🚨 Important Notice: v0.17.0 Release and Future Direction

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.

Why this change?

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.

What's Changed

• 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

New Contributors

• @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