Release Notes for LiveSwitch Server v1.23
1.23.0.1341
Release date: February 26, 2025
Client SDK
Bug Fixes
- [LS1-2745] [ClientSDK] Introduced a new error code {{WebSocketMediaServerChannelVideoStreamCapacityReached}} indicating that the capacity was reached due to the number of video participants present in a channel. Error {{WebSocketMediaServerChannelClientCapacityReached}} was removed.
- [LS1-2688] [MAUI] [Android] Fixed an issue that caused the Android app to crash due to a memory leak.
- [LS1-2570] [LS1-2772] [Media-Over-WebSockets] Fixed a bug that, under certain conditions, prevented participants from rejoining a session.
- [LS1-1792] [Media-Over-WebSockets] Fixed a bug when detecting if AudioEncoder is available in Safari Tech Preview which doesn’t completely support that feature yet.
- [LS1-2775] [Media-Over-WebSockets] Fixed a bug that, under certain conditions, prevented user video from being displayed properly when rejoining a session.
- [LS1-2743] [Media-Over-WebSockets] Fixed a bug that occured when rendering remote video when an invalid frame rate value is provided in the remote video encoding.
- [LS1-2644] [Media-Over-WebSockets] Fixed an issue preventing participants from joining in the receive-only audio-only mode.
- [LS1-2809] Addressed a recent regression in Chrome 133 where muting remote audio did not work.
Improvements
- [LS1-2750] [Android] [Examples] - Updated Android Example to indicate the best practice of application backgrounding on Android 15.
- [LS1-2771] [Examples] The .NET and Web examples have been updated to create an audio-only connection when creating a Media-Over-WebSockets with audio and video and receiving the error {{ErrorCode.WebSocketMediaServerChannelVideoStreamCapacityReached}}.
- [LS1-2552] Removed support for deprecated TLS 1.0 and 1.1 for WebRTC TCP Relay connections, Media-Over-WebSockets, and WebSocket Signalling. Continuing to use TLS 1.2.
- [LS1-2720] [ClientSDK] When publishing media via Media-Over-WebSockets, the default bitrate for audio encoding was lowered to 16kbps, and the default bitrate for video encoding was lowered to 256kbps.
- [LS1-2726] [ClientSDK] When receiving Media-Over-WebSockets, comfort noise packets will not be sent through the WebSocket to save some bandwidth.
- [LS1-2645] [LS1-2602] [Media-Over-WebSockets] Media handling for WebSocket connections has been moved to a Web Worker.
- [LS1-2563] [MAUI] [iOS] Added support for Hybrid Globalization on iOS.
- [LS1-2734] [Media-Over-WebSockets] Added round trip time tracking to the Media-Over-WebSockets transport layer and made available in Connection Stats.
- [LS1-2735] [Media-Over-WebSockets] Enhanced WebSocket connection monitoring and cleanup by adding round-trip time tracking, improving error context in messages, and implementing a more robust connection shutdown sequence.
- [LS1-2632] [Media-Over-WebSockets] When closing a connection fixed an issue where a Worker was not being terminated.
- [LS1-2680] [Web] Added Screen Wake Lock API support in Web Example to prevent screen locking during active WebRTC connections, particularly benefiting mobile users in audio-only conferences where screen locks could previously interrupt ongoing calls.
- [LS1-2724] Media-over-WebSockets now support unlimited audio-only connections while maintaining configurable video connection limits (defaulting to 10 video connections per client).
Media Server
Bug Fixes
- [LS1-2552] Removed support for deprecated TLS 1.0 and 1.1 for WebRTC TCP Relay connections, Media-Over-WebSockets, and WebSocket Signalling. Continuing to use TLS 1.2.
- [LS1-2686] Fixed a bug causing a null reference exception in the case where hex dump recording was enabled in a channel.
- [LS1-2699] Fixed an SDP bundling issue where the Media Server incorrectly set the connection address to 0.0.0.0 in IPv6-only deployments.
Improvements
- [LS1-2781] [WebSocket Media Server] Updated logging to ensure more contextual information is available.
Gateway
Breaking Changes
- [LS1-2740] [Breaking Change] Fixed OData routing validation warnings that appear at Gateway startup by simplifying route names. Routes using 'FM.LiveSwitch.Model.SipInboundMappingConfig' pattern will be removed in version 1.24 - please migrate to use 'SipInboundMappingConfig' pattern. Old routes: api/v2.0/FM.LiveSwitch.Model.SipInboundMappingConfig - New routes: api/v2.0/SipInboundMappingConfig
Bug Fixes
- [LS1-2552] Removed support for deprecated TLS 1.0 and 1.1 for WebRTC TCP Relay connections, Media-Over-WebSockets, and WebSocket Signalling. Continuing to use TLS 1.2.
- [LS1-2737] Enhanced stability and performance by fixing memory leaks in Redis message handling, improving cancellation token management, and adding better logging for WebSocket connections.
- [LS1-2729] Fixed misleading null reference error during PostgreSQL SSL handshake; users encountering the "Cannot validate remote certificate" error should verify their configuration includes "TrustedServerCertificateSha1Fingerprints" with valid certificate fingerprints.
Improvements
- [LS1-2752] Optimized logging of webhook errors.