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CHANGELOG.md

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Changelog

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

[v0.10.9] - 2020-12-23

  • Replaced Travis CI with GitHub Actions [PR-2486]
  • Fixed data channel messages potentially getting stuck in case of burst transfers (thanks @afshin2003!) [PR-2427]
  • Fixed simulcast issues when renegotiating PeerConnections [Issue-2466]
  • Added configurable TURN REST API timeout (thanks @evorw!) [PR-2470]
  • Added support for recording of binary data channels [PR-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]
  • Added option to override connections limit in HTTP transport plugin [PR-2489]
  • Added options to enable libmicrohttpd debugging in HTTP transport plugin (thanks @evorw!) [PR-2471]
  • Fixed a few compile and runtime issues in WebSocket event handler
  • Refactored postprocessing management of timestamps to fix some key issues [PR-2345]
  • Fixed postprocessing of audio recordings containing RTP silence suppression packets [PR-2467]
  • Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)

[v0.10.8] - 2020-11-23

  • Added differentiation between IPv4 and IPv6 NAT-1-1 addresses [PR-2423]
  • Made NACK buffer cleanup on outgoing keyframe disabled by default but configurable [PR-2402]
  • Added support for simulcast and TWCC to Duktape and Lua plugins [PR-2409]
  • Fixed rare crash in AudioBridge plugin when leaving a room [Issue-2432]
  • Fixed codec names not being updated in the SIP plugin after renegotiations (thanks @ihusejnovic!) [PR-2417]
  • Fixed crash in SIP plugin when handling REGISTER challenges without WWW-Authenticate headers [Issue-2419]
  • Added option to SIP plugin to let users CANCEL pending transactions without waiting for a 1xx [PR-2434]
  • Added option to enforce CORS on the server side in both HTTP and WebSocket transport plugins [PR-2410]
  • Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)

[v0.10.7] - 2020-10-30

  • Fixed SDP negotiation when client uses max-bundle [Issue-2390]
  • Added optional JSEP flag to invert processing order of simulcast "rid" in SDP [PR-2385]
  • Fixed broken rid-based simulcast when using less than 3 substreams
  • Fixed occasional misleading "private IP" warning on startup (thanks @npikimasu!) [PR-2386]
  • Added "plugin-offer mode" to AudioBridge [PR-2366]
  • Fixed occasional deadlock when sending SUBSCRIBE messages via SIP plugin [PR-2387]
  • Fixed occasional SIGABRT in RabbitMQ transport (thanks @david-goncalves!) [PR-2380]
  • Fixed broken RTP parsing in janus-pp-rec when there were too many extensions (thanks @isnumanagic!) [PR-2411]
  • Fixed occasional segfault when post-processing G.722 mjr recordings
  • Added configurable simulcast encodings to janus.js (thanks @fippo!) [PR-2393]
  • Updated old Insertable Streams APIs in janus.js and e2etest.js
  • Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)

[v0.10.6] - 2020-10-05

  • 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!)

[v0.10.5] - 2020-09-08

  • Fixed occasional crash in event handlers [Issue-2312]
  • Fixed occasional crash in VideoRoom plugin [Issue-2318]
  • Fixed missing PLI when switching Streaming mountpoint [Issue-2333]
  • Fixed broken recordings in VideoCall plugin (thanks @SamyCookie!) [PR-2325]
  • Fixed "kick" not working in TextRoom plugin (thanks @backface!) [PR-2332]
  • Fixed occasional post-processing issues with incomplete mjr files (thanks @SamyCookie!) [PR-2356]
  • Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)

[v0.10.4] - 2020-08-07

  • Fixed usrsctp vulnerability by using internal hashmap in SCTP code [PR-2302]
  • Fixed some issues when using BoringSSL for DTLS (thanks @fancycode!) [PR-2278]
  • Added support for multiple nat-1-1 addresses (thanks @fancycode!) [PR-2279]
  • Fixed negotiation issue on Firefox when Janus is built without datachannels [PR-2281]
  • Fixed small memory leaks when dealing with local candidates (thanks @fancycode!) [PR-2288]
  • Fixed occasional segfault in VideoRoom when failing to setup a new subscriber [Issue-2277]
  • Fixed potential deadlock in AudioBridge when switching rooms (thanks @JeckLabs!) [PR-2280]
  • Fixed small memory leak in AudioBridge (thanks @JeckLabs!) [PR-2298]
  • Fixed occasional segfault in VideoCall when hanging up calls [Issue-2300]
  • Fixed occasional curl hiccups with RTSP on some cameras
  • Added reconnect mechanism to RabbitMQ event handler (thanks @david-goncalves!) [PR-2267]
  • Extended MQTT support in transport and event handler to v5 (thanks @feymartynov!) [PR-2273]
  • Added settings to configure MQTT buffers in the transport plugin (thanks @feymartynov!) [PR-2286]
  • Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)

[v0.10.3] - 2020-07-09

  • Fixed occasional crashes in VideoRoom related to subscribers activity [PR-2236] [PR-2253]
  • Fixed AudioBridge compilation issues when libogg is missing (thanks @ffontaine!) [PR-2238]
  • Fixed broken SRTP forwarders in AudioBridge [PR-2258]
  • Fixed occasional segfaults due to race conditions in SIP plugin [PR-2247]
  • Fixed occasional recording issues in Janus and Duktape plugins
  • Added timeout (120s) on idle connections in HTTP transport
  • Fixed Opus recordings occasionally being way too large than the source file when processed via janus-pp-rec (thanks @neilkinnish!) [PR-2250]
  • Added a new web demo to use canvas elements as a media source for PeerConnections [PR-2261]
  • Other smaller fixes and improvements (thanks to all who contributed pull requests and reported issues!)

[v0.10.2] - 2020-06-17

  • Fixed sscanf-related security issues [PR-2229]
  • Fixed some RTP extensions not working after renegotiations [Issue-2192]
  • Fixed occasionally broken simulcast behaviour [PR-2231]
  • Fixed "switch" request not taking simulcast/SVC into account in VideoRoom and Streaming plugins [Issue-2219]
  • Fixed inability to ask for random ports when creating Streaming plugin mountpoints with simulcast support [PR-2225]
  • Fixed occasional crashes in SIP plugin when using helpers [PR-2216]
  • Updated Duktape dependencies to v2.5, and fixed Duktape plugin relaying text data as binary [PR-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]
  • Added initial support for end-to-end encryption via Insertable Streams [PR-2074]
  • Fixed security issues when processing SDPs [PR-2214]
  • Fixed occasional codec profile negotiation issues (thanks @groupboard!) [PR-2212]
  • Fixed occasional segfaults when hanging up VideoRoom subscribers
  • Fixed RTSP issues when fmtp is missing (thanks @lionelnicolas!) [PR-2190]
  • Fixed RTSP not following redirects, when used (thanks @lionelnicolas!) [PR-2195]
  • Fixed SRTP-SDES and renegotiation issues in NoSIP plugin (thanks @ihusejnovic!) [PR-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]
  • Added new callback to let plugins know when the datachannel first becomes available, and then any time it's writable (empty buffers) [PR-2060]
  • Added support for data channel subprotocols [PR-2157]
  • Added new event handler for GrayLog using GELF (thanks @mirkobrankovic!) [PR-1788]
  • Added per-user override of global room 'audio_active_packets' and 'audio_level_average' properties to AudioBridge and VideoRoom (thanks @mirkobrankovic!) [PR-2158]
  • Notify speaker that started/stopped talking too, when talking events are triggered in VideoRoom and AudioBridge (thanks @maxboehm!) [PR-2172]
  • Allow listing of private rooms/mountpoints if an admin_key is used (thanks @robby2016!) [PR-2161]
  • Fixed RTCP support not triggering PLIs for new simulcast mountpoint viewers [Issue-2156]
  • Fixed occasional issue binding multicast mountpoints (thanks @PaulKerr!) [PR-2167]
  • Fixed buffering of keyframes not working in Streaming plugin (thanks @TomFFF!) [PR-2170]
  • Added support for buffering of keyframes to RTSP mountpoints too (thanks @lionelnicolas!) [PR-2180]
  • Fixed renegotiation support in SIP plugin when audio/video is added (thanks @ihusejnovic!) [PR-2164] [PR-2173]
  • Fixed menus in html documentation when using Doxygen >= 1.8.14 (thanks @i8-pi!) [PR-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]
  • 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]
  • Fixed rare deadlock when wrapping up Streaming plugin mountpoints [PR-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]
  • Added support for RTP extensions (audio-level, video-orientation) to NoSIP plugin [Issue-2152]
  • Added option to configure ciphers suite for secure WebSockets (thanks @agclark81!) [PR-2135]
  • Added timer to janus.js to avoid spamming onmute/onunmute events and flashing videos [PR-2147]
  • Added a new tool to convert .pcap captures to .mjr recordings [PR-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