Redirecting a SIP call to a stream (SIP as Stream function)¶

Overview¶

A SIP call made through the WCS server can be captured into a stream on the server when the call is created. Then this call can be played in a browser using any method supported by WCS.
The stream captured from a SIP call can be republished to an RTMP server using the REST query /push/startup, just like any media stream on the WCS server.

Operation flowchart¶

The browser starts a call using the /call/startup REST query
WCS connects to the SIP server
The SIP server sends the RTP stream of the call to WCS
The second browser requests playback of the call stream
The second browser receives the WebRTC stream

Quick manual on testing¶

For this test we use:
- two SIP accounts;
- the softphone to answer the call;
- the REST-client in Chrome browser;
- the Player web application to play the stream.
Open the REST client. Send the /call/startup query to the WCS server and specify the following as query parameters:
- parameters of your SIP account the call will be made from;
- the stream name to republish the call to (the toStream parameter), for example, call_stream1;
- the name of your second SIP account the call will be made to
Receive and answer the incoming call on the softphone:
Open the Player web application and in the Stream field specify the name of the stream the call is redirected to (in our example: call_stream1):
Click Play. The stream starts playing:
To terminate the call, send /call/terminate from the REST client to the WCS server and pass the call id in the parameters:

Call flow¶

Below is the call flow when using the SIP as RTMP example to create the call and the Player example to play it

Sending the REST query /call/startup:
sendREST()
```
                              
```

ba/examples/demo/sip/sip-as-rtmp-4/sip-as-rtmp-4.js#L314">code
href="#__codelineno-0-1">function startCall() { ... var url = field("restUrl") + "/call/startup"; callId = generateCallID(); ... var RESTCall = {}; RESTCall.toStream = field("rtmpStream"); RESTCall.hasAudio = field("hasAudio"); RESTCall.hasVideo = field("hasVideo"); RESTCall.callId = callId; RESTCall.sipLogin = field("sipLogin"); RESTCall.sipAuthenticationName = field("sipAuthenticationName"); RESTCall.sipPassword = field("sipPassword"); RESTCall.sipPort = field("sipPort"); RESTCall.sipDomain = field("sipDomain"); RESTCall.sipOutboundProxy = field("sipOutboundProxy"); RESTCall.appKey = field("appKey"); RESTCall.sipRegisterRequired = field("sipRegisterRequired"); for (var key in RESTCall) { setCookie(key, RESTCall[key]); } RESTCall.callee = field("callee"); var data = JSON.stringify(RESTCall); sendREST(url, data); startCheckCallStatus(); }

Establishing a connection to the SIP server

Receiving a confirmation from the SIP server

The RTP stream of the call is sent to the WCS server

The browser establishes connection to the 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:
CONNECTION_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(){
     ...
 });

Request 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();

Receiving an event from the server confirming successful 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);
     ...
 });
 stream.play();

Sending audio and video stream via WebRTC

Stopping playing 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 an event from the server confirming 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();

Sending the /call/terminate REST query:
sendREST() code

function hangup() {
    var url = field("restUrl") + "/call/terminate";
    var currentCallId = { callId: callId };
    var data = JSON.stringify(currentCallId);
    sendREST(url, data);
}

Sending the command to the SIP server

Receiving confirmation from the SIP server

SIP as stream recording¶

All streams captured from SIP calls can be recorded on server. To do this, set the following parameters in flashphoner.properties file:

sip_single_route_only=true
sip_record_stream=true

The following codecs are supported:

Video: H264
Audio: opus, PCMA (alaw), PCMU (ulaw)

Stream recording is described here in details.

Known issues¶

1. Stream captured from SIP call can not be played, if RTP session is not initialized for this stream¶

Symptoms

SIP stream is published on server, but can not be played

Solution

Enable RTP session initializing with the following parameter

rtp_session_init_always=true

2. Freezes may occur, audio may be out of sync with video when republishing a SIP call stream as RTMP¶

Symptoms

Freezes and audio/video out of sync are observed while playing an RTMP stream republished by /push/startup REST query from a SIP call

Solution

a) in WCS builds before 5.2.1541 add the delay to audio/video generator start

generate_av_start_delay=1000

b) update WCS to build 5.2.1541 where the issue was fixed

3. RTP traffic buffering should be enabled in some cases when republishing SIP as Stream or SIP as RTMP¶

Symptoms

Audio and video may be out of sync when playing a SIP call stream

Solution

Update WCS to build 5.2.1910 and enable RTP traffic buffering

rtp_in_buffer=true