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.dev 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.