Example of delivery of a video stream from SIP to RTMP server with sound injecting to the stream
This example demonstrates how to make a call to SIP, receive audio and video traffic from SIP in response, inject sound from file to the received video stream and then redirect the 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-2
sip-as-rtmp-2.js - a script dealing with REST queries to the WCS server
sip-as-rtmp-2.html - example page
The example may be tested at this URL:
https://host:8888/client2/examples/demo/sip/sip-as-rtmp-2/sip-as-rtmp-2.html
where host is WCS server address.
Analyzing the code
To analyze the code get sip-as-rtmp-2.js file version with hash c306c1bbf49bfcbd8e24be927ae95f63b7dbaaba that can be found here and is availabe to download in build 0.5.28.2747.
1. Load the test RTMP player on the page for further RTMP stream playback test.
function loadPlayer() { detectFlash(); var attributes = {}; attributes.id = "player"; attributes.name = "player"; attributes.styleclass="center-block"; var flashvars = {}; var pathToSWF = "../../dependencies/rtmp_player/player.swf"; var elementId = "player"; var params = {}; params.menu = "true"; params.swliveconnect = "true"; params.allowfullscreen = "true"; params.allowscriptaccess = "always"; params.bgcolor = "#777777"; swfobject.embedSWF(pathToSWF, elementId, "350", "400", "11.2.202", "expressInstall.swf", flashvars, params, attributes); }
2. 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 }); }
3. Making outgoing call with REST-request /call/startup
Call data (RESTCall) are collected from the boxes on page
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();
4. Getting the SIP call status with /call/find REST query.
function getStatus() { var url = field("restUrl") + "/call/find"; currentCallId = { callId: callId }; $("#callTrace").text(callId + " >>> " + field("rtmpUrl")); var data = JSON.stringify(currentCallId); sendREST(url, data); }
5. Sending DTMF signal with /call/send_dtmf REST query.
function sendDTMF(value) { var url = field("restUrl") + "/call/send_dtmf"; var data = {}; data.callId = callId; data.dtmf = value; data.type = "RFC2833"; data = JSON.stringify(data); sendREST(url, data); }
6. Re-publishing the SIP call stream to an RTMP server with sound file injecting to the stream by /push/startup REST query
function startRtmpStream() { if (!rtmpStreamStarted) { rtmpStreamStarted = true; var url = field("restUrl") + "/push/startup"; var RESTObj = {}; var options = {}; if ($("#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; sendREST(url, JSON.stringify(RESTObj), startupRtmpSuccessHandler, startupRtmpErrorHandler); sendDataToPlayer(); startCheckTransponderStatus(); } }
7. Getting RTMP stream status with /push/find REST query.
function getTransponderStatus() { var url = field("restUrl") + "/push/find"; var RESTObj = {}; // By default transponder's stream name will contain prefix "rtmp_" RESTObj.streamName = "rtmp_" + field("rtmpStream"); RESTObj.rtmpUrl = field("rtmpUrl"); sendREST(url, JSON.stringify(RESTObj), transponderStatusSuccessHandler, transponderStatusErrorHandler); }
8. Mute/unmute stream 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 with /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); } }
9. Injecting additional sound to RTMP stream.
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); } }
10. 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); }