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Implementing Client Side WebRTC using Sipml5 javascript

Step 1:

Download and require Sipml5 library function.

Step 2:

Initialize sipml5 Engine in your web page :

var readyCallback = function(e) {
  // function called when sipml is successfully initialised.
  createSipStack(); // calling this function to create sip stack(see below)

var errorCallback = function(e) {
  // function called when error occured during sipml initialisation.

SIPml.init(readyCallback, errorCallback);

Step 3:

Create Sip Stack :-
Sip Stack is an object that must be created before making/receiving call and sms. Creating Sip stack is an asynchronous process, so you need to create an event listener function to get state change notification.

var sipStack;

function EventListener(e) {

  * e.type ;type of event listener
  * e.session ; current event session
  * e.getSipResponseCode() ; event response code
  * e.description ; event description

  if(e.type == 'started'){
     // successfully started the stack.
  } else if(e.type == 'i_new_call'){
     // when new incoming call comes.
        // incoming call Id ; e.newSession.getRemoteFriendlyName()

     if(callSession || incomingCallSession) {

        e.newSession.hangup(); // hanging up new call when caller is in another outgoing call.

     } else {

        incomingCallSession = e.newSession;
                          audio_remote: document.getElementById('audio-remote'),
                          events_listener: { events: '*', listener: EventListener }
        acceptCall(); // accepts call

  } else if(e.type == 'connecting') {

        if(e.session == registerSession) {
           // registering session.
        } else if(e.session == callSession) {
           // connecting outgoing call.
        } else if(e.session == incomingCallSession) {
           // connecting incoming call.

  } else if(e.type == 'connected') {

        if(e.session == registerSession) {
           // successfully registed.
        } else if(e.session == callSession) {
           // successfully connected call
        } else if(e.session == incomingCallSession) { 
           // incoming call connected

  } else if(e.type == 'terminated') {

     * e.getSipResponseCode()=603 ; call declined without any answer
     * e.getSipResponseCode()=487 ; caller terminated the call
     * e.getSipResponseCode()=-1 ; call answered and hanguped by caller/callee 
     * e.getSipResponseCode()=200 ; user unregistered

        if(e.session == registerSession) {
           // client unregistered
        } else if(e.session == callSession) {
           callSession = null;
           //outgoing call terminated.
        } else if(e.session == incomingCallSession) { 
           incomingCallSession = null;
           // incoming call terminated


function createSipStack() {
  sipStack = new SIPml.Stack({
                    realm: '', // mandatory domain name
                    impi: 'test', // mandatory authorisation name
                    impu: '', // mandatory sip uri
                    password: 'password', //optional
                    display_name: ' test name', // optional
                    websocket_proxy_url: 'wss://', // optional
                    outbound_proxy_url: 'udp://', // optional
                    enable_rtcweb_breaker: true, // optional
                    events_listener: { events: '*', listener: EventListener } /* optional , '*' means all events */

     sipStack.start(); // starting sip stack

Step 4:

Register :-
Register is used to register the sip to server. It is not mandatory to register to receive call/sma.

var registerSession;

function register() { // register function
  registerSession = sipStack.newSession('register', {
                       expires: 300, // expire time, optional
                        events_listener: { events: '*', listener: EventListener } /* optional, '*' means all events. */
   registerSession.register(); // registering session.

Step 5:
Make Outgoing call.

var callSession;

function makeCall() {
  callSession = sipStack.newSession('call-audiovideo', {
  /* audio and video will not be played if you didnt give values to audio_remote,video_remote and for video_local. */
                          audio_remote: document.getElementById('video-remote'),
                          events_listener: { events: '*', listener: EventListener }

Step 6:

Receiving incoming call :-
"i_new_call" event is generated when a new incoming call arrives. This event handler should be added when sip stack is created.

function acceptCall() {
  // accept incoming call.

Step 7:

Hangup or Reject a call

function hangupCall() { // call this function to hangup /reject a call.
  if(callSession) {
     callSession.hangup(); // hangups outgoing call.
  } else if(incomingCallSession) {
     incomingCallSession.reject(); // rejects incoming call.

Step 8:

Unregistering Client.

function unregister() { // call this function to unregister this function


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