CHANGELOG.md

Changelog

v2.48.1

• Desktop: Add new Audio Device Module option

  • Implement ADM using cubeb
  • Update ADM switch to support a startup flag
  • Use min_latency to limit latency requested
  • Skip building ringrtc ADM on linux aarch64

• Add CallLink AdminAction logs to improve debugging

• Android: Package libraries unstripped by default

v2.48.0

• Desktop: Allow VideoSupport to accept a MediaStream

• Group Calls: Avoid creating a client if one already exists

• Update to webrtc 6613c

  • Desktop/Mac: check for a channel change for input only

• Update node and ios dependencies

v2.47.1

• Group Calls: Allow connection to a TCP+TLS server candidate

• Update dependencies

v2.47.0

• Update to webrtc 6613a

• Update PeekInfo::unique_pending_users to maintain order from SFU

• Desktop: Add function to get CallID from EraID on GroupCall object

• Retain old ratchet secrets to allow for out-of-order decryption

• Call Sim: Add group calling support

• logs-notebook: Parse system stats

• Simplify running WebRTC tests

v2.46.2

• Update to webrtc 6478k

  • iOS: Fixed issue with missing network interfaces

• Android: Disable the use of sessionId for Oboe

• Update to Rust 1.80.1

• Build improvements

v2.46.1

• CI: Upload desktop symbols to GCS

• Update to webrtc 6478j

  • Reduce kDefaultMinPixelsPerFrame
  • Revert "Add Rust_setIncomingAudioMuted"
  • Revert "Log more info when select fails"

• Update dependencies

• Build improvements

v2.46.0

• Send audio enabled status to remote device in 1:1 calls

• Update to webrtc 6478i

  • Add ice switch reason to logging
  • Android: Oboe ADM uninitialize on stop
  • Don't get stats from unused transceivers

• Build improvements

v2.45.0

• Call links: Add restrictions to create call link API

• Update to webrtc 6478h

  • Android: Add new OboeStream class for reliability

v2.44.5

• Update to webrtc 6478g
  • Android: Oboe refinements and latency improvements

v2.44.4

• Group Calls: Reduce log noise

• Run dump_syms on CI

• Support overlapping memory copy for decrypt

• Enable sending dependency descriptor in group calls

• Send encrypted TOC byte in group calls

• Update dependencies

• Update to webrtc 6478f

  • Improve network type detection on macOS
  • Enable sending dependency descriptor in group calls
  • Mark audio packets as having an encrypted TOC byte
  • Fix ios device orientation left/right assignment

v2.44.3

• Android: Add audio device module based on Oboe

• Update to webrtc 6478e

  • Add audio device module for android based on Oboe
  • Remove support for setting mobile aec
  • Simplify handling of audio callbacks

• Desktop ADM: Resolve dependency cycle and other improvements

v2.44.2

• Desktop ADM: Add support for switching to RingRTC ADM

• Fix Python deprecation warning

• Update to webrtc 6478b

  • ringrtc: Add stub ADM
  • Revert "Enable sending dependency descriptor in group calls"

v2.44.1

• Update to webrtc 6478a

  • Update to WebRTC 6478 (m126)
  • Enable sending dependency descriptor in group calls

• Group Calls: Enable sending dependency descriptor

• Refactored protobuf to own crate

• Call Sim: Refactoring

• Update dependencies

v2.44.0

• Remove reliable payload type, reuse existing data payload type

• Update to webrtc 6261l

  • Remove code for supporting SDES
  • Propagate externally-negotiated keys
  • Only attempt to relay connections to addresses that are globally unique
  • Remove lbred experiment
  • Test fixes

• Make it more convenient to build and run unit tests.

• iOS: Remove SignalCoreKit dependency

• Code Cleanup

v2.43.0 (not released)

• Add support for reliable Admin Actions (approve, deny, remove, block)

• Propagate errors starting camera

• Desktop: Add receivedAtData argument to handleAutoEndedIncoming

• Update to webrtc 6261j

  • Resolve warnings from delay settings

v2.42.0

• Add support for reporting rtc_stats to client application

• Update to webrtc 6261i

  • Support for reporting rtc_stats
  • Enable per-layer PLI for screen sharing

v2.41.0

• Call links: Add Call Link state to PeekInfo

• Update to webrtc 6261g

  • Update video settings
  • iOS: Match WebRTC acknowledgments filename

• iOS: Update builds and tests

• Update dependencies and documentation

v2.40.1

• iOS: Raised hands array can be empty

v2.40.0

• Group Calls: Support multi-recipient message sending

• Group Calls: Update bitrate limits for screen sharing

• Update to webrtc 6261e

  • Update to use Opus 1.5

v2.39.3

• Update to webrtc 6261d

  • Add receive support for encrypted TOC byte
  • Add logging when select fails

• Add receive support for encrypted TOC byte

• Update dependencies

v2.39.2

• Group Calls: Apply removal of demux IDs separately

• Log notebook improvements

• Build improvements

• Update to Rust 1.76.0

v2.39.1

• Call Sim: Add jitter buffer config

• Don't probe when close to the max probe rate

• Group Calls: Synchronize access to last_height_by_demux_id

• Update dependencies

v2.39.0

• Update to WebRTC m122

• Desktop: Update IceServer fields to be optional

• Add receive support for dependency descriptor to determine unencrypted length

• Group Calls: Handle client_status in sfu.join()

• Call links: Replace update revocation API with an explicit delete API

• Update dependencies

v2.38.0

• Update to webrtc 6099c

  • Accept list of IceServers for Turn configuration

• Desktop: Accept list of IceServers for Turn configuration

• Enable "First Ready" Turn pruning policy

v2.37.1

• Update to webrtc 6099b

  • Apply upstream m120 change to JsepTransportController

• Call Sim: Add PESQ and PLC MOS support

v2.37.0

• Update to WebRTC m120

• Desktop: added_time and speaker_time are not optional

• Desktop: Support installing via npm

• Update dependencies

v2.36.0

• Use unified plan for group calls

• Update jni crate to 0.21.1

• Desktop: Remove legacy call message fields

v2.35.0

• Update zkgroup to 0.37.0

• Code Cleanup

• Desktop: Always use the Windows ADM2

• Android: Generate assets/acknowledgments/ringrtc.md as part of build

• Make lints on CI slightly faster

• Build improvements and dependency updates

• Update to Rust 1.74.1

v2.34.5

• Use unified plan for 1:1 calls

• iOS: Make trivial RemoteDeviceState for IndividualCalls

• iOS: Make isUsingFrontCamera publically readable

• Call Sim: Add deterministic loss handling and lbred test

• Build webrtc using github actions

v2.34.4

• Fetch build artifacts using a proxy where necessary

• Update to WebRTC m118

v2.34.3

• Update to webrtc 5845j

  • Add low bitrate redundancy support
  • Lower port allocation step delay
  • Prune TurnPorts on a per-server basis
  • Unregister sink properly when closing

• Call Sim: Improvements for running large test sets

v2.34.2

• Group Calls: Propagate demux_id to LocalDeviceState

v2.34.1

• Cleanup logging

• Desktop: Remove device preloading to avoid permission prompt

v2.34.0

• Group Calls: Add Hand Raise feature

• Electron: Allow ICE server hostname to be set

• Build improvements and dependency updates

v2.33.0

• Update to webrtc 5845h

  • Add Rust_setIncomingAudioMuted
  • Update libvpx dependency

• Group Calls: Add Reactions feature

• Group Calls: Prevent comfort noise from getting stuck on

• Replace TaskQueueRuntime with Actors

• Call Sim: Speed up chart generation

v2.32.1

• Desktop: Downgrade dependency for client

v2.32.0

• Add callback for low upload bandwidth in a video call

• Increase max video receive resolution for desktop

• Update webrtc to 5845f

  • Disable audio and media flow by default
  • Allow configuration of audio jitter buffer max target delay

• iOS: Stop building for Catalyst

• Call links: Add reset-approvals to test client

• Update Rust to 1.72.1

• Build improvements and dependency updates

v2.31.2

• Update webrtc to 5845c

  • Update the hardcoded PulseAudio device name to "Signal Calling"
  • Add more audio control and safe defaults
  • Add accessor for bandwidth estimate

• Update webrtc to 5845d

  • Disable early initialization of recording

• Generate license files for WebRTC builds

• Call Sim: Add test iterations and mos averaging

• Add more audio configuration and control

• Improve builds on GitHub Actions

• Build webrtc on AWS for android, ios, linux, mac

v2.31.1

• Update tag for build automation

v2.31.0

• Group Calls: Separate PeekInfo device counts on in/excluding pending devices

• Desktop: Migrate to deviceCountIncluding/ExcludingPendingDevices as well

• Update to WebRTC m116

• Desktop: Use stack arrays for JS arguments rather than vectors

• Build improvements; Support more build automation

• Log improvements

v2.30.0

• Add JoinState.PENDING, for call link calls with admin approval

• Group Calls: Compute send rates based on devices, not users

• CI: Only run the slow tests on the private repo

• Call Sim: Use a fixed resolution for output video

• Log notebook improvements

v2.29.1

• Electron: Disable output format limits when screensharing

v2.29.0

• Call Links: Add Admin Actions support

• Desktop: Adapt video resolutions in 1:1 calls

• Add a Call Simulator for testing

• Reference signalapp/webrtc@5615c

  • Add configuration options to support simulation
  • Support adapting video frames

• Reference signalapp/webrtc@5615d

  • Configure audio jitter buffer max delay

• Improvements to build scripts for automating WebRTC builds

• Test and logging improvements

v2.28.1

• Group Calls: Add support for TCP connections

• Call Links: Switch to X-Room-Id header

• Adjust max audio jitter buffer size to support increased packet time

• Test and logging improvements

v2.28.0

• Call Links: Implement Peek and Join support

• Refactor: BandwidthMode to DataMode

• Android: Fix exception check when throwing an error up to Java

• Improvements to make tests more reliable

v2.27.0

• Update to WebRTC 5615 (m112)

• Implement Call Link Create/Read/Update APIs

• Set audio packet time to 60ms

• Apply audio encoder configuration in group calls

• ios: Fix video capture size selection

• Refactor HTTP JSON parsing so it's more reusable

• Bump Rust toolchain to nightly-2023-03-17

• Build improvements and dependency updates

v2.26.4

• Desktop: Stop duplicate MediaStreamTracks

v2.26.3

• Remove h264 video codec support

  • Reference signalapp/webrtc@5481c

• Disable ANY address ports by default

• Build improvements

v2.26.2

• Node: Require expected calling message fields

• Log notebook improvements

v2.26.1

• Revert "Android: Increase max jitter buffer size" (from v2.25.0)

v2.26.0

• Adjustments to CallId, EraId, RingId and derivations/conversions

• Group Calls: Limit bitrate for the lowest layer

• Reference signalapp/webrtc@5481b

  • VideoAdapter: Fix scaling of very large frames
  • Log more info when video input starts

• Reference signalapp/webrtc@5481a

  • Set inactive timeout to 30s
  • rffi: Set a bandwidth limit on the lowest layer of a group call
  • Allow tcp candidates in group calls

• Log notebook improvements

• Build improvements

v2.25.2

• Node: Ensure that a frame is fully copied before sending it to WebRTC

• Node: Clean up our eslint config, and fix uncovered issues

• Log stats 2sec into a call, then every 10sec after

• Build improvements

v2.25.1

• Update to WebRTC 5481 (m110)

• Use default ptime for all bandwidth modes

• Desktop: Add workaround for slow call to enumerateDevices

• Update dependencies (Rust and Electron)

v2.25.0

• Allow SFU to return multiple ICE candidates (for IPv6 support)

• Android: Add more devices to hardware encoding blocklist

• Android: Increase max jitter buffer size

• Desktop: Initialize call endpoint lazily

• Desktop: Allow explicitly rejecting very tall or very wide frames

• Add cpu statistics to logging

• Reference signalapp/webrtc@5359d

  • Improved logging around network switch
  • Allow TURN ports to be pruned

• CI: Add "Slow Tests" that will run once every night

• Update dependencies, logging, build improvements

v2.24.0

• Desktop: Get TURN servers after call creation to improve glare handling

• Desktop: Add test cases for glare handling

• Desktop: Set a minimum frame rate for screenshare capture

• Reference signalapp/webrtc@5359c

  • Remove Android API 19 support
  • Cleanup merge diffs
  • Include candidate information for ICE route changes
  • Allow any address ports to be disabled

• Log when the selected ICE candidate pair changes

• Add debuglogs notebook for analyzing logs

• CI: Add builds and tests for all platforms

• Build improvements

v2.23.1

• Support fetching prebuilds from build-artifacts.signal.org

• Add support for setting WebRTC field trials

• Android: Add support non-vendored NDK

• Update logging, builds

v2.23.0

• Update to WebRTC 5359 (m108)

• Enable Opus DTX and set default encoding bitrate to 32kbps

• Desktop: Handle failure when entering PiP

• Desktop: Move builds to NPM

• Update dependencies, builds

v2.22.0

• Group Calls: Only allow ringing if you are the call creator

• Electron: Add callId to the call ended notification function

• Improve display of stats in logs

• Update dependencies

• Electron: Save debug information when building

v2.21.5

• Group Calls: Improve ring handling

• Group Calls: Update group membership upon unknown media keys

• Improve display of stats in logs

• Update builds and documentation

• Update Rust

v2.21.4

• iOS: Add isValidOfferMessage and isValidOpaqueRing to the API

v2.21.3

• iOS: Allow WebRTC field trials to be set

• Update dependencies, builds

v2.21.2

• Android: Fix possible crash from AndroidNetworkMonitor

• Electron: Update dependencies (neon mainly)

• Reference signalapp/webrtc@5005b

  • Cherry-pick commits to fix issues

v2.21.1

• Group Calls: Expose isHigherResolutionPending to apps

• Android: Fix race when audio levels change early

• iOS: Set deployment target to 12.2

• Other logging improvements

v2.21.0

• Update to WebRTC 5005 (m102)

• Allow clients to specify the active speaker's height

• Reference signalapp/webrtc@5005a

  • Add logging for audio device timing

v2.20.14

• Reference signalapp/webrtc@4896g
  • Windows: Support multi-channel output

v2.20.13

• Android: Remove audio level debug logging

• Group Calls: Expose decoded video height to apps

• Handle out-of-order IceCandidate and Hangup messages

• Turn off backtraces to stderr by default

v2.20.12

• Group Calls: Prefer recently received group call rings

• Reduce binary size by dropping unicode support from the regex crate

• Enforce that errors are handled on background tokio runtimes

• Update Android builds

  • Update gradle dependencies
  • Use -C linker instead of ndk toolchains

v2.20.11

• Add support for TURN over TLS

• Android: Add echo likelihood to logs

• Reference signalapp/webrtc@4896f

  • Add support for TURN over TLS
  • Enable echo detection

• Update Rust

• Update builds

v2.20.10

• Group Calls: Enable audio recording properly

v2.20.9

• Reference signalapp/webrtc@4896d
  • Have one default port allocator flags instead of two

v2.20.8

• Reference signalapp/webrtc@4896c

  • Remove bitrate multiplier

• Electron: Add logging to video support

v2.20.7

• Log PeerConnection ICE gathering errors

• Let rust core enable media (playback and recording), not clients

v2.20.6

• Prioritize VP9 and H.264 hardware codecs for 1:1 calls

• Add more logging for checking connectivity and group call issues

• Update parse_log.py utility for more debugging

• Reference signalapp/webrtc@4896b

  • Cherry-pick upstream fixes for network crash and iOS audio/logging

• Update Android builds

v2.20.5

• Fix a deadlock when calling set_network_route

v2.20.4

• Remove old video frames when re-enabling video

• Use less bandwidth when using TURN relays

• Improve support when developing on M1 chips

• Avoid notifying remote ringing in case of accepted before connected

• Process remote status events received before the call is accepted

• Android: Allow local video recording to be started while ringing

• Reference signalapp/webrtc@4896a

  • Fix issue with opus frame length for AudioSendStream

• Adjust logging

v2.20.3

• iOS: Fix mapping of log output

v2.20.2

• Update to WebRTC 4896 (M100)

• Disable transport-cc for audio

v2.20.1

• Add VP9 codec support and enable for Android hardware/Electron

• Add state for ConnectingAfterAccepted to fix connect/accept race on caller's end

• Group Calls: Fire peek changed events even if the call is empty

• Reference signalapp/webrtc@4638j

  • Reduce more noise from error/warning logs

• Update dependencies, builds, and ci

v2.20.0

• Clean up "lite" interfaces

• Add recall support

• Fix typos

• Add WebRTC error and warning logs to RingRTC logging

• Reference signalapp/webrtc@4638i

  • Reduce noise from error/warning logs

v2.19.2

• Introduce a "lite" part of RingRTC

v2.19.1

• Android: Add default enum for audio processing

v2.19.0

• Group Calls: Increase max send bitrate for large calls

• Group Calls: Use v2 frontend api and remove notion of endpoint_id

• Reference signalapp/webrtc@4638h

  • Android: Add Aec3/AecM switch
  • Windows: Workaround for multi-channel input

• Android: Add aec switch and remove legacy default

• Electron: Bubble up more DemuxIds

• Update Rust and dependencies

v2.18.1

• Fix group call rate constant

• iOS: Fix audio level api for group calls and tests

v2.18.0

• Update Audio Level API to specify desired interval

• Electron: Use WebCodecs to capture and send video

• Reference signalapp/webrtc@4638f

  • Group Calls: Enable 3rd spatial layer for video

• Update dependencies

v2.17.2

• Electron: Revert new state and fix issue with prering ended handling

v2.17.1

• Electron: Fix incoming call notifications for better call history

• Reference signalapp/webrtc@4638e

  • Mac: Fix stereo playout bug

• Update dependencies

v2.17.0

• Add API to get the incoming and outgoing audio levels

v2.16.1

• Node: Optimize use of CanvasVideoRenderer.renderVideoFrame

• Node: Update builds and logging

v2.16.0

• Group Calls: Leave via RTP instead of HTTP

• Group Calls: Don't use DTLS

• Group Calls: Increase default max receive rate

v2.15.0

• Android: Add audio processing options (to control AEC/NS)

• Android: Improve JNI/Rust interfaces

• Remove legacy Multi-Ring checks and hangup

v2.14.3

• Avoid handling RTP Data before accepted

• Reference signalapp/webrtc@4638c

  • Port crash fix

v2.14.2

• Don't terminate a 1:1 call because of transient RTP data error

• Reference signalapp/webrtc@4638b

  • Make it possible to share an APM between PeerConnections (ensures AEC/NS operation)

v2.14.1

• Desktop: Clear out the incoming video frame to avoid rendering old data

• iOS: Delete the dSYMs out of the built xcframework

v2.14.0

• Update WebRTC to 4638 (M95)

• Further improvements to WebRTC pointer management

• Replace DataChannel with direct RTP data

• Logging/Testing/Build improvements

v2.13.6

• Use SetAudioPlayout() function for group calls

v2.13.5

• Improve how WebRTC pointer is tracked across FFI

• Update Rust

• Update dependencies

• Update builds

v2.13.4

• Electron: Use Neon's Channel to avoid polling for events/logs

• Desktop: Allow logger to be initialized multiple times

• Enable the use of the SetAudioPlayout() function to start playout after accept

• Reference signalapp/webrtc@4389k

  • Initialize ADM playout before starting

v2.13.3

• iOS & Android: Pass PeerConnectionFactory down to Rust for group calls

• Desktop: Fix an issue generating device lists on Windows

• Add test client for group calls

• Adjust some interfaces between RingRTC and WebRTC

• Reference signalapp/webrtc@4389j

  • Cleanup iOS interfaces

v2.13.2

• Desktop: Update local preview source object correctly

• Android: Build Java against the same SDK/NDK that WebRTC uses

v2.13.1

• Desktop: Add support for auto-ended call timestamps

• Desktop: Formatting and other updates

• Android: Fix signature for new argument

v2.13.0

• Desktop: Option to use new or default audio device module on Windows

• Reference signalapp/webrtc@4389i

  • Support new Windows ADM

• Desktop: Support glare scenarios

• Request updated membership proof for group calls at least once a day

• Request bitrate constraints for group calls according to BandwidthMode

• Fix PeerConnectionFactory leaks

• iOS: Remove dependency on PromiseKit

• Android: Enable a Hardware AEC blocklist and fix a memory leak

• Android: Native PeerConnectionFactory uses AndroidNetworkMonitor and JavaAudioDeviceModule

v2.12.0

• Enable ICE continual gathering

• Add signaling for the removal of ICE candidates

• Add notifications for network route changes

• Adjust ringing timeout to 60 seconds

• iOS: Fixes to address resource leaks

• Reference signalapp/webrtc@4389h

  • iOS: AudioSession adjustments for volume issues

• Update builds and documentation

v2.11.1

• Update Group Ringing feature

v2.11.0

• Add Group Ringing feature

• Reference signalapp/webrtc@4389f

• Remove DTLS and SDP

v2.10.8

• Group Calling: Reduce notifications for active speakers

• Android: Modify NDK dependencies and use armv7 instead of arm

• Update logging

v2.10.7

• iOS: Add support for building for Catalyst

• iOS: Update builds

• Update dependencies

v2.10.6

• Electron: Use Buffer everywhere we used to use ArrayBuffer

• iOS: Update builds and tests to support M1 iOS simulator

• Update to Rust nightly

v2.10.5

• Screenshare: Allow screenshare without a camera

v2.10.4

• Screenshare: Add optimizations

v2.10.3

• Screenshare: Fix bandwidth for group call

v2.10.2

• Screenshare: Fix sending of status

v2.10.1

• Screenshare: Fixes for legacy clients

• Build Fixes: Support older Linux distros and other optimizations

• Reference signalapp/webrtc@4389c

v2.10.0

• Add Screensharing feature

• Electron: Support alternative target architectures

v2.9.7

• Electron: Rebuild (no functional changes)

v2.9.6

• Revert change for shared picture ID in WebRTC

v2.9.5

• Reference signalapp/webrtc@4389a

• Update dependencies

• Update builds and tests

v2.9.4

• Add statistics to monitor connection information

• Reference signalapp/webrtc@4183l

• Adjust logging and build issues

v2.9.3

• Electron: Update neon to use n-api runtime

• CI optimizations and lint improvements

v2.9.2

• Electron: Update to version 11

• Android: Add setOrientation() API

• Update contributing readme

v2.9.1

• Electron: Fix Windows build

v2.9.0

• Add very low bandwidth support for audio

• Remove SCTP

• Update documentation

v2.8.10

• Android: Fix JNI out of memory issues for large groups

v2.8.9

• Android: Fix memory issues for Direct Calling

• Electron: Fix issue where camera was not released

v2.8.8

• iOS: Fix issue when ending a Group Call

v2.8.7

• Group Calling: Fix issue with video resolution requests

v2.8.6

• Update Group Calling feature

• Reference signalapp/webrtc@4183h

v2.8.5

• Android: Improve stability for Group Calling

v2.8.4

• Update Group Calling feature

v2.8.3

• Update Group Calling feature

v2.8.2

• Update Group Calling feature

• Android: Add more devices to hardware encoder blacklist

• Reference signalapp/webrtc@4183g

v2.8.1

• Electron: Fix video track setting

v2.8.0

• Add Group Calling feature

• Reference signalapp/webrtc@4183f

• Update Rust dependencies

• Update builds and documentation

v2.7.4

• Electron: Fix debug build

v2.7.3

• Refactor calling code (non-functional improvements)

• Update opus codec settings

• Update builds and documentation

v2.7.2

• Electron: Expose more message types

v2.7.1

• Reference signalapp/webrtc@4183a

  • Electron: Should prevent early microphone access

• Electron: Do not stretch video if different resolution

v2.7.0

• Update Rust dependencies

• Implement "V4" protocol with protobufs; deprecate SDP

• Electron: Improve logging and handling of device selection

v2.6.0

• Reference signalapp/webrtc@4183

• Implement "V3" protocol; deprecate DTLS

• Fix offer-busy handling and support better glare experience

• Electron: Fix issue when sending busy would end current call

v2.5.2

• Electron: Mac minimum sdk and os set to 10.10

v2.5.1

• Electron: Improve device selection on Windows

• Fix message queue issue

v2.5.0

• Disable processing of incoming RTP before incoming call is accepted

• Electron: A/V device selection support

• Implement low bandwidth mode support

v2.4.3

• iOS: Update video support

v2.4.2

• Reference signalapp/webrtc@4147d

v2.4.1

• Fixes for release

v2.4.0

• Reference signalapp/webrtc@4147b

• Implement data channel support over RTP; deprecate SCTP

• Add audio statistics logging

• Minor fixes and improvements

v2.3.1

• Fix for call request support

• Fix to ensure hangups sent

v2.3.0

• Reference signalapp/webrtc@4103

• Add support for call request permissions

v2.2.0

• Reference signalapp/webrtc@4044g

• iOS: Remove 32-bit support, require 11.0 target

v2.1.1

• Reference signalapp/webrtc@4044f

v2.1.0

• Implement native interface

• Reference signalapp/webrtc@4044e

• Minor API improvements (call, proceed, receivedOffer)

v2.0.3

• Android: Use video sink for remote video stream

v2.0.2

• Reference signalapp/webrtc@4044d

v2.0.1

• Reference signalapp/webrtc@4044c

  • Fixes a call forking bug
  • Improves connectivity using PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS
  • Cherry picked updates from WebRTC

• Disable TURN port pruning

• Fix glare handling before connection

v2.0.0

• Add Multi-Ring feature

• Android: Fix video encoder crash on some devices

• Update build documentation

• Update Rust dependencies

v1.3.1

• Fix issue preventing some calls from ringing

v1.3.0

• Update build documentation

• Reference signalapp/webrtc@4044

v1.2.0

• Move to vendored WebRTC at signalapp/webrtc

• Reference signalapp/webrtc@3987, includes cherry picked updates from WebRTC 4044

v1.1.0

• Disable unused audio codecs and RTP header extensions

• Adjust settings and logging

• iOS: Minor optimizations

v1.0.2

• Cherry pick updates from WebRTC 4044

v1.0.1

• Android: improve logging

v1.0.0

• Add Call Manager component

v0.3.3

• Update WebRTC to 3987

• Update Rust dependencies

• iOS: build system improvements

v0.3.2

• iOS: Fix iOS 13 issue with camera capture

v0.3.1

• Android: Filter list of cameras when switching cameras

v0.3.0

• Update WebRTC to m79

• Android: Improve WebRTC debug logging

v0.2.0

• Improve logging on Android

• Build system improvements

v0.1.9

• Make termination a two-phase close and dispose operation

v0.1.8

• Improve logging on Android

• Patch WebRTC M78 for AudioRecord regression

v0.1.7

• Add integration tests

• Build system fixes and clean up

v0.1.6

• Android: Use an application supplied logging object

v0.1.5

• Update WebRTC to m78

• Add integration tests

• Build system fixes and clean up

v0.1.4

• Update Makefile targets for 'clean' and 'distclean'

• Simplify the IceReconnecting logic

• Remove non-critical DataChannel error callbacks

v0.1.3

• Add IceReconnectingState

v0.1.2

• iOS Support

• Update WebRTC to m77

v0.1.1

• Initial Release

• Based on WebRTC release m76