Full Changelog: https://github.com/webrtc-rs/webrtc/compare/v0.17.0...v0.20.0-alpha.1

Full Changelog: https://github.com/webrtc-rs/webrtc/compare/v0.17.0...v0.17.1

v0.17.x is the final feature release of the Tokio-coupled async WebRTC implementation.

• v0.17.x branch: A dedicated branch will be created for v0.17.x that will receive bug fixes only (no new features).
• Master branch: Will transition to a new Sans-IO based architecture built on top of webrtc-rs/rtc.

The project is shifting toward a Sans-IO WebRTC implementation that decouples the protocol logic from any specific async runtime. This new architecture will:

• ✅ Support multiple async runtimes (Tokio, smol, async-std, etc.)
• ✅ Provide a clean, protocol-centric Sans-IO core via webrtc-rs/rtc
• ✅ Enable a truly runtime-agnostic, async-friendly WebRTC implementation in Rust

If you need Tokio-specific stability, please use the v0.17.x branch. If you want to adopt the new runtime-agnostic approach, follow development on the master branch.

• Add support for multi codec negotiation by @rainliu in https://github.com/webrtc-rs/webrtc/pull/741
• tokio-test dev-dependencies is used for only webrtc-util by @Razzwan in https://github.com/webrtc-rs/webrtc/pull/742
• chore: fix a typo "dail"->"dial" by @link2xt in https://github.com/webrtc-rs/webrtc/pull/744
• Remove Seek requirement for some writers by @Yesterday17 in https://github.com/webrtc-rs/webrtc/pull/743
• Perf: Changed RR and SR ticker behavior to Skip to avoid overdue reports and catchup bursts by @Havunen in https://github.com/webrtc-rs/webrtc/pull/745
• PR Submission for Issue #749 by @Wandalen in https://github.com/webrtc-rs/webrtc/pull/750
• Make psk callback async-capable by @mdelete in https://github.com/webrtc-rs/webrtc/pull/751
• Add markdown files for basic explanation on WebRTC by @amitnos123 in https://github.com/webrtc-rs/webrtc/pull/756
• Document feature flags in lib.rs by @amitnos123 in https://github.com/webrtc-rs/webrtc/pull/759
• Certain enhancements, from my experience I wish I knew sooner by @ris-work in https://github.com/webrtc-rs/webrtc/pull/760
• feat(srtp): add AES CM 256 crypto profiles by @RRRadicalEdward in https://github.com/webrtc-rs/webrtc/pull/764
• refactor(dtls): replace bincode serialization crate with rkyv by @MikeRomaniuk in https://github.com/webrtc-rs/webrtc/pull/767
• feat: Add H.264 High Profile codec support by @sachaarbonel in https://github.com/webrtc-rs/webrtc/pull/768
• Accept unknown bandwidth types in SDP parser per RFC 8866 by @alps3325 in https://github.com/webrtc-rs/webrtc/pull/770

• @Razzwan made their first contribution in https://github.com/webrtc-rs/webrtc/pull/742
• @link2xt made their first contribution in https://github.com/webrtc-rs/webrtc/pull/744
• @Yesterday17 made their first contribution in https://github.com/webrtc-rs/webrtc/pull/743
• @Havunen made their first contribution in https://github.com/webrtc-rs/webrtc/pull/745
• @Wandalen made their first contribution in https://github.com/webrtc-rs/webrtc/pull/750
• @mdelete made their first contribution in https://github.com/webrtc-rs/webrtc/pull/751
• @amitnos123 made their first contribution in https://github.com/webrtc-rs/webrtc/pull/756
• @RRRadicalEdward made their first contribution in https://github.com/webrtc-rs/webrtc/pull/764
• @MikeRomaniuk made their first contribution in https://github.com/webrtc-rs/webrtc/pull/767
• @sachaarbonel made their first contribution in https://github.com/webrtc-rs/webrtc/pull/768
• @alps3325 made their first contribution in https://github.com/webrtc-rs/webrtc/pull/770

Full Changelog: https://github.com/webrtc-rs/webrtc/compare/v0.14.0...v0.17.0

• Fix handling of AUD NAL units with size 2 by @Kleptine in https://github.com/webrtc-rs/webrtc/pull/670
• fix panics when parsing rtcp packets with invalid block sizes by @TimeToogo in https://github.com/webrtc-rs/webrtc/pull/678
• Implement ChaCha20_Poly1305 cipher suite by @wdouglass in https://github.com/webrtc-rs/webrtc/pull/675
• Make SRTP AES_256_GCM actually work by @TimeToogo in https://github.com/webrtc-rs/webrtc/pull/677
• Fix deadlock in data-channels-flow-control example. by @recap in https://github.com/webrtc-rs/webrtc/pull/679
• Handle replay after wrapping in replaydetector by @anders-avos in https://github.com/webrtc-rs/webrtc/pull/681
• Revert "Fix deadlock in data-channels-flow-control example. " by @rainliu in https://github.com/webrtc-rs/webrtc/pull/682
• stun: simplify MappedAddress Display implement by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/683
• stun: simplify stop_with_error by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/684
• stun: simplify UnknownAttributes Display by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/685
• turn: client binding direct use Option value by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/686
• turn: use ok_or() to simplify get_allocation() call by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/687
• turn: use ok_or() to simplify command_tx by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/689
• turn: introduced DEFAULT_MAX_RETRIES by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/692
• mdns: simplify resource pack function by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/693
• turn: simplify result_ch_rx by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/691
• dtls: refactor cipher suite error handling with ok_or() by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/694
• style: add blank lines between methods in DTLSConn impl by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/695
• refactor: remove unnecessary Cow::Borrowed wrapper in URL query parsing by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/696
• refactor(ice): replace magic numbers with named constants for STUN/TURN ports by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/697
• stun: use format! macro instead of string concatenation in Uri display by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/698
• ice/examples: improve error handling and formatting in ping_pong example by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/699
• refactor: apply clippy fixes across workspace by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/703
• mdns: allow reuse_port on non-Unix platforms when feature is enabled by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/702
• update dependencies rand by @a-wing in https://github.com/webrtc-rs/webrtc/pull/707
• util: Simplify timeout error handling in Buffer by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/708
• interceptor: Use futures::future::join_all in stats interceptor by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/710
• dtls: Add packet length validation in CBC decryption by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/711
• stun: simplify URI parsing logic by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/712
• do not used fixed sctp-ports 5000, but consult the negotiated ports by @KillingSpark in https://github.com/webrtc-rs/webrtc/pull/706
• interceptor: simplify NACK generator close_rx error handling by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/714
• sctp: Make most chunk handling errors non-fatal by @anders-avos in https://github.com/webrtc-rs/webrtc/pull/716
• Add function to export key data from DTLS connection by @TimeToogo in https://github.com/webrtc-rs/webrtc/pull/718
• [ice] Fix unhandled overflow panic in listen_udp_in_port_range by @lucamuscat in https://github.com/webrtc-rs/webrtc/pull/719
• fmtp: add Unicode case-folding support for parameter compariso by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/721
• Make use of the max-message-size SDP attribute in datachannels by @lucamuscat in https://github.com/webrtc-rs/webrtc/pull/722
• stun: simplify XorMappedAddress Display implementation by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/725
• Updated logo background by @mvel96 in https://github.com/webrtc-rs/webrtc/pull/730
• stun: simplify connection retrieval by @cppcoffee in https://github.com/webrtc-rs/webrtc/pull/731
• Require rand version 0.9.1 or greater, as rand::distr::Alphabetic does not exist in 0.9.0, which is currently allowed by @lucamuscat in https://github.com/webrtc-rs/webrtc/pull/732
• Remove unused dependencies by @lucamuscat in https://github.com/webrtc-rs/webrtc/pull/733
• sample_builder: fix panic on timestamp wrap by @ArtemMartus in https://github.com/webrtc-rs/webrtc/pull/735

• @Kleptine made their first contribution in https://github.com/webrtc-rs/webrtc/pull/670
• @TimeToogo made their first contribution in https://github.com/webrtc-rs/webrtc/pull/678
• @wdouglass made their first contribution in https://github.com/webrtc-rs/webrtc/pull/675
• @cppcoffee made their first contribution in https://github.com/webrtc-rs/webrtc/pull/683
• @lucamuscat made their first contribution in https://github.com/webrtc-rs/webrtc/pull/719
• @ArtemMartus made their first contribution in https://github.com/webrtc-rs/webrtc/pull/735

Full Changelog: https://github.com/webrtc-rs/webrtc/compare/v0.13.0...v0.14.0

• revert the support of Aes128CmHmacSha1_32 and AeadAes256Gcm
• peer_connection: Only warn about unhandled incmoing RTP track by @haaspors in https://github.com/webrtc-rs/webrtc/pull/642
• Remove unused ErrTurnCredentials as it is no longer used. It was used… by @ris-work in https://github.com/webrtc-rs/webrtc/pull/644
• ice: Fix USE-CANDIDATE in controlled agent by @anders-avos in https://github.com/webrtc-rs/webrtc/pull/647
• track_local: Get rid of some unnecessary Mutexes and Options by @haaspors in https://github.com/webrtc-rs/webrtc/pull/646
• stun client: remove unnecessary mut by @petar-dambovaliev in https://github.com/webrtc-rs/webrtc/pull/652
• peer_connection: Make sure all packets are read through interceptor by @haaspors in https://github.com/webrtc-rs/webrtc/pull/648
• Use any available curve for dtls connection by @Mierunski in https://github.com/webrtc-rs/webrtc/pull/654
• docs: update Stream logo with latest brand asset by @tyaga001 in https://github.com/webrtc-rs/webrtc/pull/655
• updating the throughput calculation in data_channel_flow_control by @recap in https://github.com/webrtc-rs/webrtc/pull/659
• Fix: Invoke onClose handler upon receiving a notification by @harry0349 in https://github.com/webrtc-rs/webrtc/pull/666
• Change MIME_TYPE_HEVC from video/HEVC to video/H265 (#141) by @bzld-acn in https://github.com/webrtc-rs/webrtc/pull/668

• @ris-work made their first contribution in https://github.com/webrtc-rs/webrtc/pull/644
• @petar-dambovaliev made their first contribution in https://github.com/webrtc-rs/webrtc/pull/652
• @Mierunski made their first contribution in https://github.com/webrtc-rs/webrtc/pull/654
• @tyaga001 made their first contribution in https://github.com/webrtc-rs/webrtc/pull/655
• @recap made their first contribution in https://github.com/webrtc-rs/webrtc/pull/659
• @harry0349 made their first contribution in https://github.com/webrtc-rs/webrtc/pull/666
• @bzld-acn made their first contribution in https://github.com/webrtc-rs/webrtc/pull/668

Full Changelog: https://github.com/webrtc-rs/webrtc/compare/v0.12.0...v0.13.0