/baresip/modules/avfilter.so /usr/lib64/baresip/modules/avformat.so /usr/lib64/baresip/modules/fakevideo.so /usr/lib64/baresip/modules/gst.so /usr/lib64/baresip/modules/sdl.so /usr/lib64/baresip/modules/selfview.so /usr/lib64/baresip/modules/swscale.so /usr/lib64/baresip/modules/vidbridge.so /usr/lib64/baresip/modules/vidinfo.so /usr/lib64/baresip/modules/vp8.so /usr/lib64/baresip/modules/vp9.so /usr/lib64/baresip/modules/x11.so /usr/share/licenses/baresip-video /usr/share/licenses/baresip-video/LICENSE
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Fabrice Bellet, Mon Aug 3 00:14:01 2026