# Changelog
All notable changes to this project will be documented in this file.
The format is based on [Keep a Changelog](https://keepachangelog.com/en/1.0.0/),
and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0.html).
## [Unreleased]
## [0.7.32](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.31...rust-sdks/livekit@0.7.32) - 2026-02-16
### Fixed
- fix full_reconnect downgrade & don't ignore Leave messages ([#893](https://github.com/livekit/rust-sdks/pull/893))
### Other
- turn single peerconnection off by default ([#897](https://github.com/livekit/rust-sdks/pull/897))
- ensure signal connections times out properly and retries ([#895](https://github.com/livekit/rust-sdks/pull/895))
- added Single Peer Connection support to Rust ([#888](https://github.com/livekit/rust-sdks/pull/888))
- set the simulcast codec & layers ([#891](https://github.com/livekit/rust-sdks/pull/891))
## [0.7.31](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.30...rust-sdks/livekit@0.7.31) - 2026-02-10
### Other
- don't use clamp as the ultimate_kbps can be lower than 300 ([#886](https://github.com/livekit/rust-sdks/pull/886))
- pre-connect the publisher PC when an RPC handler is registered ([#880](https://github.com/livekit/rust-sdks/pull/880))
## [0.7.30](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.29...rust-sdks/livekit@0.7.30) - 2026-02-09
### Fixed
- fix the RPC race condition ([#865](https://github.com/livekit/rust-sdks/pull/865))
### Other
- update proto & fix CI ([#871](https://github.com/livekit/rust-sdks/pull/871))
- Use workspace dependencies & settings ([#856](https://github.com/livekit/rust-sdks/pull/856))
- Upgrade protocol to v1.44.0 ([#857](https://github.com/livekit/rust-sdks/pull/857))
- Expose participant's permission to ffi layer ([#824](https://github.com/livekit/rust-sdks/pull/824))
## [0.7.29](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.28...rust-sdks/livekit@0.7.29) - 2026-01-15
### Fixed
- ensure Room.creation_time is ms ([#822](https://github.com/livekit/rust-sdks/pull/822))
### Other
- try setting x-google-start-bitrate for vp9 ([#820](https://github.com/livekit/rust-sdks/pull/820))
## [0.7.28](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.27...rust-sdks/livekit@0.7.28) - 2025-12-19
### Added
- *(ParticipantInfo)* export kind details ([#813](https://github.com/livekit/rust-sdks/pull/813))
## [0.7.27](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.26...rust-sdks/livekit@0.7.27) - 2025-12-17
### Other
- Handle server initiated mute request ([#812](https://github.com/livekit/rust-sdks/pull/812))
## [0.7.26](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.25...rust-sdks/livekit@0.7.26) - 2025-12-04
### Added
- *(connector)* initial service impl ([#790](https://github.com/livekit/rust-sdks/pull/790))
### Fixed
- fix mute/unmute events for LocalTrack. ([#799](https://github.com/livekit/rust-sdks/pull/799))
### Other
- Add RoomEvent::TokenRefreshed ([#803](https://github.com/livekit/rust-sdks/pull/803))
## [0.7.25](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.24...rust-sdks/livekit@0.7.25) - 2025-11-20
### Other
- perform full reconnect if resume fails ([#792](https://github.com/livekit/rust-sdks/pull/792))
- E2E RPC tests ([#769](https://github.com/livekit/rust-sdks/pull/769))
- Remove unused dependencies ([#761](https://github.com/livekit/rust-sdks/pull/761))
## [0.7.24](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.23...rust-sdks/livekit@0.7.24) - 2025-10-27
### Other
- Expose set video quality ([#759](https://github.com/livekit/rust-sdks/pull/759))
## [0.7.23](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.22...rust-sdks/livekit@0.7.23) - 2025-10-23
### Other
- add h265 codec support ([#762](https://github.com/livekit/rust-sdks/pull/762))
## [0.7.22](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.21...rust-sdks/livekit@0.7.22) - 2025-10-22
### Other
- License check ([#746](https://github.com/livekit/rust-sdks/pull/746))
- Remove participant check for data packets ([#757](https://github.com/livekit/rust-sdks/pull/757))
- clamp connection timeout and fixed the comment ([#748](https://github.com/livekit/rust-sdks/pull/748))
- put examples in root Cargo workspace ([#731](https://github.com/livekit/rust-sdks/pull/731))
## [0.7.21](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.20...rust-sdks/livekit@0.7.21) - 2025-10-13
### Added
- *(e2ee)* add data channel encryption ([#708](https://github.com/livekit/rust-sdks/pull/708))
### Fixed
- fix some potential audio issues, clean up the code a bit, and suppress some warnings ([#737](https://github.com/livekit/rust-sdks/pull/737))
- do not log 'signal client closed: "stream closed"' on disconnect ([#727](https://github.com/livekit/rust-sdks/pull/727))
### Other
- Test participant disconnect ([#732](https://github.com/livekit/rust-sdks/pull/732))
- Increase RPC max RT time to 7s ([#729](https://github.com/livekit/rust-sdks/pull/729))
- E2E audio test ([#724](https://github.com/livekit/rust-sdks/pull/724))
## [0.7.20](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.19...rust-sdks/livekit@0.7.20) - 2025-10-03
### Other
- updated the following local packages: libwebrtc
## [0.7.19](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.18...rust-sdks/livekit@0.7.19) - 2025-09-29
### Fixed
- apply original participant fields in data messages ([#709](https://github.com/livekit/rust-sdks/pull/709))
### Other
- Implement Display and Error for RpcError ([#719](https://github.com/livekit/rust-sdks/pull/719))
- Fix intermittently failing E2E reliability test ([#718](https://github.com/livekit/rust-sdks/pull/718))
- Do not modify raw packets ([#714](https://github.com/livekit/rust-sdks/pull/714))
- Add send_bytes method ([#691](https://github.com/livekit/rust-sdks/pull/691))
- Disable opus red for e2ee enabled clients ([#706](https://github.com/livekit/rust-sdks/pull/706))
- Upgrade protocol to v1.41.0 ([#703](https://github.com/livekit/rust-sdks/pull/703))
## [0.7.18](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.17...rust-sdks/livekit@0.7.18) - 2025-09-09
### Other
- Add option to enable audio preconnect buffer in SDK & FFI ([#700](https://github.com/livekit/rust-sdks/pull/700))
- Data channel reliability ([#688](https://github.com/livekit/rust-sdks/pull/688))
## [0.7.17](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.16...rust-sdks/livekit@0.7.17) - 2025-09-03
### Other
- updated the following local packages: livekit-api, libwebrtc
## [0.7.16](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.15...rust-sdks/livekit@0.7.16) - 2025-07-31
### Other
- updated the following local packages: livekit-api
# Changelog
## [0.7.15](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.14...rust-sdks/livekit@0.7.15) - 2025-07-16
### Other
- remove published tracks when the room is closed ([#677](https://github.com/livekit/rust-sdks/pull/677))
## [0.7.14](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.13...rust-sdks/livekit@0.7.14) - 2025-06-23
### Fixed
- `audio_frame_ms` didn't work expectedly ([#671](https://github.com/livekit/rust-sdks/pull/671))
## [0.7.13](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.12...rust-sdks/livekit@0.7.13) - 2025-06-17
### Other
- Expose room updates, support MoveParticipant (protocol 15) ([#662](https://github.com/livekit/rust-sdks/pull/662))
## [0.7.12](https://github.com/livekit/rust-sdks/compare/rust-sdks/livekit@0.7.11...rust-sdks/livekit@0.7.12) - 2025-06-11
### Fixed
- fix duration overflow ([#654](https://github.com/livekit/rust-sdks/pull/654))
### Other
- Remove debouncer when fast_publish is enabled ([#649](https://github.com/livekit/rust-sdks/pull/649))
## [0.7.9] - 2025-04-08
### Added
- High-level data streams API
### Deprecated
- Low-level data stream packet events
## [0.7.8] - 2025-03-21
### Fixed
- Revert the RPC change, need more robust way
## [0.7.7] - 2025-03-18
### Fixed
- Move RPC handlers to room
## [0.7.6] - 2025-02-28
### Added
- Add audio filter support
## [0.7.5] - 2025-02-06
### Fixed
- Fix a dependency issue with an older version of the libwebrtc crate
## [0.7.4] - 2025-02-03
### Added
- Support for track subscription permissions
## [0.7.3] - 2025-01-17
### Added
- Add an API to set buffer_amount_low_threshold for DataChannel
- Update RoomInfo to contain buffer_amount_low_threshold for DataChannel
### Fixed
- Wait for the buffered amount to become low before sending data during publish_data for Reliable Data Channel
## [0.7.2] - 2025-01-04
### Added
- bump
### Fixed
- Fixed deadlock with nested RPC calls
## [0.7.1] - 2024-12-14
### Added
- bump libwebrtc to m125
## 0.9.5 (2026-10-12)
### Fixes
- In-flight incoming data streams now error with `AbnormalEnd` when the room closes. Before, the stream readers ended cleanly, so `read_all()` returned truncated content as if it were complete.
- Keep pre-encoded video decodable when WebRTC drops frames: the passthrough encoder disables WebRTC's frame dropper, and after a frame is dropped before it, holds delta frames and requests a keyframe until the next keyframe arrives.
- Add `PeerConnection::set_bitrate` to set the send bitrate limits of a peer connection.
- Public API tokens minted from an API key and secret now grant every thread action (list, create, read, write, delete) across the project. Before, they carried only `all`, so creating or reading a thread was refused.
- Re-apply dynacast publishing layers after every publisher answer, so that layers paused by dynacast stay paused after a renegotiation.
- `Room::close` now leaves the room `Disconnected` and emits `RoomEvent::Disconnected` before it returns, and `Disconnected` is final: a reconnect still finishing in the background (a queued resume, or the republish after a full reconnect) can no longer bring a closed room back to `Connected` or emit `Reconnected` after `Disconnected`. Before, the room's event task could drop the engine's `Disconnected` event when the close signal won the race, so a closed room kept reporting `Connected` (or `Reconnecting`) about half the time.
#### `VideoGrants` gains the `agent` grant and `Claims` the `kind` claim (with
`AccessToken::with_kind`), which the Go, Python and JS SDKs already carry. An
agent worker's token is `VideoGrants { agent: true }` and a simulated job's
participant token is `kind: "agent"`; neither could be minted from Rust before.
`livekit-uniffi` exposes both: `TokenOptions.kind`, `Claims.kind`, and `agent`
on its `VideoGrants` record.
**Breaking:** the four grants the server infers when absent -- `can_publish`,
`can_subscribe`, `can_publish_data`, `can_update_own_metadata` -- are now
`Option<bool>`, as in the Go and JS SDKs. `None` leaves the decision to the
server, and the new getters (`can_publish()`, `can_subscribe()`,
`can_publish_data()`, `can_update_own_metadata()`) read a token the way the
server does, `can_publish_data` falling back to `can_publish` included. Code
that set these fields writes `Some(..)`; code that read them uses the getters.
The same fields are optional on the `livekit-uniffi` record, and
`livekit-api` re-exports the crate as `livekit_api::access_token`, so both
carry the change.
Nothing at its default is written into the token any more: unset claims and
grants are omitted, as the server's own `omitempty` grants are. Verification
of existing tokens is unchanged.
## 0.9.4 (2026-10-07)
### Fixes
- Scale the `x-google-start-bitrate` hint by connection setup time: the 1 Mbps camera cap now applies to connections that set up within 1.5 s and ramps linearly down to 300 kbps at 3.5 s or slower; screen share stays uncapped.
- Increase the local and remote data track event buffers so a burst of track lifecycle events cannot fill the channel and deadlock room disconnect.
- Manage VAAPI displays across capability probes and encoder lifecycles so driver resources and threads are released without invalidating active encoders.
- Release local published tracks when a server-initiated room closure has already closed the peer connection.
- RoomClient::send_data now returns a Send future by dropping its RNG before awaiting.
- Never wait on the peer when closing a signal stream: a half-open socket (no FIN/RST) left the reader parked in `recv()` forever, so the resume never dialled and `Room::close` hung. `SignalStream::close` now aborts the parked reader and bounds the writer's Close flush.
## 0.9.3 (2026-09-24)
### Fixes
- Avoid panics when malformed RTC error headers contain non-ASCII text.
- Java version in libwebrtc was bumped by Google, downgrade it again for Unity 2022 build compatibility - #1456 (@MaxHeimbrock)
- refactor(signaling): explicit signal lifecycle state machine - #1402 (@lukasIO)
- Upgrade to prost 0.14 across the whole project - #1447 (@1egoman)
## 0.9.2 (2026-09-22)
### Fixes
- Correct the AppKit framework name so macOS linking works with case-sensitive SDK filesystems.
- Cleanup unused dependencies
- Fix reliable data channel replay: keep the full retry buffer across resumes, drop duplicate reliable packets, and ignore replayed chunks on uncompressed streams instead of failing with `MissedChunk`.
- Stop answering a server-initiated Leave (room deleted, duplicate identity) with a client Leave. The server has already ended the session and is closing the signalling socket, so the reply only ever produced the warning "dropping pass-through signal — no stream available" on every such disconnect.
#### Fix the RTC engine leaking when a room is dropped without being closed.
The engine's event task held a strong reference to the engine, while the signal that stops
that task lives inside the engine itself, so the two kept each other alive. An engine
dropped without an explicit `close()` released nothing: the session, both peer connections,
the WebRTC runtime, and the signal client with its open websocket all stayed resident for
the lifetime of the process, and the server kept its half of the session because the socket
was never closed. The task now holds a weak reference and stops on its own once the engine
is gone.
#### `EncryptionError::Failed` and `DecryptionError::Failed` carry a `reason` string and are no longer `flat_error`,
so a foreign `EncryptionProvider` or `DecryptionProvider` returning an error no longer aborts the process with
"Can't lift flat errors" -- a failed data track decrypt (no E2EE manager, key mismatch, corrupt frame) now
drops the frame and leaves the room connected.
#### Report the reconnect reason to the server when resuming.
Resumes previously sent no reason, so server-side telemetry could not attribute why Rust
clients reconnect — every resume looked like `RR_UNKNOWN`. The engine now records what caused
the episode (signal disconnected, publisher failed, subscriber failed) and reports it on each
resume attempt. The v0 signalling path was also missing the `reconnect_reason` query parameter
entirely, so it would not have been reported even if a reason had been supplied.
#### Fix resume reporting success for a PeerConnection that had not recovered.
A resume decided recovery from `PeerConnectionState`, which keeps reading `Connected` for tens
of seconds after the far end goes away. A resume could therefore emit `Resumed` — and so
`RoomEvent::Reconnected` with `ConnectionState::Connected` — for a session whose subscriber
transport was dead, leaving applications with no signal that they had stopped receiving media.
A resume now requires each transport to have entered `Connected` since the resume began, or to
have held it throughout, rather than trusting the state it currently reports.
#### `VideoGrants` gains the `agent` grant and `Claims` the `kind` claim (with
`AccessToken::with_kind`), which the Go, Python and JS SDKs already carry. An
agent worker's token is `VideoGrants { agent: true }` and a simulated job's
participant token is `kind: "agent"`; neither could be minted from Rust before.
`livekit-uniffi` exposes both: `TokenOptions.kind`, `Claims.kind`, and `agent`
on its `VideoGrants` record.
**Breaking:** the four grants the server infers when absent -- `can_publish`,
`can_subscribe`, `can_publish_data`, `can_update_own_metadata` -- are now
`Option<bool>`, as in the Go and JS SDKs. `None` leaves the decision to the
server, and the new getters (`can_publish()`, `can_subscribe()`,
`can_publish_data()`, `can_update_own_metadata()`) read a token the way the
server does, `can_publish_data` falling back to `can_publish` included. Code
that set these fields writes `Some(..)`; code that read them uses the getters.
The same fields are optional on the `livekit-uniffi` record, and
`livekit-api` re-exports the crate as `livekit_api::access_token`, so both
carry the change.
Nothing at its default is written into the token any more: unset claims and
grants are omitted, as the server's own `omitempty` grants are. Verification
of existing tokens is unchanged.
#### Write the `x-google-start-bitrate` hint once per publisher connection, and exempt screen share from the 1 Mbps cap.
libwebrtc reads this fmtp parameter per m-section but applies it to the shared `Call` (`WebRtcVideoSendChannel::ApplyChangedParams` -> `SetSdpBitrateParameters`), where `RtpBitrateConfigurator` holds one config for the whole peer connection. It retains `start_bitrate_bps` and re-applies it on network route changes (`RtpTransportControllerSend::OnNetworkRouteChanged`), so a WiFi-to-cellular handover re-seeds the estimator from the original hint with no renegotiation. Rewriting the value on later offers was therefore at best a no-op and at worst a restart of a converged bandwidth estimator; it is now written only on the first offer that carries local video, and only once that offer is accepted locally. A full reconnect builds a new peer connection and seeds the new estimator again. The initial offer sent with the JoinRequest in single PC mode never carries the hint: it is created before any track is published, so no target bitrate exists yet.
Screen share is no longer capped at 1 Mbps, matching client-sdk-js and client-sdk-android: unlike camera content, a screen share is published at a high bitrate so text stays legible, and a conservative start costs more than a brief overshoot.
## 0.9.1 (2026-09-09)
### Features
#### Moves the RPC implementation into a new `livekit-rpc` crate, alongside the existing
`livekit-data-stream` and `livekit-datatrack` crates.
**Breaking:** the `livekit::rpc` module is gone. The RPC types it held are unchanged and
still re-exported from `livekit::participant` and the prelude, so most code needs no edit;
code that spelled the module out (`use livekit::rpc::RpcError;`) should import from
`livekit::participant` or the prelude instead. `RpcClientManager`, `RpcServerManager` and
`HandleRequestOptions` remain reachable under `livekit::participant` but are now
`#[doc(hidden)]`: they are internal SDK API and were never usable without the (private)
transport trait.
Within the `livekit` crate itself, RPC types are now imported from `livekit-rpc` directly
rather than through those re-exports.
The new crate does not depend on `libwebrtc`, so its unit tests run without building WebRTC.
The transport seam that made this possible was already in place; the only change to it is
that `RpcTransport::publish_data` now returns a message-only `RpcTransportError` instead of
`livekit::RoomError`, mirroring `livekit_data_stream::api::SendError`.
Also fixes four latent bugs found while moving the code:
- An RPC call to a participant who disconnects mid-call now fails promptly with
`RecipientDisconnected`. Pending calls were never purged on disconnect, so the caller
waited out its full response timeout (15s by default) and got `ResponseTimeout` instead.
- A server reporting a version that is not valid semver no longer panics the calling task.
An unparseable version is no longer treated as evidence that the server is too old.
- A v1 `RpcResponse` carrying a compressed payload, or no value at all, now fails with an
`ApplicationError` instead of resolving the caller with an empty successful response.
- Removed an unguarded `unwrap` when building a v1 response packet, by giving the function
a signature that cannot represent the invalid state.
Also drops the `semver` dependency from `livekit`, which was only used by the RPC client.
### Fixes
- Load the Jetson MMAPI encoder's runtime libraries (libnvbufsurface, libv4l2/libnvv4l2) lazily via dlopen instead of linking them, so an aarch64 binary built with Jetson support also loads on non-Jetson ARM systems and falls back to other encoders there.
- log warning if signal messages get dropped during reconnect - #1391 (@lukasIO)
#### Own the add_ice_candidate completion state
`PeerConnection::add_ice_candidate` captured `ctx` and `on_complete` by reference in the
completion lambda it hands to libwebrtc. That completion runs asynchronously on the signaling
thread and is deferred behind the operations chain whenever it is busy, for example while a
`SetRemoteDescription` is in flight, so it could execute after the calling frame had returned
and dereference freed stack memory (a crash on the signaling thread on the first ICE candidate
in practice). The lambda now owns its state through a `shared_ptr`.
## 0.9.0 (2026-09-08)
### Breaking Changes
- Removes livekit-runtime and converts this package to be tokio only again - #1375 (@1egoman)
### Fixes
- Add data streams v2 to exposed uniffi interface - #1286 (@1egoman)
- Add the `PASSTHROUGH` encoding preset and remove the unused `UpdateEgressRequest` from the generated protocol
- Fix pre-encoded frame segfault on macOS
- Handle capture of dmabuf using existing capture path
- Add `self_test_http_get` / `self_test_ws_echo` / `has_http_client` / `has_ws_client` UniFFI exports so foreign hosts can exercise the transport seam end-to-end.
#### Make AdmProxy worker-thread-affine: all platform ADM access now happens on the WebRTC worker thread, matching the ADM threading contract.
- The platform ADM is now created lazily on the first PlatformAudio acquire on all platforms, so apps that never use platform audio never construct it.
- Fixes Android platform recording delivering no audio: the audio transport was never registered on the lazily created ADM.
- Fixes a shutdown race by keeping the runtime threads alive as long as Rust can reach the audio device controller.
- Adds a `platform_audio` example exercising the PlatformAudio API and the worker-thread marshaling.
#### Add agent guidance for detecting and preventing memory-lifecycle regressions in
Rust, FFI, and native WebRTC code.
#### Close peer connections before awaiting signal teardown
`SessionInner::close` released the peer connections only after two awaits that can block
indefinitely, so cancelling `close()` — for example by wrapping it in a timeout — left the
transports open and their ICE UDP sockets bound for the lifetime of the process. Long-lived
clients eventually exhausted their file descriptors. The transports are now closed before
the first await, which makes the teardown safe to cancel.
#### Moves the internal region-discovery cache into a new `livekit-region` crate. No
public API or behaviour change.
#### Moves the signalling client into a new `livekit-signaling` crate. livekit-api
re-exports it under the historical `livekit_api::signal_client` path, now marked
deprecated: it is internal SDK API, and dependents should use livekit-signaling
directly. livekit-api no longer depends on livekit-net.
Also drops two dependencies that were declared but never used: `scopeguard` and
`bytes`.
#### Fix CUDA and FFI resource cleanup during SDK shutdown.
NVIDIA encoder and decoder factories now share a reference-counted CUDA context
and destroy it when the final factory is dropped. FFI shutdown now releases
leftover handles one at a time so nested `drop_handle` calls do not re-enter
`DashMap::clear()`. Adds regression coverage for FFI-handle, watcher, and
configuration cleanup during disposal.
#### Expose network_type on IceCandidateStats
Chromium's local `RTCIceCandidateStats` carries a non-standard `networkType` field (WiFi,
cellular, ethernet, etc.), but `IceCandidateStats` had no place to put it, so it was silently
dropped during `get_stats()` deserialization. Adds `network_type: Option<String>` to the struct;
non-breaking since it already derives `#[serde(default)]`.
#### Fix room-session and data-channel leaks across connect/disconnect cycles.
The E2EE manager callback now captures `RoomSession` weakly so the session can
drop after disconnect. Data-channel observer callbacks are cleared during RTC
teardown so the observer/callback cycle cannot keep peer connections alive.
Adds regression coverage for room-session destruction and data-channel callback
cleanup.
#### Fix native video-source lifecycle and NVENC initialization failure handling.
The raw-video keepalive task now uses a weak liveness check and defers its
black I420 buffer allocation until source liveness is confirmed, so dropping
an unused source releases its resources. `nvEncInitializeEncoder` failures now
propagate instead of leaving the encoder half-initialized.
## 0.8.4 (2026-08-25)
### 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 - #1234 (@lukasIO)
- Automatically retry webrtc build downloads
- feat: upgrade libwebrtc to m150. - #1284 (@cloudwebrtc)
- 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.
#### Moves access-token generation and verification into a new `livekit-token` crate.
`livekit_api::access_token::*` continues to resolve to the same types via a
re-export, so no consumer changes are needed.
Also fixes the `services-tokio` and `services-async` features, which used the
access-token types without declaring the `access-token` feature. Building with
`--no-default-features --features services-tokio` previously failed to compile.
## 0.8.3 (2026-08-10)
### Features
- Add `other_sdks` field to propagate additional SDK metadata to the server.
### Fixes
- Only advertise internal H264 decode formats if the decoder works - #1313 (@MaxHeimbrock)
## 0.8.2 (2026-08-03)
### Fixes
- Add a unified `EgressClient::start_egress` that calls the v2 `Egress.StartEgress` RPC with a `StartEgressRequest`, alongside the existing per-type helpers.
- Fix a publisher-transport deadlock during renegotiation. When another negotiation was requested while an offer was awaiting its answer, `set_remote_description` re-entered `create_and_send_offer` while holding the transport's non-reentrant inner mutex, permanently wedging publishing.
- Move the concept of "internal" data streams into `livekit` crate from `livekit-data-stream` - #1304 (@1egoman)
#### Fix H.264, H.265, and AV1 NVENC sessions so live bitrate and frame rate updates
reconfigure the hardware without restarting the encoder.
## 0.8.1 (2026-07-29)
### Fixes
- allow for audio filters to be registered after initial room connection - #1273 (@lukasIO)
- Caching of tokio backend reqwest http client - #1285 (@MaxHeimbrock)
- Add data streams v2 - #1192 (@1egoman)
- Ensure participant disconnects are synthesized after connection resume - #1250 (@lukasIO)
## 0.8.0 (2026-07-27)
### Breaking Changes
#### Route LiveKit signalling through a pluggable transport (new `livekit-net` crate).
The signalling WebSocket and the two pre-connect HTTP GETs (validate, region discovery) now go through pluggable transport traits (`WsClient` for the WebSocket, `HttpClient` for request/response) resolved from a process-global registry with independent slots — a consumer can bring only HTTP, or only WebSocket. The new `livekit-net` crate owns the WebSocket/HTTP/TLS stack behind those traits and ships native (tokio / async-std) backends. Native builds are unchanged in behavior.
**Breaking (`livekit-api`, and `livekit` via `EngineError::Signal`):**
- `SignalError::WsError` is removed — `tungstenite` is no longer part of the public API. A failed WebSocket handshake now surfaces its HTTP status as `SignalError::Client`/`Server`; transport connection and close failures surface as the new `SignalError::Connection(String)` / `SignalError::Closed` variants (previously all collapsed into `Timeout`).
- `SignalError` is now `#[non_exhaustive]`, and gains a `SignalError::TransportNotConfigured` variant — returned when no transport is registered (host/foreign builds must call `livekit_net::set_ws_client` / `set_http_client` before connecting). This is a permanent configuration error; callers must not retry.
- The signalling WebSocket/HTTP/TLS crates are no longer transitive dependencies of `livekit-api`; TLS features delegate to `livekit-net`. Existing `signal-client-tokio` / `-async` / `-dispatcher` and TLS feature names are unchanged.
### Fixes
- Address typo in parsing rpc server version - #1268 (@1egoman)
- Data tracks schema metadata support.
- Emit black keepalive frames from NativeVideoSource instead of uninitialized memory. webrtc::I420Buffer::Create leaves the pixel planes uninitialized, so the pre-capture keepalive frames could leak recycled heap contents (often fragments of earlier frames from the same process) to subscribers as the first keyframes - #1271 (@eh-steve)
- Add NVIDIA NVENC AV1 encoding when the GPU reports AV1 encode support.
## 0.7.53 (2026-07-17)
### Features
- Add a pre-encoded video publish path: a passthrough video encoder and encoded video frame buffer in webrtc-sys, and `EncodedVideoFrame`/`EncodedVideoCodec`/`EncodedFrameType` publish APIs with a `VideoEncoderBackend::PreEncoded` backend in libwebrtc. WebRTC rate-control targets and keyframe requests are forwarded to encoded sources, and pre-encoded AV1 and H265 access units are validated on ingest.
### Fixes
- Emit room EOS when the underlying LiveKit room event channel closes after a server-initiated disconnect, and ignore duplicate disconnect events during teardown.
- Don't log an expected publisher data channel close as unexpected - #1224 (@longcw)
#### Simplify x-google-start-bitrate logic and update degradation preference defaults
- Start bitrate: use min(90% of target, 1 Mbps) instead of adaptive network hints
- Remove slow connection detection and network quality hints on reconnect
- Default degradation preference by track source:
- Camera: MaintainFramerate (smoother video)
- Screenshare: MaintainResolution (clarity for text/UI)
- Other: Balanced
## 0.7.52 (2026-07-14)
### Fixes
- Make some fields public for data track types
- Refactor data tracks E2EE interface
- refactor: extract data-stream logic and shared types into new `livekit-common` and `livekit-data-stream` crates (public API unchanged; types are re-exported from `livekit`)
- Use concrete type for data track manager output events
- Add an opt-in zero-playout-delay mode for native video subscribers, expose it through the `local_video` subscriber's `--low-latency` flag, and isolate subscriber diagnostics from frame-driven video rendering.
## 0.7.51 (2026-07-09)
### Fixes
- feat: auto failover APIs with LK Cloud - #1196 (@davidzhao)
- Fix for dynacast error - #1213 (@MaxHeimbrock)
- Fix malformed RTC error handling
- Handle data track SID reassignment
- introduce LiveKitAPI construct, added smoke tests - #1220 (@davidzhao)
- Turn single peerconnection off by default - #1206 (@cnderrauber)
## 0.7.50 (2026-06-30)
### Features
- Add `user_data` support to frame metadata, allowing arbitrary application-supplied bytes to be attached to a video frame via the `PTF_USER_DATA` packet trailer feature.
#### Improve initial video quality by setting `x-google-start-bitrate` SDP hint for all video codecs (VP8, VP9, AV1, H264, H265) and defaulting to `MaintainResolution` degradation preference.
This addresses the issue where video starts blurry for several seconds before improving, by:
1. Telling WebRTC's bandwidth estimator to start at 70% of target bitrate instead of ramping up from ~300kbps
2. Preferring frame drops over resolution reduction when bandwidth is constrained
The `DegradationPreference` option is now exposed via FFI for Python, C++, Unity, and Node SDKs.
#### Add `MaintainFramerateAndResolution` to `DegradationPreference` enum to align with WebRTC M144.
- `MAINTAIN_FRAMERATE_AND_RESOLUTION` is now the recommended value (replaces deprecated `DISABLED`)
- `DISABLED` is deprecated but still supported for backwards compatibility
- Both values map to the same behavior: maintain framerate and resolution, dropping frames if needed
### Fixes
- Fix AV1 subscriber decode when packet trailers are enabled.
- Improve log messages around plugin loading - #1186 (@lukasIO)
## 0.7.49 (2026-06-24)
### Fixes
- harden reconnect behaviour - #1148 (@lukasIO)
## 0.7.48 (2026-06-23)
### Features
- Rename user facing APIs for Packet Trailer to Frame Metadata.
### Fixes
- Upgrade protocol to v1.48.0
## 0.7.47 (2026-06-19)
### Fixes
- fix: escalate to full reconnect if connection failed during a resume - #1175 (@davidzhao)
## 0.7.46 (2026-06-17)
### Features
#### Make GLib an opt-in dependency
`webrtc-sys` no longer links against `glib-2.0`/`gobject-2.0`/`gio-2.0` by default.
Breaking: Wayland screen sharing now requires the `glib-main-loop` feature on `livekit` (or `libwebrtc`).
### Fixes
- Add track publishing doc example
- Fix silent subscription failures in single-pc mode when the SFU reuses an existing empty transceiver for a new remote track. Also make `RtpTransceiver::mid()` safe to call on transceivers that haven't been negotiated yet — libwebrtc is built with `-fno-exceptions`, so `std::optional::value()` aborted the process instead of throwing.
- Add `LK_DISABLE_NVDEC` to bypass NVIDIA NVDEC decoder registration when the environment variable is set.
- return DeviceNotFound when device is not there for set_recording_devi… - #1155 (@xianshijing-lk)
#### Add dynacast support - #1003 (@chenosaurus, @stephen-derosa)
This includes a minor breaking change for `libwebrtc`: `RtpParameters` now
contains additional RTP sender state that must be preserved when round-tripping
through `set_parameters()`.
## 0.7.45 (2026-06-09)
### Fixes
- Fix NVIDIA encoder I420 uploads to copy each plane using its actual source stride, avoiding chroma corruption when source frames use padded YUV planes. Also fix the `local_video` publisher reusing mutable I420 frame storage after handing frames to WebRTC.
- Reject oversized data messages before they break the data channel.
- Add per-publication video encoder backend selection. Add a video encoder backend availability query. Remove `LIVEKIT_PREFERRED_HW_ENCODER` in favor of per-publication backend selection.
## 0.7.44 (2026-06-03)
### Features
- Add rpc max_round_trip_latency and move to builder pattern - #1127 (@1egoman)
### Fixes
- [allow(dead_code)] for dead function in room module - #1128 (@stephen-derosa)
- Send publisher offer with join request to accelerate connection - #996 (@cnderrauber)
## 0.7.43 (2026-05-29)
### Fixes
- bump protocol to v1.46.4 - #1121 (@lukasIO)
- Add native video pipeline timing instrumentation for local video measurements, exposing local publish and subscribe timing through async streams and subscriber overlay GPU upload and receive-to-GPU latency metrics through explicit timing observers.
## 0.7.42 (2026-05-21)
### Features
- Introduce pipeline options for remote data tracks, support multiple in-flight frames.
### Fixes
- Filter internal data streams out of livekit-ffi interface - #1112 (@1egoman)
#### feat: add Android application context initialization for PlatformAudio support.
Android requires `ContextUtils.initialize(applicationContext)` before WebRTC audio components can be created. This change:
- Adds `livekit_ffi_initialize_android_context()` C FFI function for Unity and other FFI consumers
- Uses `CreateAndroidAudioDeviceModule()` instead of generic `CreateAudioDeviceModule()` on Android
- Handles empty device GUIDs on Android (falls back to index 0)
- Documents Android-specific limitations: single default device, no app-level device selection
Platform notes:
- Android device enumeration returns only one "default" device with empty name/GUID
- Audio routing (speaker/earpiece/Bluetooth) is controlled by Android's AudioManager, not WebRTC
## 0.7.41 (2026-05-20)
### Fixes
- Bugfix: Always emit Disconnected on engine close - #1096 (@MaxHeimbrock)
- Support for large RPC messages using data streams - #1013 (@1egoman)
## 0.7.40 (2026-05-14)
### Fixes
- feat: add scalability mode for AV1/VP9. - #1076 (@cloudwebrtc)
- Add `LIVEKIT_PREFERRED_HW_ENCODER` to prefer `nvenc` or `vaapi` hardware video encoding when both are available.
- Relocate unrelated types out of `livekit-protocol`
#### Get WebRTC ADM into Rust - #1037 (@xianshijing-lk)
This PR introduces platform audio device management via WebRTC's Audio Device Module (ADM).
#### Features
- **ADM Proxy**: New `AdmProxy` class that switches between Dummy ADM (synthetic mode) and Platform ADM (real audio I/O)
- **PlatformAudio API**: High-level Rust API for microphone capture and speaker playout with AEC/AGC/NS
- **Device enumeration**: List and select recording/playout devices by index or GUID
- **Mode switching**: Seamlessly switch between synthetic mode (FFI callbacks) and platform mode (native speakers) while audio is active
- **FFI platform audio support**: Expose platform audio device enumeration and selection through `livekit-ffi`
- **Audio processing**: Configure echo cancellation, noise suppression, and auto gain control with platform-specific defaults (hardware on iOS, software elsewhere)
#### Audio Modes
| Mode | Recording | Playout | Use Case |
|------|-----------|---------|----------|
| Synthetic | NativeAudioSource | Dummy ADM + FFI | Unity audio, agents |
| Platform | Platform ADM mic | Platform ADM speakers | VoIP with AEC |
#### API
```rust
// Create PlatformAudio for microphone/speaker access
let audio = PlatformAudio::new()?;
// Enumerate and select devices
for i in 0..audio.recording_devices() as u16 {
println!("Mic {}: {}", i, audio.recording_device_name(i));
}
audio.set_recording_device(0)?;
// Create audio track for publishing
let track = LocalAudioTrack::create_audio_track("mic", audio.rtc_source());
```
## 0.7.39 (2026-05-11)
### Fixes
- fix: Sync inner.enabled state for E2EE manager. - #1073 (@cloudwebrtc)
- Upgrade protocol to v1.45.8
## 0.7.38 (2026-05-10)
### Fixes
- Bump `rustls-webpki` to 0.103.13, addressing [GHSA-82j2-j2ch-gfr8](https://github.com/advisories/GHSA-82j2-j2ch-gfr8)
- Fix missing `libwebrtc.jar` for Android builds, harden build scripts
- fix: derive `simulcasted` from non-deprecated TrackInfo fields - #1052 (@cloudwebrtc)
- fix race in download_webrtc to reduce flaky build - #1047 (@hechen-eng)
- Improve WebRTC build scripts and add external_audio_source patch - #1053 (@xianshijing-lk)
- support SimulateScenario through FFI to improve testing - #1069 (@davidzhao)
- TEL-464: reduce redundant resampling in audio filter - #1019 (@hechen-eng)
## 0.7.37 (2026-04-23)
### Features
#### Add support for frame level packet trailer
##890 by @chenosaurus
- Add support to attach/parse frame level timestamps & frame ID to VideoTracks as a custom payload trailer.
- Breaking change in VideoFrame API, must include `frame_metadata` or use VideoFrame::new().
### Fixes
- Add device-info crate and send device_info to telemetry - #982 (@maxheimbrock)
- Fix data track packet format issue breaking E2EE
- Fix unbound send queue that can cause latency in data track messages - #1032 (@chenosaurus)
- Fix for raw stream drop called from non tokio thread like Unity .NET GC - #1016 (@MaxHeimbrock)
## 0.7.36 (2026-04-02)
### Features
- Initial support for data tracks
### Fixes
#### use the bounded buffer for video stream
##956 by @xianshijing-lk
Before this PR, it uses an unbounded buffer for video stream, that will cause multiple problems:
1, video will be lagged behind if rendering is slow or just wake up from background
2, it will be out of sync with audio
This PRs provides options to set a bounded buffer for video stream, and use 1 buffer as the default option.
## 0.7.35 (2026-03-31)
### Features
- Expose participant active event, state, and joined at
### Fixes
- Upgrade to thiserror 2
## 0.7.34 (2026-03-22)
### Features
#### E2EE: allow setting key_ring_size and key_derivation_algorithm, update webrtc to m144
##921 by @onestacked
This PR uses [this webrtc-sdk PR](https://github.com/webrtc-sdk/webrtc/pull/224) to configure the KDF.
I've tested this with https://codeberg.org/esoteric_programmer/matrix-jukebox and it is compatible with Element Call.
Fixed: https://github.com/livekit/rust-sdks/issues/796
### Fixes
- Add disconnectReason to Room::close
- End-to-end testing for video streams
- Fix H.264 codec matching
#### add bounded buffer to audio_stream, and use 10 frames as the default
##945 by @xianshijing-lk
#### fix PC timeout when connecting with can_subscribe=false
##955 by @s-hamdananwar
When a participant connects with `canSubscribe=false` in their token, the server sends `subscriber_primary=false` in the JoinResponse and does not send a subscriber offer. This results in `wait_pc_connection` timing out as it is expecting a subscriber PC even when the publisher PC is primary. This PR will skip waiting for subscriber PC when `subscriber_primary=false`.
#### Send client os and os_version from rust
##952 by @MaxHeimbrock
Adds [os_info](https://crates.io/crates/os_info) crate as dependency and sends the data for client connections.
## 0.7.33 (2026-03-13)
### Fixes
#### enhanced build configuration to support macOS and iOS platforms with proper system library linking
##847 by @SchmErik
#### fix video track subscription in single peer connection mode
##914 by @xianshijing-lk