Skip to content

From another WCS server via WebRTC

Overview

WCS can capture on demand a WebRTC video stream published by another WCS server. The captured stream can be than played on any of supported platforms using any of supported technologies. WebRTC stream capturing is managed using REST API.

Operation flowchart

  1. The browser connects to the WCS1 server via Websocket and sends the publishStream command.
  2. The browser captures the microphone and the camera and sends the WebRTC stream to the server.
  3. The REST client sends to the WCS2 server the /pull/pull query.
  4. WCS2 requests the stream from WCS1.
  5. WCS2 receives the WebRTC stream from WCS1.
  6. The second browser establishes a connection to the WCS2 server via Websocket and sends the playStream command.
  7. The second browser receives the WebRTC stream and plays this stream on the page.

REST API

REST query must be an HTTP/HTTPS POST request as follows:

  • HTTP: http://test.flashphoner.com:8081/rest-api/pull/rtmp/pull
  • HTTPS: https://test.flashphoner.com:8444/rest-api/pull/rtmp/pull

Where:

  • test.flashphoner.com is the address of the WCS server
  • 8081 is the standard REST / HTTP port of the WCS server
  • 8444 is the standard HTTPS port
  • rest-api is the required part of the URL
  • /pull/rtmp/pull is the REST method used

REST queries and responses

REST method Request body Response body Response status Description
`/pull/pull`
{ 
    "uri":"wss://demo.flashphoner.com:8443",
    "localStreamName": "testStream",
    "remoteStreamName": "testStream"
}
409 Conflict 500 Internal error Pull the WebRTC stream at the specified URL
`/pull/find_all`
[
    {
        "localMediaSessionId": "5a072377-73c1-4caf-abd3",
        "remoteMediaSessionId": null,
        "localStreamName": "testStream",
        "remoteStreamName": "testStream",
        "uri": "wss://demo.flashphoner.com:8443",
        "status": "NEW"
    }
]
200 OK 404 Not found 500 Internal error Find all pulled WebRTC streams
`/pull/terminate`
{ 
   "uri":"wss://demo.flashphoner.com:8443",
   "localStreamName": "testStream",
   "remoteStreamName": "testStream"
}
200 OK 404 Not found 500 Internal error Terminate the pulled WebRTC stream

Parameters

Parameter Description Example
uri Websocket URL of WCS server `wss://demo.flashphoner.com:8443`
localMediaSessionId Session identifier `5a072377-73c1-4caf-abd3`
remoteMediaSessionId Session identifier on the remote server `12345678-abcd-dead-beaf`
localStreamName Local name assigned to the captured stream. By this name the stream can be requested from the WCS server `testStream`
remoteStreamName Captured stream name on the remote server `testStream`
status Current stream status `NEW`

Configuration

By default, WebRTC stream is pulled over unsecure Websocket connection, i.e. WCS server URL has to be set as ws://demo.flashphoner.com:8080. To use Secure Websocket, the parameter must be set in flashphoner.properties file

wcs_agent_ssl=true

This change has to be made on both WCS servers: the server that publishes the stream and the server the stream is pulled to.

Quick manual on testing

  1. For this test we use:
  2. two WCS servers;
  3. the Chrome browser and a REST-client to send queries to the server;
  4. the Two Way Streaming web application to publish the stream;
  5. the Player web application to play the captured stream in the browser.

  6. Open the Two Way Streaming web application and publish the stream on the first server

  7. Open the REST client. Send the /pull/pull query to the second WCS server and specify the following parameters:

  8. URL of the WCS server the stream is captured from;
  9. stream name published on the server;
  10. local stream name

  11. Make sure the second server captured the stream. To do this, send the /pull/find_all query:

  12. Open the Player web application and put in the local stream name into the Stream field, then click Start:

Call flow

Below is the call flow when using the Two Way Streaming example to publish a stream on one WCS server and playing that stream on another WCS server

two_way_streaming.html

two_way_streaming.js

player.html

player.js

  1. Establishing connection to the server
    Flashphoner.createSession() code

    Flashphoner.createSession({urlServer: url}).on(SESSION_STATUS.ESTABLISHED, function (session) {
      setStatus("#connectStatus", session.status());
      onConnected(session);
    }).on(SESSION_STATUS.DISCONNECTED, function () {
      setStatus("#connectStatus", SESSION_STATUS.DISCONNECTED);
      onDisconnected();
    }).on(SESSION_STATUS.FAILED, function () {
      setStatus("#connectStatus", SESSION_STATUS.FAILED);
      onDisconnected();
    });
    

  2. Receiving from the server an event confirming successful connection
    SESSION_STATUS.ESTABLISHED code

    Flashphoner.createSession({urlServer: url}).on(SESSION_STATUS.ESTABLISHED, function (session) {
      setStatus("#connectStatus", session.status());
      onConnected(session);
    }).on(SESSION_STATUS.DISCONNECTED, function () {
      ...
    }).on(SESSION_STATUS.FAILED, function () {
      ...
    });
    

  3. Publishing the stream
    Stream.publish() code

    session.createStream({
      name: streamName,
      display: localVideo,
      cacheLocalResources: true,
      receiveVideo: false,
      receiveAudio: false
    }).on(STREAM_STATUS.PUBLISHING, function (stream) {
      ...
    }).on(STREAM_STATUS.UNPUBLISHED, function () {
      ...
    }).on(STREAM_STATUS.FAILED, function () {
      ...
    }).publish();
    

  4. Receiving an event confirming successful publishing of the stream
    STREAM_STATUS.PUBLISHING code

    session.createStream({
      name: streamName,
      display: localVideo,
      cacheLocalResources: true,
      receiveVideo: false,
      receiveAudio: false
    }).on(STREAM_STATUS.PUBLISHING, function (stream) {
      setStatus("#publishStatus", STREAM_STATUS.PUBLISHING);
      onPublishing(stream);
    }).on(STREAM_STATUS.UNPUBLISHED, function () {
      ...
    }).on(STREAM_STATUS.FAILED, function () {
      ...
    }).publish();
    

  5. Sending the stream via WebRTC to the server

  6. Sending the /pull/pull REST query to the second server

  7. Requesting the stream from the first server

  8. Sending the stream via WebRTC to the second server

  9. Establishing connection to the second server
    Flashphoner.createSession() code

    Flashphoner.createSession({urlServer: url}).on(SESSION_STATUS.ESTABLISHED, function(session){
      setStatus(session.status());
      //session connected, start playback
      playStream(session);
    }).on(SESSION_STATUS.DISCONNECTED, function(){
      setStatus(SESSION_STATUS.DISCONNECTED);
      onStopped();
    }).on(SESSION_STATUS.FAILED, function(){
      setStatus(SESSION_STATUS.FAILED);
      onStopped();
    });
    

  10. Receiving from the server and event confirming successful connection
    SESSION_STATUS.ESTABLISHED code

    Flashphoner.createSession({urlServer: url}).on(SESSION_STATUS.ESTABLISHED, function(session){
        setStatus(session.status());
        //session connected, start playback
        playStream(session);
    }).on(SESSION_STATUS.DISCONNECTED, function(){
        ...
    }).on(SESSION_STATUS.FAILED, function(){
        ...
    });
    

  11. Requesting to play the stream
    Stream.play() code

    stream = session.createStream(options).on(STREAM_STATUS.PENDING, function(stream) {
        var video = document.getElementById(stream.id());
        if (!video.hasListeners) {
            video.hasListeners = true;
            video.addEventListener('playing', function () {
                $("#preloader").hide();
            });
            video.addEventListener('resize', function (event) {
                var streamResolution = stream.videoResolution();
                if (Object.keys(streamResolution).length === 0) {
                    resizeVideo(event.target);
                } else {
                    // Change aspect ratio to prevent video stretching
                    var ratio = streamResolution.width / streamResolution.height;
                    var newHeight = Math.floor(options.playWidth / ratio);
                    resizeVideo(event.target, options.playWidth, newHeight);
                }
            });
        }
        ...
    });
    stream.play();
    

  12. Receiving an event confirming successful capturing and playing of the stream
    STREAM_STATUS.PLAYING code

    stream = session.createStream(options).on(STREAM_STATUS.PENDING, function(stream) {
        ...
    }).on(STREAM_STATUS.PLAYING, function(stream) {
        $("#preloader").show();
        setStatus(stream.status());
        onStarted(stream);
    }).on(STREAM_STATUS.STOPPED, function() {
        ...
    }).on(STREAM_STATUS.FAILED, function(stream) {
        ...
    }).on(STREAM_STATUS.NOT_ENOUGH_BANDWIDTH, function(stream){
        ...
    });
    stream.play();
    

  13. Sending the stream via WebRTC to the playing client

  14. Stopping playback of the stream
    Stream.stop() code

    function onStarted(stream) {
        $("#playBtn").text("Stop").off('click').click(function(){
            $(this).prop('disabled', true);
            stream.stop();
        }).prop('disabled', false);
        ...
    }
    

  15. Receiving an event confirming successful unpublishing of the stream
    STREAM_STATUS.STOPPED code

    stream = session.createStream(options).on(STREAM_STATUS.PENDING, function(stream) {
        ...
    }).on(STREAM_STATUS.PLAYING, function(stream) {
        ...
    }).on(STREAM_STATUS.STOPPED, function() {
        setStatus(STREAM_STATUS.STOPPED);
        onStopped();
    }).on(STREAM_STATUS.FAILED, function(stream) {
        ...
    }).on(STREAM_STATUS.NOT_ENOUGH_BANDWIDTH, function(stream){
        ...
    });
    stream.play();
    

  16. Stopping publishing the stream
    Stream.stop() code

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

  17. Receiving an event confirming successful unpublishing of the stream
    STREAM_STATUS.UNPUBLISHED code

    session.createStream({
        name: streamName,
        display: localVideo,
        cacheLocalResources: true,
        receiveVideo: false,
        receiveAudio: false
    }).on(STREAM_STATUS.PUBLISHING, function (stream) {
        ...
    }).on(STREAM_STATUS.UNPUBLISHED, function () {
        setStatus("#publishStatus", STREAM_STATUS.UNPUBLISHED);
        onUnpublished();
    }).on(STREAM_STATUS.FAILED, function () {
        ...
    }).publish();