Release packages

[knope-bot]

Important

Merging this pull request will create these releases

libwebrtc 0.3.46 (2026-08-24)

Features

feat: enable WARP (SPED + SNAP) by default, gated by the server

WARP is now always enabled on the client and negotiated with the SFU: SPED
(DTLS-in-STUN) via the WebRTC-IceHandshakeDtls field trial, and SNAP
(SCTP-INIT-in-SDP) via the RtcConfiguration.enable_sctp_snap field. When the
server does not enable WARP it is not advertised and the connection falls back to
plain DTLS/SCTP, so there is no client-side toggle.

BREAKING CHANGE: libwebrtc::RtcConfiguration is now #[non_exhaustive] and has a
new enable_sctp_snap field. Construct it from RtcConfiguration::default() and set
the fields you need instead of a struct literal.

Fixes

• Automatically retry webrtc build downloads
• feat: upgrade libwebrtc to m150. - feat: upgrade libwebrtc to m150. #1284 (@cloudwebrtc)

fix: bump libwebrtc to webrtc-b9233c3-2 so TURN/TLS can use the OS trust store

WebRTC validates TURN/TLS against a small set of anchors compiled into
rtc_base/ssl_roots.h, generated in 2023 from Google's own PKI list. It has no
Amazon, Starfield Services or ISRG roots, so a relay-only connection to a TURN
server fronted by AWS ACM or Let's Encrypt times out with unknown_ca even
though the host OS trusts the chain.

This build picks up webrtc-sdk/webrtc#277, which falls back to the operating
system's trust store when the built-in anchors yield no path. It sits in
rtc_base below every SDK, so the C++ API that webrtc-sys binds is covered
without any Rust-side change.

webrtc-sys 0.3.43 (2026-08-24)

Fixes

• Automatically retry webrtc build downloads
• feat: upgrade libwebrtc to m150. - feat: upgrade libwebrtc to m150. #1284 (@cloudwebrtc)

fix: bump libwebrtc to webrtc-b9233c3-2 so TURN/TLS can use the OS trust store

WebRTC validates TURN/TLS against a small set of anchors compiled into
rtc_base/ssl_roots.h, generated in 2023 from Google's own PKI list. It has no
Amazon, Starfield Services or ISRG roots, so a relay-only connection to a TURN
server fronted by AWS ACM or Let's Encrypt times out with unknown_ca even
though the host OS trusts the chain.

This build picks up webrtc-sdk/webrtc#277, which falls back to the operating
system's trust store when the built-in anchors yield no path. It sits in
rtc_base below every SDK, so the C++ API that webrtc-sys binds is covered
without any Rust-side change.

feat: enable WARP (SPED + SNAP) by default, gated by the server

WARP is now always enabled on the client and negotiated with the SFU: SPED
(DTLS-in-STUN) via the WebRTC-IceHandshakeDtls field trial, and SNAP
(SCTP-INIT-in-SDP) via the RtcConfiguration.enable_sctp_snap field. When the
server does not enable WARP it is not advertised and the connection falls back to
plain DTLS/SCTP, so there is no client-side toggle.

BREAKING CHANGE: libwebrtc::RtcConfiguration is now #[non_exhaustive] and has a
new enable_sctp_snap field. Construct it from RtcConfiguration::default() and set
the fields you need instead of a struct literal.

webrtc-sys-build 0.3.19 (2026-08-24)

Fixes

• Automatically retry webrtc build downloads
• feat: upgrade libwebrtc to m150. - feat: upgrade libwebrtc to m150. #1284 (@cloudwebrtc)

fix: bump libwebrtc to webrtc-b9233c3-2 so TURN/TLS can use the OS trust store

WebRTC validates TURN/TLS against a small set of anchors compiled into
rtc_base/ssl_roots.h, generated in 2023 from Google's own PKI list. It has no
Amazon, Starfield Services or ISRG roots, so a relay-only connection to a TURN
server fronted by AWS ACM or Let's Encrypt times out with unknown_ca even
though the host OS trusts the chain.

This build picks up webrtc-sdk/webrtc#277, which falls back to the operating
system's trust store when the built-in anchors yield no path. It sits in
rtc_base below every SDK, so the C++ API that webrtc-sys binds is covered
without any Rust-side change.

livekit-ffi 0.12.76 (2026-08-24)

Fixes

• differentiate signal connection errors correctly from timeouts - Add e2e signalling tests #1234 (@lukasIO)
• Automatically retry webrtc build downloads
• feat: upgrade libwebrtc to m150. - feat: upgrade libwebrtc to m150. #1284 (@cloudwebrtc)
• fix(uniffi): register the Bytes custom type once, in livekit-common - fix(uniffi): register the Bytes custom type once, in livekit-common #1343 (@pblazej)

fix: bump libwebrtc to webrtc-b9233c3-2 so TURN/TLS can use the OS trust store

WebRTC validates TURN/TLS against a small set of anchors compiled into
rtc_base/ssl_roots.h, generated in 2023 from Google's own PKI list. It has no
Amazon, Starfield Services or ISRG roots, so a relay-only connection to a TURN
server fronted by AWS ACM or Let's Encrypt times out with unknown_ca even
though the host OS trusts the chain.

This build picks up webrtc-sdk/webrtc#277, which falls back to the operating
system's trust store when the built-in anchors yield no path. It sits in
rtc_base below every SDK, so the C++ API that webrtc-sys binds is covered
without any Rust-side change.

feat: enable WARP (SPED + SNAP) by default, gated by the server

WARP is now always enabled on the client and negotiated with the SFU: SPED
(DTLS-in-STUN) via the WebRTC-IceHandshakeDtls field trial, and SNAP
(SCTP-INIT-in-SDP) via the RtcConfiguration.enable_sctp_snap field. When the
server does not enable WARP it is not advertised and the connection falls back to
plain DTLS/SCTP, so there is no client-side toggle.

BREAKING CHANGE: libwebrtc::RtcConfiguration is now #[non_exhaustive] and has a
new enable_sctp_snap field. Construct it from RtcConfiguration::default() and set
the fields you need instead of a struct literal.

livekit-uniffi 0.1.9 (2026-08-24)

Fixes

• differentiate signal connection errors correctly from timeouts - Add e2e signalling tests #1234 (@lukasIO)
• Expose RemoteDataTrack.set_pipeline_options and RemoteDataTrackPipelineOptions (max_partial_frames) over UniFFI, matching the JS and Rust SDKs
• Register the Bytes UniFFI custom type once in livekit-common and borrow it from each component with uniffi::use_remote_type!, so the converter is emitted once and the post-generation Swift workaround is no longer needed

livekit-common 0.1.2 (2026-08-24)

Fixes

• Register the Bytes UniFFI custom type once in livekit-common and borrow it from each component with uniffi::use_remote_type!, so the converter is emitted once and the post-generation Swift workaround is no longer needed

livekit-data-stream 0.1.3 (2026-08-24)

Fixes

• fix(uniffi): register the Bytes custom type once, in livekit-common - fix(uniffi): register the Bytes custom type once, in livekit-common #1343 (@pblazej)

livekit-token-source 0.1.2 (2026-08-24)

Features

• Adds optional caching version of the token sources

livekit-datatrack 0.1.14 (2026-08-24)

Fixes

• Register the Bytes UniFFI custom type once in livekit-common and borrow it from each component with uniffi::use_remote_type!, so the converter is emitted once and the post-generation Swift workaround is no longer needed

livekit 0.8.4 (2026-08-24)

Features

feat: enable WARP (SPED + SNAP) by default, gated by the server

WARP is now always enabled on the client and negotiated with the SFU: SPED
(DTLS-in-STUN) via the WebRTC-IceHandshakeDtls field trial, and SNAP
(SCTP-INIT-in-SDP) via the RtcConfiguration.enable_sctp_snap field. When the
server does not enable WARP it is not advertised and the connection falls back to
plain DTLS/SCTP, so there is no client-side toggle.

BREAKING CHANGE: libwebrtc::RtcConfiguration is now #[non_exhaustive] and has a
new enable_sctp_snap field. Construct it from RtcConfiguration::default() and set
the fields you need instead of a struct literal.

Fixes

• differentiate signal connection errors correctly from timeouts - Add e2e signalling tests #1234 (@lukasIO)
• Automatically retry webrtc build downloads
• feat: upgrade libwebrtc to m150. - feat: upgrade libwebrtc to m150. #1284 (@cloudwebrtc)
• fix(uniffi): register the Bytes custom type once, in livekit-common - fix(uniffi): register the Bytes custom type once, in livekit-common #1343 (@pblazej)

fix: bump libwebrtc to webrtc-b9233c3-2 so TURN/TLS can use the OS trust store

WebRTC validates TURN/TLS against a small set of anchors compiled into
rtc_base/ssl_roots.h, generated in 2023 from Google's own PKI list. It has no
Amazon, Starfield Services or ISRG roots, so a relay-only connection to a TURN
server fronted by AWS ACM or Let's Encrypt times out with unknown_ca even
though the host OS trusts the chain.

This build picks up webrtc-sdk/webrtc#277, which falls back to the operating
system's trust store when the built-in anchors yield no path. It sits in
rtc_base below every SDK, so the C++ API that webrtc-sys binds is covered
without any Rust-side change.

livekit-api 0.6.4 (2026-08-24)

Fixes

• differentiate signal connection errors correctly from timeouts - Add e2e signalling tests #1234 (@lukasIO)
• fix(uniffi): register the Bytes custom type once, in livekit-common - fix(uniffi): register the Bytes custom type once, in livekit-common #1343 (@pblazej)

[devin-ai-integration]

✅ Devin Review: No Issues Found

Devin Review analyzed this PR and found no bugs or issues to report.