continue : Streaming / broadcasting Live Video call to non webrtc supported browsers and media players

This blog is in continuation to the attempts / outcomes and problems in building a WebRTC to RTP media framework that successfully stream / broadcast WebRTC content to non webrtc supported browsers ( safari / IE ) / media players ( VLC )

Attempt 4: Stream the content to a WebRTC endpoint which is hidden in a video call . Pick the stream from vp8 object URL send to a streaming server

This process involved the following components :

  • WebRTC API : simplewebrtc on Chrome
  • Transfer mechanism from client to Streaming server:  webrtc media channel

Problems : No streaming server is qualified to handle a direct webrtc input and stream it on network .

Attempt 4.1 : Stream the content to a WebRTC endpoint . Do WebRTC Endpoint to RTP Endpoint bridge using Kurento APIs. 

Use the RTP port and ip address to input into a ffmpeg or gstreamer or VLC terminal command and out put a live H264 stream on another ip and port address .  

This process involved the following components :

  • API : Kurento
  • Transfer mechanism : HTML5 webrtc client -> application server hosting java -> media server -> application for webrtc media to RTP media conversation -> RTP player

Screenshots of attempts with Wowza to stream from a ip and port


problems :

  • The stream was black ie no video content .

Attempt 4.2 : Build a WebRTC Endpoint to Http endpoint in kurento and force the video audio encoding to be that of H264 and PCMU.

code for adding constraints to output media and forcing choice of codecs

MediaPipeline pipeline = kurento.createMediaPipeline();
    WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(pipeline).build();
    HttpGetEndpoint httpEndpoint=new HttpGetEndpoint.Builder(pipeline).build();

    org.kurento.client.Fraction fr= new org.kurento.client.Fraction(1, 30);         
    VideoCaps vc= new VideoCaps(VideoCodec.H264,fr);

    AudioCaps ac= new AudioCaps(AudioCodec.PCMU, 65536);


code for using gstreamer filter to force the output in raw format . It is a alternate solution to above

//basic media operation of 1 pipeline and 2 endpoinst
MediaPipeline pipeline = kurento.createMediaPipeline();
WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(pipeline).build();
RtpEndpoint rtpEndpoint = new RtpEndpoint.Builder(pipeline).build();

//adding Gstream filters 
GStreamerFilter filter1 = new GStreamerFilter.Builder(pipeline, "videorate max-rate=30").withFilterType(FilterType.VIDEO).build();
GStreamerFilter filter2 = new GStreamerFilter.Builder(pipeline, "capsfilter caps=video/x-h264,width=1280,height=720,framerate=30/1").withFilterType(FilterType.VIDEO).build();
GStreamerFilter filter3 = new GStreamerFilter.Builder(pipeline, "capsfilter caps=audio/x-mpeg,layer=3,rate=48000").withFilterType(FilterType.AUDIO).build();

//connecting all poin ts to one another 
webRtcEndpoint.connect (filter1); 
filter1.connect (filter2); 
filter2.connect (filter3); 
filter3.connect (rtpEndpoint);

// RTP SDP offer and answer
String requestRTPsdp = rtpEndpoint.generateOffer();

problem : The output is still webm

Attempt 5  : Use a RTP SDP Endpoint ( ie a SDP file valid for a given session ) and use it to play the WebRTC media over Wowza streaming server

This process involved the following components

  1. WebRTC Stream and object URL of the blob containing VP8 media
  2. Kurento  WebRTC Endpoint  bridge to generate SDP
  3. Wowza Streaming server

code for kurento to generate a SDP file from WebRTC to RTP bridge

@RequestMapping(value = "/rtpsdp", method = RequestMethod.POST)
private String processRequestrtpsdp(@RequestBody String sdpOffer)
throws IOException, URISyntaxException, InterruptedException {

//basic media operation of 1 pipeline and 2 endpoinst
MediaPipeline pipeline = kurento.createMediaPipeline();
WebRtcEndpoint webRtcEndpoint = new WebRtcEndpoint.Builder(pipeline).build();
RtpEndpoint rtpEndpoint = new RtpEndpoint.Builder(pipeline).build();

//connecting all poin ts to one another 
webRtcEndpoint.connect (rtpEndpoint);

// RTP SDP offer and answer
String requestRTPsdp = rtpEndpoint.generateOffer();

// write the SDP conector to an external file
PrintWriter out = new PrintWriter("/tmp/test.sdp");

HttpGetEndpoint httpEndpoint = new HttpGetEndpoint.Builder(pipeline).build();
PlayerEndpoint player = new PlayerEndpoint.Builder(pipeline, requestRTPsdp).build();

// Playing media and opening the default desktop browser;
String videoUrl = httpEndpoint.getUrl();
System.out.println(" ------- video URL -------------"+ videoUrl);

// send the response to front client
String responseSdp = webRtcEndpoint.processOffer(sdpOffer);

return responseSdp;

problems : wowza doesnt not recognize the WebRTC SDP and play the video

screenshot of wowza with SDP input

Screenshot from 2015-01-30 15:28:59

Attempt 5.1 : Use a RTP SDP Endpoint ( ie a SDP file valid for a given session ) and use it to play the WebRTC media over Default Ubuntu media player 

SDP file formed contains contents such as :

o=- 3631611195 3631611195 IN IP4
s=Kurento Media Server
c=IN IP4
t=0 0
m=audio 42802 RTP/AVP 98 99 0
a=rtpmap:98 OPUS/48000/2
a=rtpmap:99 AMR/8000/1
a=rtpmap:0 PCMU/8000
a=ssrc:2713728673 cname:user59375791@host-ad1117df
m=video 35946 RTP/AVP 96 97 100 101
a=rtpmap:96 H263-1998/90000
a=rtpmap:97 VP8/90000
a=rtpmap:100 MP4V-ES/90000
a=rtpmap:101 H264/90000
a=ssrc:93449274 cname:user59375791@host-ad1117df

problem : deformed media

screenshot of playing from a SDP file

Screenshot from 2015-01-29 17:42:21

Attempt 5.2 : Use a RTP SDP Endpoint ( ie a SDP file valid for a given session ) and use it to play the WebRTC media over VLC using socket input

problem : nothing plays

screenshot of VLC connected to play from socket and failure to play anything

Screenshot from 2015-01-21 17:49:52

Attempt 5.3: Create a WebRTC endpoint and connected it to RTP endpoint via media pipelines . Also make the RTP SDP offer and answering the same . Play with ffnpeg / ffplay / gst playbin

String requestRTPsdp = rtpEndpoint.generateOffer();

Write the requestRTPsdp to a file and obtain a RTP connector endpoint with Application/SDP .It plays okay with gst playbin ( 10 secs without audio )

Successful attempt to play from a gst playbin

gst-launch -vvv playbin uri=file:///tmp/test.sdp 

donekurento streaming

but refuses to be played by VLC , ffplay and even wowza . The error generated with

ffmpeg -i test.sdp -vcodec copy -acodec copy -f mpegts output-file.ts


ffmpeg -re -i test.sdp -vcodec h264 -acodec mp3 -f mpegts “udp://″


Could not find codec parameter for stream1 ( video:h263, none ) .Other errors types are , Could not write header for output file <incorrect codec parameter > output file is empty nothing was encoded

Error screenshots of trying to play the RTP SDP file with ffmpeg

ffmpeg error kurebto1 ffmpeg error kurebto2

Attempt 6 : Use a WebRTC capable media and streaming server ( eg Kurento )  to pick a live stream of VP8 . Convert the VP8 to H268  ( ffmpeg / RTP endpoint ) . Convert H268 to Mp4 using MP4 parser and pass to a streaming server  ( wowza)

In process . to be updated .

Streaming / broadcasting Live Video call to non webrtc supported browsers and media players

As the title of this article suggests I am going to pen my attempts of streaming / broadcasting Live Video WebRTC call to non WebRTC supported browsers and media players such as VLC , ffplay , default video player in Linux etc .

I am currently attempting to do this by making my own MP4 engine from WebRTC feed . However I am sharing my past experiments in hope of helping someone whose objective is not the same as mine and might get some help from these threads .

Attempt 1 : use one to many brodcasting API :

<!DOCTYPE html>
<html id=”home” lang=”en”>

<meta http-equiv=”Content-Type” content=”text/html; charset=UTF-8″>
<meta charset=utf-8>
<meta name=”viewport” content=”width=device-width, initial-scale=1.0, user-scalable=no”>
<meta name=”author” content=”altanai”>
<meta http-equiv=”X-UA-Compatible” content=”IE=edge,chrome=1″>

<link rel=”stylesheet” type=”text/css” href=”style.css”>



<table class=”visible”>
<td style=”text-align: right;”>
<input type=”text” id=”conference-name” placeholder=”Broadcast Name”>
<select id=”broadcasting-option”>
<option>Audio + Video</option>
<option>Only Audio</option>
<button id=”start-conferencing”>Start Broadcasting</button>
<table id=”rooms-list” class=”visible”></table>

<div id=”participants”></div>

<script src=”RTCPeerConnection-v1.5.js”></script>
<script src=”firebase.js”></script>
<script src=”broadcast.js”></script>
<script src=”broadcast-ui.js”></script>




It uses API The broadcast is in one direction only where the viewrs are never asked for their mic / webcam permission .

problem : The broadcast is for WebRTC browsers only and doesnt support non webrtc players / browsers

Attempt 1.1: Stream the media directly to nodejs through websocket

window.addEventListener('DOMContentLoaded', function() {

var v = document.getElementById('v');
navigator.getUserMedia = (navigator.getUserMedia || 
navigator.webkitGetUserMedia || 
navigator.mozGetUserMedia || 

if (navigator.getUserMedia) {
// Request access to video only
function(stream) {
var url = window.URL || window.webkitURL;
v.src = url ? url.createObjectURL(stream) : stream;;

var ws = new WebSocket('ws://localhost:3000', 'echo-protocol');
waitForSocketConnection(ws, function(){

console.log(" url.createObjectURL(stream)-----", url.createObjectURL(stream))

console.log("message sent!!!"); 

function(error) {
alert('Something went wrong. (error code ' + error.code + ')');
else {
alert('Sorry, the browser you are using doesn\'t support getUserMedia');

//Make the function wait until the connection is made...
function waitForSocketConnection(socket, callback){
function () {
if (socket.readyState === 1) {
console.log("Connection is made")
if(callback != null){

} else {
console.log("wait for connection...")
waitForSocketConnection(socket, callback);

}, 5); // wait 5 milisecond for the connection...

problem : The video is in form of buffer and doesnot play

Attempt 2: Record the WebRTC media ( 5 secs each ) into chunks of webm format->  transfer them to other end -> append the chunks together like a regular file 

This process involved the following components :

  • Recorder Javascript library : RecordJs
  • Transfer mechanism : Record using RecordRTC.js -> send to other end for media server -> stitching together the small webm files into big one at runtime and play
  • Programs :

Code for video recorder

navigator.getUserMedia(videoConstraints, function(stream) {

video.onloadedmetadata = function() {
video.width = 320;
video.height = 240;

var options = {
type: isRecordVideo ? 'video' : 'gif',
video: video,
canvas: {
width: canvasWidth_input.value,
height: canvasHeight_input.value

recorder = window.RecordRTC(stream, options);
video.src = URL.createObjectURL(stream);
}, function() {
if (document.getElementById('record-screen').checked) {
if (location.protocol === 'http:')
alert('<https> is mandatory to capture screen.');
alert('Multi-capturing of screen is not allowed. Capturing process is denied. Are you enabled flag: "Enable screen capture support in getUserMedia"?');
} else
alert('Webcam access is denied.');

Code for video append-er

var FILE1 = '1.webm';
var FILE2 = '2.webm';
var FILE3 = '3.webm';
var FILE4 = '4.webm';
var FILE5 = '5.webm';

var NUM_CHUNKS = 5;
var video = document.querySelector('video');

window.MediaSource = window.MediaSource || window.WebKitMediaSource;
if (!!!window.MediaSource) {
alert('MediaSource API is not available');

var mediaSource = new MediaSource();

video.src = window.URL.createObjectURL(mediaSource);

function callback(e) {

var sourceBuffer = mediaSource.addSourceBuffer('video/webm; codecs="vorbis,vp8"');

GET(FILE1, function(uInt8Array) {

var file = new Blob([uInt8Array], {type: 'video/webm'});
var i = 1;

(function readChunk_(i) {

var reader = new FileReader();

reader.onload = function(e) {

sourceBuffer.appendBuffer(new Uint8Array(;

if (i == NUM_CHUNKS) mediaSource.endOfStream();

else {
if (video.paused) {; // Start playing after 1st chunk is appended.



})(i); // Start the recursive call by self calling.

mediaSource.addEventListener('sourceopen', callback, false);
mediaSource.addEventListener('webkitsourceopen', callback, false);
mediaSource.addEventListener('webkitsourceended', function(e) {
logger.log('mediaSource readyState: ' + this.readyState);
}, false);

// function get the video via XHR
function GET(url, callback) {

var xhr = new XMLHttpRequest();'GET', url, true);
xhr.responseType = 'arraybuffer';

xhr.onload = function(e) {

if (xhr.status != 200) {
alert("Unexpected status code " + xhr.status + " for " + url);
return false;

callback(new Uint8Array(xhr.response));

Shortcoming of this approach

  1. The webm files failed to play on most of the media players
  2. The recorder can only either record video or audio file at a time .

Attempt 2.1: Record the WebRTC media ( 5 secs each ) into chunks of webm format ( RecordRTC.js) >  Use Kurento JS script ( kws-media-api,js) to make a HTTP Endpoint to recorded Webm files  -> append the chunks together like a regular file at runtime 

function getByID(id) {
return document.getElementById(id);

var recordAudio = getByID('record-audio'),
recordVideo = getByID('record-video'),
stopRecordingAudio = getByID('stop-recording-audio'),
stopRecordingVideo = getByID('stop-recording-video'),

var canvasWidth_input = getByID('canvas-width-input'),
canvasHeight_input = getByID('canvas-height-input');

var video = getByID('video');
var audio = getByID('audio');

var videoConstraints = {
audio: false,
video: {
mandatory: {},
optional: []

var audioConstraints = {
audio: true,
video: false

const ws_uri = 'ws://localhost:8888/kurento';
var URL_SMALL="http://localhost:8080/streamtomp4/approach1/5561840332.webm";

var audioStream;
var recorder;

recordAudio.onclick = function() {
if (!audioStream)
navigator.getUserMedia(audioConstraints, function(stream) {

if (window.IsChrome) stream = new window.MediaStream(stream.getAudioTracks());
audioStream = stream;

audio.src = URL.createObjectURL(audioStream);
audio.muted = true;;

// "audio" is a default type
recorder = window.RecordRTC(stream, {
type: 'audio'
}, function() {});
else {
audio.src = URL.createObjectURL(audioStream);
audio.muted = true;;
if (recorder) recorder.startRecording();

window.isAudio = true;

this.disabled = true;
stopRecordingAudio.disabled = false;

stopRecordingAudio.onclick = function() {
this.disabled = true;
recordAudio.disabled = false;
audio.src = '';

if (recorder)
recorder.stopRecording(function(url) {
audio.src = url;
audio.muted = false;;

document.getElementById('audio-url-preview').innerHTML = '<a href="' + url + '" target="_blank">Recorded Audio URL</a>';

recordVideo.onclick = function() {

function recordVideoOrGIF(isRecordVideo) {
navigator.getUserMedia(videoConstraints, function(stream) {

video.onloadedmetadata = function() {
video.width = 320;
video.height = 240;

var options = {
type: isRecordVideo ? 'video' : 'gif',
video: video,
canvas: {
width: canvasWidth_input.value,
height: canvasHeight_input.value

recorder = window.RecordRTC(stream, options);
video.src = URL.createObjectURL(stream);
}, function() {
if (document.getElementById('record-screen').checked) {
if (location.protocol === 'http:')
alert('<https> is mandatory to capture screen.');
alert('Multi-capturing of screen is not allowed. Capturing process is denied. Are you enabled flag: "Enable screen capture support in getUserMedia"?');
} else
alert('Webcam access is denied.');

window.isAudio = false;

if (isRecordVideo) {
recordVideo.disabled = true;
stopRecordingVideo.disabled = false;
} else {
recordGIF.disabled = true;
stopRecordingGIF.disabled = false;

stopRecordingVideo.onclick = function() {
this.disabled = true;
recordVideo.disabled = false;

if (recorder)
recorder.stopRecording(function(url) {
video.src = url;;
document.getElementById('video-url-preview').innerHTML = '<a href="' + url + '" target="_blank">Recorded Video URL</a>';


/*--------------------------broadcasting -----------------------------------*/

function onerror(error)
console.log( " error occured");

broadcast.onclick = function() {
var videoOutput = document.getElementById("videoOutput");

KwsMedia(ws_uri, function(error, kwsMedia)
if(error) return onerror(error);

// Create pipeline
kwsMedia.create('MediaPipeline', function(error, pipeline)
if(error) return onerror(error);

// Create pipeline media elements (endpoints & filters)
pipeline.create('PlayerEndpoint', {uri: URL_SMALL},
function(error, player)
if(error) return console.error(error);

pipeline.create('HttpGetEndpoint', function(error, httpGet)
if(error) return onerror(error);

// Connect media element between them
player.connect(httpGet, function(error, pipeline)
if(error) return onerror(error);
// Set the video on the video tag
httpGet.getUrl(function(error, url)
if(error) return onerror(error);

videoOutput.src = url;


// Start player
if(error) return onerror(error);


// Subscribe to HttpGetEndpoint EOS event
httpGet.on('EndOfStream', function(event)
console.log("EndOfStream event:", event);


problem : dissecting the live video into small the files and appending to each other on reception is an expensive , time and resource consuming process . Also involves heavy buffering and other problems pertaining to real-time streaming .

Attempt 2.2 : Send the recorded chunks of webm to a port on linux server . Use socket programming to pick up these individual files and play using  VLC player from UDP port of the Linux Server

Screenshot from 2015-01-22 15:32:51

Attempt 2.3: Send the recorded chunks of webm to a port on linux server socket . Use socket programming to pick up these individual webm files and convert to H264 format so that they can be send to a media server. 

This process involved the following components :

  • Recorder Javascript library : RecordJs
  • Transfer mechanism :WebRTC endpoint -> Call handler ( Record in chunks ) -> ffmpeg / gstreamer to put it on RTP -> streaming server like wowza – > viewers
  • Programs : Use HTML webpage Webscoket connection -> nodejs program to write content from websocket to linux socket -> nodejs program to read that socket and print the content on console

Program to transfer the webm recorder files over websocket to nodejs program

//Make the function wait until the connection is made...
function waitForSocketConnection(socket, callback){
function () {
if (socket.readyState === 1) {
console.log("Connection is made")
if(callback != null){

} else {
console.log("wait for connection...")
waitForSocketConnection(socket, callback);

}, 5); // wait 5 milisecond for the connection...

function previewFile() {
var preview = document.querySelector('img');
var file = document.querySelector('input[type=file]').files[0];
var reader = new FileReader();

reader.onloadend = function () {

preview.src = reader.result;
console.log(" reader result ", reader.result);

var video=document.getElementById("v");
console.log(" video played ");

var ws = new WebSocket('ws://localhost:3000', 'echo-protocol');

waitForSocketConnection(ws, function(){
console.log("message sent!!!"); 


if (file) {
// converts to base64 encoded string of the file data


} else {
preview.src = "";

Program for Linux Sockets sender which creates the socket for the webm files

var net = require('net');
var fs = require('fs');
var socketPath = '/tmp/tfxsocket';
var http = require('http');
var stream = require('stream');
var util = require('util');

var WebSocketServer = require('ws').Server;
var port = 3000;
var serverUrl = "localhost";

var socket;
/*--------------------------------http server -----------------------------*/
var server= http.createServer(function (request, response) {


server.listen(port, serverUrl);

console.log('HTTP Server running at ',serverUrl,port);

/*--------------------------------websocket server -----------------------------*/

var wss = new WebSocketServer({server: server});

wss.on("connection", function(ws) {
console.log("websocket connection open");

ws.on('message', function (message) {
console.log(" stream recived from broadcast client on port 3000 ");

var s = require('net').Socket();

console.log(" send the stream to socketPath",socketPath); 

ws.on("close", function() {
console.log("websocket connection close")


Program for Linux Socket Listener using nodejs and socket . Here the socket is in node /tmp/mysocket

var net = require('net');

var client = net.createConnection("/tmp/mysocket");

client.on("connect", function() {
console.log("connected to mysocket");

client.on("data", function(data) {

client.on('end', function() {
console.log('server disconnected');

Output 1: Video Buffer displayed

Screenshot from 2015-01-22 15:35:06 (copy)

Output 2 : Random data from Video displayed

Screenshot from 2015-01-23 12:57:35

ffmpeg format of transfering the content from socket to UDP IP and port

ffmpeg -i unix://tmp/mysocket -f format udp://

problems of this approach : The video was on a passing stage from the socket and contained no information as such when tried to play / show console

Attempt 3 : Send the live WebRTC stream from Kurento WebRTC endpoint to Kurento HTTP endpoint . play using  Mozilla VLC web plugin

VLC mozilla plugin can be embedded by :

autoplay=”yes” loop=”no” hidden=”no”
target=”rtp://@″ />

screenshot of failure on part of Mozilla VLC plugin to play from a WebRTC endpoint

Screenshot from 2015-01-29 10:37:06Screenshot from 2015-01-29 10:37:17

Screenshot from 2015-01-29 12:06:14

problem : VLC mozilla plugin was unable to play the video


The 4th , 5th and 6th sections of this article are in the next blog :

continue : Streaming / broadcasting Live Video call to non webrtc supported browsers and media players

Building WebRTC API , platform and deploying as a browser component

So I haven’t written anything worthy in a while , just published some posts that were lying around in my drafts . Here I write about the main thing . some thing awesome that I was trying to accomplish in the last quarter .

TFX platform

TFX Sessions is a plug and play platform for VoIP ( voice over IP ) scenarios.  Intrinsically it  is a very lightweight API package and shipped in form of a Chrome Extension . It is a turn-key solution when parties want instant audio/audio communication without any sign-in ,plugin installation or additional downloads  . Additionally TFX Sessions is packaged with some interesting plugins which enable the communicating parties to get the interactive and immersive experience as in a face to face meeting.

So here is the thing . I have been thinking of making  a utterly simple WebRTC API  that has everything needed to build bigger aggregate projects but the available solutions are either just to basic or much too complex . So I initially stardted writing my own getuserMedia APIs, but left it midway and picked up simplewebrtc API instead for want of time .Then I focused on the main crux  which was my plugin/ widget/ applications based platform .

TFX WebRTC platform architecture . socket io signalling

TFX WebRTC platform architecture . socket io signalling

Note that when I say plugin it isnt actually a plugin for chrome but for my TFX platform alone . Essentially it is any web project that wants communication over webrtc channel .

So now the platform is ready I have core APIs , widgets and signalling server. Then ofcourse came up the subject of enterprise internet blocking my communication stream . Time for TURN . I used a coturn server which mingled with my platform just as water dissolves coffee :) .

So here is the final architecture .

TFX whitepaper v2.0

TFX platform Server client components . WebRTC media and socketio communication . Build as chrome Extension

Alright so that’s there . Tada the platform is alive and kicking . Right now in beta stage however . Intensive testing going on here . However here are some screenshots that are from my own developer version .

TFX recording widget

TFX recording widget

TFX face detection and overlay widget

TFX face detection and overlay widget

TFX multilingual communication

TFX multilingual communication

TFX screen-sharing

TFX screen-sharing

TFX video Filters

TFX video Filters

TFX audio visualizer

TFX audio visualizer

TFX text messaging widget

TFX text messaging widget

TFX cross domian access . flicker here

TFX cross domain access . flicker here

TFX draw widget

TFX draw widget

TFX code widget supportes many programming languages

TFX code widget supportes many programming languages

TFX  webrtc dynamic stats

TFX webrtc dynamic stats

TFX  introduction widget

TFX introduction widget

please note that the plugins . widgets / applications described above have been made with the help of third party APIs. The detailed summary of every widget and its procedure , development , setup is to be described in my next blog .

TFX startup screen

TFX startup screen

TFX create / join room

TFX create / join room

My objective behind all this work was to come up with a modular solution which can exists as a API , platform as a service  , simple browser extension and web project .

SIP messages

1. Request Message

Request Message


REGISTER A Client use this message to register an address with a SIP server
INVITE A User or Service use this message to let another user/service participate in a session. The body of this message would include a description of the session to which the callee is being invited.
ACK This is used only for INVITE indicating that the client has received a final response to an INVITE request
CANCEL This is used to cancel a pending request
BYE A User Agent Client use this message to terminate the call
OPTIONS This is used to query a server about its capabilities


2. Response Message




1xx Provisional The request has been received and processing is continuing
2xx Success An ACK, to indicate that the action was successfully received, understood, and accepted.
3xx Redirection Further action is required to process this request
4xx Client Error The request contains bad syntax and cannot be fulfilled at this server
5xx Server Error The server failed to fulfill an apparently valid request
6xx Global Failure The request cannot be fulfilled at any server 

, based on RFC 3261

Sip server Brekeke

We used Brekeke SIP server to run our SIP applications . Although there are newer versions of Brekeke SIP server out now . More awesome than before , we prefer using the old one for the sake of not messing with legacy SIP applications . The official site for brekeke is – .

A general architecture of Brekeke SIP server is . brekeke

Here are the steps of installing and configuring a Brekeke SIP server .

Step 1: Download the Server form and run the setup file .





brekek2 brekek3 brekek6

brekeke4 brekeke5 brekeke7  brekeke9

Step 2: It is always good to give a look to README file . brekeke11

Step 3: Run the local implementation of SIP server at localhost or at port 8080brekeke12

Step 4: Important is to get the license which will help us activate the SIP server . One can obtain a free license from

Step 5 : Once the license is activates , we can goto the console screen after loggin with default username and password sa . brekeke13

Step 6 : Once we are at console , we could add/ delete / modify parameters like port , start/shutdown status etc . brekeke14 brekeke14_001Step 7 : Once the server is all setup , just add the IP and port of SIP server to SIP clients server filed . Now all the SIP request and response will be catered by this SIP Server

Steps for building and deploying WebRTC solution

Step 1  : Do it myself 

Pick any WebRTC API and run its demos . It works kool . download and run in local-machine with nodejs server . Awesome . Everything is Awesome !!

You can learn more about some WS based WebRTC API here:

If you are a diehard telecom engineer and only want SIP based WebRTC solutions go here :

step 1 of building and deploying a WebRTC solution

Step 2 : Give it to friends 

Now what good is it doing to anyone if its running locally on my machine with addresses like localhost and  . Let us put it on the cloud and atleast let my colleague / friends enjoy it .  Now I need two things

1. Cloud Web Server and Nodejs signalling server . That is okay use amazon’s Ec2. works for most of the people most of the time .

2. STUN server for address mapping and NAT . For this I have to rely on google’s default STUN server Easy and free .

step 2 : building and deploying a WebRTC solution

Note that this step only works if everyone you want to connect to is either on same intranet or on public internet without and UDP blocks / firewalls / restriction .

There you go everything is looking good from here now . But wait a sec what about step 3 . Lets see what that has to say .

Step 3: Call people in a inter network fashion 

Sure the architecture I have setup is bound to work everywhere , but wait it doesnt . Error in connectivity , errors in console , blank video are the problems that might appear . So well err things begin to get a bit complicated from here . To bypass network firewalls , corporate net policies , UDP blocks and filters we require a TURN server .

Now we have 3 options to choose from

1.  Use a wildly popular

2. Build your own TURN server with RFC 5766 , or rather easier would be to use any open source TURN server code available in Github

3. Pay and use a commercial TURN service provider or you can even use their trail version to see if things work out for you .

Remember you can use any TURN service it does not affecr your WebRTC API functionality . All we need to do is add it to Peerconnection confih configuration like

peerConnectionConfig: {
{“url”: < stunserver address >},
{“username”:”xx”,”url”:< turn server address transport=udp>,”credential”:”yy”},
{“username”:”xx”,”url”:< turn server address transport=tcp> , “credential”:”yy”}]

step 3 : Call between Inter network machines


There we go , now anyone from anywhere should be able to use our WebRTC setup for making audio , video calls or just exchanging data via DataChannel ( like screen-sharing , file transfer , messages , playing games , collaborative office work etc )  .

The setups covers scenarios wherein user is on office corporate network , home network , mobile network , no problem as long as he / she has a webrtc enables browser ( read Chrome , Mozilla , Opera ) .

It is noteworthy that ideally voice should be traversing on TCP while video and data can go around in UDP however unless restrained the WebRTC API’s self determine the best protocol to route the packets / stream .


you can read more about best WebRTC frameworks and code in this book

Current State of WebRTC

In the last few months I have been observing how the course for WebRTC is turning out so far . Unfortunately contrary to my expectations the fundamental holes in WebRTC specification are still the same with less being done to fulfill them . Ofcourse now there are abundance of interactive
WebRTC API each using a new masking method to call the same old WebRTC API function of getusermedia and peer-connection . Few of these I will list down in this blog but no concrete stable reliable  guide to setup the backbone network ( yes i am referring to Media inter conversion , relay , TURN , STUN servers ) which is left to telecom software engineer / developer to find out and configure . Instead I see many commercial service providers who claim of providing their backend for our WebRTC implementation but that in my opinion completely defeats the objective of WebRTC based communication .  WebRTC was meant to *everything you can’t do with proprietary communication tools and networks* .

Well moving on , here are some nice API implementations of WebRTC ( only for Websockets no SIP )


apprtc apprtc2 apprtc3 apprtc4 apprtc5


talky3 talky2 talky1 talky4 talky5 talky63. tokbox

tokbox2tokbox1tokbox7 tokbox6 tokbox5 tokbox4 tokbox3


IPTV ( Internet Based Television ) appliactions

We know the power of Internet protocol suit as it takes on the world of telecom . Alreday half of Communication has been transferred from legacy telecom signalling protocols like SS7 to IP based communication ( Skype , Hangouts , whatsapp , facebook call ) . The TV service providers too are largely investing in IP based systems like SIP and IMS to deliver their content over Telecom’s IP based network ( Packet switched ).

A consumer today wants HD media content anytime anywhere . The traditional TV solutions just dont match upto the expectations anymore . The IPTV provider in todays time must make investments to deliver content that is media-aware, and device-aware. Not only this it should be  personal, social, and interactive . after all its all about user  experience.

Few popular applications for IPTV solutions developers are

  • Menu overlay with detailed description of channels , categories , programs , movies
  • Replay option also referred to as timeshift . It allows a user to pause , resume and  record the show in his absence and view it later
  • Video on demand which concerns paying and viewing music albums , movies etc on demand
  • Live streaming of events such as president speech , tennis match etc .

Application that can be build around the IPTV context

  • Parental Control to realtime view , monitor and control what your child is watching on the IPTV
  • Watch the surveillance  footage from IP cameras anywhere
  • Real time communication on IPTV  with advanced features like call continuity , content sync .
IPTV overview

IPTV overview

Developing a OTT ( Over the Top ) Communication application

Market trends are really not in favor of Telecom Service /providers with increasing use of OTT application like watsapp , Facebook messenger , Google hangouts , skype  , viber , etc .

Four Depleting waves of revenue – Operator’s dilemma

  1. Messaging – OTT messaging cost operators $13.9 billion, or 9% of message revenue in 2013
  2. Voice – Voice services under threat from VOIP services like Skype, Viber
  3. OTT apps – Voice & Message apps have been the operator’s biggest headache. Its time Operator should launch its own OTT Services
  4. Data Traffic – The utilization is yet to reach its peak. Will face challenges from  WiFi access
  5. Critical Pain areas – Erosion of Operator’s revenue from voice and (especially) messaging

At this stage it is crucial for a telecom Service provider / Operator to enter the Apps market and bring forth a Messenger which is more powerful , interactive and awesome than a OTT application.  Fortunately the Operator can always couple this application with his background telecom infrastructure to provide the edge in performance and functionalists .

Let us analyse the current roadblocks for the Telecom Operator through the following figure

Road block while developing a OTT application for a Telecom Service Provider

Road block while developing a OTT application for a Telcom Service Provider

Next we find the way of solving the problems and integrating them together to form a Solution that is described in the figure below :

OTT Application for Telecom Service provider

OTT Application for Telecom Service provider

This writeup outlines the process of creating a OTT application for a Telecom Service Provider .

Components for the application include cloud Address Book , Video Chatting , Location share , Contact synchronization ,REST based thin  client , OS and device agnostic etc shown in the figure below


The Application  is designed to close knit with Operator’s own infrastructure hence the crucial entities like Network Address Book , Location Service are synced and fetched from Backend Network .

High level design of the OTT application is provided below :

Technical high level digram for developing Telecom Operator's own OTT application

Technical high level digram for developing Telecom Operator’s own OTT application

Feature Overview

Smart Address Book

  • Automatic: Get contacts from Gmail, Facebook
  • Fast search by first, last name, frequently
  •   dialed number
  • Roadmap: View calendar events
  • Personal: Get image from Gmail and display in   contacts list

Geo Location

  • Share own location during chatting
  • Get map for calculating the distance between two chat users
  • Roadmap : Trigger device (say Switch on/off AC before reaching home) from a threshold distance away from home   location


  • Ad-hoc Chat
  • Session Based Chat
  • Voice Input for texting
  • Presence information of contacts
  • RoadMap: Legacy message integration


  • Voice call to mobile
  • Voice call to PSTN
  • Video call to other @imAll user
  • Share images during voice call to other

Device agnostic

  • Compatible with IOS, windows
  • Can run as native app on ipad
  • Can run as browser client on windows
  • RoadMap: native app for android, windows phone,blackberry10


  • To upgrade the application and provide enganced and enrich service support the I propose the following roadmap.
  • From plain vanilla voice and video calling ( supported by every other OTT application ) our application should progress towards  legacy telecom support whihc included PSTN , GSM , ISDN etc . This requires backbone of telecom network and a good setup for media codec conversion to suit various legacy media codecs .
  • Ott4
  • To keep the interest of customers it is essential that the application be supported on other popular OTT services like skype  , Gtalk . for exmaple a caller should be able to make call from Skype  / Gtalk to our application .
  • Multilingual capabilities, support for larger protocol spectrum will just act like icing on the cake .
  • How does it benefit the Operator??
  1.  Saves on development cost and time
  2.  Device Agnostic OTT Applications
  3. Simplified Service deployment
  4. Saves licensing cost per client
  5. Reuses existing Messaging and   Address Book service logic.
  6. Open New Revenue Streams for operator
  7. No separate SIP stack required for the client
  8.  Faster Time to Market

Developing a Service Creation Environment for SIP Applications

I hoped of making a SIP application Development environment a year back and worked towards it earnestly . Sadly I wasn’t able to complete the job yet I have decided to share a few things about it here .

Aim :

Develop  a SCE ( Service Creation Environment ) to addresses all aspects of lifecycle of a Service, right from creation/development, orchestration, execution/delivery, Assurance and Migration/Upgrade of services.

Similar market products :

  • Open/cloud Rhino
  • Mobicents and Telestax

Limitations of open source/other market products:

  • Free versions of the Service Creation Environments do not offer High Availability.
  • High Cost of Deployment grade versions.

Solution Description

I propose a in-house Java based Service Creation Environment “SLC SCE”. The SLC SCE will enable creation of JAINSLEE based SIP  services. It can be used to develop and deploy carrier-grade applications that use SS7 and IMS based protocols such as INAP, CAP, Diameter and SIP as well as IT / Web protocols such as HTTP and XML.


  • Service Agility
  • Significantly Lower price points
  • Open Standards eliminate Legacy SCP Lock-in


Java-based service creation environment (SCE) – 1.5 Months

Graphical User Interface (GUI) and schematic representations to help in the design, maintenance and support of applications – 1.5 months

SIP Resource Adapter – 1 month


In essence it encompasses the idea of developing the following

  1. SIP stack
  2. Javascript API’s
  3. Java Libraries for calling SIP stack
  4. Eclipse plugin to work with the SIP application development process
  5. Visual Interface to view the logic of application and possible errors / flaws
  6. SDKs (  Service Development Kit) , which are development Environment themselves

Extra Effort required to put in to make the venture successful

  1. Demo applications for basic SIP logic like Call screening , call rerouting .
  2. tutorial to create , deploy and run application from scratch . Aimed at all sections ie web developer , telecom engineer , full stack developer etc .
  3. Some opensource implementation on public repositories like Github , Google code , SourceForge
  4. Perform active problem solving on Stackoverflow , CodeRanch , Google groups and  other forums .


Call Continuity from Mobile GSM network to WebRTC

In  the present age of IP telephony when telecom convergence is the big thing all around the world , need of the hours is to enable fixed and mobile Service Providers ( SP )  to monetize the subscriber’s phone number by extending it to new web based services.SPs can offer a WebRTC Communicator endpoint that uses the same phone number as the subscriber’s fixed or mobile phone.

Advanced features enable calls to be transferred between fixed-line, mobile and WebRTC endpoints.

Find the diagram depicting this below :

Transfer mobile callto WebRTC session

Transfer mobile callto WebRTC session

SPs can offer 3rd Party WebRTC endpoints to access the user’s phone number and subscription . E.g. enable web applications such as Facebook, Amazon or Netflix to allow their users to make/receive calls or messages directly from the web applications

Revenue Streams :

  • monthly fee for access to WebRTC endpoints and for receiving calls from by 3rd Party WebRTC endpoints
  • One time upgrade fees for Accessing the Web service integration with telecom network like a plan upgrade

Brownie points

  • No software is required to be downloaded on the subscriber’s computer, tablet or mobile phone
  • No desktop support required for the service provider

Plans For Consumer Customers:

  • Subscribers can use the WebRTC endpoints on their computers, tablets or mobile phones as a fixed-line device at home, as a desktop solution when away from home and to avoid international tolls when traveling
  • Subscribers can connect their web services (e.g. Websites , Facebook, Amazon, Netflix) to their fixed or mobile services subscriptions using their SP-provided phone number

Plans For SP Enterprise Customers:

  • Enterprises can deploy a WebRTC endpoint for their employees that provides a single corporate communications endpoint that can be connected to any of the corporation’s UC/PBX and Call Recording systems
  • Employees can use the WebRTC endpoint as their office phone at work, home or when traveling
  • Connects to all leading UC/PBX and Recording platforms simultaneously
  • Enterprises can deploy a single WebRTC endpoint across all their UC/PBX and Recording platforms – current and future
  • Easy for IT departments to deploy – no software is required to be downloaded to employees’ computers, tablets or mobile phones
  • Enables corporate policies and features from the WebRTC endpoint including
  • Displaying the corporate identity
  • Routing calls via corporate networks
  • Tracking and Recording calls and messages

Service Harmonization between various telecom generations

I shall be editing this post to discuss more on the process of Service Harmonization to save the Telecom Service Provider the trouble of rewriting call logic with every telecom generation evolution ie IN to SIP to Web based .

The Service Harmonization Layer does the job of holding all new and legacy services while providing uniform interface to interact with access network regardless of the back-end Call program logic .

Diagrammatic depiction of roles of Service Harmonization

Role of Service Broker and Service Harmonization Layer in Telecom Network

Role of Service Broker and Service Harmonization Layer in Telecom Network


Simple words :

Nodejs lets you write web apps that use Javascript on both the server and the client, so you don’t need to know multiple programming languages to program your website. It’s also really good at handling real-time concurrent web applications, which makes it a great choice for a lot of modern web apps.

Technically :

Node.js is different from JavaScript development in a browser . Technically speaking it makes use of Google’s V8 VM, the same runtime environment for JavaScript that Google Chrome uses.

  • cross-platform runtime environment and a library for running applications written in JavaScript
  • uses non-blocking I/O and asynchronous events.

Nodejs just runs on one  CPU core processor in an asynchronous, single-threaded, event-driven execution model.It contains a built-in asynchronous I/O library for file, socket and HTTP communication.

HTTP and socket support allows Node.js to act as a web server without additional web server software such as Apache.

Node.js vs traditional server-side scripting environments (eg: PHP, Python, Ruby, etc).


The steps to setup the nodejs environment are as follows :

  1. Get a web browser . I am using chrome v35 on ubuntu and windows.
  2. Get the installation of nodejs from this site


It is available in form of windows installer , macintosh installer , linux binaries and from source code . Lets us just use linux binaries .

  1. Note the location of nodejs installation there should be an executable file there name nodejs.nodejs
  2. To start nodejs , just goto terminal in this location and type “ node “.

To load a script type “ node <name of script>.js

…………………………………..CLI ( command Line Interface……………………..

nodejs (1)


Another simple example for function call for console output .Here we are trying to call a function from another function  . First example is to call print function through now function . The second example is the definition of print function inline inside parameter list of now function .


function print(status) {
function now(func2name, value) {
now(say, “Running”);


altanai@tcs:~/nodejsscripts$ node consoletest.js


This code passes the function print as the first parameter to the now function. The print function is called inside now function .

Another way to achieve the above logic through function-inplace


function now(func2name, value) {
now(function(status){ console.log(status) }, “Running”);


altanai@tcs:~/nodejsscripts$ node consoletest2.js


……………………………………… Different script Modules/Files ………………..

Make a js file server.js

var http = require(“http”);
function start() {
function onRequest(request, response) {
 console.log(“Request received for Http on server.js.”);
 response.writeHead(200, {“Content-Type”: “text/plain”});
 response.write(“Running onRequest logic from server.js”);
console.log(“Inside server.js”);
exports.start = start;


Make another js file which is the main file to be loaded onto nodejs. Main.js

var server = require(“./server”);
console.log(“Inside main.js”);

start this file from node

console output

nodejs (2)

web output

nodejs (3)

………………………………………… HTTP Server …………………………………

Make a Javascript file for creating a HTTP server and displaying some text on webpage as well as console . Lets us name it helloworld.js. The code in that file is

var http = require(‘http’);
http.createServer(function (request, response) {
response.writeHead(200, {‘Content-Type': ‘text/plain’});
response.end(‘Display text on webpage – Hello World\n’);  
/*check this address in browser */
console.log(‘Display text on console – Server running ‘);

/* check terminal screen */

Run it on console using command “node helloworld.js”

nodejs (5)

Check output in browser

nodejs (4)

Explanation :

The code for creation of HTTP server is

var http = require(“http”);
var server = http.createServer();

Web Services

  • HTTP and XML is the basis for Web services

Advertisement Engine with WebRTC

  • WSDL stands for Web Services Description Language
  • It specifies the location of the service and the operations (or methods) the service exposes.
  •  XML-based language for describing Web services.

  • SOAP stands for Simple Object Access Protocol
  • SOAP is an XML based protocol for accessing Web Services.
  • SOAP is based on XML

  • UDDI stands for Universal Description, Discovery and Integration
  • UDDI is a directory service where companies can search for Web services.
  • UDDI is described in WSDL
  • UDDI communicates via SOAP

  • RDF stands for Resource Description Framework
  • RDF is a framework for describing resources on the web
  • RDF is written in XML
uses :Web services can offer application-components like: currency conversion, weather reports, or even language translation as services.

sip server officesip

officesip0 officesip1 officesip2 officesip3 officesip4 officesip5 officesip6 officesip7


officesip9 officesip10 officesip11 officesip13