SIP as RTMP 4¶
Example of stream capturing from SIP call, RTMP stream pulling from other server, streams mixing and re-publishing to a third party RTMP server¶
This example demonstrates how to make a call to SIP, receive audio and video traffic from SIP in response, capture RTMP stream from other server, mix audio streams, inject mixer sound to the SIP call and then redirect the SIP stream to a third-party RTMP server for further broadcasting
The code of the example¶
This example is a simple REST client written on JavaScript, available at:
/usr/local/FlashphonerWebCallServer/client2/examples/demo/sip/sip-as-rtmp-4
- sip-as-rtmp-4.js -Â a script dealing with REST queries to the WCS server
- sip-as-rtmp-4.html -Â example page
The example may be tested at this URL:
https://host:8888/client2/examples/demo/sip/sip-as-rtmp-4/sip-as-rtmp-4.html
where host is WCS server address.
Analyzing the code¶
To analyze the code get sip-as-rtmp-4.js file version with hash ecbadc3 that can be found here and is availabe to download in build 2.0.212.
1. REST / HTTP queries sending¶
Sending is done using POST method with ContentType: application/json by AJAX query using jQuery framework.
function sendREST(url, data, successHandler, errorHandler) {
console.info("url: " + url);
console.info("data: " + data);
$.ajax({
url: url,
beforeSend: function ( xhr ) {
xhr.overrideMimeType( "text/plain;" );
},
type: 'POST',
contentType: 'application/json',
data: data,
success: (successHandler === undefined) ? handleAjaxSuccess : successHandler,
error: (errorHandler === undefined) ? handleAjaxError : errorHandler
});
}
2. Making outgoing call with REST-request /call/startup¶
Connection and call data (RESTCall) are collected from the boxes on page
var url = field("restUrl") + "/call/startup";
callId = generateCallID();
$("#sipCallId").val(callId);
...
var RESTCall = {};
RESTCall.toStream = field("rtmpStream");
RESTCall.hasAudio = field("hasAudio");
RESTCall.hasVideo = field("hasVideo");
RESTCall.callId = callId;
RESTCall.sipLogin = field("sipLogin");
RESTCall.sipAuthenticationName = field("sipAuthenticationName");
RESTCall.sipPassword = field("sipPassword");
RESTCall.sipPort = field("sipPort");
RESTCall.sipDomain = field("sipDomain");
RESTCall.sipOutboundProxy = field("sipOutboundProxy");
RESTCall.appKey = field("appKey");
RESTCall.sipRegisterRequired = field("sipRegisterRequired");
for (var key in RESTCall) {
setCookie(key, RESTCall[key]);
}
RESTCall.callee = field("callee");
var data = JSON.stringify(RESTCall);
sendREST(url, data);
startCheckCallStatus();
3. Capturing RTMP stream from other server with /pull/rtmp/pull REST query¶
var pullRtmp = function(uri, fn) {
console.log("Pull rtmp " + uri);
send(field("restUrl") + "/pull/rtmp/pull", {
uri: uri
}).then(
fn(STREAM_STATUS.PENDING)
).catch(function(e){
console.error(e);
fn(STREAM_STATUS.FAILED);
});
};
4. Stop capturing RTMP stream from other server with /pull/rtmp/terminate REST query¶
var terminateRtmp = function(uri, fn) {
console.log("Terminate rtmp " + uri);
send(field("restUrl") + "/pull/rtmp/terminate", {
uri: uri
}).then(
fn(STREAM_STATUS.STOPPED)
).catch(function(e) {
fn(STREAM_STATUS.FAILED);
console.error(e);
})
};
5. Mixer starting with /mixer/startup REST query¶
var startMixer = function(streamName) {
console.log("Start mixer " + streamName);
return send(field("restUrl") + "/mixer/startup", {
uri: "mixer://" + streamName,
localStreamName: streamName
});
};
6. Mixer stopping with /mixer/terminate REST query¶
var stopMixer = function(streamName, fn) {
console.log("Stop mixer " + streamName);
return send(field("restUrl") + "/mixer/terminate", {
uri: "mixer://" + streamName,
localStreamName: streamName
});
};
7. Adding/removing streams to mixer with /mixer/add and /mixer/remove REST queries¶
if ($(ctx).is(':checked')) {
// Add stream to mixer
send(field("restUrl") + "/mixer/add", {
uri: "mixer://" + mixerStream,
localStreamName: mixerStream,
remoteStreamName: stream
}).then(function(){
console.log("added");
});
} else {
// Remove stream from mixer
send(field("restUrl") + "/mixer/remove", {
uri: "mixer://" + mixerStream,
localStreamName: mixerStream,
remoteStreamName: stream
}).then(function(){
console.log("removed");
});
}
8. Injecting mixer stream to the SIP call with /call/inject REST query¶
function injectStreamBtn(ctx) {
var streamName = $("#injectStream").val();
if (!streamName) {
$("#injectStream").parent().addClass('has-error');
return false;
}
var $that = $(ctx);
send(field("restUrl") + "/call/inject_stream", {
callId: $("#sipCallId").val(),
streamName: streamName
}).then(function(){
$that.removeClass('btn-success').addClass('btn-danger');
$that.parents().closest('.input-group').children('input').attr('disabled', true);
}).catch(function() {
$that.removeClass('btn-danger').addClass('btn-success');
$that.parents().closest('.input-group').children('input').attr('disabled', false);
});
}
9. Re-publishing the SIP call stream to an RTMP server with /push/startup REST query¶
function startRtmpStream() {
if (!rtmpStreamStarted) {
rtmpStreamStarted = true;
var url = field("restUrl") + "/push/startup";
var RESTObj = {};
var options = {};
if ($("#mute").is(':checked')) {
options.action = "mute";
} else if ($("#music").is(':checked')) {
options.action = "sound_on";
options.soundFile = "sample.wav";
}
RESTObj.streamName = field("rtmpStream");
RESTObj.rtmpUrl = field("rtmpUrl");
RESTObj.options = options;
console.log("Start rtmp");
sendREST(url, JSON.stringify(RESTObj), startupRtmpSuccessHandler, startupRtmpErrorHandler);
sendDataToPlayer();
startCheckTransponderStatus();
}
}
10. Mute/unmute RTMP stream re-published sound¶
Mute sound with /push/mute code
function mute() {
if (rtmpStreamStarted) {
$("#mute").prop('disabled', true);
var RESTObj = {};
RESTObj.mediaSessionId = rtmpMediaSessionId;
var url = field("restUrl") + "/push/mute";
sendREST(url, JSON.stringify(RESTObj), muteSuccessHandler, muteErrorHandler);
}
}
Unmute sound /push/unmute code
function unmute() {
if (rtmpStreamStarted) {
$("#mute").prop('disabled', true);
var RESTObj = {};
RESTObj.mediaSessionId = rtmpMediaSessionId;
var url = field("restUrl") + "/push/unmute";
sendREST(url, JSON.stringify(RESTObj), muteSuccessHandler, muteErrorHandler);
}
}
11. Injecting additional sound to RTMP stream re-published.¶
Injecting sound from file with /push/sound_on code
function soundOn() {
if (rtmpStreamStarted) {
$("#music").prop('disabled', true);
var RESTObj = {};
RESTObj.mediaSessionId = rtmpMediaSessionId;
RESTObj.soundFile = "sample.wav";
RESTObj.loop = false;
var url = field("restUrl") + "/push/sound_on";
sendREST(url, JSON.stringify(RESTObj), injectSoundSuccessHandler, injectSoundErrorHandler);
}
}
Stop injecting sound from file with /push/sound_off code
function soundOff() {
if (rtmpStreamStarted) {
$("#music").prop('disabled', true);
var RESTObj = {};
RESTObj.mediaSessionId = rtmpMediaSessionId;
var url = field("restUrl") + "/push/sound_off";
sendREST(url, JSON.stringify(RESTObj), injectSoundSuccessHandler, injectSoundErrorHandler);
}
}
12. Hangup the SIP call with /call/terminate REST query.¶
function hangup() {
var url = field("restUrl") + "/call/terminate";
var currentCallId = { callId: callId };
var data = JSON.stringify(currentCallId);
sendREST(url, data);
}