CHANGELOG.md

Changelog

All notable changes to this project will be documented in this file.

The format is based on Keep a Changelog, and this project adheres to Semantic Versioning.

[Unreleased]

0.7.32 - 2026-02-16

Fixed

• fix full_reconnect downgrade & don't ignore Leave messages (#893)

Other

• turn single peerconnection off by default (#897)
• ensure signal connections times out properly and retries (#895)
• added Single Peer Connection support to Rust (#888)
• set the simulcast codec & layers (#891)

0.7.31 - 2026-02-10

Other

• don't use clamp as the ultimate_kbps can be lower than 300 (#886)
• pre-connect the publisher PC when an RPC handler is registered (#880)

0.7.30 - 2026-02-09

Fixed

• fix the RPC race condition (#865)

Other

• update proto & fix CI (#871)
• Use workspace dependencies & settings (#856)
• Upgrade protocol to v1.44.0 (#857)
• Expose participant's permission to ffi layer (#824)

0.7.29 - 2026-01-15

Fixed

• ensure Room.creation_time is ms (#822)

Other

• try setting x-google-start-bitrate for vp9 (#820)

0.7.28 - 2025-12-19

Added

• (ParticipantInfo) export kind details (#813)

0.7.27 - 2025-12-17

Other

• Handle server initiated mute request (#812)

0.7.26 - 2025-12-04

Added

• (connector) initial service impl (#790)

Fixed

• fix mute/unmute events for LocalTrack. (#799)

Other

• Add RoomEvent::TokenRefreshed (#803)

0.7.25 - 2025-11-20

Other

• perform full reconnect if resume fails (#792)
• E2E RPC tests (#769)
• Remove unused dependencies (#761)

0.7.24 - 2025-10-27

Other

• Expose set video quality (#759)

0.7.23 - 2025-10-23

Other

• add h265 codec support (#762)

0.7.22 - 2025-10-22

Other

• License check (#746)
• Remove participant check for data packets (#757)
• clamp connection timeout and fixed the comment (#748)
• put examples in root Cargo workspace (#731)

0.7.21 - 2025-10-13

Added

• (e2ee) add data channel encryption (#708)

Fixed

• fix some potential audio issues, clean up the code a bit, and suppress some warnings (#737)
• do not log 'signal client closed: "stream closed"' on disconnect (#727)

Other

• Test participant disconnect (#732)
• Increase RPC max RT time to 7s (#729)
• E2E audio test (#724)

0.7.20 - 2025-10-03

Other

• updated the following local packages: libwebrtc

0.7.19 - 2025-09-29

Fixed

• apply original participant fields in data messages (#709)

Other

• Implement Display and Error for RpcError (#719)
• Fix intermittently failing E2E reliability test (#718)
• Do not modify raw packets (#714)
• Add send_bytes method (#691)
• Disable opus red for e2ee enabled clients (#706)
• Upgrade protocol to v1.41.0 (#703)

0.7.18 - 2025-09-09

Other

• Add option to enable audio preconnect buffer in SDK & FFI (#700)
• Data channel reliability (#688)

0.7.17 - 2025-09-03

Other

• updated the following local packages: livekit-api, libwebrtc

0.7.16 - 2025-07-31

Other

• updated the following local packages: livekit-api

Changelog

0.7.15 - 2025-07-16

Other

• remove published tracks when the room is closed (#677)

0.7.14 - 2025-06-23

Fixed

• audio_frame_ms didn't work expectedly (#671)

0.7.13 - 2025-06-17

Other

• Expose room updates, support MoveParticipant (protocol 15) (#662)

0.7.12 - 2025-06-11

Fixed

• fix duration overflow (#654)

Other

• Remove debouncer when fast_publish is enabled (#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

0.7.48 (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

// 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
• 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 to configure the KDF.

I've tested this with https://codeberg.org/esoteric_programmer/matrix-jukebox and it is compatible with Element Call.

Fixed: #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 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