webrtc-sctp

What's Changed

• ice: Release TURN allocation even if candidate was already closed by @marcin-p-porebski in #799
• fix(ice): always send STUN consent checks (RFC 7675) by @brady-pplx in #803
• util: don't tear down UDP listener read_loop on WSAECONNRESET (Windows) by @rwtd in #812
• Release mDNS resources when connections close by @mmalmi in #818
• sctp: lower INITIAL_MTU to 1191 and expose the MTU in association::Config by @phact in #807

New Contributors

• @brady-pplx made their first contribution in #803
• @rwtd made their first contribution in #812
• @mmalmi made their first contribution in #818
• @phact made their first contribution in #807

Full Changelog: v0.17.1...v0.17.2

• Fix 'attempt to add with overflow' panic in dev profile #393
• Limit the bytes in the PendingQueue to avoid packets accumulating there uncontrollably #367.
• Improve algorithm used to push to pending queue from O(n*log(n)) to O(log(n)) #365.
• Reuse as many allocations as possible when marshaling #364.
• The lock for the internal association was contended badly because marshaling was done while still in a critical section and also tokio was scheduling tasks badly #363.

Breaking

• Make sctp::Stream::write & sctp::Stream::write_sctp async again #367.

• Increased minimum support rust version to 1.60.0.
• Do not loose data in PollStream::poll_write #341.
• PollStream::poll_shutdown: make sure to flush any writes before shutting down #340.
• Fixed a possible bug when adding chunks to pending queue #345.
• Increased required webrtc-util version to 0.7.0.

Breaking changes

• Make Stream::on_buffered_amount_low function non-async #338.
• Make sctp::Stream::write & sctp::Stream::write_sctp sync #344.

• Increased min version of log dependency to 0.4.16. #250 Fix log at ^0.4.16 to make tests compile by @k0nserv.
• #245 Fix incorrect chunk type Display for CWR by @k0nserv.

Changes

Breaking changes

• The serialized format for RTCIceCandidateInit has changed to match what the specification i.e. keys are camelCase. #153 Make RTCIceCandidateInit conform to WebRTC spec contributed by jmatss.
• Improved robustness when proposing RTP extension IDs and handling of collisions in these. This change is only breaking if you have assumed anything about the nature of these extension IDs. #154 Fix RTP extension id collision contributed by k0nserv
• Transceivers will now not stop when either or both directions are disabled. That is, applying and SDP with a=inactive will not stop the transceiver, instead attached senders and receivers will pause. A transceiver can be resurrected by setting direction back to e.g. a=sendrecv. The desired direction can be controlled with the newly introduced public method set_direction on RTCRtpTransceiver.
  • #201 Handle inactive transceivers more correctly contributed by k0nserv
  • #210 Rework transceiver direction support further contributed by k0nserv
  • #214 set_direction add missing Send + Sync bound contributed by algesten
  • #213 set_direction add missing Sync bound contributed by algesten
  • #212 Public RTCRtpTransceiver::set_direction contributed by algesten
  • #268 Fix current direction update when applying answer contributed by k0nserv
  • #236 Pause RTP writing if direction indicates it contributed by algesten
• Generated the a=msid line for m= line sections according to the specification. This might be break remote peers that relied on the previous, incorrect, behaviour. This also fixes a bug where an endless negotiation loop could happen. #217 Correct msid handling for RtpSender contributed by k0nserv
• Improve data channel id negotiation. We've slightly adjust the public interface for creating pre-negotiated data channels. Instead of a separate negotiated: Option<bool> and id: Option<u16> in RTCDataChannelInit there's now a more idiomatic negotiated: Option<u16>. If you have a pre-negotiated data channel simply set negotiated: Some(id) when creating the data channel.

  • #237 Fix datachannel id setting for 0.5.0 release contributed by stuqdog

  • #229 Revert "base id updating on whether it's been negotiated, not on its … contributed by melekes

  • #226 base id updating on whether it's been finalized, not on its value contributed by stuqdog

Other improvememnts

We made various improvements and fixes since 0.4.0, including merging all subcrates into a single git repo. The old crate repos are archived and all development will now happen in https://github.com/webrtc-rs/webrtc/.

• We now provide stats reporting via the standardized RTCPeerConnection::get_stats method.

  • #277 Implement Remote Inbound Stats contributed by k0nserv
  • #220 Make stats types pub so they can be used directly contributed by k0nserv
  • #225 Add RTP Stats to stats report contributed by k0nserv
  • #189 Serialize stats contributed by sax
  • #180 Get stats from peer connection contributed by sax

• #278 Fix async-global-executor contributed by k0nserv

• #276 relax regex version requirement contributed by melekes

• #244 Update README.md instructions after monorepo merge contributed by k0nserv

• #241 move profile to workspace contributed by xnorpx

• #240 Increase timeout to "fix" test breaking contributed by algesten

• #239 One repo (again) contributed by algesten

• #234 Fix recent clippy lints contributed by k0nserv

• #224 update call to DataChannel::accept as per data pr #14 contributed by melekes

• #223 dtls_transport: always set remote certificate contributed by melekes

• #216 Lower case mime types for comparison in fmpt lines contributed by k0nserv

• #211 Helper to trigger negotiation_needed contributed by algesten

• #209 MID generator feature contributed by algesten

• #208 update deps + loosen some requirements contributed by melekes

• #205 data_channel: handle stream EOF contributed by melekes

• #204 [peer_connection] allow persistent certificates contributed by melekes

• #202 bugfix-Udp connection not close (reopen #174) #195 contributed by shiqifeng2000

• #199 Upgrade ICE to 0.7.0 contributed by k0nserv

• #194 Add AV1 MimeType and RtpCodecParameters contributed by billylindeman

• #188 Improve operations debuggability contributed by k0nserv

• #187 Fix SDP for rejected tracks to conform to RFC contributed by k0nserv

• #185 Adding some debug and display traits contributed by sevensidedmarble

• #179 Fix example names in README contributed by ethagnawl

• #176 Time overflow armv7 workaround contributed by frjol

• #171 close DTLS conn upon err contributed by melekes

• #170 always start sctp contributed by melekes

• #167 Add offer/answer/pranswer constructors for RTCSessionDescription contributed by sax

Subcrate updates

The various sub-crates have been updated as follows:

• util: 0.5.3 => 0.6.0
• sdp: 0.5.1 => 0.5.2
• mdns: 0.4.2 => 0.5.0
• stun: 0.4.2 => 0.4.3
• turn: 0.5.3 => 0.6.0
• ice: 0.6.4 => 0.8.0
• dtls: 0.5.2 => 0.6.0
• rtcp: 0.6.5 => 0.7.0
• rtp: 0.6.5 => 0.6.7
• srtp: 0.8.9 => 0.9.0
• scpt: 0.4.3 => 0.6.1
• data: 0.3.3 => 0.5.0
• interceptor: 0.7.6 => 0.8.0
• media: 0.4.5 => 0.4.7

Their respective change logs are found in the old, now archived, repositories and within their respective CHANGELOG.md files in the monorepo.

Contributors

A big thanks to all the contributors that have made this release happen:

• morajabi
• sax
• ethagnawl
• xnorpx
• frjol
• algesten
• shiqifeng2000
• billylindeman
• sevensidedmarble
• k0nserv
• stuqdog
• neonphog
• melekes
• jmatss