Added #

• RTCP Extended Reports (XR) - RFC 3611 - Full implementation of RTCP XR (PT=207):
  • RtcpExtendedReport - Main XR packet wrapper
  • ReceiverReferenceTimeBlock (BT=4) - Enables RTT for non-senders
  • DlrrBlock (BT=5) - Delay Since Last Receiver Report for RTT calculation
  • StatisticsSummaryBlock (BT=6) - Aggregate QoS metrics
  • 33 new tests for XR parsing/serialization
  • 3 new XR performance tests

Performance #

• DataChannel timing matches werift - Changed SCTP sendData() to use fire-and-forget _transmit(), eliminating blocking in DataChannel.send() and open()

Fixed #

• DTLS race condition - Await initialization in setRemoteDescription to prevent DTLS handshake race
• RTCP lenient handling - Skip unknown RTCP packet types gracefully (matches werift behavior)
• Test stability - Tagged flaky concurrent tests as slow for isolated execution

Tests #

• 2661 tests passing (including 41 performance tests)

Performance #

• ICE candidate parsing 2.8x faster - Replaced split() with indexOf() + substring():

  • Host candidate: 1.8M → 5.0M ops/sec
  • Round-trip: 1.4M → 2.8M ops/sec
  • Gap vs werift reduced from 7x to 2.6x

• Codec parameter parsing optimized - RTCRtpCodecParameters.name and .contentType now use indexOf() instead of split('/') to avoid List allocation

Added #

• Performance regression test suite - 38 new tests in test/performance/:

  • SRTP encrypt/decrypt throughput
  • RTP/RTCP parse/serialize
  • SDP parse/serialize
  • STUN message handling
  • SCTP queue operations
  • DTLS anti-replay window
  • VP8/H.264 depacketization
  • ICE candidate parse/serialize

• Benchmark infrastructure - Compare webrtc_dart vs werift:

  • ./benchmark/run_perf_tests.sh - Run Dart performance tests
  • ./benchmark/run_werift_benchmarks.sh - Run werift comparison
  • ./benchmark/compare.sh - Side-by-side comparison
  • Micro-benchmarks in benchmark/micro/

Tests #

• 2625 tests passing (including 38 performance tests)

Changed #

• Documentation clarity - Clarified server-side WebRTC positioning:
  • README: Explicitly positioned as server-side library (like Pion, aiortc, werift)
  • Added comparison table: browser WebRTC vs server-side capabilities
  • Removed misleading "W3C API compatibility" claims
  • Clarified that media capture and codec encoding/decoding require external tools

Tests #

• 2587 tests passing

Breaking Changes (with backward compatibility) #

• W3C WebRTC API naming - All core classes renamed to match W3C standard:

  • RtcPeerConnection -> RTCPeerConnection
  • DataChannel -> RTCDataChannel
  • RtpTransceiver -> RTCRtpTransceiver
  • RtpSender -> RTCRtpSender
  • RtpReceiver -> RTCRtpReceiver
  • Candidate -> RTCIceCandidate
  • SessionDescription -> RTCSessionDescription

• Backward compatibility preserved - Old names available via deprecated typedefs. Existing code continues to work with deprecation warnings.

Added #

• RTCDTMFSender - Full DTMF (Dual-Tone Multi-Frequency) support:

  • insertDTMF() method with configurable duration and gap
  • toneBuffer property for queued tones
  • ontonechange event callback
  • canInsertDTMF property
  • Supports tones: 0-9, A-D, *, #
  • RTP telephone-event payload (RFC 4733)

• W3C property aliases - Standard naming alongside Dart conventions:

  • RTCDataChannel.id (alias for streamId)
  • RTCDataChannel.readyState (alias for state)
  • RTCIceCandidate.address (alias for host)
  • RTCIceCandidate.protocol (alias for transport)
  • RTCIceCandidate.usernameFragment (alias for ufrag)
  • MediaStreamTrack.readyState (alias for state)

• Missing W3C methods:

  • RTCRtpSender.replaceTrack() - Replace track without renegotiation
  • RTCRtpSender.getStats() - Per-sender statistics
  • RTCRtpReceiver.getParameters() - Receiver RTP parameters
  • RTCRtpReceiver.getStats() - Per-receiver statistics
  • RTCRtpTransceiver.currentDirection - Actual negotiated direction
  • RTCIceCandidate.toJSON() - JSON serialization
  • RTCSessionDescription.toJSON() - JSON serialization

• MediaStreamTrack constraints API:

  • getSettings() - Current track settings
  • getCapabilities() - Device capabilities
  • getConstraints() - Applied constraints
  • applyConstraints() - Apply new constraints

• Transport properties:

  • RTCRtpSender.transport - DTLS transport reference
  • RTCRtpReceiver.transport - DTLS transport reference

Changed #

• All examples updated to use W3C API names
• All tests updated to use W3C API names (no deprecation warnings)
• Documentation updated for W3C API focus
• Merged W3C_COMPAT_PLAN.md and REFACTOR.md into ROADMAP.md

Tests #

• 2587 tests passing (up from 2537)
• Added DTMF unit tests (12 tests)
• Added browser interop tests for DTMF
• All browser interop tests passing (Chrome, Firefox, Safari)

Clarification #

webrtc_dart is a server-side WebRTC library. It provides W3C-compatible API naming (RTCPeerConnection, etc.) for the WebRTC transport layer. As with other server-side implementations (Pion, aiortc, werift), media capture (getUserMedia) and codec encoding/decoding require external tools like FFmpeg or GStreamer.

Fixed #

• RTP sender stream replay - Fixed SSRC handling when replaying streams (matching werift's replaceRTP pattern):
  • Added onSourceChanged subscription to reset SSRC tracking on source change
  • Changed SSRC filtering to accept new SSRCs instead of dropping packets
  • Enables seamless replay of media streams without reconnection

Tests #

• 2537 tests passing

Fixed #

• SCTP flow control improvements - Multiple fixes for real-time streaming:
  • Non-blocking sendData(): Removed blocking that caused complete stalls when T3 timer was running
  • Congestion window check for new chunks: Added missing cwnd check in _transmit() to prevent flooding
  • First SACK handling: Fixed _lastSackedTsn to be nullable, accepting first SACK correctly
  • Increased initial cwnd from 4380 to 65536 bytes for better real-time performance

Added #

• DataChannel bufferedAmount API - Per W3C WebRTC spec:
  • bufferedAmount: Returns bytes queued but not yet ACKed (tracked at SCTP level per-stream)
  • bufferedAmountLowThreshold: Configurable threshold for flow control
  • onBufferedAmountLow: Event fires when buffer drains below threshold
  • Applications can wait for bufferedAmount == 0 before disconnecting to ensure all data is delivered

Tests #

• 2537 tests passing

Fixed #

• SCTP fragment reassembly - Fixed critical bug where fragmented SCTP messages (>1024 bytes) were silently dropped. DataChannel messages larger than userDataMaxLength (1024 bytes) are now properly reassembled from multiple DATA chunks.
• Forward TSN reassembly integration - Added missing reassembly logic to _handleForwardTsn: advances stream sequence numbers, delivers pending messages, and prunes obsolete chunks from all inbound streams.

Added #

• InboundStream class - New lib/src/sctp/inbound_stream.dart implementing per-stream fragment reassembly matching werift's InboundStream pattern. Handles ordered/unordered delivery, out-of-order chunks, and TSN wraparound.

Tests #

• Added 8 fragment reassembly tests covering: unfragmented, 2/3-fragment, out-of-order, multiple messages, missing fragments, duplicates, large messages
• 2537 tests passing

Fixed #

• DTLS close race condition - Fixed "Cannot add to a closed controller" error when browser disconnects during async processing. Added double-check guard (!isClosed && !errorController.isClosed) before all errorController.add() calls in socket.dart, server.dart, and client.dart.

Tests #

• Added regression test: "close during async processing does not throw"
• 2434 tests passing

Fixed #

• Return audio for bundlePolicy:disable - Fixed three issues preventing return audio from reaching Ring cameras:
  • MID migration bug: _mediaTransports map wasn't updated when Ring's SDP answer migrates MIDs
  • Closure capture bug: onSendRtp/onSendRtcp callbacks captured mid at creation time instead of looking up dynamically
  • Codec payload type bug: Sender's codec.payloadType wasn't updated from SDP answer, causing PT mismatch
• ICE connectivity for bundlePolicy:disable - Improved candidate pair priority for separate audio/video transports

Added #

• debugLabel for IceToDtlsAdapter - Helps trace which transport receives DTLS packets during debugging
• Public export for nonstandard MediaStreamTrack - Enables custom track implementations

Tests #

• 2433 tests passing (up from 2431)
• All browser interop tests passing on Chrome, Firefox, Safari

Fixed #

• Firefox browser test - Added STUN server configuration for ICE connectivity
• Safari save_to_disk tests - Added synthetic audio (Web Audio API) and video (canvas) for headless testing
• Safari sendrecv_answer test - Added frame counting fallback for WebKit headless

Changed #

• Browser test infrastructure - Added 1s delay between tests for reliable port cleanup
• H264 test - Skip Firefox (H264 encoding not supported in headless mode)
• VP9 test - Skip Safari (VP9 codec not supported by browser)

Tests #

• 22/22 browser interop tests passing on Chrome, Firefox, and Safari
• Full browser parity achieved across all three major browsers

Added #

• 30 example files matching werift patterns - Complete parity with werift TypeScript examples:
  • save_to_disk/: vp8, vp9, h264, opus, av1x, rtp, pipeline (all use WebSocket + MediaRecorder)
  • mediachannel/simulcast/: offer, answer, multiple, multiple_answer, twcc
  • mediachannel/sendonly/offer.dart - GStreamer + WebSocket pattern
  • mediachannel/sendrecv/offer.dart - Header extensions + echo pattern

Fixed #

• All analyzer warnings resolved - dart analyze reports "No issues found!"
  • Migrated deprecated addTransceiverWithTrack to polymorphic addTransceiver API
  • Fixed conditional assignment style (??= operator)
  • Fixed dead code and unused variables in tests
  • Fixed catchError return types in transaction tests

Changed #

• Deprecated addTransceiverWithTrack - Use addTransceiver(track, direction: ...) instead
  • Matches werift's polymorphic API pattern
  • Updated all examples, interop servers, and tests

Tests #

• 2431 tests passing (up from 2262)
• 22/22 Chrome browser interop tests passing
• All examples verified against werift equivalents

Fixed #

• SCTP RFC 4960 padding - Chunks must be padded to 4-byte boundaries; fixes DataChannel failures with certain label lengths
• Analyzer warnings - Removed unused fields, imports, and dead null-aware expressions across interop tests

Added #

• waitForReady() API - Wait for PeerConnection async initialization before createDataChannel
• createAnswer extmap support - Answer SDP copies header extension mappings from offer (critical for browser RTP parsing)
• createAnswer rtcp-fb support - Answer SDP copies RTCP feedback attributes from offer (NACK, PLI, transport-cc)
• Transceiver direction matching - Creates transceivers with matching direction when remote is sendrecv
• Header extension ID extraction - Sets mid/abs-send-time/transport-cc extension IDs on sender from remote SDP
• Comprehensive browser interop tests - Playwright test suite for Chrome, Firefox, Safari

Changed #

• Improved RTP session handling for answerer pattern
• Enhanced header extension regeneration for RTP forwarding

Tests #

• 2262 tests passing (up from previous release)
• Browser interop: DataChannel, media sendonly/recvonly/sendrecv, save-to-disk, simulcast, TWCC
• All major browsers verified: Chrome (full), Safari (full), Firefox (with browser-as-offerer pattern)

Added #

• Configurable logging via Dart logging package:

  • WebRtcLogging class with hierarchical loggers per component (ICE, DTLS, SCTP, RTP, etc.)
  • WebRtcLogging.enable() / WebRtcLogging.disable() for global control
  • Selective logging: WebRtcLogging.ice.level = Level.FINE
  • Backward compatible: deprecated webrtcDebug flag still works

• Ring camera example (example/ring/):

  • Video streaming server connecting to Ring cameras via WebRTC
  • Forwards video to browser clients
  • Documentation for setup and audio/video handling

• SRTP-CTR cipher support (AES_CM_128_HMAC_SHA1_80):

  • Required for Ring camera compatibility
  • Refactored SRTP key derivation for both GCM and CTR modes
  • Fixed SRTCP index handling and authentication

• API enhancements:

  • Candidate.copyWith() for trickle ICE with sdpMLineIndex/sdpMid
  • RtpTransceiver codec preference support
  • MediaStreamTrack.clone() method

Changed #

• Migrated 284 debug call sites from custom debugLog() to standard logging
• Improved SRTP cipher architecture with separate CTR and GCM implementations

Tests #

• SRTP CTR cipher tests (542 lines)
• SRTP GCM cipher tests (541 lines)
• SRTP RFC 7714 test vectors (476 lines)
• Server handshake tests (406 lines)
• Extended peer connection tests
• Total: 2262 tests passing

Added #

• Expanded public API exports in webrtc_dart.dart:
  • Configuration types: PeerConnectionState, SignalingState, IceConnectionState, IceGatheringState, RtcConfiguration, IceServer, IceTransportPolicy, RtcOfferOptions
  • Media parameters: Complete RTP parameters API (RTCRtpParameters, RTCRtpCodecParameters, RTCRtpEncodingParameters, etc.)
  • RTCP feedback: REMB (src/rtcp/psfb/remb.dart) and TWCC (src/rtcp/rtpfb/twcc.dart)
  • RTP extensions: Header extension handling (src/rtp/header_extension.dart)
  • RTCP packet types (src/srtp/rtcp_packet.dart)
  • Transport layer: IntegratedTransport, DtlsTransport, SctpAssociation
  • STUN protocol: Message and attribute handling for advanced use cases
  • Binary utilities: random16, random32, bufferXor, bufferArrayXor

Changed #

• Core API exports now use explicit show clause for better API documentation
• Improved package API completeness to match werift-webrtc structure

Added #

• Test coverage improvements: 2171 tests, 80% code coverage
• New test files for DTLS, stats, and media components:
  • DTLS handshake message tests (finished, alert, random, client_key_exchange)
  • Extended master secret extension tests
  • Transport and certificate stats tests
  • Media parameters tests (RTCRtpEncodingParameters, RTCRtpSendParameters)
  • Processor interface tests (CallbackProcessor, AVProcessor mixin)

Changed #

• Updated README with accurate test count and coverage metrics

Changed #

• Upgrade pointycastle from 3.9.1 to 4.0.0
• Apply dart format to all source files
• Add example/example.md for pub.flutter-io.cn Example tab
• Add quickstart examples matching README inline code

Fixed #

• Remove unnecessary casts for pointycastle 4.0.0 compatibility
• Fix curly brace style in certificate_request.dart

Initial release - complete Dart port of werift-webrtc v0.22.2.

Features #

Core Protocols

• STUN message encoding/decoding with MESSAGE-INTEGRITY and FINGERPRINT
• ICE candidate model (host, srflx, relay, prflx)
• ICE checklists, connectivity checks, nomination
• ICE TCP candidates and mDNS obfuscation
• ICE restart support
• DTLS 1.2 handshake (client and server) with certificate authentication
• SRTP/SRTCP encryption (AES-GCM)
• RTP/RTCP stack (SR, RR, SDES, BYE)
• SCTP association over DTLS
• DataChannel protocol (reliable/unreliable, ordered/unordered)
• SDP parsing and generation

Video Codec Depacketization

• VP8 depacketization
• VP9 depacketization with SVC support
• H.264 depacketization with FU-A/STAP-A
• AV1 depacketization with OBU parsing

RTCP Feedback

• NACK (Generic Negative Acknowledgement)
• PLI (Picture Loss Indication)
• FIR (Full Intra Request)
• REMB (Receiver Estimated Max Bitrate)

Retransmission

• RTX packet wrapping/unwrapping
• RetransmissionBuffer (128-packet circular buffer)
• RTX SDP negotiation

TURN Support

• TURN allocation with 401 authentication (RFC 5766)
• Channel binding (0x4000-0x7FFF)
• Permission management
• Send/Data indications
• ICE integration with relay candidates

TWCC (Transport-Wide Congestion Control)

• Transport-wide sequence numbers (RTP header extension)
• Receive delta encoding/decoding
• Packet status chunks
• Bandwidth estimation algorithm

Simulcast

• RID (Restriction Identifier) support (RFC 8851)
• RTP header extension parsing for RID/MID
• SDP simulcast attribute parsing
• RtpRouter for RID-based packet routing

Quality Features

• Jitter buffer with configurable sizing
• RED (Redundancy Encoding) for audio (RFC 2198)
• Media Track Management (addTrack, removeTrack, replaceTrack)
• Extended getStats() API (ICE, transport, data channel stats)

Media Recording

• WebM container support
• MP4 container support (fMP4)
• EBML encoding/decoding

Browser Compatibility

• Chrome: Tested and working
• Firefox: Tested and working
• Safari: Tested and working

Test Coverage #

1658 tests covering all implemented components.

Acknowledgments #

This is a Dart port of werift-webrtc by Yuki Shindo.