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Tagged versions and Changelog

There are different tagged versions on the Janus repository. We usually tag a new version any time a breaking change and/or a set of comprehensive changes and fixes is going to be merged/applied to Janus, and so the Changelog below can act as a simple and quick summary of which changes are available in each version.

It's very important to point out, though, that tagged version are NOT to be considered stable versions. This is a common misunderstanding. The only version we consider stable is master as it's the only branch where we continuously provide fixes and enhancements: this is particularly important in the WebRTC world, where it's not uncommon to see features breaking overnight due to changes in how browsers and other WebRTC devices implement things. As such, again, a tagged version is only a way to take a snapshot of where Janus was at a specific point in time, and before a more or less major change occurred. While you're free to stick to tagged versions for your deployments (e.g., because that's how provisioning is usually done in your company), please notice we will ignore issues and reports addressing any other branch that is not master: due to lack of time and resources, we simply cannot go and investigate issues we may have fixed already, so if you're experiencing issues, make sure you can replicate them on master as well first.

# Changelog

All notable changes to this project will be documented in this file.


## [v0.10.2] - 2020-06-17

- Fixed sscanf-related security issues [[PR-2229](https://github.com/meetecho/janus-gateway/pull/2229)]
- Fixed some RTP extensions not working after renegotiations [[Issue-2192](https://github.com/meetecho/janus-gateway/issues/2192)]
- Fixed occasionally broken simulcast behaviour [[PR-2231](https://github.com/meetecho/janus-gateway/pull/2231)]
- Fixed "switch" request not taking simulcast/SVC into account in VideoRoom and Streaming plugins [[Issue-2219](https://github.com/meetecho/janus-gateway/issues/2219)]
- Fixed inability to ask for random ports when creating Streaming plugin mountpoints with simulcast support [[PR-2225](https://github.com/meetecho/janus-gateway/pull/2225)]
- Fixed occasional crashes in SIP plugin when using helpers [[PR-2216](https://github.com/meetecho/janus-gateway/pull/2216)]
- Updated Duktape dependencies to v2.5, and fixed Duktape plugin relaying text data as binary [[PR-2233](https://github.com/meetecho/janus-gateway/pull/2233)]
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.10.1] - 2020-06-11

- Added initial support for AV1 and H.265 video codecs [[PR-2120](https://github.com/meetecho/janus-gateway/pull/2120)]
- Added initial support for end-to-end encryption via Insertable Streams [[PR-2074](https://github.com/meetecho/janus-gateway/pull/2074)]
- Fixed security issues when processing SDPs [[PR-2214](https://github.com/meetecho/janus-gateway/pull/2214)]
- Fixed occasional codec profile negotiation issues (thanks @groupboard!) [[PR-2212](https://github.com/meetecho/janus-gateway/pull/2212)]
- Fixed occasional segfaults when hanging up VideoRoom subscribers
- Fixed RTSP issues when fmtp is missing (thanks @lionelnicolas!) [[PR-2190](https://github.com/meetecho/janus-gateway/pull/2190)]
- Fixed RTSP not following redirects, when used (thanks @lionelnicolas!) [[PR-2195](https://github.com/meetecho/janus-gateway/pull/2195)]
- Fixed SRTP-SDES and renegotiation issues in NoSIP plugin (thanks @ihusejnovic!) [[PR-2196](https://github.com/meetecho/janus-gateway/pull/2196)]
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.10.0] - 2020-06-01

- Added support for negotiation of codec profiles (mainly VP9 and H.264) [[PR-2080](https://github.com/meetecho/janus-gateway/pull/2080)]
- Added new callback to let plugins know when the datachannel first becomes available, and then any time it's writable (empty buffers) [[PR-2060](https://github.com/meetecho/janus-gateway/pull/2060)]
- Added support for data channel subprotocols [[PR-2157](https://github.com/meetecho/janus-gateway/pull/2157)]
- Added new event handler for GrayLog using GELF (thanks @mirkobrankovic!) [[PR-1788](https://github.com/meetecho/janus-gateway/pull/1788)]
- Added per-user override of global room 'audio_active_packets' and 'audio_level_average' properties to AudioBridge and VideoRoom (thanks @mirkobrankovic!) [[PR-2158](https://github.com/meetecho/janus-gateway/pull/2158)]
- Notify speaker that started/stopped talking too, when talking events are triggered in VideoRoom and AudioBridge (thanks @maxboehm!) [[PR-2172](https://github.com/meetecho/janus-gateway/pull/2172)]
- Allow listing of private rooms/mountpoints if an admin_key is used (thanks @robby2016!) [[PR-2161](https://github.com/meetecho/janus-gateway/pull/2161)]
- Fixed RTCP support not triggering PLIs for new simulcast mountpoint viewers [[Issue-2156](https://github.com/meetecho/janus-gateway/issues/2156)]
- Fixed occasional issue binding multicast mountpoints (thanks @PaulKerr!) [[PR-2167](https://github.com/meetecho/janus-gateway/pull/2167)]
- Fixed buffering of keyframes not working in Streaming plugin (thanks @TomFFF!) [[PR-2170](https://github.com/meetecho/janus-gateway/pull/2170)]
- Added support for buffering of keyframes to RTSP mountpoints too (thanks @lionelnicolas!) [[PR-2180](https://github.com/meetecho/janus-gateway/pull/2180)]
- Fixed renegotiation support in SIP plugin when audio/video is added (thanks @ihusejnovic!) [[PR-2164](https://github.com/meetecho/janus-gateway/pull/2164)] [[PR-2173](https://github.com/meetecho/janus-gateway/pull/2173)]
- Fixed menus in html documentation when using Doxygen >= 1.8.14 (thanks @i8-pi!) [[PR-2155](https://github.com/meetecho/janus-gateway/pull/2155)]
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.9.5] - 2020-05-18

- Fixed sessions not being cleaned up when disabling session timeouts and the transport disconnects (thanks @nicolasduteil!) [[PR-2143](https://github.com/meetecho/janus-gateway/pull/2143)]
- Added option to keep candidates with private host addresses when using nat-1-1, and advertize them too instead of just replacing them
- Added auth token, if available, to 'attached' event (handlers) and to Admin API (handle_info)
- Added new API to start/stop recording a VideoRoom as a whole, and a new option to prevent participants from starting/stopping their own recording (thanks @wheresjames!) [[PR-2137](https://github.com/meetecho/janus-gateway/pull/2137)]
- Fixed rare deadlock when wrapping up Streaming plugin mountpoints [[PR-2141](https://github.com/meetecho/janus-gateway/pull/2141)]
- Fixed rare deadlock when destroying AudioBridge rooms
- Added synchronous request to check if an announcement is playing in the AudioBridge
- Fixed AudioBridge announcement not waking up sleeping forwarder
- Added global room mute/unmute support to AudioBridge
- Added configurable DSCP support for outgoing RTP packets to SIP and NoSIP plugins (thanks @GerardM22!) [[PR-2150](https://github.com/meetecho/janus-gateway/pull/2150)]
- Added support for RTP extensions (audio-level, video-orientation) to NoSIP plugin [[Issue-2152](https://github.com/meetecho/janus-gateway/issues/2152)]
- Added option to configure ciphers suite for secure WebSockets (thanks @agclark81!) [[PR-2135](https://github.com/meetecho/janus-gateway/pull/2135)]
- Added timer to janus.js to avoid spamming onmute/onunmute events and flashing videos [[PR-2147](https://github.com/meetecho/janus-gateway/pull/2147)]
- Added a new tool to convert .pcap captures to .mjr recordings [[PR-2144](https://github.com/meetecho/janus-gateway/pull/2144)]
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.9.4] - 2020-05-04

- Updated code not to wait forever for local candidates when half-trickling and sending an SDP out
- Fixed occasional CPU spiking issues when dealing with ICE failures (thanks @sjkummer!)
- Fixed occasional stall when gathering ICE candidates (thanks @wheresjames!)
- Fixed the incorrect value being set via DSCP, when configured
- Fixed occasional race condition when hanging up VideoRoom subscribers
- Fixed Audiobridge and Streaming plugins not playing the last chunk of .opus files (thanks @RSATom!)
- Fixed duplicate subscriptions (and SRTP/SRTCP errors) on multiple watch requests in Streaming plugin
- Updated Streaming and TextRoom plugins to stop using legacy datachannel negotiation
- Fixed occasional crash in HTTP transport when dealing with unknown requests
- Fixed occasional disconnect in WebSockets (thanks @tomnotcat!)
- Made RabbitMQ exchange type configurable in both transport and event handler (thanks @voicenter!)
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.9.3] - 2020-04-22

- 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!)


## [v0.9.2] - 2020-03-26

- 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
- Fixed double unlock when listing private rooms in AudioBridge
- Made AudioBridge prebuffering property configurable, both per-room and per-participant
- Added G.711 support to AudioBridge (both participants and RTP forwarders)
- Added called URI to 'incomingcall' and 'missed_call' events in SIP plugin (in case the registered user is associated with multiple public URIs)
- Fixed race conditions and leaks in VideoCall and VoiceMail plugins
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.9.1] - 2020-03-10

- Added configurable global prefix for log lines
- Implemented better management of remote candidates with invalid addresses
- Added subtype property to differentiate some macro-types in event handlers
- Improved detection of H.264 keyframes (thanks @cameronlucas3!)
- Added configurable support for strings as unique IDs in AudioBridge, VideoRoom, TextRoom and Streaming plugins
- Fixed small memory leak when creating Streaming mountpoints dynamically
- Fixed segfault when trying to start a SIP call with a non-existing refer_id (thanks @tmatth!)
- Fixed errors negotiating video in SIP plugin when multiple video profiles are provided
- Updated SIP plugin transfer code to answer with a 202 right away, instead of sending a 100 first (which won't work with proxies)
- Added several features and fixes several nits in SIP demo UI
- Fixed janus.js error callback not being invoked when an HTTP error happens trying to attach to a plugin (thanks @hxl-dy!)
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.9.0] - 2020-02-21

- Refactored core-plugin callbacks
- Added RTP extensions termination
- Removed requirement to enable ICE Lite to use ICE-TCP, even though it may cause issues (thanks @sjkummer!)
- Added support for transport-wide CC on outgoing streams (feedback still unused, though)
- Dynamically update NACK queue size depending on RTT
- Fixed risk of RTP header memory misalignment when dealing with rtx packets
- Users muted in AudioBridge by an admin are now notified as well (thanks @klanjabrik!)
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.8.2] - 2020-02-04

- Added Travis CI integration (thanks @fippo for kickstarting it!)
- New configuration property to add protected folders not to save recordings and pcap captures to
- Fixed rare race condition when joining and destroying a VideoRoom session
- Improved parsing of headers in RTSP messages (thanks @kefir266!)
- Fixed segfault in AudioBridge when leaving a room before PeerConnection is ready
- Fixed '500' errors being sent in response to incoming OPTIONS in the SIP plugin (thanks @ycherniavskyi!)
- Fixed helpers not being able to send SUBSCRIBE requests in SIP plugin
- Added option to fix audio skew compensation, if present, to janus-pp-rec
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.8.1] - 2020-01-13

- Added binary data support to data channels
- Fixed segfault at startup if event handlers or loggers directory couldn't be opened (thanks @kazzmir!)
- Fixed potential segfault when closing logging at shutdown
- Allowed RTCP ports to be picked randomly using 0, in Streaming plugin
- Fixed occasional memory leak when destroying mountpoints in Streaming plugin
- Fixed memory leak in SIP plugin
- Updated 'referred_by' field to contain the value of SIP referred-by header, and not just the URI (thanks @pawnnail!)
- Don't keep TextRoom plugin loaded if data channels were not compiled
- Removed SIPre plugin from the repo
- Fixed late initialization of janus.js constructor callbacks
- Changed janus.js to use sendBeacon instead of XHR when closing/refreshing page
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.8.0] - 2019-12-12

- Added changelog file to the repo and docs (thanks @oscarvadillog!)
- Added new category of plugins for modular logging (stdout and file still there, and part of the core)
- Removed option to enable rtx (now always supported, when negotiated)
- Added gzip compression helper method to the core utils
- Fixed RTSP SETUP issues when url contains query string parameters
- Added option to gzip events when using the Sample Event Handler
- Streamlined janus.js (thanks @oscarvadillog!)
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.7.6] - 2019-11-27

- Split SDP lines when parsing on line feed only, and trim carriage feed instead (\n instead of \r\n)
- Reduced default twcc_period (how often to send feedback when using Transport Side BWE) from 1s to 200ms
- Added option to skip (and disable) unreachable STUN/TURN server at startup (thanks @sjkummer!)
- Fixed video desynchronization when doing G.722/iSac audio
- Other generic fixes on A/V desync
- Added support for multiple concurrent calls for the same account to the SIP plugin
- Added support for blind and attended transfers to the SIP plugin
- Added way to inject custom Contact params in REGISTER to the SIP plugin
- Added way to intercept non-standard headers in SIP messages to SIP plugin (thanks @ihusejnovic!)
- Fixed missing SIP CANCEL when hanging up outgoing unanswered calls in SIP plugin
- Added support for domain names (and IPv6) to RTP forwarders in AudioBridge and VideoRoom
- Fixed broken b=TIAS SDP attribute support for Firefox in VideoRoom (thanks @MvEerd!)
- Fixed and improved VP9 SVC support in VideoRoom and Streaming plugins
- Added IPv6 support to Streaming plugin
- Fixed potential segfault in Streaming plugin (thanks @garry81!)
- Fixed occasional latching issues for RTSP in Streaming plugin
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.7.5] - 2019-10-28

- Added warning at startup if libnice version is outdated (at least 0.1.15 recommended)
- Added option to specify CWD when launching Janus as a daemon (thanks @l7s!)
- Extended the STUN test via Admin API to support binding to a specific port, and return the public one
- Fixed simulcast issue when needing to automatically drop to lower layers
- Fixed potential endless loop when binding ports in the Streaming plugin
- Made creating Streaming mountpoints more asynchronous (especially for RTSP)
- Added support for SIP SUBSCRIBE/NOTIFY to SIP plugin
- Added ability to add custom headers to SIP BYE (thanks @mmujic!)
- Added option to specify IP to bind to for media in SIP plugin (thanks @razvancrainea!)
- Fixed occasional segfault when leaving a VideoRoom
- Added audio level dBov average to talk events in VideoRoom plugin (thanks @aconchillo!)
- Added new synchronous API to mute other participants in the AudioBridge plugin (thanks @klanjabrik!)
- Fixed typo in SDP processing in Duktape/JavaScript plugin, and tied Duktape logging to the one in the Janus core (thanks @l7s!)
- Tied Lua logging to the one in the Janus core
- Added command line option to janus-pp-rec to specify the output format (thanks @rscreene!)
- Added new WebSocket and Nanomsg event handlers
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.7.4] - 2019-09-06

- Fixed duplicate values in config that could result in wrong property being used
- Fixed occasional race condition when processing SDPs (thanks @Bug-Fairy!)
- Fixed broken SDP when rejecting audio/video m-line
- Fixed Admin API not responding after sending messages to unresponsive plugins
- Fixed some issues with RTSP support in Streaming plugin
- Added option to keep recording Streaming mountpoints even when disabled
- Allow SIP plugin to negotiate SRTP separately for audio and video
- Fixed autoaccept_reinvites=FALSE not working when accepting calls in SIP plugin, and improved re-INVITEs support in general (thanks @pawnnail!)
- Added possibility to have different addresses for remote audio and video in SIP, SIPre and NoSIP plugins (thanks @pawnnail!)
- Make sure remote addresses are reset when call ends in SIP, SIPre and NoSIP plugins (thanks @pawnnail!)
- Added SIP Reason Header (RFC3326) info to "hangup" event in SIP plugin, if available (thanks @ihusejnovic!)
- Added method to list participants in a TextRoom (thanks @mtltechtemp!)
- Added method to send a room announcement in TextRoom plugin
- Fixed occasional segfault in TextRoom when using Admin API to send requests (thanks @MvEerd!)
- Added support for MQTT v5, and fixed reconnection issue (thanks @feymartynov!)
- Fixed occasional crashes when using more than one event handler at the same time
- Added configurable bitrate values for rid-based simulcast to janus.js (thanks @vivaldi-va!)
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.7.3] - 2019-07-10

- Added Admin API method to make synchronous requests to plugins
- Fixed broken media when removing/adding it again in renegotiations
- Fixed several issues related to datachannels
- Fixed occasional memory leak in the core when ending sessions from plugins (thanks @uxmaster!)
- Changed Janus API 'slowlink' event to use lost packets instead of NACKs, and made it configurable with a dynamic threshold
- Fixed broken SDES length in compound RTCP packets (thanks @glenn-hpcnt!)
- Fixed DTLS window size support in the core (thanks @garry81!)
- Added status messages to MQTT transport (thanks @feymartynov!)
- Changed default for sender-side bandwidth estimation in VideoRoom to TRUE
- Fixed occasional segfaults when using RTP forwarders with RTCP support
- Added VideoRoom RTP forwarder events to event handlers notifications
- Added a configurable RTP range to the Streaming plugin settings
- Fixed broken H.264 simulcast support in Streaming plugin
- Refactored janus-pp-rec to support command line options
- Fixed occasional segfault when post-processing VP8 recordings
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.7.2] - 2019-06-07

- Removed requirement for both sdpMid and sdpMLineIndex to be in trickle messages
- Set ICE remote credentials when receiving remote SDP, instead of later
- Fixed occasional segfaults when using WebSocket as a transport
- Fixed segfault in WebSocket transport when using ACL
- Added new Admin API messages to destroy a session, detach a handle and hangup a PeerConnection (same as Janus API)
- Fixed leak when RTP forwarding with RTCP feedback in the VideoRoom plugin
- Added support for third spatial layer when using VP9 SVC in VideoRoom (assuming EnabledByFlag_3SL3TL is used)
- Fixed segfault when changing rooms in AudioBridge
- Made sure the SIP stack doesn't accept new calls until the previous one has freed all resources
- Fixed occasional segfault when pushing SIP messages to event handlers
- Added option to locally cleanup handles when destroying a session in janus.js
- Fixed exception in janus.js when using datachannels
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.7.1] - 2019-05-20

- Added experimental debug mode with disabled WebRTC encryption (to use with the --disable-webrtc-encryption in Chrome unstable)
- Added Janus API ping/pong mechanism to Admin API as well
- Added Admin API methods to check address resolving capabilities and test a provided STUN server
- Added check on ICE gathering process start (fixes issue with exhausted port range)
- Added support for temporal layer in H.264 simulcast via frame marking
- Made sure a PLI is sent on all layers, when simulcast is used
- Fixed a crash when using event handlers in SIP plugin
- Fixed some race conditions on hangups in SIP plugin
- Added option to lock RTP forwarding functionality via an admin key/secret (VideoRoom and AudioBridge)
- Fixed regression in Streaming plugin RTCP support
- Added option to override payload type for RTSP mountpoints in Streaming plugin
- Fixed a few issues saving permanent mountpoints in Streaming plugin
- Separated checks for PeerConnection and getUserMedia support in janus.js (since plain HTTP hides getUserMedia now)
- Added sanity checks on createOffer/createAnswer in janus.js
- Fixed regression in simulcasting when doing SDP munging in janus.js
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.7.0] - 2019-05-10

- Added support for multiple datachannel streams in the same PeerConnection
- Forced DTLS 1.2 on older OpenSSL versions
- Added first integration of SDP support in the fuzzers
- Fixed several leaks in SDP utils
- Explicitly disabled support for encrypted RTP extensions (was causing SDP inconsistencies)
- Added count of incoming retransmissions to Admin API and Event Handlers stats
- Improved check for H.264 keyframe (thanks bwerther!)
- Modified "cap REMB" behavior to "replace REMB"
- Fixed missing notification of lurkers when first joining VideoRoom with notify_join=TRUE
- Improved support for incoming re-INVITEs in SIP plugin
- Fixed check in WebSocket transport that could lead to crashes
- Fixed occasional segfaults when postprocessing H.264 recordings
- Added new callback to janus.js to intercept the SDP before it is sent, e.g., for munging purposes (thx @carlcc!)
- Fixed direction property error in janus.js on Safari (thx @alienpavlov!)
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.6.3] - 2019-03-20

- Removed folder with self-signed certificate (unneeded and confusing)
- Added many fixes and improvements to the RTCP code
- Fixed typos that caused issues when sending retransmissions using RFC4588
- Fixed typo when sending empty RR coupled with REMB
- Made sure the CNAME is always the same for all m-lines in an SDP
- Added support for mid RTP extension
- Improved support for rid-based simulcasting
- Fixed publish errors in MQTT transport and event handler
- Fixed issue when switching Streaming mountpoints powered by helper threads
- Added info on whether VideoRoom publisher is simulcasting to join events
- Added option for new VideoRoom subscribers to specify simulcast substream/layer to subscribe to in join request (before it was configure-only)
- Added type definitions for janus.js (thanks Elias!)
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.6.2] - 2019-03-04

- Added RTP/RTCP fuzzing targets and tools
- Fixed occasional crash when pushing the local SDP to event handlers, when enabled
- Fixed NACK issue when receiving an out of order keyframe
- Added option to configure the TWCC feedback period
- Added option to include opaqueID in Janus API events
- Added option to negotiate Opus inband FEC in the VideoRoom
- Added option to specify temporary extension when recording AudioBridge rooms, and event handler notification for when recording is over
- Fixed occasional playout issue after recording, using Record&Play demo
- Fixed typo in janus.js that affected replacing audio tracks in renegotiations
- Changed default maxev (number of events in long poll results) to 10 in janus.js
- Updated path of getDisplayMedia in janus.js to reflect current spec (thanks cb22!)
- Fixed ambiguous check in Janus.isWebrtcSupported in janus.js
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.6.1] - 2019-02-11

- Added several fixes to RTP/RTCP parsing after fuzzing tests
- Added fixes to keyframe detection after fuzzing tests
- Fixed some demos not working after update to Chrome 72
- Fixed occasional crashes when saving .jfcg files (e.g., saving permanent Streaming mountpoints)
- Added new Admin API command to temporarily stop/resume accepting sessions (e.g., for draining servers)
- Fixed recordings sometimes not closed/destroyed/renamed when hanging up SIP sessions
- Added option to SIP/SIPre/NoSIP plugin to override c= IP in SDP
- Fixed missing RTSP support in Streaming plugin if TURN REST API was disabled in configure
- Fixed Streaming plugin not returning complete information on secret-less mountpoints (thanks @Musashi178!)
- Fixed missing .jfcg support in Duktape plugin (thanks @fbertone!)
- Updated janus.js to use transceivers for Chrome >=72
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.6.0] - 2019-01-07

- Changed default configuration format to libconfig (INI still supported but deprecated)
- Fixed several RTCP parsing issues that could lead to crashes (thanks to Fippo for bringing fuzzying to our attention!)
- Added support to clang compiler (needed for fuzzying)
- Fixed rtx packets ending up in retransmission buffer (thanks glenn-hpcnt!)
- Fixed occasional crash when cleaning NACK buffer (thanks tmatth!)
- Fixed loop termination warning when handling event handlers (thanks tmatth!)
- Fixed occasional invalid rtx payload type
- Fixed local SDP notification to event handlers
- Fixed typo in link quality calculation
- Fixed occasional crash in SIP plugin
- Added option to provide custom headers in SIP 200 OK as well (thanks ihusejnovic!)
- Fixed typo in Range header when sending RTSP PLAY in Streaming plugin (thanks Phil1972!)
- Made MQTT and RabbitMQ configuration files more consistent with other ones (thanks manifest!)
- Added support for Last Will and Testament to MQTT event handler (thanks 0nkery!)
- Fixed broken video when post-processing recordings with high-profile H.264
- Fixed missing success callback in sendDtmf JS method (thanks nevcos!)
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.5.0] - 2018-11-20

- Refactored core to have a persistent GMainLoop/thread per handle
- Added option to share static number of GMainLoop/thread instances for multiple handles
- Better management of incoming RTCP packets before passing them to plugins
- Updated TURN REST API to support both "key" and "api" as parameters
- Added support for dumping directly to .pcap, rather than text first via text2pcap
- Fixed occasional missing notifications of temporal layer changes, when doing simulcast
- Fixed occasional crash in TextRoom plugin
- Fixed crashes in Duktape plugin after some iterations
- Added .mjr metadata to media files when postprocessing the recordings, if supported by the container
- Fixed datachannels not working in Streaming demo, when configured
- Fixed dangling "Publish" button in VideoRoom demo
- Better management of timeout notifications when using websockets in janus.js (thanks @nevcos!)
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.4.5] - 2018-10-16

- Switched to GMutex for locks by default (changeable in configure)
- Fixed missing sdpMid in some trickle candidates, which could break full-trickle support
- Fixed missing TWCC info when handling rtx duplicates (thanks garry81!)
- Fixed H.264 keyframe detection and broken H.264 simulcast code
- Fixed bug in skew compensation code
- Fixed occasional crashes when closing PeerConnections in AudioBridge
- Fixed broken Record-Route usage in SIP plugin (thanks Dan!)
- Removed outdated autoack property from SIP plugin
- Switched from SET_PARAMETER to OPTIONS as an RTSP keep-alive (thanks cnzjy!)
- Fixed missing endianness for RTP packets in postprocessor, which caused problems on MacOS
- Fixed crash in postprocessor when handling high(er) H.264 profiles (e.g., Safari 12)
- Fixed multiple "First keyframe" log lines when postprocessing video
- Added support for parsing a few RTP extensions in the postprocessor
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.4.4] - 2018-09-28

- Added several important fixes to NACK and retransmission code
- Fixed connectivity establishment when only available candidates are prflx
- Fixed some leaks in TWCC code
- Fixed missing information when reporting TWCC reports (thanks Kangsik!)
- Made the timeout for trickle candidates configurable
- Added support for mDNS candidates (see draft-ietf-rtcweb-mdns-ice-candidates)
- Added option to configure the DTLS retransmission timer (BoringSSL only)
- Optimized DTLS writes by removing a copy on each send (thanks Joachim!)
- Added option to override codecs to negotiate in EchoTest
- Added H.264 simulcasting support to plugins that did VP8 simulcast already
- Added VP9/SVC support to the Streaming plugin
- Improved the way simulcast streams can be recorded and forwarded
- Added partial RTCP support to RTP forwarders (thanks Adam!)
- Fixed occasional segfaults in the VideoRoom when forcing private IDs (thanks tugtugtug!)
- Added option to use helper threads for Streaming plugin mountpoints
- Fixed a couple of errors in the RTSP support of the Streaming plugin (thanks nu774!)
- Several fixes in the NoSIP plugin (thanks Dmitry!)
- Fixed broken SIP MESSAGE support in SIP plugin
- Fixed occasional segfaults in SIP and SIPre plugins (thanks mharcar!)
- Fixed broken recording support in the VideoCall plugin (thanks codebot!)
- Fixed potential deadlock in Lua and Duktape plugins (thanks Gabriel!)
- Fixed memory leaks in VideoRoom, AudioBridge and TextRoom
- Added new MQTT event handler (thanks Olle!)
- Made HTTP REST API optionally more consistent with other transports
- Added new flag to postprocessor for just printing the JSON header
- Fixed occasional segfaults when processing recordings
- Added getDisplayMedia() support to janus.js
- Added better support to constraints when screensharing (thanks Sol!)
- Added better iOS devices support to janus.js and the demos
- Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)


## [v0.4.3] - 2018-08-27

- Fixed occasional crash when closing PeerConnections
- Fixed way of negotiating datachannels in Firefox Nightly
- Fixed broken check when enabling TURN REST API
- Fixed occasional crash when post-processing H.264 recordings (thanks Thomas!)
- Fixed occasional issue when creating PID file
- Fixed broken SDES generation (thanks Garry!)
- Added new Duktape plugin to write plugin logic in JavaScript
- Fixed occasional crash in VideoCall plugin when declining calls
- Added basic RTCP support to the Streaming plugin (thanks Adam!)
- Added basic RTCP support to RTP forwarders in the VideoRoom plugin
- Added new Nanomsg transport
- Changed the way libwebsockets logging is configured
- Updated janus.js to use promises for WebRTC APIs (thanks Philipp!)
- Some more bug fixes and improvements


## [v0.4.2] - 2018-06-18

- Fixed ICE loop not terminating at times, and spiking the CPU
- Fixed compilation against older OpenSSL versions (thanks Joachim!)
- Added option to statically enable locking debug via command line or configuration file
- Fixed occasional crash in VideoRoom when destroying rooms
- Fixed VideoRoom not closing subscribers PeerConnections when publisher goes away, if so configured
- Fixed SRTP errors when resuming VideoRoom subscribers that were paused for a long time
- Added new option to really force a cap on the bitrate in VideoRoom rooms
- Fixed recording not being started for VideoRoom publishers media added in a renegotiation
- Fixed occasional crash in AudioBridge when closing PeerConnections under load
- Added Opus FEC support to AudioBridge (thanks Eric!)
- Fixed pipe socket initialization in Streaming plugin (thanks Adam!)
- Added systemd support to Unix Sockets transport plugin (thanks Adam!)
- WebSocket connection is no longer torn down in case of a Janus session timeout
- Added options to configure keep-alive and long-poll timers in janus.js
- Some more bug fixes and improvements


## [v0.4.1] - 2018-05-29

- Single thread per PeerConnection, instead of two
- Fixed issue with API secret, where sessions would be created anyway
- Cleanup of ICE related code (thx Joachim!)
- Removed ad-hoc thread for SCTP code
- Fixed deadlock in VideoRoom plugin
- Fixed segfault in SIPre plugin
- Fixed leaks when using event handlers (thx zgjzzhw!)
- Fixed some missing events when closing PeerConnections
- Fixed broken dependencies mechanism in janus.js (thx Philippe!)
- Some more bug fixes and improvements


## [v0.4.0-broken] - 2018-05-22

- Changed memory management to use reference counters
- New plugin to write application logic in Lua
- Added mechanism to reclaim sessions after a reconnection (thx Geige!)
- Fixed broken renegotiations when upgrading from audio-only to audio-video
- Fixed typo in evaluation of RTT from RTCP packets
- Fixed crash when SRTP profile is missing in DTLS handshake
- Improved and streamlined a few events (event handlers), e.g., selected-pair
- Added new "external" events (event handlers), for events pushed via Admin API
- Fixed deadlock when joining a VideoRoom with notify_join=true
- Fixed some info not saved permanently in some plugins when editing
- Added media latching to RTSP streams setup in the Streaming plugin
- Fixed an issue with simulcast support in the Streaming plugin
- Fixed occasional unexpected WebSockets disconnects when using the Streaming plugin
- Fixed Streaming plugin not returning bound ports when creating mountpoints with random ones (port=0)
- Improved and streamlined documentation for all plugins
- Added option to limit ciphers/protocols in HTTP and WebSockets (thx Alexander!)
- Added transceivers support to janus.js for proper renegotiations in Firefox
- More bug fixing and general cleanup (thx to mtdxc, fancycode and others!)
- Added a way to support other screensharing extensions in janus.js in a programmatic way (thx Sol!)


## [v0.3.1] - 2018-04-04

- Changed threading model for processing requests in the core
- Added support for SRTP AES-GCM to core and SIP/SIPre/NoSIP plugins
- Changed set of ciphers negotiated in DTLS, disabling weaker ones (thanks Chad!)
- Added option to specify passphrase when dealing with certificates/keys
- Added ability for Admin API requests to tweak Event Handlers
- Integrated link quality stats info (thanks Piter!)
- Added support for storage-less authentication via Signed Tokens (thanks Sol!)
- Added option to force TCP for SIP messages in the SIP plugin
- Added option to not fail RTSP mountpoint creation right away if backend is not up
- Added SSL/TLS support to the MQTT transport (thanks Andrei!)
- Added new request to edit some Streaming mountpoint properties (thanks Rob!)
- Fixed management of DTMF in janus.js
- Updated management of constraints in janus.js (thanks Igor!)
- Bug fixing and general improvements


## [v0.3.0] - 2018-02-23

- Implemented renegotiations and ICE restarts
- Bundle and rtcp-mux now are always forced
- Added support to Transport Wide CC sender-side BWE (thanks Sergio!)
- Added SRTP support to Streaming mountpoints
- Implemented a skew compensation algorithm in the Streaming plugin
- Added SRTP support to RTP forwarders
- Implemented support for RFC4588 (rtx/90000 retransmissions)
- Janus can now do full-trickle too
- SIP plugin now supports 407 (proxy authentication)
- Fixed post-processing of G.711 recordings
- Added versioning info to janus-pp-rec
- Several fixes and cleanup


## [v0.2.6] - 2017-12-19

- New SIP plugin based on libre, SIPre (janus.plugin.sipre), and related demo
- New NoSIP plugin, that can be used with legacy applications (like SIP) without doing any signalling itself
- VideoRoom can now support multiple codecs at the same time, instead of being forced to choose just one per media type
- Plugins now record streams specifying the actual codec in use, instead of making assumptions (e.g., like Record&Play did with Opus and VP8)
- Streaming plugin now allows you to temporarily pause audio and/or video delivery via "configure" requests
- Removed RTCP BYE as a trigger to shutdown a PeerConnection (fixes Firefox 52 issues)
- Added RTCP support for simulcast SSRCs
- Fixed parsing of Firefox simulcast offer when order of attributes was different than expected
- Improved RTP headers rewriting in case of SSRC changes (e.g., context switches)
- Improved performance of the ICE send threads/loops and computation of transfer rates, by getting rid of all list traversals
- Added support for MSG_EOR in SCTP datachannels
- Added "exchange" support to RabbitMQ transport
- Added new info to Event Handlers (server info in "started" event, and server name in "emitter")
- Added RabbitMQ Event Handler
- You can now add additional constraints for a PeerConnection when invoking createOffer and createAnswer in janus.js
- Fixed occasional problems when postprocessing .mjr recordings, especially long ones, and Opus recordings
- Several bug and typo fixes, in both core and janus.js


## [v0.2.5] - 2017-10-23

- VP8 simulcasting supported in a few plugins (you may have experimented with it on the online demos already);
- VP9 SVC is also available (VideoRoom only);
- VideoRoom and Streaming plugins allow you to subscribe to a subset of the feed's media (e.g., only get audio even though feed is audio/video);
- automatic fallback in the VideoRoom to subset of the media in case of unsupported codecs (e.g., Safari joining VP8 room falls back to audio only);
- added option to override rtpmap and fmtp SDP attributes for RTSP mountpoints in the Streaming plugin;
- added support for other codecs besides opus and VP8 in Record&Play plugin;
- added option to have a static RTP forwarder for an AudioBridge room in the configuration file;
- added possibility to specify an RTP range to use in the SIP plugin;
- implemented text2pcap support to dump incoming and outgoing unencrypted RTP/RTCP traffic for debugging purposes;
- added support to G.722 in postprocessor;
- made sure that each m-line now has its own a=end-of-candidates attribute;
- fixed crash in websockets transport plugin when SSL was enabled on both APIs;
- added support to ping/pong mechanism in websockets transport plugin;
- switched from addstream to addtrack in janus.js;
- decoupled the dependencies in janus.js to allow for dynamic override of some features;
- added support to build JavaScript modules out of janus.js.


## [v0.2.4] - 2017-07-28

- binding to some or all interfaces/families has been fixed in the HTTP transport;
- the Access-Control-Allow-Origin return value is now configurable in the HTTP transport;
- fixed occasional slow WebSocket request management when DNS was involved;
- there's a new timer before we return an ICE failed (as due to trickling there may be a success shortly after a temporary failure);
- the frequency of media stats notifications (event 32) in event handlers has been made configurable (default is still 1s);
- event handlers now notify about each local and remote candidate as well;
- the admin.html demo page now prompts you with the password (although you can still hardcode it in the page, as before);
- several changes in the SIP plugin: support for offerless INVITEs, early media (183+SDP), outbound proxies, and fixes to some POLLERR messages;
- added support for LibreSSL as an alternative to OpenSSL and BoringSSL;
- added a=end-of-candidates to all m-lines, since we half-trickle (fixes Edge support);
- fixed a race condition in the TextRoom plugin;
- fixed the way janus.js used getStats, in particular for Firefox;
- fixed device selection demo;
- several smaller fixes derived from a static analysis of the code via Coverity.


## [v0.2.3] - 2017-06-12

- A few janus.js fixes (among which a small fix to get it working with Safari, and the possibility to add mic audio when screensharing);
- Several RTCP related enhancements in the Streaming plugin;
- Support for on-hold in SIP plugin;
- Fixed MQTT transport when credentials are needed;
- Improved "kick" in VideoRoom (needs forcing of private_id when creating room);
- Possibility to create Streaming mountpoints with random ports, instead of specifying them via API;
- Optional "talking" events in AudioBridge and VideoRoom;
- Possibility to force BUNDLE/rtcp-mux per handle via API (no need to wait for complete negotiation);
- Several bug fixes, a couple of them to nasty race conditions that finally got solved.


## [v0.2.2] - 2017-03-08

- ACL/Kick support in VideoRoom/AudioBridge/TextRoom
- Man pages for Janus and post-processor
- Opaque identifiers for Event handlers + Transport related events
- Ability to specify SSRC + payload type when using RTP forwarders
- Ability to relay datachannels in Streaming plugin
- Ability to send some TextRoom commands (e.g., create/list/etc.) via Janus API instead of only datachannels
- Configurable session timeouts
- Configurable "no-media" timeouts
- Optional temporary extension for recordings until they're done
- cleanup and bug fixing


## [v0.2.1] - 2016-12-13

- Missing info


## [v0.2.0] - 2016-10-10

- Missing info


## [v0.1.2] - 2016-09-05

- Missing info


## [v0.1.1] - 2016-06-15

- Missing info


## [v0.1.0] - 2016-05-27

- Missing info


## [v0.0.9] - 2015-11-11

- First release
Last updated on Tue Jun 23 2020 — Janus WebRTC Server © Meetecho 2014-2020