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rtp_activity_video=false

If Chrome browser sends empty video due to web camera conflict

Some Chrome versions does not return an error if web camera is busy, but publish a stream with empty video (black screen). In this case, stream publishing can be stopped by two ways: using JavaScript and HTML5 on client, or using server settings.

Stopping a stream with empty video on client side

Videotrack that Chrome browsers creates for busy web camera, stops after no more than one second publishing, then stream is send without a videotrack. In this case videotrack state (readyState variable) changes to ended, and corresponding  onended event is generated that can be catched by web application. To use this event:

1. Add to web application script the registartion function for onended event handler, in which stream pub;ishing is stopped with stream.stop()

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RTP activity can be checked for publishing streams only, not for playing streams.

Disable tracks activity checking by stream name

Since build 5.2.1784 it is possible to disable video and audio tracks activity checking for the streams with names matching a regular expression

rtp_activity_audio_exclude=stream1
rtp_activity_video_exclude=stream1

The feature may be useful for streams in which a media traffic can stop for a long time, for example, screen sharing streams from an application window

rtp_activity_audio_exclude=.*-screen$
rtp_activity_video_exclude=.*-screen$

In this case tracks activity checking will not be applied to the tracks named like conference-123-user-456-screen 

If Chrome browser sends empty video due to web camera conflict

Some Chrome versions does not return an error if web camera is busy, but publish a stream with empty video (black screen). In this case, stream publishing can be stopped by two ways: using JavaScript and HTML5 on client, or using server settings.

Stopping a stream with empty video on client side

Videotrack that Chrome browsers creates for busy web camera, stops after no more than one second publishing, then stream is send without a videotrack. In this case videotrack state (readyState variable) changes to ended, and corresponding  onended event is generated that can be catched by web application. To use this event:

1. Add to web application script the registartion function for onended event handler, in which stream pub;ishing is stopped with stream.stop()

function addVideoTrackEndedListener(localVideo, stream) {
    var    videoTrack.onended = function extractVideoTrack(eventlocalVideo) {;
    if (videoTrack && videoTrack.readyState == 'ended') {
        console.error("Video source error. Disconnect...");
        stream.stop();
    } else if (videoTrack) {
        videoTrack.onended = function (event) {
            console.error("Video source error. Disconnect...");
            stream.stop();
        };
    }
}

2. Add function to remove event handler when stream is stopped

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    session.createStream({
        name: streamName,
        display: localVideo,
        cacheLocalResources: true,
        receiveVideo: false,
        receiveAudio: false,
        videoContentHint: "detail"
        ...
    }).publish();

By default, this In WebSDK builds before 2.0.242 this option is set to detail by default and forces browsers to keep the publishing resolution as set in constraints. However, browser can drop FPS in this case when publishing stream from som USB web cameras. If FPS should be kept mo matter to resolution, the option should be set to to motion

    session.createStream({
        name: streamName,
        display: localVideo,
        cacheLocalResources: true,
        receiveVideo: false,
        receiveAudio: false,
        videoContentHint: "motion"
        ...
    }).publish();

Since WebSDK build 2.0.242, videoContentHint  is set to motion by default. The detail or text values should be set only for screen sharing streaming in browser.

Since WebSDK build 2.0.204 videoContentHint selection is available in Media Device example

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Audio bitrate should be more than 60000 bps to publish stereo in Opus codec from browser. This can be done by setting Opus codec parameters on server client side

opus_formats = maxaveragebitrate=64000;stereo=1;sprop-stereo=1;

In this case, Firefox browser publishes stereo audio without additional setup.

Stereo audio publishing in Chrome based browsers

A certain client setup is required to publish stereo audio from Chrome. Thre is two ways to set this up depending on client implementation

Using Web SDK

If Web SDK is used in project, it is necessary to set the fiollowing constraint option:

session.createStream({
    name: streamName,
    session.createStream({
    name: streamName,
    display: remoteVideo,
    constraints: {
        audio: {
            stereobitrate: true64000
        },
        ...
    }
    ...
}).publish();

Using Websocket API

If only Websocket API or on server side

opus_formats = maxaveragebitrate=64000;stereo=1;sprop-stereo=1;

In this case, Firefox browser publishes stereo audio without additional setup.

Stereo audio publishing in Chrome based browsers

A certain client setup is required to publish stereo audio from Chrome. Thre is two ways to set this up depending on client implementation

Using Web SDK

If Web SDK is used in project, it is necessary to disable echo cancellationset the fiollowing constraint option:

session.createStream({
    name: streamName,
    display: remoteVideolocalVideo,
    constraints: {
        audio: {
            echoCancellation: false,
            googEchoCancellationstereo: falsetrue
        },
        ...
    }
    ...
}).publish();

Using Websocket API

If echo cancellation is enabled, Chrome will publish mono audio even if stereo is set in Opus codec options.only Websocket API is used in project, it is necessary to disable echo cancellation

    var constraints = {
        audio: {
            echoCancellation: false,
            googEchoCancellation: false
        },
        ...
    };
    ...
    navigator.getUserMedia(constraints, function (stream) {
        ...
    }, reject);

If echo cancellation is enabled, Chrome will publish mono audio even if stereo is set in Opus codec options.

How to bypass an encrypted UDP traffic blocking

Sometimes an encrypted UDP mediatraffic may be blocked by ISP. In this case, WebRTC stream publishing over UDP will fail with Failed by RTP activity error. To bypass this, it is recommended to use TCP transport at client side

session.createStream({
    name: streamName,
    display: localVideo,
    transport: "TCP"
    ...
}).publish();

Another option is to use external or internal TURN server or publish a stream via RTMP or RTSP.

Redundancy support while publishing audio

Since build 5.2.1969 a redundancy is supported while publishing audio data (RED, RFC2198). This allows to reduce audio packet loss when using opus codec. The feature may be enabled with the following parameter

codecs=red,opus,alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,h264,vp8,flv,mpv

Note that red codec should be set before opus codec. In this case a browser supporting RED will send a redundancy data in audio packets. Note that audio publishing bitrate will be raised.

RED should be excluded for the cases when it cannot be applied:

codecs_exclude_sip=red,mpeg4-generic,flv,mpv
codecs_exclude_sip_rtmp=red,opus,g729,g722,mpeg4-generic,vp8,mpv

Known issues

1. If the web app is inside an iframe element, publishing of the video stream may fail.

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