LiveKit - Open source, distributed video/audio rooms over WebRTC

Overview

LiveKit - Open source, distributed video/audio rooms over WebRTC

LiveKit is an open source project that provides scalable, multi-user conferencing over WebRTC. It's designed to give you everything you need to build real time video/audio capabilities in your applications.

Features

  • Horizontally scalable WebRTC Selective Forwarding Unit (SFU)
  • Modern, full-featured client SDKs for JS, iOS, Android
  • Built for production - JWT authentication and server APIs
  • Robust networking & connectivity, over UDP & TCP
  • Easy to deploy - pure Go & single binary
  • Advanced features - speaker detection, simulcasting, selective subscription, moderation APIs.

Documentation & Guides

Docs & Guides at: https://docs.livekit.io

Try it live

Head to our playground and give it a spin. Build a Zoom-like conferencing app in under 100 lines of code!

SDKs & Tools

Client SDKs:

Server SDKs:

Tools:

Installing

From source

Pre-requisites:

  • Go 1.15+ is installed
  • GOPATH/bin is in your PATH
  • protoc is installed and in PATH

Then run

git clone https://github.com/livekit/livekit-server
cd livekit-server
./bootstrap.sh
mage

Docker

LiveKit is published to Docker Hub under livekit/livekit-server

Running

Creating API keys

LiveKit utilizes JWT based access tokens for authentication to all of its APIs. Because of this, the server needs a list of valid API keys and secrets to validate the provided tokens. For more, see Access Tokens guide.

Generate API key/secret pairs with:

./bin/livekit-server generate-keys

or

docker run --rm livekit/livekit-server generate-keys

Store the generate keys in a YAML file like:

APIwLeah7g4fuLYDYAJeaKsSE: 8nTlwISkb-63DPP7OH4e.nw.J44JjicvZDiz8J59EoQ+

Starting the server

In development mode, LiveKit has no external dependencies. You can start LiveKit by passing it the keys it should use in LIVEKIT_KEYS. LiveKit could also use a config file or config environment variable LIVEKIT_CONFIG

: " ./bin/livekit-server --dev ">
LIVEKIT_KEYS=": " ./bin/livekit-server --dev

or

: " \ livekit/livekit-server \ --dev \ --node-ip= ">
docker run --rm \
  -p 7880:7880 \
  -p 7881:7881 \
  -p 7882:7882/udp \
  -e LIVEKIT_KEYS=": " \
  livekit/livekit-server \
  --dev \
  --node-ip=<machine-ip>

When running with docker, --node-ip needs to be set to your machine's local IP address.

The --dev flag turns on log verbosity to make it easier for local debugging/development

Creating a JWT token

To create a join token for clients, livekit-server provides a convenient subcommand to create a development token. This token has an expiration of a month, which is useful for development & testing, but not appropriate for production use.

./bin/livekit-server --key-file <path/to/keyfile> create-join-token --room "myroom" --identity "myidentity"

Sample client

To test your server, you can use our example web client (built with our React component)

Enter generated access token and you are connected to a room!

Deploying for production

LiveKit is deployable to any environment that supports docker, including Kubernetes and Amazon ECS.

See deployment docs at https://docs.livekit.io/guides/deploy

Contributing

We welcome your contributions to make LiveKit better! Please join us on Slack to discuss your ideas and/or submit PRs.

License

LiveKit server is licensed under Apache License v2.0.

Comments
  • Client is missing published tracks in certain conditions

    Client is missing published tracks in certain conditions

    Describe the bug @bekriebel reported this in slack. When users are joining the room at the same time from across different regions (where latency is a bigger concern), sometimes clients are reporting TrackSubscriptionFailure events with the following logs:

    could not find published track PA_8qxvzPbj3G3R TR_QKFBfQyMBW6Q
    addSubscribedMediaTrack @ RemoteParticipant.js?f400:71
    eval @ RemoteParticipant.js?f400:78
    setTimeout (async)
    addSubscribedMediaTrack @ RemoteParticipant.js?f400:77
    eval @ RemoteParticipant.js?f400:78
    setTimeout (async)
    addSubscribedMediaTrack @ RemoteParticipant.js?f400:77
    eval @ RemoteParticipant.js?f400:78
    

    According to @bekriebel, he's able to produce this if the server instance is located in a node far way from him. (Frankfurt to Seattle)

    Server

    • Version: 0.13.6

    Client

    • SDK: JS
    • Version: 0.13.6
    bug 
    opened by davidzhao 31
  • Recording support?

    Recording support?

    The server looks very promising since it's supported with mobile and web SDKs.

    However I could not see any section about recording in the documentation. Do you plan to add support for recording + processing recordings?

    enhancement 
    opened by postacik 21
  • Audio tracks do not support stereo sound

    Audio tracks do not support stereo sound

    Describe the bug When using LiveKit, at least with the Javascript SDK, received audio tracks are always downsampled to mono, regardless of how they are sent.

    I would like to use LiveKit for some higher-quality audio purposes. For this, I think some changes may be needed to allow the audio quality to be controlled either by server configurations, or preferably client-side.

    Server

    • Version: [1.2.0]
    • Environment: Docker image VPS hosted & local dev

    Client

    • SDK: js
    • Version: 1.3.0

    To Reproduce Steps to reproduce the behavior:

    1. two clients are connected to room
    2. One client publishes an audio track, specifying AudioCaptureOptions and TrackPublishOptions: <track>.mediaStreamTrack.getSettings()
        AudioCaptureOptions = {
            autoGainControl: false,
            echoCancellation: false,
            noiseSuppression: false,
            channelCount: 2,
          }
          
          TrackPublishOptions = {
            audioBitrate: 256_000,
          };
    
    1. 2nd client receives the track
    2. Validate on the sending side that two channels are being captured: <track>.mediaStreamTrack.getSettings()
    {
        "autoGainControl": false,
        "channelCount": 2,
        "deviceId": "web-contents-media-stream://2884:4",
        "echoCancellation": false,
        "latency": 0.042666,
        "noiseSuppression": false,
        "sampleRate": 48000,
        "sampleSize": 16
    }
    
    1. See that on the receiving side, only a single channel is received:
    {
        "channelCount": 1,
        "deviceId": "TR_Asstr5zE4jUefc",
        "latency": 0.01,
        "sampleRate": 48000,
        "sampleSize": 16
    }
    

    Expected behavior If a track is published as stereo audio, it should also be received as such.

    Screenshots N/A

    Additional context I think at least part of this is coming from the fact that the SDPFmtpLine is not specifying stereo=1 here: https://github.com/livekit/livekit/blob/1371108a46ac090d9ae24bea012650e70086ee10/pkg/rtc/mediaengine.go#L13, though there may be more changes needed, especially to support the new opus/red codec. It may also be helpful to add maxaveragebitrate=510000 to ensure the maximum bitrate can be allowed.

    opened by bekriebel 18
  • Lots and lots of race conditions

    Lots and lots of race conditions

    Describe the bug building and running the server with -race reveals a lot of race conditions going on.

    Server

    • Version: 0.15.6
    • Environment: local dev

    Client

    • SDK: flutter
    • Version: 0.5.6

    To Reproduce Steps to reproduce the behavior:

    1. build the server with -race
    2. two clients are connected to room (one can be the go server sdk)
    3. See error

    Expected behavior No race conditions should occur

    Screenshots Lots. E.g.

    ==================
    WARNING: DATA RACE
    Write at 0x00c005dda910 by goroutine 114:
    github.com/livekit/livekit-server/pkg/rtc.(*MediaTrack).ToProto()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/mediatrack.go:103 +0xa6
    github.com/livekit/livekit-server/pkg/rtc.(*UpTrackManager).ToProto()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/uptrackmanager.go:76 +0x1c1
    github.com/livekit/livekit-server/pkg/rtc.(*ParticipantImpl).ToProto()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/participant.go:360 +0x698
    github.com/livekit/livekit-server/pkg/rtc.ToProtoParticipants()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/utils.go:51 +0xdd
    github.com/livekit/livekit-server/pkg/rtc.(*Room).broadcastParticipantState()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/room.go:705 +0x93
    github.com/livekit/livekit-server/pkg/rtc.(*Room).onTrackPublished()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/room.go:592 +0x84
    github.com/livekit/livekit-server/pkg/rtc.(*Room).onTrackPublished-fm()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/room.go:590 +0x6d
    github.com/livekit/livekit-server/pkg/rtc.(*ParticipantImpl).handleTrackPublished()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/participant.go:1405 +0xa3
    github.com/livekit/livekit-server/pkg/rtc.(*ParticipantImpl).mediaTrackReceived()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/participant.go:1393 +0x484
    github.com/livekit/livekit-server/pkg/rtc.(*ParticipantImpl).onMediaTrack()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/participant.go:920 +0xce
    github.com/livekit/livekit-server/pkg/rtc.(*ParticipantImpl).onMediaTrack-fm()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/participant.go:910 +0x4d
    github.com/pion/webrtc/v3.(*PeerConnection).onTrack·dwrap·70()
    /home/matthew/go/pkg/mod/github.com/pion/webrtc/[email protected]/peerconnection.go:459 +0x58
    
    Previous read at 0x00c005dda910 by goroutine 111:
    reflect.Value.Bool()
    /nix/store/j8zd71jnc6r7lhh45jwk9ywygr4w68c9-go-1.17.8/share/go/src/reflect/value.go:285 +0x51
    google.golang.org/protobuf/internal/impl.fieldInfoForScalar.func1()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/internal/impl/message_reflect_field.go:286 +0x28f
    google.golang.org/protobuf/internal/impl.(*messageState).Range()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/internal/impl/message_reflect_gen.go:48 +0x21e
    google.golang.org/protobuf/internal/order.RangeFields()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/internal/order/range.go:50 +0x21a
    google.golang.org/protobuf/encoding/protojson.encoder.marshalMessage()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/encoding/protojson/encode.go:223 +0x452
    google.golang.org/protobuf/encoding/protojson.encoder.marshalSingular()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/encoding/protojson/encode.go:304 +0x6c8
    google.golang.org/protobuf/encoding/protojson.encoder.marshalValue()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/encoding/protojson/encode.go:248 +0x18f
    google.golang.org/protobuf/encoding/protojson.encoder.marshalMessage.func1()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/encoding/protojson/encode.go:232 +0x213
    google.golang.org/protobuf/internal/order.RangeFields()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/internal/order/range.go:60 +0x3d9
    google.golang.org/protobuf/encoding/protojson.encoder.marshalMessage()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/encoding/protojson/encode.go:223 +0x452
    google.golang.org/protobuf/encoding/protojson.MarshalOptions.marshal()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/encoding/protojson/encode.go:136 +0x1cb
    google.golang.org/protobuf/encoding/protojson.MarshalOptions.Marshal()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/encoding/protojson/encode.go:110 +0xa4
    google.golang.org/protobuf/encoding/protojson.Marshal()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/encoding/protojson/encode.go:39 +0xa5
    github.com/livekit/protocol/webhook.(*notifier).Notify()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/webhook/notifier.go:50 +0x73
    github.com/livekit/livekit-server/pkg/telemetry.(*telemetryServiceInternal).notifyEvent.func1()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/telemetry/telemetryserviceinternalevents.go:278 +0x8f
    github.com/gammazero/workerpool.worker()
    /home/matthew/go/pkg/mod/github.com/gammazero/[email protected]/workerpool.go:243 +0x34
    github.com/gammazero/workerpool.startWorker·dwrap·6()
    /home/matthew/go/pkg/mod/github.com/gammazero/[email protected]/workerpool.go:234 +0x39
    

    In general, there seem to be a lot involving how protobuf msgs are constructed.

    ==================
    WARNING: DATA RACE
    Write at 0x00c005dda928 by goroutine 114:
    github.com/livekit/livekit-server/pkg/rtc.(*MediaTrack).ToProto()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/mediatrack.go:111 +0x217
    github.com/livekit/livekit-server/pkg/rtc.(*MediaTrackSubscriptions).AddSubscriber()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/mediatracksubscriptions.go:270 +0x153c
    github.com/livekit/livekit-server/pkg/rtc.(*MediaTrackReceiver).AddSubscriber()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/mediatrackreceiver.go:198 +0x3d4
    github.com/livekit/livekit-server/pkg/rtc.(*MediaTrack).AddSubscriber()
    <autogenerated>:1 +0x77
    github.com/livekit/livekit-server/pkg/rtc.(*UpTrackManager).AddSubscriber()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/uptrackmanager.go:127 +0x67b
    github.com/livekit/livekit-server/pkg/rtc.(*ParticipantImpl).AddSubscriber()
    <autogenerated>:1 +0xb9
    github.com/livekit/livekit-server/pkg/rtc.(*Room).onTrackPublished()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/room.go:615 +0x745
    github.com/livekit/livekit-server/pkg/rtc.(*Room).onTrackPublished-fm()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/room.go:590 +0x6d
    github.com/livekit/livekit-server/pkg/rtc.(*ParticipantImpl).handleTrackPublished()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/participant.go:1405 +0xa3
    github.com/livekit/livekit-server/pkg/rtc.(*ParticipantImpl).mediaTrackReceived()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/participant.go:1393 +0x484
    github.com/livekit/livekit-server/pkg/rtc.(*ParticipantImpl).onMediaTrack()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/participant.go:920 +0xce
    github.com/livekit/livekit-server/pkg/rtc.(*ParticipantImpl).onMediaTrack-fm()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/rtc/participant.go:910 +0x4d
    github.com/pion/webrtc/v3.(*PeerConnection).onTrack·dwrap·70()
    /home/matthew/go/pkg/mod/github.com/pion/webrtc/[email protected]/peerconnection.go:459 +0x58
    
    Previous read at 0x00c005dda928 by goroutine 129:
    google.golang.org/protobuf/internal/impl.pointer.Elem()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/internal/impl/pointer_unsafe.go:119 +0x3f7
    google.golang.org/protobuf/internal/impl.(*MessageInfo).marshalAppendPointer()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/internal/impl/encode.go:136 +0x3a9
    google.golang.org/protobuf/internal/impl.appendMessageSliceInfo()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/internal/impl/codec_field.go:485 +0x20e
    google.golang.org/protobuf/internal/impl.(*MessageInfo).marshalAppendPointer()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/internal/impl/encode.go:139 +0x482
    google.golang.org/protobuf/internal/impl.appendMessageSliceInfo()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/internal/impl/codec_field.go:485 +0x20e
    google.golang.org/protobuf/internal/impl.(*MessageInfo).marshalAppendPointer()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/internal/impl/encode.go:139 +0x482
    google.golang.org/protobuf/internal/impl.appendMessageInfo()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/internal/impl/codec_field.go:238 +0x190
    google.golang.org/protobuf/internal/impl.(*MessageInfo).initOneofFieldCoders.func4()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/internal/impl/codec_field.go:96 +0x105
    google.golang.org/protobuf/internal/impl.(*MessageInfo).marshalAppendPointer()
    google.golang.org/protobuf/internal/impl.(*MessageInfo).marshal()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/internal/impl/encode.go:107 +0xd0
    google.golang.org/protobuf/internal/impl.(*MessageInfo).marshal-fm()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/internal/impl/encode.go:100 +0xd4
    google.golang.org/protobuf/proto.MarshalOptions.marshal()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/proto/encode.go:163 +0x3b9
    google.golang.org/protobuf/proto.Marshal()
    /home/matthew/go/pkg/mod/google.golang.org/[email protected]/proto/encode.go:79 +0x59
    github.com/livekit/livekit-server/pkg/service.(*WSSignalConnection).WriteResponse()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/service/wsprotocol.go:84 +0x105
    github.com/livekit/livekit-server/pkg/service.(*RTCService).ServeHTTP.func2()
    /home/matthew/go/pkg/mod/github.com/livekit/[email protected]/pkg/service/rtcservice.go:230 +0x4d4
    

    There's not a shortage of them.

    Data races can lead to unexpected behaviour and very hard to debug problems. These should all be solved, most likely by judiciously adding mutexes, or using atomics.

    enhancement good first issue 
    opened by msackman 15
  • Data messages not sending in subscribe only mood

    Data messages not sending in subscribe only mood

    In my case I wanted user can only participants without sharing microphone or webcam. It's like subscribe only/listen only mood. But in that mood data messages aren't sending. I'm getting this error in my JS client:

    Uncaught (in promise) DOMException: An attempt was made to use an object that is not, or is no longer, usable

    Am I misunderstanding somewhere? But if that user connect with webcam then messages are sending as expected.

    opened by jibon57 15
  • Support TURN server without TLS cert

    Support TURN server without TLS cert

    Hi, we're trying to understand why is the TLS cert required for the TURN server.

    We're considering deploying in DigitalOcean with k8s and their load balancer. The load balancer will terminate the SSL before it reaches the cluster.

    However, following the Helm chart and also the server source code, it seems like the TLS cert is necessary. Is there any way to run TURN without the cert?

    opened by alvinthen 15
  • Add support for Redis Sentinel

    Add support for Redis Sentinel

    This is a rebase of #232 from @heilerich onto the current mainline. I don't have a sentinel install available at the moment to test myself, but my client will be testing this out later in the week.

    add client sentinel configuration and client init rewire add configuration sample

    opened by bekriebel 13
  • High activity rooms cause `

    High activity rooms cause `"error": "channel is full"` errors

    Hi,

    We had a live call with over 90 people. Things were going fine early on, but then we noticed users weren't able to publish video/audio, and suddenly some users who did publish would come through as a black screen. The error we were seeing over and over in the server looks like this:

    
    Apr 2, 2021 @ 14:27:11.777 | <14>1 2021-04-02T18:27:10.858904Z - - - - - 	/workspace/pkg/routing/redisrouter.go:311 |  
    -- | -- | --
    
      |   | Apr 2, 2021 @ 14:27:11.777 | <14>1 2021-04-02T18:27:10.858898Z - - - - - github.com/livekit/livekit-server/pkg/routing.(*RedisRouter).redisWorker |  
    
      |   | Apr 2, 2021 @ 14:27:11.777 | <14>1 2021-04-02T18:27:10.859405Z - - - - - 2021-04-02T18:27:10.859Z	ERROR	routing/redisrouter.go:311	error processing signal message	{"error": "channel is full"} |  
    
      |   | Apr 2, 2021 @ 14:27:11.777 | <14>1 2021-04-02T18:27:10.859435Z - - - - - github.com/livekit/livekit-server/pkg/routing.(*RedisRouter).redisWorker |  
    
      |   | Apr 2, 2021 @ 14:27:11.777 | <14>1 2021-04-02T18:27:10.859444Z - - - - - 	/workspace/pkg/routing/redisrouter.go:311 |  
    
      |   | Apr 2, 2021 @ 14:27:11.776 | <14>1 2021-04-02T18:27:10.854218Z - - - - - github.com/livekit/livekit-server/pkg/routing.(*RedisRouter).redisWorker |  
    
      |   | Apr 2, 2021 @ 14:27:11.776 | <14>1 2021-04-02T18:27:10.855512Z - - - - - github.com/livekit/livekit-server/pkg/routing.(*RedisRouter).redisWorker |  
    
      |   | Apr 2, 2021 @ 14:27:11.776 | <14>1 2021-04-02T18:27:10.855518Z - - - - - 	/workspace/pkg/routing/redisrouter.go:311 |  
    
      |   | Apr 2, 2021 @ 14:27:11.776 | <14>1 2021-04-02T18:27:10.854223Z - - - - - 	/workspace/pkg/routing/redisrouter.go:311 |  
    
      |   | Apr 2, 2021 @ 14:27:11.776 | <14>1 2021-04-02T18:27:10.858236Z - - - - - 	/workspace/pkg/routing/redisrouter.go:311 |  
    
      |   | Apr 2, 2021 @ 14:27:11.776 | <14>1 2021-04-02T18:27:10.856528Z - - - - - github.com/livekit/livekit-server/pkg/routing.(*RedisRouter).redisWorker |  
    
      |   | Apr 2, 2021 @ 14:27:11.776 | <14>1 2021-04-02T18:27:10.858231Z - - - - - github.com/livekit/livekit-server/pkg/routing.(*RedisRouter).redisWorker
    
    

    It looks like the channel used for the socket is filling up and once that happens, the connected user doesn't receive any new events and can't publish any new events either

    opened by atbe 12
  • Handling multiple IPs exposed on the same network interfaces

    Handling multiple IPs exposed on the same network interfaces

    @davidzhao @boks1971

    I have kept all the logs this time, please check the attachment. Here are some of the errors I've highlighted. The binary I'm running is compiled from the latest source code.

    Thanks again to the LiveKit team for providing such a great product.

    2022-04-07T13:15:15.015+0800 ERROR livekit rtc/signalhandler.go:24 could not handle answer {"room": "random_newl_2_1649308512137", "roomID": "RM_4yvUv4YjLvs6", "participant": "14986346", "pID": "PA_48uPq8yM9kvh", "error": "could not set remote description: InvalidModificationError: invalid proposed signaling state transition: stable->SetRemote(answer)->stable", "errorVerbose": "InvalidModificationError: invalid proposed signaling state transition: stable->SetRemote(answer)->stable\ncould not set remote description\ngithub.com/livekit/livekit-server/pkg/rtc.(*ParticipantImpl).HandleAnswer\n\t/root/livekit-server/pkg/rtc/participant.go:508\ngithub.com/livekit/livekit-server/pkg/rtc.HandleParticipantSignal\n\t/root/livekit-server/pkg/rtc/signalhandler.go:23\ngithub.com/livekit/livekit-server/pkg/service.(*RoomManager).rtcSessionWorker\n\t/root/livekit-server/pkg/service/roommanager.go:424\nruntime.goexit\n\t/usr/local/go/src/runtime/asm_amd64.s:1571"} github.com/livekit/livekit-server/pkg/rtc.HandleParticipantSignal /root/livekit-server/pkg/rtc/signalhandler.go:24 github.com/livekit/livekit-server/pkg/service.(*RoomManager).rtcSessionWorker /root/livekit-server/pkg/service/roommanager.go:424 2022-04-07T13:15:15.016+0800 INFO livekit service/rtcservice.go:225 source closed connection {"room": "random_newl_2_1649308512137", "participant": "14986346", "connID": "CO_Dyb7KRKVgoHi"}

    2022-04-06T23:10:18.447+0800 ERROR livekit.mux mux/mux.go:121 mux: ending readLoop dispatch error io: read/write on closed pipe {"room": "random_newl_2_1649257798784", "roomID": "RM_Puc9WfPrKE4n", "participant": "14956419", "pID": "PA_xa4imABDP6uT", "transport": "PUBLISHER"} github.com/pion/webrtc/v3/internal/mux.(*Mux).readLoop /root/godev/pkg/mod/github.com/pion/webrtc/[email protected]/internal/mux/mux.go:121

    2022-04-07T02:17:17.038+0800 ERROR livekit logger/logger.go:34 could not add down track {"participant": "14999873", "pID": "PA_5pDHbgHW2YPi", "error": "DownTrack already exist"} github.com/livekit/protocol/logger.Errorw /root/godev/pkg/mod/github.com/livekit/[email protected]/logger/logger.go:34 github.com/livekit/livekit-server/pkg/rtc.(*MediaTrackReceiver).AddSubscriber /root/livekit-server/pkg/rtc/mediatrackreceiver.go:209 github.com/livekit/livekit-server/pkg/rtc.(*UpTrackManager).AddSubscriber /root/livekit-server/pkg/rtc/uptrackmanager.go:126 github.com/livekit/livekit-server/pkg/rtc.(*Room).onTrackPublished /root/livekit-server/pkg/rtc/room.go:616 github.com/livekit/livekit-server/pkg/rtc.(*ParticipantImpl).handleTrackPublished /root/livekit-server/pkg/rtc/participant.go:1441 github.com/livekit/livekit-server/pkg/rtc.(*ParticipantImpl).mediaTrackReceived /root/livekit-server/pkg/rtc/participant.go:1429 github.com/livekit/livekit-server/pkg/rtc.(*ParticipantImpl).onMediaTrack /root/livekit-server/pkg/rtc/participant.go:925

    2022-04-07T05:04:32.750+0800 ERROR livekit rtc/transport.go:358 could not set local description {"room": "random_newl_2_1649279054705", "roomID": "RM_JDMFLq9Hhmw7", "participant": "14988207", "pID": "PA_seQeypW9vamq", "transport": "SUBSCRIBER", "error": "InvalidStateError: connection closed"} github.com/livekit/livekit-server/pkg/rtc.(*PCTransport).createAndSendOffer /root/livekit-server/pkg/rtc/transport.go:358 github.com/livekit/livekit-server/pkg/rtc.(*PCTransport).CreateAndSendOffer /root/livekit-server/pkg/rtc/transport.go:297 github.com/livekit/livekit-server/pkg/rtc.(*PCTransport).Negotiate.func1 /root/livekit-server/pkg/rtc/transport.go:288 2022-04-07T05:04:32.750+0800 ERROR livekit rtc/transport.go:289 could not negotiate {"room": "random_newl_2_1649279054705", "roomID": "RM_JDMFLq9Hhmw7", "participant": "14988207", "pID": "PA_seQeypW9vamq", "transport": "SUBSCRIBER", "error": "InvalidStateError: connection closed"} github.com/livekit/livekit-server/pkg/rtc.(*PCTransport).Negotiate.func1 /root/livekit-server/pkg/rtc/transport.go:289

    2022-04-07T06:08:05.528+0800 ERROR livekit.pc [email protected]/peerconnection.go:1582 Incoming unhandled RTP ssrc(2731095130), OnTrack will not be fired. incoming SSRC failed Simulcast probing {"room": "random_newl_2_1649282864935", "roomID": "RM_embtrPdkqKjF", "participant": "14974334", "pID": "PA_ZRUNQot4VdEd", "transport": "PUBLISHER"} github.com/pion/webrtc/v3.(*PeerConnection).undeclaredMediaProcessor.func1.1 /root/godev/pkg/mod/github.com/pion/webrtc/[email protected]/peerconnection.go:1582

    2022-04-07T06:39:35.790+0800 ERROR livekit.sctp [email protected]/association.go:2427 [0xc002ec9180] retransmission failure: T1-init {"room": "random_newl_2_1649284430786", "roomID": "RM_wKVjGU9kvq84", "participant": "14732048", "pID": "PA_YpHsJfdx7fTu", "transport": "SUBSCRIBER"} github.com/pion/sctp.(*Association).onRetransmissionFailure /root/godev/pkg/mod/github.com/pion/[email protected]/association.go:2427 github.com/pion/sctp.(*rtxTimer).start.func1 /root/godev/pkg/mod/github.com/pion/[email protected]/rtx_timer.go:158

    2022-04-07T00:22:52.557+0800 ERROR livekit rtc/mediatracksubscriptions.go:407 could not write RTCP {"room": "random_newl_2_1649262155533", "roomID": "RM_tvXi7pjd2Dtr", "participant": "14994548", "pID": "PA_MUCz9nMyTHgR", "trackID": "TR_VCLqHaSgq2cp7k", "error": "io: read/write on closed pipe"}

    livekit-server.log.tar.gz

    opened by CensorKoo 11
  • Quickly muting and unmuting an audio track adds a 1-2 second delay to the audio

    Quickly muting and unmuting an audio track adds a 1-2 second delay to the audio

    Describe the bug When an audio track is disabled and re-enabled quickly, it will often cause a delay to be introduced into the audio track. The delay is a noticeable 1-2 seconds and it causes the audio track to be off-sync with the video track if both are published. If multiple clients are connected, the delay often arises in all of the connected clients, but not always the same amount of delay. If either the published or subscriber disconnects and reconnects, the delay will be fixed.

    Server

    • Version: [0.14.2]
    • Environment: Docker image on Fly.io

    Client

    • SDK: js
    • Version: 0.14.3

    To Reproduce Steps to reproduce the behavior:

    1. Connect two clients to a room using the client-js sample app. This is easiest seen in two windows on one machine, but reproduces across multiple machines as well. It is best to use a low-latency server so normal audio has minimal delay.
    2. Mute the audio of one client so you are only testing from a single source, referred to from here on out as the sender.
    3. Notice that on the receiver has minimal delay and the audio is in sync with the video stream (I tested by snapping my fingers for a good indicator).
    4. Use the Disable Audio button to mute track on the sender, wait a few seconds and re-enable the audio. Note that no delay is introduced.
    5. Disable the audio track and then quickly enable it again (double tap the button)
    6. Check the audio delay on the receiver, often times a 1-2 second delay will be introduced

    It may take multiple tries of step 5 to get the issue to reproduce. If I tap the button 10 times in quick succession, (5 mute/unmute cycles), it reproduces for me almost every time. It's possible that the issue also reproduces when not muting and unmuting quickly, but I was not able to observe this.

    Expected behavior Audio delay should not be introduced by disabling/enabling the audio track. Audio and video should remain in sync.

    Screenshots N/A

    Additional context It's possible this is a client-js bug and not livekit-server, but I do not have a good setup to test this at the moment. I did not notice anything relevant in the client or server-side logs.

    opened by bekriebel 11
  • Lingering connections even after a client socket is disconnected

    Lingering connections even after a client socket is disconnected

    I'm not sure what to include for this when it comes to logs because I can't pinpoint this down to a specific error

    When there is a flood of new connections, some of the participants never get cleaned out when their socket disconnects

    I ran this command to see how many connections were open on the server:

    # cat /proc/net/tcp | wc -l
    25
    

    Although one of the rooms I'm currently in reports 926 participants and the redis records for all the disconnected users are still there.

    This causes a miscount in how many participants are actually in the call that cannot be cleaned up without manually clearing redis, or starting a new room.

    Please let me know if there's something I can include to make this bug more clear. I may be able reproduce it on the livekit sample if I spam the server with lots of connections all at once.

    The only thing I see in the logs repeatedly is

    2021-04-05T03:51:51.950Z        ERROR   routing/redisrouter.go:317      error processing signal message{"error": "channel is full"}
    github.com/livekit/livekit-server/pkg/routing.(*RedisRouter).redisWorker
            /workspace/pkg/routing/redisrouter.go:317
    2021-04-05T03:51:51.950Z        ERROR   routing/redisrouter.go:317      error processing signal message{"error": "channel is full"}
    github.com/livekit/livekit-server/pkg/routing.(*RedisRouter).redisWorker
            /workspace/pkg/routing/redisrouter.go:317
    

    when this occurred

    opened by atbe 11
  • Minor clean up of media track & friends module

    Minor clean up of media track & friends module

    • ClearAllReceivers instead of InitiateClose. Basically, it is clearing all receivers and removing all subscribers. So, use that.
    • Use state in MediaTrackReceiver. Although, it is only open or closed and a isClosed boolean should suffice, mirroring remote tracks for consistency
    opened by boks1971 0
  • Update golang.org/x/sync digest to 8fcdb60

    Update golang.org/x/sync digest to 8fcdb60

    Mend Renovate

    This PR contains the following updates:

    | Package | Type | Update | Change | |---|---|---|---| | golang.org/x/sync | require | digest | 7f9b162 -> 8fcdb60 |


    Configuration

    📅 Schedule: Branch creation - "on the first day of the month" (UTC), Automerge - At any time (no schedule defined).

    🚦 Automerge: Disabled by config. Please merge this manually once you are satisfied.

    Rebasing: Whenever PR becomes conflicted, or you tick the rebase/retry checkbox.

    🔕 Ignore: Close this PR and you won't be reminded about this update again.


    • [ ] If you want to rebase/retry this PR, click this checkbox.

    This PR has been generated by Mend Renovate. View repository job log here.

    opened by renovate[bot] 0
  • Error

    Error "Could not connect PeerConnection after timeout"

    Hi,

    Since last update I got a error for every client outside my LAN who try to connect to a room a/v of my livekit server (self-host) : "Could not connect PeerConnection after timeout". If I try inside my LAN it's ok (192.168.x.x). I have try to disable, my UFW, my fail2ban and the firewall of my Internet Box, activate the DMZ. I have make a new fresh install of livekit. I have other service (https) without any problems to connect. All service are NAT forwardings.

    Repoduce the error : Go to : https://example.livekit.io/ or https://livekit.io/connection-test LiveKit URL : https://my.hombrew.dns/ Token : Generate with "docker run --rm -v$PWD:/output livekit/generate --local"

    => Ok for me (LAN), error for client (INTERNET).

    For information : Old post for the configuration with docker "https://github.com/livekit/livekit/issues/687"

    Thanks for any help, I become mad..

    Configuration :

    cat docker-compose.yaml
    version: '3'
    
    services:
      livekit:
        image: livekit/livekit-server:latest
        command: --config /etc/livekit.yaml --node-ip xxx
        container_name: livekit
        restart: unless-stopped
        security_opt:
          - no-new-privileges:true
        ports:
          - "7881:7881"
          - "7882:7882/udp"
        networks:
          - proxy
        volumes:
          - ./livekit.yaml:/etc/livekit.yaml:ro
        labels:
          - "traefik.enable=true"
          - "traefik.http.routers.livekit.entrypoints=web,websecure"
          - "traefik.http.routers.livekit.rule=Host(`xxx.xxx.xxx`)"
          - "traefik.http.routers.livekit.service=livekit"
          - "traefik.http.routers.livekit.tls=true"
          - "traefik.http.routers.livekit.tls.domains[0].main=xxx.xxx.xxx"
          - "traefik.http.routers.livekit.tls.domains[0].sans=*-*.xxx.xxx"
          - "traefik.http.services.livekit.loadbalancer.server.port=7880"
          - "traefik.docker.network=proxy"
    
    networks:
      proxy:
        external: true
    
    cat livekit.yaml
    port: 7880
    rtc:
        udp_port: 7882
        tcp_port: 7881
        use_external_ip: false
    keys:
        XXX: XXX
    logging:
        json: false
        level: info
    
    sudo ufw status
    Status: active
    To                         Action      From
    --                         ------      ----
    XXX
    7881/tcp                   ALLOW       Anywhere
    7882/udp                   ALLOW       Anywhere
    XXX
    7881/tcp (v6)              ALLOW       Anywhere (v6)
    7882/udp (v6)              ALLOW       Anywhere (v6)
    

    NAT : Capture

    Error from https://livekit.io/connection-test
    Connecting to signal connection via WebSocket
    
    Connected to server, version 1.2.3.
    
    Establishing WebRTC connection
    
    Warning: ports need to be open on firewall in order to connect.
    
    Error: could not connect after timeout
    
    SKIPPED: Can connect via TURN
    
    Warning: No TURN servers configured.
    
    Can publish audio
    
    Error: could not connect after timeout
    
    Can publish video
    
    Error: could not connect after timeout
    
    Resuming connection after interruption
    
    Warning: could not connect after timeout
    
    Fail
    

    Log : livekit_log.txt

    opened by CaosFR 9
  • How to implement C# client SDK in Unity for windows build?

    How to implement C# client SDK in Unity for windows build?

    Hi, I want to use LiveKit in unity for windows build. Can you help me implement C# SDK? Is it possible?

    I founded this Unity package for WebRTC (experimental): https://docs.unity3d.com/Packages/[email protected]/manual/index.html

    Can I integrate LiveKit with this package for windows or other platforms?

    opened by EbiPenMan 1
  • RedisStore: make UnlockRoom atomic

    RedisStore: make UnlockRoom atomic

    according to redis manual, releasing lock(get, check and delete redis key) should be an atomic operation (via lua script)

    if redis.call("get",KEYS[1]) == ARGV[1] then
        return redis.call("del",KEYS[1])
    else
        return 0
    end
    

    otherwise, we may encounter this situation

    • client A tried to get lock and succeed

    • the lock is about to expired

    • client A tried to unlock

    • redis get and check token pass

      val, err := s.rc.Get(s.ctx, key).Result()
      	if err == redis.Nil {
      		// already unlocked
      		return nil
      	} else if err != nil {
      		return err
      	}
      
      	if val != uid {
      		return ErrRoomUnlockFailed
      	}
      
    • unfortunately, beforing deleting the lock, client A blocked, could be paused by gc or kernel schedule (everything is possible in distributed system!)

    • the lock expired while client A is still blocked

    • client B tried to get lock and succeeded

    • client A woke up

    client A deleted the lock

    return s.rc.Del(s.ctx, key).Err()
    
    opened by MaxnSter 1
Releases(v1.2.3)
  • v1.2.3(Sep 27, 2022)

    Several stability improvements / edge case handling along with stereo Opus support

    Added

    • Supervisor framework to improve edge case & error handling #1005 #1006 #1010 #1017
    • Support for stereo Opus tracks #1013
    • Allow CORS responses to be cached to allow faster initial connection #1027

    Fixed

    • Fixed SSRC mix-up for simulcasted tracks during session resume #1014
    • Fixed screen corruption for non-simulcasted tracks, caused by probing packets #1020
    • Fixed Handling of Simple NALU keyframes for H.264 #1016
    • Fixed TCPMux & UDPMux mixup when multiple host candidates are offered #1036

    Changed

    • Webhook requests are now using Content-Type application/webhook+json to avoid eager JSON parsing #1025
    • Don't automatically add STUN servers when explicit Node IP has been set #1023
    • Automatic TCP and TURN/TLS fallback is now enabled by default #1033

    Removed

    • Fully removed references to VP9. LiveKit is focused on AV1. #1004
    Source code(tar.gz)
    Source code(zip)
    checksums.txt(591 bytes)
    livekit_1.2.3_linux_amd64.tar.gz(7.83 MB)
    livekit_1.2.3_linux_arm64.tar.gz(7.20 MB)
    livekit_1.2.3_linux_armv7.tar.gz(7.35 MB)
    livekit_1.2.3_windows_amd64.zip(7.74 MB)
    livekit_1.2.3_windows_arm64.zip(7.13 MB)
    livekit_1.2.3_windows_armv7.zip(7.34 MB)
  • v1.2.1(Sep 13, 2022)

    v1.2.1 is a bugfix release

    Added

    • Accepts existing participant ID on reconnection attempts #988

    Fixed

    • Fixed ICE restart during candidate gathering #963
    • Ensure TrackInfoAvailable is fired after information is known to be ready #967
    • Fixed layer handling when publisher pauses layer 0 (FireFox is has a tendency to pause lowest layer) #984
    • Fixed inaccurate participant count due to storing stale data #992

    Changed

    • Protect against looking up dimensions for invalid spatial layer #977
    • Improvements around migration handling #979 #981 #982 #995
    • Consistent mapping between VideoQuality, rid, and video layers #986
    • Only enable TCP/TURN fallback for supported clients #997
    Source code(tar.gz)
    Source code(zip)
    checksums.txt(591 bytes)
    livekit_1.2.1_linux_amd64.tar.gz(7.80 MB)
    livekit_1.2.1_linux_arm64.tar.gz(7.17 MB)
    livekit_1.2.1_linux_armv7.tar.gz(7.32 MB)
    livekit_1.2.1_windows_amd64.zip(7.72 MB)
    livekit_1.2.1_windows_arm64.zip(7.10 MB)
    livekit_1.2.1_windows_armv7.zip(7.31 MB)
  • v1.2.0(Aug 27, 2022)

    We are excited to introduce two major quality improvement efforts in this release!

    First, we've added support for Opus RED. This brings a major step up to audio quality over lossy networks. Even with 50% packet loss, audio still comes out clear and free of robotic artifacts.

    The other major feature is the ability to automatically switch to TCP or TURN/TLS when UDP connection is failing. Certain routers/firewalls would initially let through UDP packets, but then either rate limit or block them from continuing. We can detect these scenarios and switch to TCP for that participant. To enable this, set rtc.allow_tcp_fallback: true in config.

    Added

    • Support for NACK with audio tracks #829
    • Allow binding HTTP server to specific address, binds to localhost in dev mode #831
    • Packet stats from TC (#832)
    • Automatic connectivity fallback to TCP & TURN (#872 #873 #874 #901 #950)
    • Support for client-side ping/pong messages (#871)
    • Support for setCodecPreferences for clients that don't implement it (#916)
    • Opus/RED support: redundant audio transmission is enabled by default (#938 #940)

    Fixed

    • Fixed timing issue in DownTrack.Bind/Close (#833)
    • Fixed TCPMux potentially blocking operations (#840)
    • Fixed ICE restart while still in ICE gathering (#895)
    • Fixed Websocket connection hanging if node isn't available to accept connection (#923)
    • Fixed ICE restart/resume in single node mode (#930)
    • Fixed client disconnected in certain conditions after ICE restart (#932)

    Changed

    • Move to synchronously handle subscriber dynacast status (#834)
    • Retransmit DD extension in case packets were missed (#837)
    • Clean up stats workers (#836)
    • Use TimedVersion for subscription permission updates (#839)
    • Cleaned up logging (#843 #865 #910 #921)
    • track_published event now includes the participant's ID and identity (#846)
    • Improve synchronization of track publishing/unpublish path (#857)
    • Don't re-use transceiver when pending negotiation (#862)
    • Dynacast and media loss proxy refactor (#894 #902)
    • PCTransport refactor (#907 #944)
    • Improve accuracy of connection quality score (#912 #913)
    • Docker image now builds with Go v1.19
    Source code(tar.gz)
    Source code(zip)
    checksums.txt(591 bytes)
    livekit_1.2.0_linux_amd64.tar.gz(7.78 MB)
    livekit_1.2.0_linux_arm64.tar.gz(7.16 MB)
    livekit_1.2.0_linux_armv7.tar.gz(7.31 MB)
    livekit_1.2.0_windows_amd64.zip(7.70 MB)
    livekit_1.2.0_windows_arm64.zip(7.09 MB)
    livekit_1.2.0_windows_armv7.zip(7.30 MB)
  • v1.1.2(Jul 11, 2022)

    Added

    • Returns reason when server disconnects a client (#801 #806)
    • Allow livekit-server to start without keys configuration (#788)
    • Added recovery from negotiation failures (#807)

    Fixed

    • Fixed synchronization issues with Dynacast (#779 #802)
    • Fixed panic due to timing in Pion's ICE agent (#780)
    • ICELite is disabled by default, improving connectivity behind NAT (#784)
    • Fixed EgressService UpdateLayout (#782)
    • Fixed synchronization bugs with selective subscriptions & permissions (#796 #797 #805 #813 #814 #816)
    • Correctly recover from ICE Restart during an negotiation attempt (#798)

    Changed

    • Improved Transceiver re-use to avoid renegotiation (#785)
    • Close room if recorder is the only participant left (#787)
    • Improved connection quality score stability & computation (#793 #795)
    • Set layer state to stopped when paused (#818)

    Removed

    • Removed deprecated RecordingService - Egress should be used instead (#811)
    Source code(tar.gz)
    Source code(zip)
    checksums.txt(591 bytes)
    livekit_1.1.2_linux_amd64.tar.gz(7.36 MB)
    livekit_1.1.2_linux_arm64.tar.gz(6.81 MB)
    livekit_1.1.2_linux_armv7.tar.gz(6.92 MB)
    livekit_1.1.2_windows_amd64.zip(7.35 MB)
    livekit_1.1.2_windows_arm64.zip(6.80 MB)
    livekit_1.1.2_windows_armv7.zip(6.97 MB)
  • v1.1.0(Jun 22, 2022)

    [1.1.0] - 2022-06-21

    Added

    • Add support for Redis Sentinel (#707)
    • Track participant join total + rate in node stats (#741)
    • Protocol 8 - fast connection support (#747)
    • Simulate switch candidate for network connectivity with poor UDP performance (#754)
    • Allow server to disable codec for certain devices (#755)
    • Support for on-demand multi-codec publishing (#762)

    Fixed

    • Fixed unclean DownTrack close when removed before bound. (#736)
    • Do not munge VP8 header in place - fixes video corruption (#763)

    Changed

    • Reintroduce audio-level quantization to dampen small changes (#732)
    • Allow overshooting maximum when there are no bandwidth constraints. (#739)
    • Improvements to upcoming multi-codec simulcast (#740)
    • Send layer dimensions when max subscribed layers change (#746)
    • Use stable TrackID after unpublishing & republishing (#751)
    • Update egress RPC handler (#759)
    • Improved connection quality metrics (#766 #767 #770 #771 #773 #774 #775)
    Source code(tar.gz)
    Source code(zip)
  • v1.0.2(May 28, 2022)

    [1.0.2] - 2022-05-27

    What's changed

    • Fixed edge cases where streams were not allocated (#701)
    • Fixed panic caused by concurrent modifications to stats worker map (#702 #704)
    • Batched subscriber updates to reduce noise in large rooms (#703 #729)
    • Fixed potential data race conditions (#706 #709 #711 #713 #715 #716 #717 #724 #727)
    • /debug/pprof endpoint when running in development mode (#708)
    • When audio tracks are muted, send blank frames to induce silence (#710)
    • Fixed stream allocator not upgrading streams after downgrading (#719)
    • Fixed repeated AddSubscriber potentially ignored (#723)
    • Fixed ListEgress API sometimes returning not found (#722)
    Source code(tar.gz)
    Source code(zip)
  • v1.0.0(May 18, 2022)

    1.0 Release!

    Read more about the release on our blog

    What's new

    • Improved stats around NACKs (#664)
    • Internal structures in preparation for AV1 SVC support (#669)

    What's changed

    • Supports participant identity in permissions API (#633)
    • Fixed concurrent access of stats worker map (#666 #670)
    • Do not count padding packets in stream tracker (#667)
    • Fixed TWCC panic under heavy packet loss (#668)
    • Change state to JOINED before sending JoinResponse (#674)
    • Improved frequency of stats update (#673)
    • Send active speaker update during initial subscription (#676)
    • Updated DTLS library to incorporate security fixes (#678)
    • Improved list-nodes command (#681)
    • Improved screen-share handling in StreamTracker (#683)
    • Inject silence opus packets when muted (#682)
    Source code(tar.gz)
    Source code(zip)
  • v0.15.7(May 3, 2022)

    Features

    • Supports IPv6 networks by default #571
    • NodeSelector to support sort options #599 (thanks @bekriebel)
    • Supports adaptiveStream flag - starts stream in a paused state for adaptive stream capable clients #623 #631

    Changes

    • Disallow identity that is an empty string #580
    • Returns Participant.region to clients for multi-region deployments #585
    • TrackIDs indicates the type and source of track #586
    • Reduce contention during session starts #614
    • Improved docker connectivity with using srflx candidates #624
    • Exposes Participant.isPublisher to indicate publisher vs subscriber #643
    • Reduced memory usage of internal stats accounting #645
    • Callback improvements #655 #652 #651

    Bugfixes

    • Improved available layer tracking #575
    • Avoid locking in callback #588
    • Prevent negative timestamp difference #595
    • Avoid locking when flushing DownTrack #594
    • Fixes server locking up sometimes with TCP connections #606
    • Fixed dynacast settings lost after ICE restart #620
    • Increase sizes of message queues to ensure delivery reliability #638 #641
    • Fixed connections silently disconnecting due to aggressive nomination #642 #644
    • Fixed memory leaks in MessageChannel #646
    • Correctly determine number of CPUs in a non-linux environment #653
    • Fixed node-ip parameter being ignored, leading to connectivity issues in local env #661
    Source code(tar.gz)
    Source code(zip)
  • v0.15.6(Mar 29, 2022)

    Features

    • Enable the ability to filter out certain network interfaces to avoid duplicate candidates #502
    • Support for Redis TLS connections #482 (thanks @alexbeattie42)
    • Client configuration system for detecting device specific issues/limitations #452
    • Supports TrackPublished and TrackUnpublished webhooks, along with other webhooks improvements #535
    • Unpublish tracks automatically when publish permissions are revoked for a participant #545
    • Support for upcoming Egress service

    Changes

    • Quality improvements to congestion controller: more stable stream allocations #532 #544 #549 #551 #557
    • Congestion controller now defaults to not pausing video by default #554
    • Passes serverRegion back when a participant is joining #479
    • Improved handling of simulcasted screenshares #503
    • Speaker events are now only emitted for audio level changes on microphone tracks #553 (thanks @sibis)
    • Dynacast now throttles downgrade events to reduce unnecessary changes #556 #558
    • Enable size limits to room & participant metadata #566

    Bugfixes

    • Fixed potential race condition when creating RTC room #485 (thanks @b20132367)
    • Fixed panic when writing to closed RTCP channel #495
    • Fixed RTCP worker stopped due to nil packets #504
    • Prevent StreamTracker from declaring base layer video to be stopped incorrectly #530
    • Fixed connection stall when non-primary peer connection becomes disconnected #537 #548
    • Fixed timestamp jump upon layer switch #543
    • Fixed deadlocks within Pion mux with 3.1.27 #555
    • Compatibility with Go 1.18
    • Fixed connectivity with Firefox when no tracks are subscribed #565
    • Always re-issue token to prevent client disconnecting before refresh interval #569
    Source code(tar.gz)
    Source code(zip)
  • v0.15.5(Mar 2, 2022)

    Changes in 0.15.5

    • Improved default speaker detection sensitivity #427
    • Improved handling of network congestion #429 #433
    • Use padding to probe instead of higher layers #434
    • Throttle retransmissions to prevent RTX storm #440
    • Include NACK ratio in congestion detection #443
    • Fixed stream update sending incorrect publisher ID #432
    • Fixed issue where screensharing would pause with Chrome 97+ #437
    • Fixed allocation retry in TURN #445
    • Avoid deadlocks in room close #451
    • Close websocket instead of hang on connection failure #458
    • Disable default node limits #472
    Source code(tar.gz)
    Source code(zip)
  • v0.15.4(Feb 9, 2022)

    Bring your own TURN servers #409

    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.

    Bugfixes and improvements

    • Fixed deadlock causing underlying buffer to become full #413
    • Disabled SRTP replay protection to improve Firefox compatibility #394
    • Improve connection reliability over links with longer latency #405
    • Lower DTLS retransmission interval to improve initial connection speed #414
    • Disable use of ICELite by default #397 #408
    • Smoother dynamic broadcast transition #389
    • Thread safety with map traversal #388
    • Use a single buffered channel for RTCP messages #418
    • Use message versions to better prevent race #399
    • Simplification/improvement of sfu.Buffer #398 #402
    • Improved context with logging #391 #416
    Source code(tar.gz)
    Source code(zip)
  • v0.15.3(Jan 29, 2022)

    Ability to disable room auto-creation #361

    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.

    Automatic token refresh #365

    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.

    RoomService returns only after operation is complete #362

    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.

    Other Changes

    • Use ICE-Lite to let clients take controlling role #322
    • Code refactoring for improved re-use
    • Simulate scenarios to allow client tests #330
    • Prevent multiple resume notifications for track changes #334
    • Enable CORS for RoomService #335
    • Integrated logging with Pion (#341)
    • Fixed missing tracks during long latency links #346
    • Fixed race condition when the room is closing when another participant is joining at the same time #370
    • Improved transceiver-reuse, avoid sending potentially un-decodable frames to clients #382
    • Honor auto-subscribe for participants who's given subscribe permissions after joining #381
    Source code(tar.gz)
    Source code(zip)
  • v0.15.2(Jan 6, 2022)

    Changes in 0.15.2

    Dynacast

    Ability to dynamically publish only layers that are being subscribed, significantly improving resource consumption on publishing clients. #295

    Scoped speaker updates

    Speaker updates will only be sent to subscribers. Other participants in the room will not receive updates. #280 #301

    List rooms by name(s)

    The ability to list rooms that match a particular set of inputs #290

    Webhook event uuid and timestamp

    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

    Track MIME type

    TrackInfo now includes a MIME type field that identifies the codec used (i.e. video/h264 or video/vp8) #292

    Participant name

    Ability to attach a participant name in addition to identity. This should be set inside of the JWT token #293

    Configurable congestion control

    The ability to disable congestion control #305. This option could be set in configuration.

    Bugfixes

    • Close RTCP channel after published tracks are fully closed #286
    • Fix rare deadlock when waiting on a participant that stopped publishing #288
    • Handle IP resolution failure instead of silently failing #289
    • Fixed recording service requests for specific URL 7b0db1f3446c7bfa13ed7080d5a5b7435ad58110
    Source code(tar.gz)
    Source code(zip)
  • v0.15.1(Dec 22, 2021)

    Downstream congestion control

    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.

    Publisher-only mode

    When a participant connects without subscribe permission, server will use the publisher PeerConnection as the default #198

    Improved connection quality updates

    Connection quality updates now supports audio-only participants, with a MOS-style scoring.

    Other changes

    • bugfix: participant is always present when webhook is triggered #200
    • Room.numParticipant will reflect number of participants in a room #199
    • configurable limit for max number of tracks before a node marks itself unavailable #197
    • bugfix: send correct simulcast information in TrackInfo #218
    • docker image uses Go 1.17 #223
    • support for updated recorder protocol (to match livekit-recorder v0.3.12)
    • support for custom simulcast layers #238
    • cleaned up logging to include context #252
    • external TURN/TLS termination #168
    • improve video quality in simulcast layer selection #283
    Source code(tar.gz)
    Source code(zip)
  • v0.14.2(Nov 19, 2021)

    Bugfix release

    Lots of bugs squished and packed with improvements in the core SFU.

    • Improved health checks to avoid sending traffic to dead nodes #177 #183
    • Fixed compatibility with arm64 #178
    • For transceiver re-use, fixed retained frames from previous track #179
    • Fixed edge cases with forwarding incorrect picture id #180
    • Fixed deadlocks when multiple (>20) participants join at the exact same time #189 90f3c43dc583f5dbd244842bd3df6143e74deac7
    • Fixed connection quality updates not utilizing publisher loss stats #186
    • Fixed Room API breakage #190
    • Improved audio level indicator with Opus DTX #159
    • Supports both H.264 and VP8 by default, including mixing tracks from the same participant #128
    Source code(tar.gz)
    Source code(zip)
  • v0.14.0(Nov 5, 2021)

    Connection Quality Updates

    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.

    Source code(tar.gz)
    Source code(zip)
  • v0.13.7(Nov 1, 2021)

    What's Changed

    • Fixes missing tracks when >3 participants join at the same time #163 (thanks to @bekriebel for reporting & verifying)

    Full Changelog: https://github.com/livekit/livekit-server/compare/v0.13.5...v0.13.7

    Source code(tar.gz)
    Source code(zip)
  • v0.13.5(Oct 20, 2021)

    Bugfix release

    • Update to pion v3.1.5, fixed handling of mixing simulcast/non-simulcast tracks 43079866a289bd8ca52cc26225958363e74ee711
    • Improve bandwidth estimation by sending abs-send-time #149
    • Correctly forward Track.Source #150
    Source code(tar.gz)
    Source code(zip)
  • v0.13.4(Oct 15, 2021)

  • v0.13.3(Oct 14, 2021)

    Region aware routing

    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)

    Recording revamp

    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 support

    Opus DTX is enabled by default in this version, significantly reducing audio bandwidth utilization.

    Other change & bugfixes

    • Added routing metrics to be exposed via Prometheus #139
    • Enable Opus FEC with publisher in the room when subscribers are experiencing high loss #142
    • Transceiver re-use: with clients supporting protocol 4, livekit will re-use existing transceivers to avoid it ballooning. #145
    • Support for Source attribute in TrackInfo #146
    Source code(tar.gz)
    Source code(zip)
  • v0.13.1(Oct 5, 2021)

    • Fixed NACK handling when simulcast is enabled 797d2607c45e6986a74ef23c98ab26183687aa6a
    • Fixed client rejoining in single-node mode cdb04248fb0af7c167d7c168142597d5d831d524
    • Upgraded to Pion v3.1.1
    • Room Metadata support #126
    Source code(tar.gz)
    Source code(zip)
  • v0.13.0(Sep 22, 2021)

    Features

    Protocol 3

    support for protocol 3, where subscriber connection becomes the primary one. This speeds up session establishment for participants that aren't publishing.

    Graceful termination

    When running in multi-node, server will now terminate gracefully, allowing remaining rooms on the node to drain. #116

    Other changes

    • Fixed mute/unmute on simulcasted tracks with less than 3 layers #114
    • Support incremental speaker updates #120
    • Webhooks for when recording is finished #125
    Source code(tar.gz)
    Source code(zip)
  • v0.12.5(Sep 6, 2021)

    Changes since v0.12.0

    • option to disable server-initiated mute/unmute (supported with JS SDK) #107
    • fixed mute/unmute loop when JS client changes mute states quickly #107
    • support for load aware node selection #94
    • support for sendData room API #88
    • windows development support #101
    • fixed panic when simulcasting low res video #102
    • recorder to use message bus #108
    • various interface updates #97 #103 #104 #105 #106
    Source code(tar.gz)
    Source code(zip)
  • v0.12.0(Aug 10, 2021)

    Feature release v0.12.0

    Webhooks

    LiveKit can now notify your server when room events take place. See webhooks docs for configuration and details. The following server SDK versions include support for receiving webhooks:

    • server-sdk-go v0.6.0
    • server-sdk-js v0.5.1

    Recording support

    We've also included support for our upcoming recording feature. When released, it'll work with v0.12 and above.

    Source code(tar.gz)
    Source code(zip)
  • v0.11.4(Aug 1, 2021)

    Bugfix release

    Changes

    • Default TLS port updated to match rfc5766 #68
    • STUN servers are sent to clients automatically #69
    • Preparing for recording mode #70
    • Fixed external IP discovery #72
    • Fixed case where subscriber could be placed on an unavailable layer upon joining b8e1cbe4f57ebbae6676d3c744f0ae0e3eb64965
    Source code(tar.gz)
    Source code(zip)
  • v0.11.1(Jul 23, 2021)

    Bugfixes

    • Fixed force_tcp flag, correctly suppress UDP candidates when enabled #62
    • Fixed participant actions with Room API in single node configuration #67
    • Fixed participants kicked out of the room sometimes when adaptive simulcast is used (f3a17a151f8641fe00851c2d047776f72d67677e)

    Improvements

    Huge shoutout to @hn8 for the contributions!

    • TURN/UDP support for improved connectivity #61
    • Updated logger, consistent field names #57 #60
    • Ability to have invisible participants (preparing for recorder) #65
    Source code(tar.gz)
    Source code(zip)
  • v0.11.0(Jul 16, 2021)

    We are introducing a new feature with v0.11 that significantly improve LiveKit's handling of simulcast, particularly on the publisher side. (#51) With v0.11, publishers can now indicate which layers they are actively publishing. This enables the SFU to place subscribers on currently active layers. Publishers could stop publishing to a particular layer due to bandwidth or CPU constraints.

    JS SDK > v0.10.x supports adaptive publishing

    Also in this release:

    • Improved connection error messaging with validate API (https://github.com/livekit/livekit-server/commit/53bc65285c9ba4130db2f058356721a2dbe00115)
    • Fixed Safari compatibility for H.264 rooms (https://github.com/livekit/livekit-server/commit/4ce29799cfd4fc4e39d96b3dda62f1f547d3892f)
    • Use protobuf for initial roomJoin message #52
    Source code(tar.gz)
    Source code(zip)
  • v0.10.6(Jul 13, 2021)

    Improvements

    • Fixes down track resync on unmute

    Changes

    • Exposes /debug/rooms endpoint when running in dev mode, which displays room and participant state along with down track stats
    Source code(tar.gz)
    Source code(zip)
  • v0.10.5(Jul 11, 2021)

    Improvements

    • Fix glitch during layer switch with H.264 simulcast
    • Handle client reconnect after server has been restarted (#43)
    • Enhancements to active-speaker detection (#44)
    • Improves handling of Node IP in container environments (#48)

    Changes

    • When ports are not explicitly configured, and --dev flag is used, single port mode will be used to make it easier to map ports via Docker.
    Source code(tar.gz)
    Source code(zip)
  • v0.10.4(Jul 7, 2021)

    Changes:

    • Use multi-port mode by default (#40)
    • Optimized SFU send-loop to fully utilize all CPUs
    • Embedded TURN/TLS for strict corporate firewalls.
    Source code(tar.gz)
    Source code(zip)
Owner
LiveKit
Open source platform for real-time audio and video
LiveKit
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