Changelog for janus-gateway-0.12.3-19.78.i586.rpm :

* Sun Jun 26 2022 ecsosAATTopensuse.org- Update to version 0.12.3:
* Updated Changelog (0.12.3)
* Use inet_pton instead of inet_net_pton
* Only reset rid when processing video m-line (fixes #2992)
* Add new shared JavaScript file for settings in demos (see #2991)
* Fixed broken VP8 payload descriptor parsing when 7-bit PictureID are used
* Fixed typo in destroy request of Streaming plugin
* Fixed exception when adding helper in SIP plugin demo
* Fixed missing contact header in SUBSCRIBE (#2973) and crash in SIP plugin when freeing a session while a subscription is active (2974)
* Fixed negotiation of RTP extensions when direction is involved
* Improved check on when to send playout-delay extension
* Fixed missing checks on auth challenges in SIP plugin
* Keep track of extensions when storing packets for retransmission (see #2981)
* Fixed issues/PRs links in ChangeLog
* Bumped to version 0.12.3 (legacy)
* Sun Jun 26 2022 ecsosAATTopensuse.org- Update to version 0.12.2:
* Updated Changelog (0.12.2)
* Fixed RED parsing not returning blocks when only primary data is available
* Link to -lresolv explicitly when building websockets transport
* Fix build with libressl >= 3.5.0 (see #2980)
* Fix address size in Streaming plugin RTCP sendto call (#2976)
* Make SIP timer T1X64 configurable (see #2972)
* Added custom headers for SIP SUBSCRIBE requests (see #2971)
* Fixed spaces instead of tabs
* Added synchronous request to start/stop recording single participant in VideoRoom
* Fixed typo in stereo support in EchoTest plugin
* Added configurable property to put a cap to task threads (see #2964)
* Return an error when attempting to postprocess a non-MJR file
* Disable IPv6 in WebSockets transport if binding to IPv4 address explicitly (see #2969)
* Honor \"audio\", \"data\", \"video\" flag changes in subscriber updates. (#2963)
* Abort DTLS handshake if DTLSv1_handle_timeout returns an error
* Fixed error message being displayed incorrectly when creating a mountpoint
* Bumped to version 0.12.2 (legacy)
* Sun Jun 26 2022 ecsosAATTopensuse.org- Update to version 0.12.1:
* Updated Changelog (0.12.1)
* Fix misaligned access in pp-rec when parsing VP8/VP9 frame resolution
* Fix error message when creating a session (see #2890)
* Also return reason header protocol and cause if present in BYE in the SIP plugin (see #2935)
* Better error when trying to rejoin on an existing VideoRoom handle
* Extend H.265 keyframe checks to more NALs (see #2323)
* videoroom: always default {min,max}_delay to -1 (#2936)
* Check if IPv6 is disabled to avoid failure when creating forwarder socket (see #2915 and #2916)
* Reset rids when renegotiating SDPs (see #2927 and #2931)
* Fix segfault in UNIX transport teardown when pathnames have different sizes.
* Fix highest sequence number not being properly initialized in the RTCP context (see #2920)
* Added new request to SIP plugin to reset the establishing flag if it\'s stuck
* Fixed typo in config sample (missing quote) (see #2912)
* Bumped to version 0.12.1 (legacy)
* Sun Jun 26 2022 ecsosAATTopensuse.org- Update to version 0.12.0:
* Updated Changelog (0.12.0)
* Fixed typos (see #2909)
* Add quirk for RTSP servers (#2909)
* Fix transport-wide CC feedback when simulcast SSRCs are missing (see #2908)
* Update version number in npm package (see #2901)
* Add support for playout-delay RTP extension (see #2895)
* Fixed leftover variable in janus.js
* Fix some other build warnings (output should be clean now).
* Added missing fix related to #2894 (0.x)
* Fix definition of trylock wrapper when using pthreads (see #2894)
* Link to Duktape engine as a library (see #2886)
* Remove distinction between simulcast and simulcast2 in janus.js (see #2887)
* Added checks when inserting RTP extensions to avoid buffer overflows
* Fix some build warnings.
* Keep track of when simulcast substreams are disabled in SDP (see #2888)
* Fixed broken RTP when no extensions are negotiated (see #2881)
* Link to CONTRIBUTING file from README instead of ISSUE_TEMPLATE folder
* Fixed crash at startup when not able to connect to RabbitMQ server
* Bumped to version 0.12.0 (legacy)
* Fix links in Changelog
* Make 0.x more apparent in README
* Clarify this is the 0.x branch in the README, and update links to demos/docs
* Preliminary changes for 0.x branch of Janus
* Sun Jun 26 2022 ecsosAATTopensuse.org- Update to version 0.11.8:
* Updated Changelog (0.11.8)
* Change automatic allocation on static loops from round robin to least loaded (#2878)
* Fixed outdated janus-pp-rec option description
* Changed default distance in postprocessor to 0, and removed unneeded DTX flag
* Check continuity of Opus packets when postprocessing (see #2880, for DTX)
* Reset extensions struct when not used
* fix PCMA/PCMU RTP forwarding in audiobridge - incorrect RTP header offset (#2875)
* Fixed runtime error
* Update RTP extensions for outgoing packets in the PeerConnection loop (fixes #2867) (#2869)
* Added link to FOSDEM2022 presentation to FAQ
* Fix last stats before closing PeerConnection not being sent to handlers (replaces #2873) (#2874)
* janus: add yet another missing NULL check for opusred (#2872)
* Add another missing NULL check on stream when setting opus RED
* Modified issues template
* janus: add missing NULL check on stream (#2865)
* Fixed broken duration in spatial AudioBridge recordings
* Fixed ambiguity in AudioBridge documentation (fixes #2863)
* Add new API to bulk start/stop MJR-based recordings in AudioBridge (#2862)
* Remove deprecated issue template
* Update issue templates
* Fixed missing check after merge
* Initial support for AV1-SVC Dependency Descriptor (#2741)
* Initial integration of RED for audio (#2685)
* Fixed broken recordings in NoSIP plugin
* Add a couple of checks after static analysis
* Bumbed to version 0.11.8
* Sun Jun 26 2022 ecsosAATTopensuse.org- Update to version 0.11.7:
* Updated Changelog (0.11.7)
* Fixed problems compiling post-processor with older versions of FFmpeg
* Added option to print extended header in janus-pp-rec (inspired by #2838) (#2858)
* Make rtp port range variables static in ice.c to prevent clashes with plugins (#2860)
* Added number of subscribers in response to listpartipants (fixes #2856)
* Added more checks before dereferencing
* Make record directory changeable via edit in AudioBridge and VideoRoom
* Allow pcap2mjr to autodetect SSRC
* Validate call_id when handling calls and messages (see #2853)
* Add strerror to errno-related log lines in SCTP code
* Protect updates to callid in SIP plugin (see #2853)
* Fixed broken \'sips\' in SIP plugin with Sofia >= 1.13 (see #2683)
* Disable sips by default in SIP plugin (fixes issue with Sofia >= 1.13, see #2683)
* fix saving signed_tokens field when room is permanent (#2843)
* Change SDP syntax for AV1 from \"AV1X\" to \"AV1\" (fixes #2844)
* Handle sdp in 180 sip message same as 183 (#2849)
* html: update webrtc-adapter to 8.1.1 (#2848)
* janus.d.ts typing fixes (#2847)
* Add reason when failing to open DTLS cert/key file (see #2845)
* Updated year in demos and docs
* Add NULL checks for stream and component in janus_ice_media_stopped (fixes #2840)
* janus.d.ts missing typing (#2837)
* Remove unneeded codec configuration in janus-pp-rec (fixes #2833)
* Add configurable expected loss to AudioBridge to actually send FEC (#2802)
* Fixed error in Changelog
* Small code style tweaks
* Take note of video orientation extension when recording video in SIP plugin (#2836)
* Add strlcat variant that uses memccpy for writing SDPs (#2835)
* Fixed missing XSS mitigation (see #2817)
* Bumbed to version 0.11.7
* Fixed warning
* Sun Jun 26 2022 ecsosAATTopensuse.org- Update to version 0.11.6:
* Updated Changelog (0.11.6)
* Reject SDPs with invalid media types, and realloc less often when writing one
* Moved from linux/ip.h to netinet/ip.h (#2831)
* Fix syntax error in VideoRoom (#2832)
* Add transport-cc to video rtcp-fb list too
* Fixed extra CRLF being appended when adding extensions in offers
* Added Janode presentation to the docs
* Allow VideoRoom to choose whether signed tokens should be used (#2825)
* Add missing error code in SIP plugin (fixes #2830)
* conf: add exit_on_dl_error option (#2828)
* Protect removal from hashtables when destroying SIP sessions (fixes #2818) (#2823)
* Added more videos to the documentation
* Removed loop initial declarations (fixes errors on some compilers)
* Send receiving:false notifications right after renegotiating (see #2807 and #2808) (#2824)
* Added additional check when mixing in AudioBridge
* Added basic history support to TextRoom plugin (#2814)
* Fix potential Cross-site Scripting (XSS) exploits in demos (#2817)
* Fixed typos
* Create SECURITY.md (fixes #2815)
* Fix warning and remove provided token from error response
* add typescript client lib (#2813)
* Fix signed token auth not work on join to the videoroom #2810 (#2812)
* Disable slowlink events by default (#2803)
* plugins/janus_sip.c: MESSAGE Authentication and Deliver Status Report (#2786)
* Added support for multiopus to janus-pp-rec
* add support for custom datachannel options in janus.js (#2806)
* Added janode to the list of resources in the docs
* html: update webrtc-adapter to 8.1.0 (#2798)
* Fixed incorrect info in SIP plugin documentation
* Grow buffer as needed when generating SDPs (fixes #2791, see #2793) (#2797)
* Added missing unrefs (fixes #2795)
* utils: add janus_strlcat helper (#2792)
* Initial support for DTX (EchoTest, VideoRoom) (#2789)
* Added new presentation video to the documentation
* Fixed typos in Streaming demo
* Fix plain HTTP links
* Bumbed to version 0.11.6
* Sun Jun 26 2022 ecsosAATTopensuse.org- Update to version 0.11.5:
* Reverted version bump
* Bumbed to version 0.11.6
* Updated Changelog (0.11.5)
* Small tweaks to #2785
* Allow media statistic events to get dispatched as one event per stream instead of dedicated ones per media (#2785)
* Fix WebSockets admin unix sockets, and streamlined parsing code (#2787)
* Avoid ICE local setup if handle has been destroyed (fix some memory leaks)
* Print time of when video is rotated, if available
* Add info on incoming REMB values to Admin API and Event handlers
* Add the token used to join the room to the info sent to event handlers (#2773)
* Ensure we access room uniformly and safely across all the VideoRoom code (#2782)
* Added some IDE notes to the docs (see #2784)
* Fixed systemd sample in documentation (#2783)
* Add pause/resume recording functionality to Record&Play and SIP plugins (#2724)
* Insert private ID mapping only when participant is added to the list (fixes #2781)
* Fix missing unlock/unref in case of publisher errors (fixes #2780).
* Make janus.js pass linter (#2772)
* Add API to force Janus to use TURN (#2774)
* Use prepend with reverse list in the sdp parsing (Credit to OSS-Fuzz) (#2776)
* handle host being in ignore list and enforced list (#2768)
* Fixed broken upsampling in AudioBridge
* Add query string to force codec when joining AudioBridge demo
* Moved pragma precondition from the middle of an if (fixes #2766)
* Fixed AudioBridge plain RTP thread sometimes exiting prematurely
* Clarify in documentation that the mobile SDK is abandoned
* Fix compilation warning on openSUSE
* Bumbed to version 0.11.5
* Sun Jun 26 2022 ecsosAATTopensuse.org- Update to version 0.11.4:
* Updated Changelog (0.11.4)
* Doxygen tweaks
* Fixed compilation warning
* Fix streaming mountpoint name generation when starting with 0 (#2764)
* Added new project to resources in the docs
* Added plain RTP participants documentation to AudioBridge
* Clearer indication of payload types when using plain RTP participants in AudioBridge
* Fixed optional arguments in Duktape datachannel relay methods
* Fixed datachannel protocol not being sent to plugins for incoming messages (fixes #2753)
* Fix partial/broken ACL support in TextRoom plugin (#2763)
* Configurable mechanism for manually setting static event loop to use for new handles (see #2450) (#2684)
* Use crypto safe random numbers (#2738)
* Adds reconnect() to the types definition (#2745)
* Added support for forwarding groups in AudioBridge (#2653)
* Fixed README (see #2744)
* Support for abs-send-time RTP extension (#2721)
* Fix sample event handler behaviour (#2743)
* Fixed link to usrsctp in documentation
* Fixed indentation in README
* Update README.md (#2744)
* Fixed broken links in Changelog
* Fixed broken links in Changelog
* Fixed typo in comment
* Fixed incoming_header_prefixes not working for helper sessions in SIP plugin
* Fixed incoming_header_prefixes not working for helper sessions in SIP plugin
* Videoroom API \"list\" endpoint: update docs, add is_private (#2715)
* Improve websocket event handler reconnection handling on fail
* Log connection error reason, in WS event handler
* Set new remote credentials after restarting ICE (fixes #2672) (#2729)
* Rename macro constant in ws event handler.
* Improve ws event handler reconnection and set an upper bound for the backoff (fixes #2734).
* Better SRTP-SDES negotiation in SIP/NoSIP plugins (fixes #2726) (#2727)
* Wake up the lws loop when destroying the WebSocket transport.
* Extend clang matching rule (let autoconf match afl-clang-fast in oss-fuzz).
* Revert \"Remove unneeded definition of RAND_bytes from the RTP fuzzer.\"
* Remove unneeded definition of RAND_bytes from the RTP fuzzer.
* [janus-pp-rec] activate keyframe logic for VP9 (#2730)
* Fix SIP plugin doxygen command for section reference (#2732)
* Manually add SIP Contact header for calls when using Sofia >= 1.13 (fixes #2439, replaces #2597) (#2708)
* Use PKG_CONFIG env variable instead of calling pkg-config directly (fixes #2713)
* Fix sending BYE (#2709)
* demos: update webrtc adapter 8.0.0 (#2702)
* Fixed some warnings when compining SIP plugin
* Add ability to specify recordings folder in AudioBridge (#2707)
* Fix potential race condition when reclaiming sessions in HTTP plugin
* Small fixes after static analysis
* Bumbed to version 0.11.4
* Sun Jun 26 2022 ecsosAATTopensuse.org- Update to version 0.11.3:
* Updated Changelog (0.11.3)
* Fix missing secret check on room enable_recording (#2706)
* Fixed AudioBridge recording stop/start not working properly on empty rooms (see #2674)
* Addess notes to #2674 (missing validation, extra timestamp in static filename)
* New enable_recording API to dynamically start/stop AudioBridge recordings (#2674)
* Fix deadlock on mountpoint destroy during RTSP reconnect (#2700)
* Fix SIP plugin missed_call event call_id field name typo (#2703)
* Fix memory leak on aborted RTSP connection (#2699)
* Fixed broken switching when using different payload types in Streaming plugin (#2692)
* Fixes on JavaScript code snippets (#2695)
* Fixed typo in docs
* Added missing semicolon (#2693)
* Optionally allow IPv6 link-local addresses to be gathered too (#2689)
* Additional target formats for some recorded codecs (fixes #2658) (#2680)
* Fix streaming plugin mutex unlock when disabling mountpoint (#2690)
* Fix SIP plugin unhold request docs typo (#2688)
* minor adjustment to the audiobridge docs (#2687)
* fix: [janus_sip] Fix \"call_id\" property in \"missed_call\" events (#2679)
* Fix status vector parsing for incoming twcc feedbacks (resolves #2677).
* Fixed race condition in VideoRoom
* Fixes variable name.
* Clarify that libnice 0.1.18 is recommended
* Spatial audio support in AudioBridge via stereo mixing (#2446)
* Cleanup avformat-based preprocessors (#2665)
* Fixed broken simulcast support in VideoCall plugin (#2671)
* feat: support for custom call-id in subscribe request + add \'call_id\' property to subscribe & notify related events (#2664)
* Fixed missing macro when using pthread mutexes (fixes #2666)
* Fixed warning
* Remove duplicated flag for fuzzing coverage.
* janus-pp-rec: support HEVC AP(aggregation packet) (#2662)
* Fixed out of bounds array access
* feat: support for SUBSCRIBE expiry (Expires header) in sip plugin (#2661)
* Fixed types
* RabbitMQ Transport Reconnect Logic (#2651)
* Add per-participant recording options in AudioBridge to join API as well
* Small simulcast-related demo tweaks
* Make sure the source exists, before scheduling nice_agent_close_async (fixes #2655)
* janus.js - renegotiate with external stream (#2604)
* createOffer crashes in Firefox (on Mac) when re-negotiating p2p (eg. mute audio) (#2656)
* Fixed broken g_strsplit limit in VideoRoom when parsing supported codecs (fixes #2657)
* Fixed missing properties in permanent AudioBridge config saves
* Add support for plain RTP participants in AudioBridge (#2464)
* Bugfix: make sure that every destroyed plugin handle is \'detached\' (#2652)
* Bumbed to version 0.11.3
* Sun Jun 26 2022 ecsosAATTopensuse.org- Update to version 0.11.2:
* Updated Changelog (0.11.2)
* Added reference to AudioBridge announcements before using them in the mix
* Added support for datachannel label/protocol to Lua and Duktape plugins (#2641)
* Remove support for framemarking RTP extension (#2640)
* Prevent race conditions on socket close in SIP and NoSIP plugins (#2599)
* Fixed overflow runtime error
* Fix for race condition between VideoRoom publisher leaving and subscriber hanging up (fixes #2582) (#2637)
* Send PLI when starting a paused stream (#2645)
* Prevent too high shift exponent
* Fixed type of seq/ts in file-based Streaming mountpoint threads
* Fix missing g_thread_unref when a streaming helper thread quits.
* Added custom headers for SIP INFO request (#2644)
* Free participant->user_id_str in case of opus enc/decoder error.
* Reject flexfec when offered, as still unsupported (see #2639)
* Don\'t add rtx ssrc if m-line is recvonly/inactive (see #2639)
* Added NULL checks for json_dumps (see #2629)
* Parse custom headers, if required, in successful REGISTER response (fixes #2636)
* Timestamp correction for janus-pp-rec (#2573)
* Don\'t chain error handler to success handler in Janus.httpAPICall (#2569)
* Add missing library link for WS event handler (fixes #2628)
* Fixed broken switch in Streaming plugin when using helper threads
* Resolves meetecho/janus-gateway#2624 jansson double referencing (#2634)
* Unlock mountpoints mutex after the spawning of helper threads.
* Don\'t fail on duplicate b= lines in the SDP (see #2558)
* Added ability to use websockets over unix sockets (#2620)
* Fix typo in getJanusToken function (#2631)
* Reference new mountpoint when switching, and check it has been destroyed
* Fix streaming plugin RTSP sample in config template (#2627)
* Removed unneeded brackets
* fix moutpoint_mutex deadlock when rtsp reconnect fail (#2542)
* Add info on moderation to list of publishers, if enabled
* mqttevh: added ability use relative to config paths in config (#2623)
* SIP plugin: SIP MESSAGE out of dialog (#2616)
* Added missing mutex initialization (fixes #2622)
* Bumbed to version 0.11.2
* Sun Jun 26 2022 ecsosAATTopensuse.org- Update to version 0.11.1:
* Fixed typo in CHANGELOG
* Updated Changelog (0.11.0)
* Added new --log-stdout flag that enabled stdout logging even when daemonizing the process (#2591)
* Added extra reference to VideoRoom
* Add substream to audio/video receiving events (fixes #2615)
* Fixed typo
* Added new video (SIP/Janus workshop) to the documentation
* Added session timeout value to Admin API info
* Initialize simulcast RTCP contexts even if SSRCs are missing (fixes #2610)
* audiobridge: GList leak fixed (#2611)
* Fixed sending responses from Janus for incoming SIP MESSAGE/SIP INFO (#2609)
* Added wss and debugging support to WebSocket event handler
* Fixed broken path parsing in WebSocket event handler (fixes #2603)
* fix memory leak in turnrest (#2606)
* Making the timeout parameters for streaming plugin for RTSP play out configurable (#2598)
* FreeBSD support (#2508)
* Added support for admin-protected custom session timeouts (#2577)
* Document \'timeout\' and \'detached\' events (fixes #2576)
* Added more videos to the list of presentations in the FAQ
* Fix warning about comparison of integers of different signedness in rtcp.
* Added check on participant destroyed flag before sending events
* Fixed typos in Changelog
* Add some checks to publisher destroyed flag.
* errors: replace strerror with locale-safe and threadsafe g_strerror (#2565)
* Fix auth when both (token, secret) modes are enabled (#2581)
* Changing default ICE nomination mode to \'aggressive\' (see #2541)
* fix rstp instead of rtsp typo (#2590)
* Fix broken calculation of out-link-quality when NACKS exceed number of sent packets (fixes #2579)
* Fixed memory leak
* Make sure the publisher hasn\'t been destroyed, before trying to relay RTCP
* Added Content type to SIP message (#2567)
* clang/ubsan fixes (#2556)
* add call_id in received sip message (#2563)
* Fixed missing mutexes around VideoRoom ACL management
* ice: fix conncheck typo (#2560)
* feat: add \"call_id\" to \"calling\", \"declining\", \"updatingcall\" & \"incomingcall\" events (#2557)
* Video moderation always returns unmuted (#2559)
* Fixed typo in keepalive-conncheck usage
* Add reference to publisher when using RTCP in forwarder
* Set specific versions for Python 3 and meson in janus-ci yml.
* Added audiocodec/videocodec supporto to \'joinandconfigure\' in VideoRoom API
* Add new option to configure ICE nomination mode, if libnice is recent enough (#2541)
* if inviting on destroy, send BYE instead of 480 response (#2554)
* Fix typo in videoroom docs.
* Fixed small leak in VideoRoom
* Initialize packet.is_rtp to false.
* Add resolution and bitrate to Record&Play playback
* Update janus.d.ts (#2553)
* Allow up to 5 (rather than 3) audio/video codecs in the same VideoRoom
* Allow forcing audio/video codec for VideoRoom publishers via query string
* Initialize VideoRoom participant recording state when room recording is active (fixes #2550)
* Fixed broken AV1 post-processing
* Renamed extern janus_callbacks variables in Lua and Duktape plugins (#2540)
* Bumped to version 0.11.1
* Sun Jun 26 2022 ecsosAATTopensuse.org- Update to version 0.10.10:
* Updated Changelog (0.10.10)
* Videoroom race condition fixes (see #2509) (#2539)
* Fix parsing of SDP to find payload type matching profiles (fixes #2544) (#2549)
* janus.js (#2548)
* Make compiler fail if implicit-function-declaration is encountered.
* Fixed non-portable call to strlcpy, and comment styles, in RabbitMQ code (see #2430)
* Fixed VideoRoom docs on ICE Restarts for subscribers (fixes #2537)
* Allow marking of RTP extensions in MJR recordings (#2527)
* Moderator based muting/unmuting of VideoRoom streams (#2513)
* Reject a=extmap-allow-mixed in SDP, when offered
* Fix code style comments, also enable routing for direct exchanges
* Configurable media direction when putting calls on-hold (SIP plugin) (#2525)
* Added starting DTLS MTU to info returned by Janus API
* Report fail if binding to a socket fails in websockets (#2534)
* fix race condition in audiobridge plugin changeroom request (#2535)
* Janus npm types upgrade (#2528)
* set webrtc-adapter verstion to 7.4.0 (#2531)
* Reduced verbosity of a few LOG_WARN messages at startup
* Feature/enhance typings (#2518)
* Fixed secret authentication on GET requests (#2524)
* Dont send bye on early dialog (#2521)
* Update Webpack instruction after webrtc-adapter dependency update (#2519)
* Close nice agent resources asynchronously (#2492)
* mqttevh: tls support implementation finished (#2517)
* Fixed broken webrtc-adapter links (see #2515)
* html: update webrtc-adapter to 7.7.0 (#2515)
* Updated year in demos and docs
* Fixed crash in WS event handler when backend is unreachable
* Bumped to version 0.10.10
* Adds back in default outgoing queue behaviour. Adds support for auto-generated queue_names
* Adds RabbitMQ options for queues, durable, exclusive and autodelete
* Check RabbitMQ admin topic in a better way
* Increase RabbitMQ logging on publish
* Fix queue_name_admin in rabbitmq transport
* Update rabbitmq logging information
* Updates RabbitMQ logic
* Sun Jun 26 2022 ecsosAATTopensuse.org- Update to version 0.10.9:
* Updated Changelog (0.10.9)
* Fixed memory leak when using announcements in AudioBridge (see #2504)
* devicetest: unused var removed (#2502)
* Increase participant\'s ref while handling kick requeest.
* Fixed typo (see #2501)
* Fixed a few compile and runtime issues in WebSocket event handler
* Fix RTP headers when leaving/joining AudioBridge rooms on same PeerConnection
* Fix occasional missing \"left\" event in audiobridge (see #2499).
* Added note on Chrome bugs that prevent multiopus demo from working
* Fixed occasional SRTP errors when pausing and then resuming Streaming plugin handles after a long time
* Replace Travis CI with GitHub Actions. (#2486)
* [janus-pp-rec] Fix potential crash when using skew compensation.
* Fix regression on RTCP for sendonly video connections (fixes #2496).
* videoroom: log invalid request name (#2495)
* Set libmicrohttpd connections limit in http transport configuration. (#2489)
* Fixed create/attach management with null optional args (fixes #2490)
* Improve detection of maximum resolution in mjr file (postprocessor) (#2487)
* Added support for binary data recordings (#2481)
* Fixed inconsistency of mySdp in janus.js (fixes #2379)
* Updated instrunctions to build libwebsockets (see #2476)
* Reset simulcast context for a videoroom publisher also when renegotiating (fixes #2466).
* Skip SDP munging in janus.js if SIM attribute is already present in the offer (see #2466).
* Do not require a cacertfile, pass null to openssl (#2485)
* Increased size of buffer used to render new prflx candidates (fixes #2480)
* Replay data channel recordings (#2468)
* Fixed warning
* [janus-pp-rec] Drop audio RTP silence suppression packets. (#2467)
* Corrected janus pp rec (#2472)
* Add an option to enable libmicrohttpd error logs (#2471)
* Fix types (#2475)
* Check simulcast layer index also when stream is video only.
* Fixed incomplete recordings after SSRC change (e.g. hold/unhold) in SIP and NoSIP plugins
* Make TURN REST API timeout configurable in janus.jcfg (#2470)
* Added custom headers to \'decline\' request (#2465)
* Adding janus-angular to the Resources page of the documentation (#2459)
* Fix autoplay policy issues on Safari (see #2455).
* Added user gesture for Safari in screensharing demo (see #2455)
* [janus-pp-rec] Enhance timestamp assignment and add payload-type option to CLI. (#2345)
* Add Custom Headers Hold Event (#2454)
* fix: disable auto ack when answering incoming call (#2447)
* Fix incompatible-pointer-types compiler warnings (#2444)
* fixing a couple of bugs in burst transfers (#2427)
* Bumped to version 0.10.9
* Sun Jun 26 2022 ecsosAATTopensuse.org- Update to version 0.10.8:
* Updated Changelog (0.10.8)
* fix: disable responses to NOTIFY requests in janus_sip plugin (response is already handled in sofia-sip) (#2441)
* Add LIBSRTP_CFLAGS to compiler flags of plugins that require srtp headers (#2442)
* [janus-pp-rec] Do not overwrite original RTP header data when attempting audio skew compensation.
* [janus-pp-rec] Use 64-bit timestamps for audio skew compensation.
* Let SIP users cancel pending transactions without waiting for a provisional response. (#2434)
* janus_streaming: fix warnings if missing libogg (#2438)
* Added new video to FAQ
* Warn if sofia-sip logs are redirected from stdout.
* Small tweaks to AudioBridge prebuffering