To another WCS server via WebRTC¶

Overview¶

WCS can rebroadcast a video stream via WebRTC to another WCS server on demand. Republishing of a WebRTC stream is managed by the REST API.

Operation flowchart¶

The browser connects to the WCS1 server via the Websocket protocol and sends the publishStream command.
The browser captures the microphone and the camera and send the WebRTC stream to the server.
The REST client sends to the WCS1 server the /pull/push query.
WCS1 publishes the stream to WCS2.
WCS2 receives the WebRTC stream from WCS1.
The second browser establishes a connection to the WCS2 server via Websocket and sends the playStream command.
The second browser receives the WebRTC stream and plays this stream on the page.

REST API¶

A REST query must be an HTTP/HTTPS POST query in the following form:

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

Where:

streaming.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 prefix
/pull/push - is the REST-method used

REST methods and responses¶

/pull/push¶

Broadcasts the WebRTC stream to the specified URL

Request example¶

POST /rest-api/pull/push HTTP/1.1
Host: localhost:8081
Content-Type: application/json

{
    "uri":"wss://demo.flashphoner.com:8443/websocket",
    "localStreamName": "testStream",
    "remoteStreamName": "testStream"
}

Response example¶

HTTP/1.1 200 OK
Access-Control-Allow-Origin: *
Content-Type: application/json

Return codes¶

Code	Reason
200	OK
404	Not found
409	Conflict
500	Internal error

/pull/find_all¶

Find all pushed streams

Request example¶

POST /rest-api/pull/find_all HTTP/1.1
Host: localhost:8081
Content-Type: application/json

Response example¶

HTTP/1.1 200 OK
Access-Control-Allow-Origin: *
Content-Type: application/json

[
    {
        "localMediaSessionId":"da157e2b-2159-40c9-9560-ae1af8d4a0b5",
        "remoteMediaSessionId":null,
        "localStreamName":"testStream",
        "remoteStreamName":"testStream",
        "uri":"wss://demo.flashphoner.com:8443/websocket",
        "status":"NEW"
    }
]

Return codes¶

Code	Reason
200	OK
404	Not found
500	Internal error

/pull/terminate¶

Terminate a pulled stream

Request example¶

POST /rest-api/pull/find_all HTTP/1.1
Host: localhost:8081
Content-Type: application/json

{
    "uri":"wss://demo.flashphoner.com:8443/websocket"
}

Response example¶

HTTP/1.1 200 OK
Access-Control-Allow-Origin: *
Content-Type: application/json

Return codes¶

Code	Reason
200	OK
404	Not found
500	Internal error

Parameters¶

Parameter	Description	Example
uri	URL of the WebRTC stream	`wss://demo.flashphoner.com:8443/websocket`
localMediaSessionId	Session identifier	`5a072377-73c1-4caf-abd3`
remoteMediaSessionId	Identifier of the session on the remote server	`12345678-abcd-dead-beaf`
localStreamName	Local name assigned to the captured stream. The stream can be fetched from the WCS server using this name	`testStream`
remoteStreamName	Name of the captured stream on the remore server	`testStream`
status	Current status of the stream	`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 file flashphoner.properties

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 pushed to.

Quick manual on testing¶

For this test we use:
- two WCS servers;
- the Chrome browser and the REST client to send queries to the server;
- the Two Way Streaming web application to publish the stream;
- the Player web application to play the captured stream in a browser.
Open the Two Way Streaming web application, publish the stream on the server
Open the REST client. Send the /pull/push query and specify in its parameters:
- the URL of the WCS server the stream is captured from
- the name of the stream published on the server
- the local name of the stream
Open the Player web application and specify the local stream name in 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 the Player example to play that stream on another WCS server.

two_way_streaming.html

two_way_streaming.js

player.html

player.js

Establishing a 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();
});

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 () {
  ...
});

Publishing the stream
Stream.publish() code

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

Receiving from the server 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();

Sending the audio-video stream via WebRTC to the server
Sending the /pull/push REST query to the server
Publishing the stream on the second server
Sending the audio-video stream via WebRTC to the second server

Establishing a 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();
});

Receiving from the server an 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(){
    ...
});

Requesting to play the stream
Stream.play() code

stream = session.createStream(options).on(STREAM_STATUS.PENDING, function(stream) {
    ...
});
stream.play();

Receiving from the server an event that confirms 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();

Sending the audio-video stream via WebRTC

Stopping the 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);
    ...
}

Receiving from the server an event confirming the playback of the stream is stopped
STRTEAM_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();

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("");
}

Receiving from the server an event that confirms 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();