-------------------------------------------------------------------
Fri Feb 11 22:27:07 UTC 2022 - michael@stroeder.com
- Update to version 0.11.8:
* Added initial (and limited) integration of RED audio (#2685)
* Added support for Two-Byte header RTP extensions (RFC8285) and, partially,
for the new Depencency Descriptor RTP extension (needed for AV1-SVC) (#2741)
* Fixed rare race conditions between sending a packet and closing a connection (#2869)
* Fix last stats before closing PeerConnection not being sent to handlers (#2874)
* Changed automatic allocation on static loops from round robin to least used (#2878)
* Added new API to bulk start/stop MJR-based recordings in AudioBridge (#2862)
* Fixed broken duration in spatial AudioBridge recordings
* Fixed broken G.711 RTP forwarding in AudioBridge (#2875)
* Fixed broken recordings in NoSIP plugin
* Fixed warnings when postprocessing Opus recordings with DTX packets
* Other smaller fixes and improvements
-------------------------------------------------------------------
Mon Jan 24 14:36:20 UTC 2022 - michael@stroeder.com
- Update to version 0.11.7:
* Added faster strlcat variant that uses memccpy for writing SDPs (#2835)
* Fixed occasional crash when updating WebRTC sessions (#2840)
* Changed SDP syntax for AV1 from "AV1X" to "AV1" (#2844)
* Fixed signed_tokens property not being saved to permanent rooms in VideoRoom (#2843)
* Made record directory changeable via "edit" in both AudioBridge and VideoRoom
* Added configurable expected loss to AudioBridge to actually send FEC (#2802)
* Fixed SIP plugin not working when using Sofia SIP >= 1.13 (#2683)
* Fixed occasional crashes in SIP plugin (#2853)
* Take note of video orientation extension when recording video in SIP plugin (#2836)
* Allow 180 besides 183 to have SDP as well (#2849)
* Fixed post-processor compilation issue with newer versions of FFmpeg (#2833)
* Added option to print extended info on MJR file as JSON in postprocessor (#2858)
* Allow pcap2mjr to autodetect SSRC
* Fixed problems compiling post-processor with older versions of FFmpeg
* Other smaller fixes and improvements
-------------------------------------------------------------------
Mon Dec 13 12:26:10 UTC 2021 - michael@stroeder.com
- Update to version 0.11.6:
* Fixed CVE-2021-4020 (see also boo#1193156):
Cross-site Scripting (XSS) vulnerability in some of the demos (#2817)
* Added strlcat helper to detect and report truncations (#2792)
* Grow buffer as needed when generating SDPs (#2797)
* Added DTX support to some plugins (#2789)
* Added option to forcibly quit Janus when getting dlopen errors (#2828)
* Fixed broken signed tokens in VideoRoom when using UUIDs (#2812)
* Added option to choose whether signed tokens should be used in the
VideoRoom when enabled in the core (#2825)
* Added configurable expected loss to AudioBridge to actually send FEC (#2802)
* Added MESSAGE authentication and out-of-dialog MESSAGE support to SIP plugin (#2786)
* Fixed potential race conditions in SIP plugin (#2823)
* Added basic history support to TextRoom plugin (#2814)
* Added support for custom datachannel options in janus.js (#2806)
* Other smaller fixes and improvements
-------------------------------------------------------------------
Mon Oct 18 16:20:34 UTC 2021 - michael@stroeder.com
- dropped obsolete 0001-include-rand-header-file.patch
- Update to version 0.11.5:
* Add API to optionally force Janus to use TURN (#2774)
* Fixed slow path on SDP parsing (#2776)
* Added event handlers option to send stats for a PeerConnection
in a single event, rather than per-media (#2785)
* Fixed occasional deadlocks on malformed requests in VideoRoom (#2780)
* Fixed AudioBridge plain RTP thread sometimes exiting prematurely
* Fixed broken upsampling when using G.711 in AudioBridge
* Add pause/resume recording functionality to Record&Play and SIP plugins (#2724)
* Fixed broken support for Unix Sockets in WebSockets Admin API (#2787)
* Added timing info for video rotation when post-processing recordings
* Added linter checks to janus.js (#2272)
* Other smaller fixes and improvements
-------------------------------------------------------------------
Thu Sep 23 05:14:47 UTC 2021 - Johannes Segitz <jsegitz@suse.com>
- Added hardening to systemd service(s) (bsc#1181400). Modified:
* janus.service
-------------------------------------------------------------------
Mon Sep 6 12:53:58 UTC 2021 - Michael Ströder <michael@stroeder.com>
- added janus-gateway-rpmlintrc
- removed systemd-related conditionals to fix obsolete-suse-version-check
- added 0001-include-rand-header-file.patch
- use %fdupes macro
- Update to version 0.11.4:
* Fixed ICE restart issues with recent versions of libnice (#2729)
* Changed randon number generators to use crypto-safe functions (#2738)
* Added support for abs-send-time RTP extension (#2721)
* Added configurable mechanism for manually setting static event loop to use for new handles (#2684)
* Fixed datachannel protocol not being sent to plugins for incoming messages (#2753)
* Added ability to specify recordings folder in AudioBridge (#2707)
* Added support for forwarding groups in AudioBridge (#2653)
* Fixed missing Contact header in SIP plugin when using Sofia >= 1.13 (#2708)
* Better SDES-SRTP negotiation in SIP and NoSIP plugins (#2727)
* Fixed WebSocket transport and event handler lagging 25/30s when shutting down or reconnecting (#2734)
* Fixed incoming_header_prefixes not working for helper sessions in SIP plugin
* Fix partial/broken ACL support in TextRoom plugin (#2763)
* Fixed potential race condition when reclaiming sessions in HTTP transport plugin
* Fixed WebSocket event handler reconnect mechanism (#2736)
* Other smaller fixes and improvements
-------------------------------------------------------------------
Tue Jun 15 13:44:32 UTC 2021 - michael@stroeder.com
- Update to version 0.11.3:
* Fixed rare crash when detaching handles (#2464)
* Added option to offer IPv6 link-local candidates as well (#2689)
* Added spatial audio support to AudioBridge via stereo mixing (#2446)
* Added support for plain RTP participants to AudioBridge (#2464)
* Added API to start/stop AudioBridge recordings dynamically
(thanks @rajneeshksoni!) (#2674)
* Fixed broken mountpoint switching when using different payload types
in Streaming plugin (#2692)
* Fixed occasional deadlock on Streaming plugin mountpoint destroy
during RTSP reconnects (thanks @lionelnicolas!) (#2700)
* Added "Expires" support to SUBSCRIBE in SIP plugin
(thanks @nicolasduteil!) (#2661)
* Added option to specify Call-ID for SUBSCRIBE dialogs in SIP plugin
(thanks @nicolasduteil!) (#2664)
* Fixed broken simulcast support in VideoCall plugin
(thanks @lucily-star!) (#2671)
* Implemented RabbitMQ reconnection logic, in both transport and event handler
(thanks @chriswiggins!) (#2651)
* Added support for renegotiation of external streams in janus.js
(thanks @kmeyerhofer!) (#2604)
* Added support for HEVC/H.265 aggregation packets (AP) to janus-pp-rec
(thanks @nu774!) (#2662)
* Refactored janus-pp-rec to cleanup the code, and use libavformat for Opus as well
(thanks @lu-zero!) (#2665)
* Added additional target formats for some recorded codecs (#2680)
* Other smaller fixes and improvements
-------------------------------------------------------------------
Mon May 03 10:24:07 UTC 2021 - michael@stroeder.com
- Update to version 0.11.2:
- Added support for relative paths in config files, currently only in
MQTT event handler (thanks @RSATom!) (#2623)
- Removed support for now deprecated frame-marking RTP extension
(#2640)
- Fixex rare race condition between VideoRoom publisher leaving and
subscriber hanging up (#2637)
- Fixed occasional crash when using announcements in AudioBridge
- Fixed rare crash in Streaming plugin when reconnecting RTSP streams
(thanks @lucylu-star!) (#2542)
- Fixed broken switch in Streaming plugin when using helper threads
- Fixed rare race conditions on socket close in SIP and NoSIP plugins
(#2599)
- Added support for out-of-dialog SIP MESSAGE requests (thanks
@ihusejnovic!) (#2616)
- Fixed memory leak when using helper threads in Streaming plugin
- Added support for datachannel label/protocol to Lua and Duktape
plugins (#2641)
- Added ability to use WebSockets transport over Unix sockets (thanks
@mdevaev!) (#2620)
- Added janus-pp-rec mechanism to correct wrong RTP timestamps in MJR
recordings (thanks @tbence94!) (#2573)
- Other smaller fixes and improvements
-------------------------------------------------------------------
Tue Apr 06 10:49:55 UTC 2021 - michael@stroeder.com
- Update to version 0.11.1:
* Add new option to configure ICE nomination mode, if libnice is recent enough (#2541)
* Added support for per-session timeout values (thanks @alg!) (#2577)
* Added support for compilation on FreeBSD (thanks @jsm222!) (#2508)
* Fixed occasional auth errors when using both API secret and stored tokens (#2581)
* Added support for stdout logging to daemon-mode as well (#2591)
* Fixed odr-violation issue between Lua and Duktape plugins (#2540)
* Fixed missing simulcast stats in Admin API and Event Handlers when using rid (#2610)
* Fixed VideoRoom recording not stopped for participants entering after global recording was started (#2550)
* Fixed 'audiocodec'/'videocodec' being ignored when joining a VideoRoom via 'joinandconfigure'
* Added content type support to MESSAGE in SIP plugin (#2567)
* Made RTSP timeouts configurable in Streaming plugin (#2598)
* Fixed incorrect parsing of backend URL in WebSockets event handler (#2603)
* Added support for secure connections and lws debugging to WebSockets event handler
* Fixed occasionally broken AV1 recordings post-processing
* Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
-------------------------------------------------------------------
Mon Feb 08 10:27:21 UTC 2021 - michael@stroeder.com
- Update to version 0.10.10:
* Reduced verbosity of a few LOG_WARN messages at startup
* Close libnice agent resources asynchronously when hanging up PeerConnections (thanks @fbellet!) [PR-2492]
* Fixed broken parsing of SDP when trying to match specific codec profiles [PR-2549]
* Added muting/moderation API to the VideoRoom plugin [PR-2513]
* Fixed a few race conditions in VideoRoom plugin that could lead to crashes [[PR-2539][#2539)]
* Send 480 instead of BYE when hanging up calls in early dialog in the SIP plugin (thanks @zayim!) [PR-2521]
* Added configurable media direction when putting calls on-hold in the SIP plugin [PR-2525]
* Fixed rare race condition in AudioBridge when using "changeroom" (thanks @JeckLabs!) [[PR-2535][#2535)]
* Fixed broken API secret management in HTTP long polls (thanks @remvst!) [PR-2524]
* Report failure if binding to a socket fails in WebSockets transport plugin (thanks @Symbiatch!) [PR-2534]
* Updated RabbitMQ logic in both transport and event handler (thanks @chriswiggins!) [PR-2430]
* Fixed segfault in WebSocket event handler when backend was unreachable
* Added TLS support to MQTT event handler (thanks @RSATom!) [PR-2517]
* Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
-------------------------------------------------------------------
Wed Dec 23 12:32:09 UTC 2020 - michael@stroeder.com
- Update to version 0.10.9:
* Replaced Travis CI with GitHub Actions [[PR-2486](#2486)]
* Fixed data channel messages potentially getting stuck in case of burst transfers (thanks @afshin2003!) [[PR-2427](#2427)]
* Fixed simulcast issues when renegotiating PeerConnections [[Issue-2466](#2466)]
* Added configurable TURN REST API timeout (thanks @evorw!) [[PR-2470](#2470)]
* Added support for recording of binary data channels [[PR-2481](#2481)]
* Fixed occasional SRTP errors when pausing and then resuming Streaming plugin handles after a long time
* Fixed occasional SRTP errors when leaving and joining AudioBridge rooms without a new PeerConnection after a long time
* Added support for playout of data channels in Record&Play plugin and demo (thanks @ricardo-salgado-tekever!) [[PR-2468](#2468)]
* Added option to override connections limit in HTTP transport plugin [[PR-2489](#2489)]
* Added options to enable libmicrohttpd debugging in HTTP transport plugin (thanks @evorw!) [[PR-2471](#2471)]
* Fixed a few compile and runtime issues in WebSocket event handler
* Refactored postprocessing management of timestamps to fix some key issues [[PR-2345](#2345)]
* Fixed postprocessing of audio recordings containing RTP silence suppression packets [[PR-2467](#2467)]
* Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
-------------------------------------------------------------------
Tue Dec 22 09:25:17 UTC 2020 - Jan Engelhardt <jengelh@inai.de>
- Do not hard-require systemd. Drop redundant wording from
description. Drop %defattr.
-------------------------------------------------------------------
Mon Dec 21 19:16:41 UTC 2020 - Michael Ströder <michael@stroeder.com>
- Initial packaging of 0.10.8 for Factory