- 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)

- 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)

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- 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  * Fixed outdated janus-pp-rec option description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

- Update to version 0.11.7:
  * Updated Changelog (0.11.7)

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- 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

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- Update to version 0.10.7:
  * Updated Changelog (0.10.7)
  * More aggressive PLI at startup when using simulcast in VideoRoom plugin
  * Configured janus-pp-rec to skip packets with unknown payload types when static payload types are expected (G.711, G.722)
  * Fixed missing initialization of AVPacket that could cause crashes when postprocessing G.722 recordings
  * Modified demos to remove hardcoded 320x240 video element slot
  * Fix RTP header buffer read (#2411)
  * Add missing unref to mDNS resolver gobject. (#2399)
  * Use PKG_CONFIG_PATH as configured for nice version (#2405)
  * Replace rand() with janus_random_uint32() (fixes #2404)
  * Fixed occasional memory leak at shutdown when frequently using timed callbacks in Lua/Duktape plugins
  * Updated insertable streams code in janus.js (and e2ee demo)
  * janus.d.ts: correct mediaState definition (#2396)
  * Minor typo fix (#2393)
  * Fixed broken rid-based simulcast for substreams<3
  * Fixed typo in AudioBridge docs (see #2391)
  * janus.js: allow configuring simulcast send encoding parameters (#2392)
  * Fixed broken indentation
  * Refresh lws 4.x connection validity for any ws incoming message.
  * fix warning about being deployed on private IP (#2386)
  * Implement mutex on rabbitmq-event to control connection (#2380)
  * Do not handle session stack mutex if helper has not been created. (#2387)
  * Fix SDP negotiation when client uses max-bundles (fixes #2390)
  * Keep extra whitespace in legacy simulcast rid SDP line
  * Removed extra whitespace in simulcast recv SDP line
  * Add JSEP flag to invert processing order of rid in SDP (#2385)
  * Fixed compilation error when using libwebsockets < 3
  * Allow AudioBridge to originate SDP offers (#2366)
  * Bumped to version 0.10.7
- Drop libwebsockets-3.patch, because now in upstream.

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- Update to version 0.10.6:
  * New mechanism to tweak/query transport plugins via Admin API [PR-2354]
  * Fixed occasional segfault when using event handlers and VideoRoom [Issue-2352]
  * Fixed occasional "Unsupported codec 'none'" log errors (thanks @neilyoung!) [PR-2357]
  * Fixed broken AudioBridge RTP forwarding when using G711 [Issue-2375]
  * Added helper threads support to RTSP mountpoints as well [PR-2361]
  * Fixed data channels not working as expected in Streaming plugin when using helper threads
  * Fixed simulcast occasionally not working in Streaming plugin until manual PLI trigger
  * Added proper fragmentation in WebSockets transport plugin [PR-2355]
  * Fixed timing resolution issue in MQTT transport (thanks @feymartynov!)) [PR-2358]
  * Fixed MQTT transport issue when trying to shutdown gracefully (thanks @feymartynov!)) [PR-2374]
  * Fixed broken configuration of Nanomsg Admin API (thanks @sdamodharan!)) [PR-2372]
  * Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)

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- Update to version 0.10.5:
  * Bugfix: prevent borked generated audio file if meetecho header is present with no RTP data next (#2356)
  * Don't print SDP errors if rtx is being negotiated for audio
  * Fixed deprecated lws semantics in WS event handler too
  * Remove deprecated libwebsockets semantics in WS transport (see #2349)
  * Added presentation on Insertable Streams to docs
  * Add missing documentation for janus-pp-rec.
  * Various minor typo fixes (#2313)
  * Fixed typo (see #2341)
  * Easy support for one-to-many scenarios in videoroomtest (#2341)
  * Kick (#2332)
  * Clear publisher codecs in videoroom hangup media.
  * Send a PLI (if supported) to the Streaming mountpoint source when switching (fixes #2333)
  * Added missing token in handle-related event (fixes #2312)
  * Fixed documentation in plugin videoroom (close #2301).
  * Remove unneeded mutex unlock that was causing a crash in the videoroom plugin (fixes #2318).
  * Bugfix: make audio/video recording in videocall working again
  * Bumped to version 0.10.5
  * Updated Changelog (0.10.4)
  * Added Janus workshop made at ClueCon 2020 to list of videos in the docs
  * Fixed definition of variable in for loop
  * Use unique IDs and internal hashtable to map SCTP associations with usrsctp (#2302)
  * Only use CURLOPT_HTTP09_ALLOWED if libcurl is >= 7.66.0 (fixes #2307)
  * Fix occasional curl hiccups with RTSP with some cameras
  * Fixed typo
  * Have VideoCall sessions reference each other, when in a call (see #2300)
  * Add more checks on peer when hanging up VideoCall session
  * Fix minor memory leak in participant inbuf of audiobridge plugin (#2298)
  * Allow specifying multiple IP addresses for 1-1 NAT. (#2279)
  * Fix candidates memory leaks (#2288)

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trigger service run

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trigger service run

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- Update to version 0.9.3:
  * Change libsrtp detection in the configure script to use pkg-config
  * Fixed compilation error with gcc10
  * Fixed RTCP issue that could occasionally lead to broken retransmissions when using rtx
  * Added option to specify DSCP Type of Service (ToS) for media streams
  * Fixed a couple of race conditions during renegotiations
  * Fixed VideoRoom and Streaming "destroy" not working properly when using string IDs
  * Fix occasional segfault in VideoRoom (thanks @cb22!)
  * Fixed AudioBridge "create" not working properly when using string IDs
  * Added support for playing Opus files in AudioBridge rooms
  * Added support to Opus files for file-based mountpoints in Streaming plugin
  * Added support for generic metadata to Streaming mountpoints
  * Streaming plugin now returns mountpoint IP address(es) in "create" and "info", when binding to specific IP/interface
  * Fixed occasional segfault when using helper threads in Streaming plugin
  * Fixed occasional race conditions in HTTP transport
  * Added support for specifying screensharing framerate in janus.js (thanks @agclark81!)
  * Cleaned up code in janus.js (thanks @alienpavlov!)
  * Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)
  * Converted HTTP transport plugin to single thread (now requires libmicrohttpd >= 0.9.59)
  * Fixed .deb file packaging (thanks @FThrum!)
  * Added foundation for aiortc-based functional testing (python)
  * Fixed occasional audio/video desync
  * Added asynchronous resolution of mDNS candidates, and an option to automatically ignore them entirely
  * Updated default DTLS ciphers (thanks @fippo!)
  * Added option to generate ECDSA certificates at startup, instead of RSA (thanks @Sean-Der!)
  * Fixed rare race condition when claiming sessions
  * Fixed rare crash in ice.c (thanks @tmatth!)
  * Fixed dangerous typo in querylogger_parameters (copy/paste error)
  * Fixed occasional deadlocks in VideoRoom (thanks @mivuDing and @agclark81!)
  * Added support for RTSP Content-Base header to Streaming plugin

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