CHANGELOG.md

Changelog

3.13.19

• Node: Fix router.createWebRtcTransport() with listenIps (PR #1330).

3.13.18

• Make transport-cc feedback work similarly to libwebrtc (PR #1088 by @penguinol).
• TransportListenInfo: "announced ip" can also be a hostname (PR #1322).
• TransportListenInfo: Rename "announced ip" to "announced address" (PR #1324).
• CI: Add macos-14.

3.13.17

• Fix prebuilt worker download (PR #1319 by @brynrichards).
• libsrtp: Update to v3.0-alpha version in our fork.

3.13.16

• Node: Add new worker.on('subprocessclose') event (PR #1307).

3.13.15

• Add worker prebuild binary for Linux kernel 6 (PR #1300).

3.13.14

• Avoid modification of user input data (PR #1285).
• ListenInfo: Add transport socket flags (PR #1291).
• ListenInfo: Ignore given socket flags if not suitable for given IP family or transport (PR #1294).
• Meson: Remove -Db_pie=true -Db_staticpic=true args (PR #1293).
• Add RTCP Sender Report trace event (PR #1267 by @GithubUser8080).

3.13.13

• worker: Do not use references for async callbacks (PR #1274).
• liburing: Enable zero copy (PR #1273).
• Fix build on musl based systems (such as Alpine Linux) (PR #1279).

3.13.12

• worker: Disable RtcLogger usage if not enabled (PR #1264).
• npm installation: Don't require Python if valid worker prebuilt binary is fetched (PR #1265).
• Update h264-profile-level-id NPM dependency to 1.1.0.

3.13.11

• liburing: Avoid extra memcpy on RTP (PR #1258).
• libsrtp: Use our own fork with performance gain (PR #1260).
• DataConsumer: Add addSubchannel() and removeSubchannel() methods (PR #1263).
• Fix Rust DataConsumer (PR #1262).

3.13.10

• tasks.py: Always include --no-user in pip install commands to avoid the "can not combine --user and --target" error in Windows (PR #1257).

3.13.9

• Update worker liburing dependency to 2.4-2 (PR #1254).
• liburing: Enable by default (PR 1255).

3.13.8

• liburing: Enable liburing usage for SCTP data delivery (PR 1249).
• liburing: Disable by default (PR 1253).

3.13.7

• Update worker dependencies (PR #1201):
  • abseil-cpp 20230802.0-2
  • libuv 1.47.0-1
  • OpenSSL 3.0.8-2
  • usrsctp snapshot ebb18adac6501bad4501b1f6dccb67a1c85cc299
• Enable liburing usage for Linux (kernel versions >= 6) (PR #1218).

3.13.6

• Replace make + Makefile with Python Invoke library + tasks.py (also fix installation under path with whitespaces) (PR #1239).

3.13.5

• Fix RTCP SDES packet size calculation (PR #1236 based on PR PR #1234 by @ybybwdwd).
• RTCP Compound Packet: Use a single DLRR report to hold all ssrc info sub-blocks (PR #1237).

3.13.4

• Fix RTCP DLRR (Delay Since Last Receiver Report) block parsing (PR #1234).

3.13.3

• Node: Fix issue when 'pause'/'resume' events are not emitted (PR #1231 by @douglaseel).

3.13.2

• FBS: LayersChangeNotification body must be optional (fixes a crash) (PR #1227).

3.13.1

• Node: Extract version from package.json using require() (PR #1217 by @arcinston).

3.13.0

• Switch from JSON based messages to flatbuffers (PR #1064).
• Add ListenInfo in all transports and send/recv buffer size options (PR #1084).
• Add optional rtcpListenInfo in PlainTransportOptions (PR #1099).
• Add pause/resume API in DataProducer and DataConsumer (PR #1104).
• DataChannel subchannels feature (PR #1152).
• Worker: Make DTLS fragment stay within MTU size range (PR #1156, based on PR #1143 by @vpnts-se).

3.12.16

• Fix IceServer crash when client uses ICE renomination (PR #1182).

3.12.15

• Fix NPM "postinstall" task in Windows (PR #1187).

3.12.14

• CI: Use Node.js version 20 (PR #1177).
• Use given PYTHON environment variable (if given) when running worker/scripts/getmake.py (PR #1186).

3.12.13

• Bump up Meson from 1.1.0 to 1.2.1 (fixes Xcode 15 build issues) (PR #1163 by @arcinston).

3.12.12

• Support C++20 (PR #1150 by @o-u-p).

3.12.11

• Google Transport Feedback: Read Reference Time field as 24bits signed as per spec (PR #1145).

3.12.10

• Node: Rename WebRtcTransport.webRtcServerClosed() to listenServerClosed() (PR #1141 by @piranna).

3.12.9

• Fix RTCP SDES (PR #1139).

3.12.8

• Export workerBin absolute path (PR #1123).

3.12.7

• SimulcastConsumer: Fix lack of "layerschange" event when all streams in the producer die (PR #1122).

3.12.6

• Worker: Add Transport::Destroying() protected method (PR #1114).
• RtpStreamRecv: Fix jitter calculation (PR #1117, thanks to @penguinol).
• Revert "Node: make types.ts only export types rather than the entire class/code" (PR #1109) because it requires typescript >= 5 in the apps that import mediasoup and we don't want to be that strict yet.

3.12.5

• DataConsumer: Fix removed 'bufferedamountlow' notification (PR #1113).

3.12.4

• Fix downloaded prebuilt binary check on Windows (PR #1105 by @woodfe).

3.12.3

Migrate npm-scripts.js to npm-scripts.mjs (ES Module) (PR #1093).

3.12.2

• CI: Use ubuntu-20.04 to build mediasoup-worker prebuilt on Linux (PR #1092).

3.12.1

• mediasoup-worker prebuild: Fallback to local building if fetched binary doesn't run on current host (PR #1090).

3.12.0

• Automate and publish prebuilt mediasoup-worker binaries (PR #1087, thanks to @barlock for his work in (PR #1083).

3.11.26

• Worker: Fix NACK timer and avoid negative RTT (PR #1082, thanks to @o-u-p for his work in (PR #1076).

3.11.25

• Worker: Require C++17, Meson >= 1.1.0 and update subprojects (PR #1081).

3.11.24

• SeqManager: Fix performance regression (PR #1068, thanks to @vpalmisano for properly reporting).

3.11.23

• Node: Fix appData for Transport and RtpObserver parent classes (PR #1066).

3.11.22

• RtpStreamRecv: Only perform RTP inactivity check on simulcast streams (PR #1061).
• SeqManager: Properly remove old dropped entries (PR #1054).
• libwebrtc: Upgrade trendline estimator to improve low bandwidth conditions (PR #1055 by @ggarber).
• libwebrtc: Fix bandwidth probation dead state (PR #1031 by @vpalmisano).

3.11.21

• Fix check division by zero in transport congestion control (PR #1049 by @ggarber).
• Fix lost pending statuses in transport CC feedback (PR #1050 by @ggarber).

3.11.20

• RtpStreamSend: Reset RTP retransmission buffer upon RTP sequence number reset (PR #1041).
• RtpRetransmissionBuffer: Handle corner case in which received packet has lower seq than newest packet in the buffer but higher timestamp (PR #1044).
• SeqManager: Fix crash and add fuzzer (PR #1045).
• Node: Make appData TS typed and writable (PR #1046, credits to @mango-martin).

3.11.19

• SvcConsumer: Properly handle VP9 K-SVC bandwidth allocation (PR #1036 by @vpalmisano).

3.11.18

• RtpRetransmissionBuffer: Consider the case of packet with newest timestamp but "old" seq number (PR #1039).

3.11.17

• Add transport.setMinOutgoingBitrate() method (PR #1038, credits to @ jcague).
• RTC::RetransmissionBuffer: Increase RetransmissionBufferMaxItems from 2500 to 5000.

3.11.16

• Fix SeqManager: Properly consider previous cycle dropped inputs (PR #1032).
• RtpRetransmissionBuffer: Get rid of not necessary startSeq private member (PR #1029).
• Node: Upgrade TypeScript to 5.0.2.

3.11.15

• RtpRetransmissionBuffer: Fix crash and add fuzzer (PR #1028).

3.11.14

• Refactor RTP retransmission buffer in a separate and testable RTC::RetransmissionBuffer class (PR #1023).

3.11.13

• AudioLevelObserver: Use multimap rather than map to avoid conflict if various Producers generate same audio level (PR #1021, issue reported by @buptlsp).

3.11.12

• Fix jitter calculation (PR #1019, credits to @alexciarlillo and @snnz).

3.11.11

• Add support for RTCP NACK in OPUS (PR #1015).

3.11.10

• Download and use MSYS/make locally for Windows postinstall (PR #792 by @snnz).

3.11.9

• Allow simulcast with a single encoding (and N temporal layers) (PR #1013).
• Update libsrtp to 2.5.0.

3.11.8

• SimulcastConsumer::GetDesiredBitrate(): Choose the highest bitrate among all Producer streams (PR #992).
• SimulcastConsumer: Fix frozen video when syncing keyframe is discarded due to too high RTP timestamp extra offset needed (PR #999, thanks to @satoren for properly reporting the issue and helping with the solution).

3.11.7

• libwebrtc: Fix crash due to invalid arrival_time value (PR #985 by @ggarber).
• libwebrtc: Replace MS_ASSERT() with MS_ERROR() (PR #988).

3.11.6

• Fix wrong PictureID rolling over in VP9 and VP8 (PR #984 by @jcague).

3.11.5

• Require Node.js >= 16 (PR #973).
• Fix wrong Consumer bandwidth estimation under Producer packet loss (PR #962 by @ggarber).

3.11.4

• Node: Migrate tests to TypeScript (PR #958).
• Node: Remove compiled JavaScript from repository and compile TypeScript code on NPM prepare script on demand when installed via git (PR #954).
• Worker: Add RTC::Shared singleton for RTC entities (PR #953).
• Update OpenSSL to 3.0.7.

3.11.3

• ChannelMessageHandlers: Make RegisterHandler() not remove the existing handler if another one with same id is given (PR #952).

3.11.2

• Fix installation issue in Linux due to a bug in ninja latest version 1.11.1 (PR #948).

3.11.1

• ActiveSpeakerObserver: Revert 'dominantspeaker' event changes in PR #941 to avoid breaking changes (PR #947).

3.11.0

• Transport: Remove duplicate call to method (PR #931).
• RTCP: Adjust maximum compound packet size (PR #934).
• DataConsumer: Fix bufferedAmount type to be a number again (PR #936).
• ActiveSpeakerObserver: Fix 'dominantspeaker' event by having a single Producer as argument rather than an array with a single Producer into it (PR #941).
• ActiveSpeakerObserver: Fix memory leak (PR #942).
• Fix some libwebrtc issues (PR #944).
• Tests: Normalize hexadecimal data representation (PR #945).
• SctpAssociation: Fix memory violation (PR #943).

3.10.12

• Fix worker crash due to std::out_of_range exception (PR #933).

3.10.11

• RTCP: Fix trailing space needed by srtp_protect_rtcp() (PR #929).

3.10.10

• Fix the JSON serialization for the payload channel rtp event (PR #926 by @mhammo).

3.10.9

• RTCP enhancements (PR #914).

3.10.8

• Consumer: use a bitset instead of a set for supported payload types (PR #919).
• RtpPacket: optimize UpdateMid() (PR #920).
• Little optimizations and modernization (PR #916).
• Fix SIGSEGV at RTC::WebRtcTransport::OnIceServerTupleRemoved() (PR #915, credits to @ybybwdwd).
• WebRtcServer: Make port optional (if not given, a random available port from the Worker port range is used) (PR #908 by @satoren).

3.10.7

• Forward abs-capture-time RTP extension also for audio packets (PR #911).

3.10.6

• Node: Define TypeScript types for internal and data objects (PR #891).
• Channel and PayloadChannel: Refactor internal with a single handlerId (PR #889).
• Channel and PayloadChannel: Optimize message format and JSON generation (PR #893).
• New C++ ChannelMessageHandlers class (PR #894).
• Fix Rust support after recent changes (PR #898).
• Modify FeedbackRtpTransport and tests to be compliant with latest libwebrtc code, make reference time to be unsigned (PR #899 by @penguinol and @sarumjanuch).

3.10.5

• RtpStreamSend: Do not store too old RTP packets (PR #885).
• Log error details in channel socket. (PR #875 by @mstyura).

3.10.4

• Do not clone RTP packets if not needed (PR #850).
• Fix DTLS related crash (PR #867).

3.10.3

• SimpleConsumer: Fix. Only process Opus codec (PR #865).
• TypeScript: Improve WebRtcTransportOptions type to not allow webRtcServer and listenIpsoptions at the same time (PR #852).

3.10.2

• Fix release contents by including meson_options.txt (PR #863).

3.10.1

• RtpStreamSend: Memory optimizations (PR #840). Extracted from #675, by @nazar-pc.
• SimpleConsumer: Opus DTX ignore capabilities (PR #846).
• Update libuv to 1.44.1: Fixes libuv build (PR #857).

3.10.0

• WebRtcServer: A new class that brings to WebRtcTransports the ability to listen on a single UDP/TCP port (PR #834).
• More SRTP crypto suites (PR #837).
• Improve EnhancedEventEmitter (PR #836).
• TransportCongestionControlClient: Allow setting max outgoing bitrate before tccClient is created (PR #833).
• Update TypeScript version.

3.9.17

• RateCalculator: Fix old buffer items cleanup (PR #830 by @dsdolzhenko).
• Update TypeScript version.

3.9.16

• SimulcastConsumer: Fix spatial layer switch with unordered packets (PR #823 by @jcague).
• Update TypeScript version.

3.9.15

• RateCalculator: Revert Fix old buffer items cleanup (PR #819 by @dsdolzhenko).

3.9.14

• NackGenerator: Add a configurable delay before sending NACK (PR #827, credits to @penguinol).
• SimulcastConsumer: Fix a race condition in SimulcastConsumer (PR #825 by @dsdolzhenko).
• Add support for H264 SVC (#798 by @prtmD).
• RtpStreamSend: Support receive RTCP-XR RRT and send RTCP-XR DLRR (PR #781 by @aggresss).
• RateCalculator: Fix old buffer items cleanup (PR #819 by @dsdolzhenko).
• DirectTransport: Create a buffer to process RTP packets (PR #730 by @rtctt).
• Node: Improve appData TypeScript syntax and initialization.
• Allow setting max outgoing bitrate below the initial value (PR #826 by @ggarber).
• Update TypeScript version.

3.9.13

• VP8: Do not discard TL0PICIDX from Temporal Layers higher than 0 (PR @817 by @jcague).
• Update TypeScript version.

3.9.12

• DtlsTransport: Make DTLS negotiation run immediately (PR #815).
• Update TypeScript version.

3.9.11

• Modify SimulcastConsumer to keep using layers without good scores (PR #804 by @ggarber).

3.9.10

• Update worker dependencies:
  • OpenSSL 3.0.2.
  • abseil-cpp 20211102.0.
  • nlohmann_json 3.10.5.
  • usrsctp snapshot 4e06feb01cadcd127d119486b98a4bd3d64aa1e7.
  • wingetopt 1.00.
• Update TypeScript version.
• Fix RTP marker bit not being reseted after mangling in each Consumer (PR #811 by @ggarber).

3.9.9

• Optimize RTP header extension handling (PR #786).
• RateCalculator: Reset optimization (PR #785).
• Fix frozen video due to double call to Consumer::UserOnTransportDisconnected() (PR #788, thanks to @ggarber for exposing this issue in PR #787).

3.9.8

• Fix VP9 kSVC forwarding logic to not forward lower unneded layers (PR #778 by @ggarber).
• Fix update bandwidth estimation configuration and available bitrate when updating max outgoing bitrate (PR #779 by @ggarber).
• Replace outdated random-numbers package by native crypto.randomInt() (PR #776 by @piranna).
• Update TypeScript version.

3.9.7

• Typing event emitters in mediasoup Node (PR #764 by @unao).

3.9.6

• TCC client optimizations for faster and more stable BWE (PR #712 by @ggarber).
• Added support for RTP abs-capture-time header (PR #761 by @oto313).

3.9.5

• ICE renomination support (PR #756).
• Update libuv to 1.43.0.

3.9.4

• Worker: Fix bad printing of error messages from Worker (PR #750 by @j1elo).

3.9.3

• Single H264/H265 codec configuration in supportedRtpCapabilities (PR #718).
• Improve Windows support by not requiring MSVC configuration (PR #741).

3.9.2

• pipeToRouter(): Reuse same PipeTransport when possible (PR #697).
• Add worker.died boolean getter.
• Update TypeScript version to 4.X.X and use target: "esnext" so transpilation of ECMAScript private fields (#xxxxx) don't use WeakMaps tricks but use standard syntax instead.
• Use more than one core for compilation on Windows (PR #709).
• Consumer: Modification of bitrate allocation algorithm (PR #708).

3.9.1

• NixOS friendly build process (PR #683).
• Worker: Emit "died" event before observer "close" (PR #684).
• Transport: Hide debug message for RTX RTCP-RR packets (PR #688).
• Update libuv to 1.42.0.
• Improve Windows support (PR #692).
• Avoid build commands when MEDIASOUP_WORKER_BIN is set (PR #695).

3.9.0

• Replaces GYP build system with fully-functional Meson build system (PR #622).
• Worker communication optimization (aka removing netstring dependency) (PR #644).
• Move TypeScript and compiled JavaScript code to a new node folder.
• Use ES6 private fields.
• Require Node.js version >= 12.

3.8.4

• OPUS multi-channel (Surround sound) support (PR #647).
• Add packetLoss stats to transport (PR #648 by @ggarber).
• Fixes for active speaker observer (PR #655 by @ggarber).
• Fix big endian issues (PR #639).

3.8.3

• Fix wrong size_t* to int* conversion in 64bit Big-Endian hosts (PR #637).

3.8.2

• ActiveSpeakerObserver: Fix crash due to a nullptr (PR #634).

3.8.1

• SimulcastConsumer: Fix RTP timestamp when switching layers (PR #626 by @penguinol).

3.8.0

• Update libuv to 1.42.0.
• Use non-ASM OpenSSL on Windows (PR #614).
• Fix minor memory leak caused by non-virtual destructor (PR #625).
• Dominant Speaker Event (PR #603 by @SteveMcFarlin).

3.7.19

• Update libuv to 1.41.0.
• C++:
  • Move header includes (PR #608).
  • Enhance debugging on channel request/notification error (PR #607).

3.7.18

• Support for optional fixed port on transports (PR #593 by @nazar-pc).
• Upgrade and optimize OpenSSL dependency (PR #598 by @vpalmisano):
  • OpenSSL upgraded to version 1.1.1k.
  • Enable the compilation of assembly extensions for OpenSSL.
  • Optimize the worker build (-O3) and disable the debug flag (-g).

3.7.17

• Introduce PipeConsumerOptions to avoid incorrect type information on PipeTransport.consume() arguments.
• Make ConsumerOptions.rtpCapabilities field required as it should have always been.

3.7.16

• Add mid option in ConsumerOptions to provide way to override MID (PR #586 by @mstyura).

3.7.15

• kind field of RtpHeaderExtension is no longer optional. It must be 'audio' or 'video'.
• Refactor API inconsistency in internal RTP Observer communication with worker.

3.7.14

• Update usrsctp to include a "possible use after free bug" fix (commit here).

3.7.13

• Fix build on FreeBSD (PR #585 by @smortex).

3.7.12

• mediasoup-worker: Fix memory leaks on error exit (PR #581).

3.7.11

• Fix DepUsrSCTP::Checker::timer not being freed on Worker close (PR #576). Thanks @nazar-pc for discovering this.

3.7.10

• Remove clang tools binaries from regular installation.

3.7.9

• Code clean up.

3.7.8

• PayloadChannel: Copy received messages into a separate buffer to avoid memory corruption if the message is later modified (PR #570 by @aggresss).

3.7.7

• Thread and memory safety fixes needed for mediasoup-rust (PR #562 by @nazar-pc).
• mediasoup-rust support on macOS (PR #567 by @nazar-pc).
• mediasoup-rust release 0.7.2.

3.7.6

• Transport: Implement new setMaxOutgoingBitrate() method (PR #555 by @t-mullen).
• SctpAssociation: Don't warn if SCTP send buffer is full.
• Rust: Update modules structure and other minor improvements for Rust version (PR #558).
• mediasoup-worker: Avoid duplicated basenames so that libmediasoup-worker is compilable on macOS (PR #557).

3.7.5

• SctpAssociation: provide 'sctpsendbufferfull' reason on send error (#552).

3.7.4

• Improve RateCalculator (PR #547 by @vpalmisano).

3.7.3

• Make worker M1 compilable.

3.7.2

• RateCalculator optimization (PR #538 by @vpalmisano).

3.7.1

• SimulcastConsumer: Fix miscalculation when increasing layer (PR #541 by @penguinol).
• Rust version with thread-based worker (PR #540).

3.7.0

• Welcome to mediasoup-rust! Authored by @nazar-pc (PRs #518 and #533).
• Update usrsctp.

3.6.37

• Fix crash if empty fingerprints array is given in webrtcTransport.connect() (issue #537).

3.6.36

• Producer: Add new stats field 'rtxPacketsDiscarded' (PR #536).

3.6.35

• XxxxConsumer.hpp: make IsActive() return true (even if Producer's score is 0) when DTX is enabled (PR #534 due to issue #532).

3.6.34

• Fix crash (regression, issue #529).

3.6.33

• Add missing delete cb that otherwise would leak (PR #527 based on PR #526 by @vpalmisano).
• router.pipeToRouter(): Fix possible inconsistency in pipeProducer.paused status (as discussed in this thread in the mediasoup forum).
• Update nlohmann/json to 3.9.1.
• Update usrsctp.
• Enhance Jitter calculation.

3.6.32

• Fix notifications from mediasoup-worker being processed before responses received before them (issue #501).

3.6.31

• Move bufferedAmount from dataConsumer.dump() to dataConsumer.getStats().

3.6.30

• Add pipe option to transport.consume()(PR #494).
  • So the receiver will get all streams from the Producer.
  • It works for any kind of transport (but PipeTransport which is always like this).
• Add LICENSE and PATENTS files in libwebrtc dependency (issue #495).
• Added worker/src/Utils/README_BASE64_UTILS (issue #497).
• Update usrsctp.

3.6.29

• Fix wrong message about rtcMinPort and rtcMaxPort.
• Update deps.
• Improve EnhancedEventEmitter.safeAsPromise() (although not used).

3.6.28

• Fix replacement of __MEDIASOUP_VERSION__ in lib/index.d.ts (issue #483).
• worker/scripts/configure.py: Handle 'mips64' (PR #485).

3.6.27

• Allow the mediasoup-worker process to inherit all environment variables (issue #480).

3.6.26

• BWE tweaks and debug logs.

3.6.25

• SCTP fixes (PR #479).

3.6.24

• Update awaitqueue dependency.

3.6.23

• Fix yet another memory leak in Node.js layer due to PayloadChannel event listener not being removed.

3.6.22

• Transport.cpp: Provide transport congestion client with RTCP Receiver Reports (#464).
• Update libuv to 1.40.0.
• Update Node deps.
• SctpAssociation.cpp: increase sctpBufferedAmount before sending any data (#472).

3.6.21

• Fix memory leak in Node.js layer due to PayloadChannel event listener not being removed (related to #463).

3.6.20

• Remove -fwrapv when building mediasoup-worker in Debug mode (issue #460).
• Add MEDIASOUP_MAX_CORES to limit NUM_CORES during mediasoup-worker build (PR #462).

3.6.19

• Update usrsctp dependency.
• Update typescript-eslint deps.
• Update Node deps.

3.6.18

• Fix ortc.getConsumerRtpParameters() RTX codec comparison issue (PR #453).
• RtpObserver: expose RtpObserverAddRemoveProducerOptions for addProducer() and removeProducer() methods.

3.6.17

• Update libuv to 1.39.0.
• Update Node deps.
• SimulcastConsumer: Prefer the highest spatial layer initially (PR #450).
• RtpStreamRecv: Set RtpDataCounter window size to 6 secs if DTX (#451)

3.6.16

• SctpAssociation.cpp: Fix OnSctpAssociationBufferedAmount() call.
• Update deps.
• New API to send data from Node throught SCTP DataConsumer.

3.6.15

• Avoid SRTP leak by deleting invalid SSRCs after STRP decryption (issue #437, thanks to @penguinol for reporting).
• Update usrsctp dep.
• DataConsumer 'bufferedAmount' implementation (PR #442).

3.6.14

• Fix usrsctp vulnerability (PR #439).

• Fix issue #435 (thanks to @penguinol for reporting).

• TransportCongestionControlClient.cpp: Enable periodic ALR probing to recover faster from network issues.

• Update nlohmann::json C++ dep to 3.9.0.

3.6.13

• RTP on DirectTransport (issue #433, PR #434):
  • New API producer.send(rtpPacket: Buffer).
  • New API consumer.on('rtp', (rtpPacket: Buffer).
  • New API directTransport.sendRtcp(rtcpPacket: Buffer).
  • New API directTransport.on('rtcp', (rtpPacket: Buffer).

3.6.12

• Release script.

3.6.11

• Transport: rename maxSctpSendBufferSize to sctpSendBufferSize.

3.6.10

• Transport: Implement maxSctpSendBufferSize.
• Update libuv to 1.38.1.

3.6.9

• Transport::ReceiveRtpPacket(): Call RecvStreamClosed(packet->GetSsrc()) if received RTP packet does not match any Producer.
• Transport::HandleRtcpPacket(): Ensure Consumer is found for received NACK Feedback packets.
• Fix issue #408.

3.6.8

• Fix SRTP leak due to streams not being removed when a Producer or Consumer is closed.
  • PR #428 (fixes issues #426).
  • Credits to credits to @penguinol for reporting and initial work at PR #427.
• Update nlohmann::json C++ dep to 3.8.0.
• C++: Enhance const correctness.

3.6.7

• ConsumerScore: Add producerScores, scores of all RTP streams in the producer ordered by encoding (just useful when the producer uses simulcast).
  • PR #421 (fixes issues #420).
• Hide worker executable console in Windows.
  • PR #419 (credits to @BlueMagnificent).
• RtpStream.cpp: Fix wrong std::round() usage.
  • Issue #423.

3.6.6

• Update usrsctp library.
• Update ESlint and TypeScript related dependencies.

3.6.5

• Set score:0 when dtx:true is set in an encoding and there is no RTP for some seconds for that RTP stream.
  • Fixes #415.

3.6.4

• gyp: Fix CLT version detection in OSX Catalina when XCode app is not installed.
  • PR #413 (credits to @enimo).

3.6.3

• Modernize TypeScript.

3.6.2

• Fix crash in Transport.ts when closing a DataConsumer created on a DirectTransport.

3.6.1

• Export new DirectTransport in types.
• Make DataProducerOptions optional (not needed when in a DirectTransport).

3.6.0

• SCTP/DataChannel termination:
  • PR #409
  • Allow the Node application to directly send text/binary messages to mediasoup-worker C++ process so others can consume them using DataConsumers.
  • And vice-versa: allow the Node application to directly consume in Node messages send by DataProducers.
• Add WorkerLogTag TypeScript enum and also add a new 'message' tag into it.

3.5.15

• Simulcast and SVC: Better computation of desired bitrate based on maxBitrate field in the producer.rtpParameters.encodings.

3.5.14

• Update deps, specially uuid and @types/uuid that had a TypeScript related bug.
• TransportCongestionClient.cpp: Improve sender side bandwidth estimation by do not reporting this->initialAvailableBitrate as available bitrate due to strange behavior in the algorithm.

3.5.13

• Simplify GetDesiredBitrate() in SimulcastConsumer and SvcConsumer.
• Update libuv to 1.38.0.

3.5.12

• SeqManager.cpp: Improve performance.
  • PR #398 (credits to @penguinol).

3.5.11

• SeqManager.cpp: Fix a bug and improve performance.
  • Fixes issue #395 via PR #396 (credits to @penguinol).
• Drop Node.js 8 support. Minimum supported Node.js version is now 10.
• Upgrade eslint and jest major versions.

3.5.10

• SimulcastConsumer.cpp: Fix IncreaseLayer() method (fixes #394).
• Udpate Node deps.

3.5.9

• libwebrtc: Apply patch by @sspanak and @Ivaka to avoid crash. Related issue: #357.
• PortManager.cpp: Do not use UV_UDP_RECVMMSG in Windows due to a bug in libuv 1.37.0.
• Update Node deps.

3.5.8

• Enable UV_UDP_RECVMMSG:
  • Upgrade libuv to 1.37.0.
  • Use uv_udp_init_ex() with UV_UDP_RECVMMSG flag.
  • Add our own uv.gyp now that libuv has removed support for GYP (fixes #384).

3.5.7

• Fix crash in mediasoup-worker due to conversion from uint64_t to int64_t (used within libwebrtc code. Fixes #357.
• Update usrsctp library.
• Update Node deps.

3.5.6

• SeqManager.cpp: Fix video lag after a long time.
  • Fixes #372 (thanks @penguinol for reporting it and giving the solution).

3.5.5

• UdpSocket.cpp: Revert uv__udp_recvmmsg() usage since it notifies about received UDP packets in reverse order. Feature on hold until fixed.

3.5.4

• Transport.cpp: Enable transport congestion client for the first video Consumer, no matter it's uses simulcast, SVC or a single stream.
• Update libuv to 1.35.0.
• UdpSocket.cpp: Ensure the new libuv's uv__udp_recvmmsg() is used, which is more efficient.

3.5.3

• PlainTransport: Remove multiSource option. It was a hack nobody should use.

3.5.2

• Enable MID RTP extension in mediasoup to receivers direction (for consumers).
  • This requires mediasoup-client 3.5.2 to work.

3.5.1

• PlainTransport: Fix event name: 'rtcpTuple' => 'rtcptuple'.

3.5.0

• PipeTransport: Add support for SRTP and RTP retransmission (RTX + NACK). Useful when connecting two mediasoup servers running in different hosts via pipe transports.
• PlainTransport: Add support for SRTP.
• Rename PlainRtpTransport to PlainTransport everywhere (classes, methods, TypeScript types, etc). Keep previous names and mark them as DEPRECATED.
• Fix vulnarability in IPv6 parser.

3.4.13

• Update uuid dep to 7.0.X (new API).
• Fix crash due wrong array index in PipeConsumer::FillJson().
  • Fixes #364

3.4.12

• TypeScript: generate es2020 instead of es6.
• Update usrsctp library.
  • Fixes #362 (thanks @chvarlam for reporting it).

3.4.11

• IceServer.cpp: Reject received STUN Binding request with 487 if remote peer indicates ICE-CONTROLLED into it.

3.4.10

• ProducerOptions: Rename keyFrameWaitTime option to keyFrameRequestDelay and make it work as expected.

3.4.9

• Add Utils::Json::IsPositiveInteger() to not rely on is_number_unsigned() of json lib, which is unreliable due to its design.
• Avoid ES6 export default and always use named export.
• router.pipeToRouter(): Ensure a single PipeTransport pair is created between router1 and router2.
  • Since the operation is async, it may happen that two simultaneous calls to router1.pipeToRouter({ producerId: xxx, router: router2 }) would end up generating two pairs of PipeTranports. To prevent that, let's use an async queue.
• Add keyFrameWaitTime option to ProducerOptions.
• Update Node and C++ deps.

3.4.8

• libsrtp.gyp: Fix regression in mediasoup for Windows.
  • libsrtp.gyp: Modernize it based on the new BUILD.gn in Chromium.
  • libsrtp.gyp: Don't include "test" and other targets.
  • Assume HAVE_INTTYPES_H, HAVE_INT8_T, etc. in Windows.
  • Issue details: sctplab/usrsctp#353
• gyp dependency: Add support for Microsoft Visual Studio 2019.
  • Modify our own gyp sources to fix the issue.
  • CL uploaded to GYP project with the fix.
  • Issue details: sctplab/usrsctp#347

3.4.7

• PortManager.cpp: Do not limit the number of failed bind() attempts to 20 since it does not work well in scenarios that launch tons of Workers with same port range. Instead iterate all ports in the range given to the Worker.
• Do not copy catch.hpp into test/include/ but make the GYP mediasoup-worker-test target include the corresponding folder in deps/catch.

3.4.6

• Update libsrtp to 2.3.0.
• Update ESLint and TypeScript deps.

3.4.5

• Update deps.
• Fix text in ./github/Bug_Report.md so it no longer references the deprecated mailing list.

3.4.4

• Transport.cpp: Ignore RTCP SDES packets (we don't do anything with them anyway).
• Producer and Consumer stats: Always show roundTripTime (even if calculated value is 0) after a roundTripTime > 0 has been seen.

3.4.3

• Transport.cpp: Fix RTCP FIR processing:
  • Instead of looking at the media ssrc in the common header, iterate FIR items and look for associated Consumers based on ssrcs in each FIR item.
  • Fixes #350 (thanks @j1elo for reporting and documenting the issue).

3.4.2

• SctpAssociation.cpp: Improve/fix logs.
• Improve Node EventEmitter events inline documentation.
• test-node-sctp.js: Wait for SCTP association to be open before sending data.

3.4.1

• Improve mediasoup-worker build system by using sh instead of bash and default to 4 cores (thanks @smoke, PR #349).

3.4.0

• Add worker.getResourceUsage() API.
• Update OpenSSL to 1.1.1d.
• Update libuv to 1.34.0.
• Update TypeScript version.

3.3.8

• Update usrsctp dependency (it fixes a potential wrong memory access).
  • More details in the reported issue: sctplab/usrsctp#408

3.3.7

• Fix version getter.

3.3.6

• SctpAssociation.cpp: Initialize the usrsctp socket in the class constructor. Fixes #348.

3.3.5

• Fix usage of a deallocated RTC::TcpConnection instance under heavy CPU usage due to mediasoup deleting the instance in the middle of a receiving iteration. Fixes #333.
  • More details in the commit: https://github.com/versatica/mediasoup/commit/49824baf102ab6d2b01e5bca565c29b8ac0fec22

3.3.4

• IPv6 fix: Use INET6_ADDRSTRLEN instead of INET_ADDRSTRLEN.

3.3.3

• Add consumer.setPriority() and consumer.priority API to prioritize how the estimated outgoing bitrate in a transport is distributed among all video consumers (in case there is not enough bitrate to satisfy them).
• Make video SimpleConsumers play the BWE game by helping in probation generation and bitrate distribution.
• Add consumer.preferredLayers getter.
• Rename enablePacketEvent() and "packet" event to enableTraceEvent() and "trace" event (sorry SEMVER).
• Transport: Add a new "trace" event of type "bwe" with detailed information about bitrates.

3.3.2

• Improve "packet" event by not firing both "keyframe" and "rtp" types for the same RTP packet.

3.3.1

• Add type "keyframe" as a valid type for "packet" event in Producers and Consumers.

3.3.0

• Add transport-cc bandwidth estimation and congestion control in sender and receiver side.
• Run in Windows.
• Rewrite to TypeScript.
• Tons of improvements.

3.2.5

• Fix TCP leak (#325).

3.2.4

• PlainRtpTransport: Fix comedia mode.

3.2.3

• RateCalculator: improve efficiency in GetRate() method (#324).

3.2.2

• RtpDataCounter: use window size of 2500 ms instead of 1000 ms.
  • Fixes false "lack of RTP" detection in some screen sharing usages with simulcast.
  • Fixes #312.

3.2.1

• Add RTCP Extended Reports for RTT calculation on receiver RTP stream (thanks @yangjinechofor for initial pull request #314).
• Make mediasoup-worker compile in Armbian Debian Buster (thanks @krishisola, fixes #321).

3.2.0

• Add DataChannel support via DataProducers and DataConsumers (#10).
• SRTP: Add support for AEAD GCM (#320).

3.1.7

• PipeConsumer.cpp: Fix RTCP generation (thanks @vpalmisano).

3.1.6

• VP8 and H264: Fix regression in 3.1.5 that produces lot of changes in current temporal layer detection.

3.1.5

• VP8 and H264: Allow packets without temporal layer information even if N temporal layers were announced.

3.1.4

• Add -fPIC in cflags to compile in x86-64. Fixes #315.

3.1.3

• Set the sender SSRC on PLI and FIR requests related thread.

3.1.2

• Workaround to detect H264 key frames when Chrome uses external encoder (related issue). Fixes #313.

3.1.1

• Improve GetBitratePriority() method in SimulcastConsumer and SvcConsumer by checking the total bitrate of all temporal layers in a given producer stream or spatial layer.

3.1.0

• Add SVC support. It includes VP9 full SVC and VP9 K-SVC as implemented by libwebrtc.
• Prefer Python 2 (if available) over Python 3. This is because there are yet pending issues with gyp + Python 3.

3.0.12

• Do not require Python 2 to compile mediasoup worker (#207). Both Python 2 and 3 can now be used.

3.0.11

• Codecs: Improve temporal layer switching in VP8 and H264.
• Skip worker compilation if MEDIASOUP_WORKER_BIN environment variable is given (#309). This makes it possible to install mediasoup in platforms in which, somehow, gcc > 4.8 is not available during npm install mediasoup but it's available later.
• Fix RtpStreamRecv::TransmissionCounter::GetBitrate().

3.0.10

• parseScalabilityMode(): allow "S" as spatial layer (and not just "L"). "L" means "dependent spatial layer" while "S" means "independent spatial layer", which is used in K-SVC (VP9, AV1, etc).

3.0.9

• RtpStreamSend::ReceiveRtcpReceiverReport(): improve rtt calculation if no Sender Report info is reported in received Received Report.
• Update libuv to version 1.29.1.

3.0.8

• VP8 & H264: Improve temporal layer switching.

3.0.7

• RTP frame-marking: Add some missing checks.

3.0.6

• Fix regression in proxied RTP header extensions.

3.0.5

• Add support for frame-marking RTP extensions and use it to enable temporal layers switching in H264 codec (#305).

3.0.4

• Improve RTP probation for simulcast/svc consumers by using proper RTP retransmission with increasing sequence number.

3.0.3

• Simulcast: Improve timestamps extra offset handling by having a map of extra offsets indexed by received timestamps. This helps in case of packet retransmission.

3.0.2

• Simulcast: proper RTP stream switching by rewriting packet timestamp with a new timestamp calculated from the SenderReports' NTP relationship.

3.0.1

• Fix crash in SimulcastConsumer::IncreaseLayer() with Safari and H264 (#300).

3.0.0

• v3 is here!

2.6.19

• RtpStreamSend.cpp: Fix a crash in StorePacket() when it receives an old packet and there is no space left in the storage buffer (thanks to zkfun for reporting it and providing us with the solution).
• Update deps.

2.6.18

• Fix usage of a deallocated RTC::TcpConnection instance under heavy CPU usage due to mediasoup deleting the instance in the middle of a receiving iteration.

2.6.17

• Improve build system by using all available CPU cores in parallel.

2.6.16

• Don't mandate server port range to be >= 99.

2.6.15

• Fix NACK retransmissions.

2.6.14

• Fix TCP leak (#325).

2.6.13

• Make mediasoup-worker compile in Armbian Debian Buster (thanks @krishisola, fixes #321).
• Update deps.

2.6.12

• Fix RTCP Receiver Report handling.

2.6.11

• Update deps.
• Simulcast: Increase profiles one by one unless explicitly forced (fixes #188).

2.6.10

• PlainRtpTransport.js: Add missing methods and events.

2.6.9

• Remove a potential crash if a single encoding is given in the Producer rtpParameters and it has a profile value.

2.6.8

• C++: Verify in libuv static callbacks that the associated C++ instance has not been deallocated (thanks @artushin and @mariat-atg for reporting and providing valuable help in #258).

2.6.7

• Fix wrong destruction of Transports in Router.cpp that generates 100% CPU usage in mediasoup-worker processes.

2.6.6

• Fix a port leak when a WebRtcTransport is remotely closed due to a DTLS close alert (thanks @artushin for reporting it in #259).

2.6.5

• RtpPacket: Fix Two-Byte header extensions parsing.

2.6.4

• Upgrade again to OpenSSL 1.1.0j (20 Nov 2018) after adding a workaround for issue #257.

2.6.3

• Downgrade OpenSSL to version 1.1.0h (27 Mar 2018) until issue #257 is fixed.

2.6.2

• C++: Remove all Destroy() class methods and no longer do delete this.
• Update libuv to 1.24.1.
• Update OpenSSL to 1.1.0g.

2.6.1

• worker: Internal refactor and code cleanup.
• Remove announced support for certain RTCP feedback types that mediasoup does nothing with (and avoid forwarding them to the remote RTP sender).
• fuzzer: fix some wrong memory access in RtpPacket::Dump() and StunMessage::Dump() (just used during development).

2.6.0

• Integrate libFuzzer into mediasoup (documentation in the doc folder). Extensive testing done. Several heap-buffer-overflow and memory leaks fixed.

2.5.6

• Producer.cpp: Remove UpdateRtpParameters(). It was broken since Consumers were not notified about profile removed and so on, so they may crash.
• Producer.cpp: Remove some maps and simplify streams handling by having a single mapSsrcRtpStreamInfo. Just keep mapActiveProfilesbecauseGetActiveProfiles()` method needs it.
• Producer::MayNeedNewStream(): Ignore new media streams with new SSRC if its RID is already in use by other media stream (fixes #235).
• Fix a bad memory access when using two byte RTP header extensions.

2.5.5

• Server.js: If a worker crashes make sure _latestWorkerIdx becomes 0.

2.5.4

• server.Room(): Assign workers incrementally or explicitly via new workerIdx argument.
• Add server.numWorkers getter.

2.5.3

• Don't announce muxId nor RTP MID extension support in Consumer RTP parameters.

2.5.2

• Enable RTP MID extension again.

2.5.1

• Disable RTP MID extension until #230 is fixed.

2.5.0

• Add RTP MID extension support.

2.4.6

• Do not close Transport on ICE disconnected (as it would prevent ICE restart on "recv" TCP transports).

2.4.5

• Improve codec matching.

2.4.4

• Fix audio codec matching when channels parameter is not given.

2.4.3

• Make PlainRtpTransport not leak if port allocation fails (related issue #224).

2.4.2

• Fix a crash in when no more RTP ports were available (see related issue #222).

2.4.1

• Update dependencies.

2.4.0

• Allow non WebRTC peers to create plain RTP transports (no ICE/DTLS/SRTP but just plain RTP and RTCP) for sending and receiving media.

2.3.3

• Fix C++ syntax to avoid an error when building the worker with clang 8.0.0 (OSX 10.11.6).

2.3.2

• Channel.js: Upgrade REQUEST_TIMEOUT to 20 seconds to avoid timeout errors when the Node or worker thread usage is too high (related to this issue).

2.3.1

• H264: Check if there is room for the indicated NAL unit size (thanks @ggarber).
• H264: Code cleanup.

2.3.0

• Add new "spy" feature. A "spy" peer cannot produce media and is invisible for other peers in the room.

2.2.7

• Fix H264 simulcast by properly detecting when the profile switching should be done.
• Fix a crash in Consumer::GetStats() (see related issue #196).

2.2.6

• Add H264 simulcast capability.

2.2.5

• Avoid calling deprecated (NOOP) SSL_CTX_set_ecdh_auto() function in OpenSSL >= 1.1.0.

2.2.4

• Fix #4: Avoid DTLS handshake fragmentation.

2.2.3

• Fix #196: Crash in Consumer::getStats() due to wrong targetProfile.

2.2.2

• Improve issue #209.

2.2.1

• Fix #209: DtlsTransport: don't crash when signaled fingerprint and DTLS fingerprint do not match (thanks @yangjinecho for reporting it).

2.2.0

• Update Node and C/C++ dependencies.

2.1.0

• Add localIP option for room.createRtpStreamer() and transport.startMirroring() [PR #199](#199).

2.0.16

• Improve C++ usage (remove "warning: missing initializer for member" [-Wmissing-field-initializers]).
• Update Travis-CI settings.

2.0.15

• Make PlainRtpTransport also send RTCP SR/RR reports (thanks @artushin for reporting).

2.0.14

• Fix #193: preferTcp not honored (thanks @artushin).

2.0.13

• Avoid crash when no remote IP/port is given.

2.0.12

• Add handled and unhandled events to Consumer.

2.0.11

• Fix #185: Consumer: initialize effective profile to 'NONE' (thanks @artushin).
• Fix #186: NackGenerator code being executed after instance deletion (thanks @baiyufei).

2.0.10

• Fix #183: Always reset the effective Consumer profile when removed (thanks @thehappycoder).

2.0.9

• Make ICE+DTLS more flexible by allowing sending DTLS handshake when ICE is just connected.

2.0.8

• Disable stats periodic retrieval also on remote closure of Producer and WebRtcTransport.

2.0.7

• Fix #180: Added missing include cmath so that std::round can be used (thanks @jacobEAdamson).

2.0.6

• Fix #173: Avoid buffer overflow in () (thanks @lightmare).
• Improve stream layers management in Consumer by using the new RtpMonitor class.

2.0.5

• Fix #164: Sometimes video freezes forever (no RTP received in browser at all).
• Fix #160: Assert error in RTC::Consumer::GetStats().

2.0.4

• Fix #159: Don’t rely on VP8 payload descriptor flags to assure the existence of data.
• Fix #160: Reset targetProfile when the corresponding profile is removed.

2.0.3

• worker: Fix crash when VP8 payload has no PictureId.

2.0.2

• worker: Remove wrong assert on Producer::DeactivateStreamProfiles().

2.0.1

• Update README file.

2.0.0

• New design based on Producers and Consumer plus a mediasoup protocol and the mediasoup-client client side SDK.

1.2.8

• Fix a crash due to RTX packet processing while the associated NackGenerator is not yet created.

1.2.7

• Habemus RTX (RFC 4588) for proper RTP retransmission.

1.2.6

• Fix an issue in buffer.toString() that makes mediasoup fail in Node 8.
• Update libuv to version 1.12.0.

1.2.5

• Add support for ICE renomination.

1.2.4

• Fix a SDP negotiation issue when the remote peer does not have compatible codecs.

1.2.3

• Add video codecs supported by Microsoft Edge.

1.2.2

• RtpReceiver: generate RTCP PLI when "rtpraw" or "rtpobject" event listener is set.

1.2.1

• RtpReceiver: fix an error producing packets when "rtpobject" event is set.

1.2.0

• RtpSender: allow disable()/enable() without forcing SDP renegotiation (#114).

1.1.0

• Add Room.on('audiolevels') event.

1.0.2

• Set a maximum value of 1500 bytes for packet storage in RtpStreamSend.

1.0.1

• Avoid possible segfault if RemoteBitrateEstimator generates a bandwidth estimation with zero SSRCs.

1.0.0

• First stable release.