End-to-end stack for WebRTC. SFU media server and SDKs.
APACHE-2.0 License
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Published by biglittlebigben 6 months ago
This release changes the default behavior when creating or updating WHIP
ingress. WHIP ingress will now default to disabling transcoding and
forwarding media unchanged to the LiveKit subscribers. This behavior can
be changed by using the new enable_transcoding
available in updated
SDKs. The behavior of existing ingresses is unchanged.
Published by davidzhao over 1 year ago
Published by davidzhao over 2 years ago
Published by davidzhao over 2 years ago
Published by davidzhao over 2 years ago
Read more about the release on our blog
Published by davidzhao over 2 years ago
Published by davidzhao over 2 years ago
Published by davidzhao over 2 years ago
Published by davidzhao over 2 years ago
You can now use any custom TURN server with LiveKit, including third-party TURN services. By setting rtc.turn_servers in the config, LiveKit will configure all connected clients to use specified TURN servers.
Published by davidzhao over 2 years ago
In some cases, you may want to prevent rooms from being created automatically. (i.e. a streamer ended a session, so viewers should not be able to join)
It's possible to disable auto-creation behavior from configuration.
For long running sessions, the session may still be running after the client's connection token had expired. livekit-server will now automatically send clients refreshed tokens so clients will always have valid tokens to reconnect with.
Previously, RoomService would return a response before the operation is actually completed. This would lead to synchronization challenge from clients. In v0.15.3, RoomService behaves like you would expect: operation is completed before it returns.
Published by davidzhao almost 3 years ago
Ability to dynamically publish only layers that are being subscribed, significantly improving resource consumption on publishing clients. #295
Speaker updates will only be sent to subscribers. Other participants in the room will not receive updates. #280 #301
The ability to list rooms that match a particular set of inputs #290
Webhook callbacks will now include an unique ID as well as timestamp of the event. This enables idempotent processing of events on the listener side: #291
TrackInfo now includes a MIME type field that identifies the codec used (i.e. video/h264
or video/vp8
) #292
Ability to attach a participant name in addition to identity. This should be set inside of the JWT token #293
The ability to disable congestion control #305. This option could be set in configuration.
Published by davidzhao almost 3 years ago
We are introducing a significant improvement to the core SFU. It now monitors each subscriber's connection for congestion, and when detected, it controls bandwidth consumption for that subscriber by switching to lower simulcast layers at reduced bitrates. In case congestion gets worse, it'll prioritize audio and pause certain video tracks.
The addition of this feature enables LiveKit to work within highly congested networks while delivering an acceptable user experience.
When a participant connects without subscribe permission, server will use the publisher PeerConnection as the default #198
Connection quality updates now supports audio-only participants, with a MOS-style scoring.
Published by davidzhao almost 3 years ago
Lots of bugs squished and packed with improvements in the core SFU.
Published by davidzhao almost 3 years ago
v0.14 introduces detection for connection quality that's performed on the server side. The SFU will gather various metrics such as packet loss, publish and subscribe success rates to determine the quality of the participant's connections. #167
By performing this check on the server side, all LiveKit clients will receive quality information with minimal effort.
Connection quality information is sent to the participant itself, as well as any other participants it's subscribed to.
JS SDK v0.14.0 supports this feature, with Android, iOS, and Flutter to follow suit next week.
Published by davidzhao almost 3 years ago
Full Changelog: https://github.com/livekit/livekit-server/compare/v0.13.5...v0.13.7
Published by davidzhao about 3 years ago
Bugfix release
abs-send-time
#149Published by davidzhao about 3 years ago
In v0.13.3 mixing simulcast and non-simulcast tracks was broken. This release addresses that.
Published by davidzhao about 3 years ago
Introducing region-aware routing. When configured, LiveKit could route traffic to nodes that are closer to the end user. See multi region support #135 #141 (thanks @bekriebel)
We've revamped our recording capabilities so that it's close to a GA release. Notable changes include RTMP simulcast support, and moving the pipelines to GStreamer from FFmpeg. Requires livekit-recorder v0.3.1 or higher #137
Opus DTX is enabled by default in this version, significantly reducing audio bandwidth utilization.
Published by davidzhao about 3 years ago
Published by davidzhao about 3 years ago
support for protocol 3, where subscriber connection becomes the primary one. This speeds up session establishment for participants that aren't publishing.
When running in multi-node, server will now terminate gracefully, allowing remaining rooms on the node to drain. #116