...

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()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 = extractVideoTrack(localVideo);
    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

function removeVideoTrackEndedListener(localVideo) {
    var videoTrack = extractVideoTrack(localVideo);
    if(videoTrack) {
        videoTrack.onended = null;
    }
}

3. Add function to extract videotrack

function extractVideoTrack(localVideo) {
    return localVideo.firstChild.srcObject.getVideoTracks()[0];
}

4. Register event handler when publishing a stream

    session.createStream({
        name: streamName,
        display: localVideo,
        ...
    }).on(STREAM_STATUS.PUBLISHING, function (stream) {
        addVideoTrackEndedListener(localVideo, stream);
        setStatus("#publishStatus", STREAM_STATUS.PUBLISHING);
        onPublishing(stream);
        ...
    }).publish();

5. Remove event handler when stopping a stream

function onPublishing(stream) {
    $("#publishBtn").text("Stop").off('click').click(function () {
        $(this).prop('disabled', true);
        removeVideoTrackEndedListener(localVideo);
        stream.stop();
    }).prop('disabled', false);
    $("#publishInfo").text("");
}

Videotrack activity checking on server side

Videotrack activity checking for streams published on server is enabled with the following parameters in flashphoner.properties file

rtp_activity_video=true

In this case, if there is no video in stream, its publishing will be stopped after 60 seconds.

Video only stream publishing with constraints

In some cases, video only stream should be published while microphone is busy, for example video is published while voice phone call. To prevent browser access request to microphone, set the constraints for video on;ly publishing:

function addVideoTrackEndedListener(localVideo, stream) {
    var videoTrack = extractVideoTrack(localVideo); session.createStream({
    if (videoTrack && videoTrack.readyState == 'ended') { name: streamName,
        console.error("Video source error. Disconnect...");display: localVideo,
        stream.stop();
    } else if (videoTrack) {constraints: {video: true, audio: false}
        videoTrack...onended
  = function }).publish(event) {
);

Audio only stream publishing

In most cases, it is enough to set the constraints to publish audio only stream:

    session.createStream({
        name: streamName,
      console.error("Video source error. Disconnect...");display: localVideo,
        constraints: {video: false,  stream.stop();audio: true}
        };...
    }
}

...

).publish();

Audio only stream publishing in Safari browser

When audio only stream is published from iOS Safari browser with constraints, browser does not send audio packets. To workaround this, a stream should be published with video, then video should be muted:

function removeVideoTrackEndedListener(localVideo) {    session.createStream({
        name: streamName,
    var   videoTrack =display: extractVideoTrack(localVideo);
,
        if(videoTrack)constraints: {video: true, audio: true}
        videoTrack.onended = null;...
    }
}

3. Add function to extract videotrack

function extractVideoTrack(localVideo) {
    return localVideo.firstChild.srcObject.getVideoTracks()[0];
}

...

).on(STREAM_STATUS.PUBLISHING, function (stream) {
        stream.muteVideo();
        ...
    }).publish();

In this case, iOS Safari browser will send emply video packets (blank screen) and audio packets.

Disable resolution constraints normalization in Safari browser

By default, WebSDK normalizes stream publishing resolution constraints set in Safari browser. In this case, if width or height is not set, or equal to 0, then the picture resolution is forced to 320x240 or 640x480. Since WebSDK build 0.5.28.2753.109 (hash 149855cc050bf7512817104fd0104e9cce760ac4), it is possible to disable normalization and pass resolution constarints to the browser as is. for example:

publishStream  =  session.createStream({
        name: streamName,
        display: localVideo,
        ....createStream({
    }).on(STREAM_STATUS.PUBLISHING, function (stream) {...
        addVideoTrackEndedListener(localVideo, stream);disableConstraintsNormalization: true,
    constraints {
   setStatus("#publishStatus", STREAM_STATUS.PUBLISHING);
        onPublishing(stream);
  video: {
      ...
    }).publish();

5. Remove event handler when stopping a stream

function onPublishing(stream) {  width: {ideal: 1024},
    $("#publishBtn").text("Stop").off('click').click(function () {
      height:  $(this).prop('disabled', true);{ideal: 768}
        removeVideoTrackEndedListener(localVideo);
},
         stream.stop();audio: true
    }
}).prop('disabled', false);on(STREAM_STATUS.PUBLISHING, function (publishStream) {
    $("#publishInfo").text("");
}

Videotrack activity checking on server side

...

...
});
publishStream.publish();

H264 encoding profiles exclusion

Since build 5.2.620 some H264 encoding profiles can be excluded from remote SDP which is sent by server to browser. The profiles to exckude should be listed in the following parameter

rtp_activity_video=truewebrtc_sdp_h264_exclude_profiles=4d,64

In this case, if there is no video in stream, its publishing will be stopped after 60 seconds.

Video only stream publishing with constraints

In some cases, video only stream should be published while microphone is busy, for example video is published while voice phone call. To prevent browser access request to microphone, set the constraints for video on;ly publishing:

    session.createStream({
        name: streamName,
        display: localVideo,
        constraints: {video: true, audio: false}
        ...
    }).publish();

Audio only stream publishing

In most cases, it is enough to set the constraints to publish audio only streamMain (4d) and High (64) profiles will be excluded, but Baseline (42) still remains:

a=rtpmap:102 H264/90000
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:125 H264/90000
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:127 H264/90000
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=rtpmap:108 H264/90000
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f

This setting may be useful if some browser encodes B-frames for example using high profiles while hardware acceleration is enabled.

By default, no profiles will be excluded from SDP if they are supported by browser

webrtc_sdp_h264_exclude_profiles=

Content type management while publishing from Chromium based browser

In some cases, most browsers based on Chromium 91 agressively assess a publishing channel quality and drop publishing resolution lower than set in constraints, even if channel is enough to publish 720p or 1080p stream. To workaround this behaviour, since WebSDK build  2.0.180 videoContentHint option was added:

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

Audio only stream publishing in Safari browser

When audio only stream is published from iOS Safari browser with constraints, browser does not send audio packets. To workaround this, a stream should be published with video, then video should be muted: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 motion

    session.createStream({
        name: streamName,
        display: localVideo,
        constraints: {videocacheLocalResources: true,
 audio:  true}
     receiveVideo: false,
  ...
    }).on(STREAM_STATUS.PUBLISHING, function (stream) {receiveAudio: false,
        stream.muteVideo();videoContentHint: "motion"
        ...
    }).publish();

In this case, iOS Safari browser will send emply video packets (blank screen) and audio packets.

Disable resolution constraints normalization in Safari browser

By default, WebSDK normalizes stream publishing resolution constraints set in Safari browser. In this case, if width or height is not set, or equal to 0, then the picture resolution is forced to 320x240 or 640x480. Since WebSDK build 0.5.28.2753.109 (hash 149855cc050bf7512817104fd0104e9cce760ac4), it is possible to disable normalization and pass resolution constarints to the browser as is. for exampleSince 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

Image Added

FPS management in Firefox browser

By default, Firefox is publishing video with maximum FPS shown by web camera driver fo requested resolution. This value is 30 FPS for most of modern web cameras. Publishing FPS can be defined more strictly if necessary. To do this, disable constraints normalization:

publishStream = session.createStream({
    ...
    disableConstraintsNormalization: true,
    constraints: {
        video: {
            width: {ideal: 1024}640,
            height: 360,
            heightframeRate: {ideal max: 15 768}
        },
        audio: true
    }
}).on(STREAM_STATUS.PUBLISHING, function (publishStream) {
    ...
});
publishStream.publish();

H264 encoding profiles exclusion

Since build 5.2.620 some H264 encoding profiles can be excluded from remote SDP which is sent by server to browser. The profiles to exckude should be listed in the following parameter

webrtc_sdp_h264_exclude_profiles=4d,64

In this case, Main (4d) and High (64) profiles will be excluded, but Baseline (42) still remains:

a=rtpmap:102 H264/90000
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f
a=rtpmap:125 H264/90000
a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
a=rtpmap:127 H264/90000
a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f
a=rtpmap:108 H264/90000
a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f

This setting may be useful if some browser encodes B-frames for example using high profiles while hardware acceleration is enabled.

By default, no profiles will be excluded from SDP if they are supported by browserNote that Foirefox can exclude the camera from the list while requesting camera and microphone access if camera driver does not provide a required combination of resolution and FPS. Also, Firefox can change a publishing resolution if there is only one resolution with required FPS in camera driver response.

Stereo audio publishing in browser

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 client side

session.createStream({
    name: streamName,
    display: remoteVideo,
    constraints: {
        audio: {
            bitrate: 64000
        },
        ...
    }
    ...
}).publish();

or on server side

webrtc_sdp_h264_exclude_profiles=

Content type management while publishing from Chromium based browser

...

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

By default, this option is set to detail 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 motion

    audio: session.createStream({
           name: streamName,
 stereo: true
        },
        ...
   display: localVideo,}
    ...
}).publish();

Using Websocket API

If only Websocket API is used in project, it is necessary to disable echo cancellation

    var constraints cacheLocalResources:= true,{
        receiveVideoaudio: false,{
             receiveAudio echoCancellation: false,
        videoContentHint: "motion"
   googEchoCancellation:   false
  ...
    }).publish();

FPS management in Firefox browser

By default, Firefox is publishing video with maximum FPS shown by web camera driver fo requested resolution. This value is 30 FPS for most of modern web cameras. Publishing FPS can be defined more strictly if necessary. To do this, disable constraints normalization:

session.createStream({
  },
        ...
    disableConstraintsNormalization: true,};
    constraints: {...
    navigator.getUserMedia(constraints,    video:function (stream) {
        ...
    width: 640,
            height: 360,
            frameRate: { max: 15 }
        }}, 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,
        audiodisplay: truelocalVideo,
    }
}).on(STREAM_STATUS.PUBLISHING, function (publishStream) {transport: "TCP"
    ...
}).publish();

Note that Foirefox can exclude the camera from the list while requesting camera and microphone access if camera driver does not provide a required combination of resolution and FPS. Also, Firefox can change a publishing resolution if there is only one resolution with required FPS in camera driver responseAnother option is to use external or internal TURN server or publish a stream via RTMP or RTSP.

Known issues

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

...

17. iOS Safari 15.1 requires from another side to enable image orientation extension when publishing H264 stream

Symptoms: webpage crashe crashes in iOS Safari 15.1 when stream publishing is started (Webkit bugs https://bugs.webkit.org/show_bug.cgi?id=232381 and https://bugs.webkit.org/show_bug.cgi?id=231505)

...

session.createStream({
    name: streamName,
    ...
    cvoExtension: true
}).publish();

and in WCS builds before 5.2.1074 disable RTP bundle support

rtp_bundle=false

Since WCS build 5.2.1074RTP bundle may not be disabled

b) use VP8 to publish a stream

...