Using webrtc2sip with Chrome and placing SIP video call to a Polycom MCU

2,124 views

Skip to first unread message

UK

unread,

Mar 28, 2013, 6:37:04 PM3/28/13

to doub...@googlegroups.com

Whats the best place to start reading on to get webrtc2sip working with Chrome Browser and enabling a SIP URI dial from the browser to dial into an MCU using H.264...

I have webrtc2sip installed on CentOS 64 and it runs:

How do I get this functionality into my Browser ?

Does it have to be dont on the same CentOS machine?

Reading tech guide, looks like there are some Chrome related steps, but H264 could still be touch to implement

Any insight would be apprecaited OR point me to the Docs please

Thx

Mamadou DIOP

unread,

Mar 28, 2013, 6:47:00 PM3/28/13

to doub...@googlegroups.com

- when building webrtc2sip, make sure to enable h264 (requires ffmpeg and x264)

- in config.xml, enable media coder: https://code.google.com/p/webrtc2sip/source/browse/trunk/config.xml?r=64#15

- in config.xml, check that h264-mp (Main Profile) and/or h264-bp (Base Profile) are/is enabled: https://code.google.com/p/webrtc2sip/source/browse/trunk/config.xml?r=64#24

- in expert view, enable RTCWeb Breaker: http://sipml5.org/expert.htm

--
You received this message because you are subscribed to the Google Groups "discuss-doubango" group.
To unsubscribe from this group and stop receiving emails from it, send an email to doubango+u...@googlegroups.com.
For more options, visit https://groups.google.com/groups/opt_out.

Ujjval Koul

unread,

Mar 28, 2013, 11:43:57 PM3/28/13

to doub...@googlegroups.com

config.xml below:

Still unclear on - How do I tie the webrtc2sip; I just compiled to my browser. Any pointers?

http://sipml5.org/expert.htm - do I need to create something like this and host on a web server and how do I activate webrtc2sip from within this?

============================================cat /opt/webrtc2sip/sbin/config.xml
<?xml version="1.0" encoding="utf-8" ?>

<config>

<debug-level>ERROR</debug-level>

<transport>udp;*;10060</transport>
<transport>ws;*;10060</transport>
<transport>wss;*;10062</transport>



<enable-rtp-symetric>yes</enable-rtp-symetric>
<enable-100rel>no</enable-100rel>
<enable-media-coder>yes</enable-media-coder>
<enable-videojb>yes</enable-videojb>
<video-size-pref>vga</video-size-pref>
<rtp-buffsize>65535</rtp-buffsize>
<avpf-tail-length>100;400</avpf-tail-length>
<srtp-mode>optional</srtp-mode>
<srtp-type>sdes;dtls</srtp-type>
<dtmf-type>rfc4733</dtmf-type>

<codecs>pcma;pcmu;gsm;vp8;h264-bp;h264-mp;h263;h263+</codecs>







<transport>c2c;*;10070</transport>
<transport>c2cs;*;10072</transport>
<database>sqlite;*</database>



</config>

============================================

Ujjval Koul

unread,

Mar 29, 2013, 1:30:01 PM3/29/13

to doub...@googlegroups.com

Any help on this appreciated.

I have the following call.htm from sipml5 ; how do I make it use webrtc2sip compiled on my centOS box to place a SIp/Video call out from the browser?

===========================================

<!DOCTYPE html>

<!--

* License: GPLv3

* This file is part of Open Source sipML5 solution <http://www.sipml5.org>

-->

<html>

<head>

<title>sipML5 live demo</title>

<!-- SIPML5 API:

DEBUG VERSION: 'SIPml-api.js'

RELEASE VERSION: 'release/SIPml-api.js'

-->

body{

padding-top: 80px;

padding-bottom: 40px;

}

.navbar-inner-red {

background-color: #600000;

background-image: none;

background-repeat: no-repeat;

filter: none;

}

.full-screen{

position: absolute;

left: 0;

top: 0;

width: 100%;

height: 100%;

overflow: hidden;

}

.normal-screen{

position: relative;

}

.call-options {

padding: 5px;

background-color: #f0f0f0;

border: 1px solid #eee;

border: 1px solid rgba(0, 0, 0, 0.08);

-webkit-border-radius: 4px;

-moz-border-radius: 4px;

border-radius: 4px;

-webkit-box-shadow: inset 0 1px 1px rgba(0, 0, 0, 0.05);

-moz-box-shadow: inset 0 1px 1px rgba(0, 0, 0, 0.05);

box-shadow: inset 0 1px 1px rgba(0, 0, 0, 0.05);

-webkit-transition-property: opacity;

-moz-transition-property: opacity;

-o-transition-property: opacity;

-webkit-transition-duration: 2s;

-moz-transition-duration: 2s;

-o-transition-duration: 2s;

}

.tab-video,

.div-video{

width: 100%;

height: 0px;

-webkit-transition-property: height;

-moz-transition-property: height;

-o-transition-property: height;

-webkit-transition-duration: 2s;

-moz-transition-duration: 2s;

-o-transition-duration: 2s;

}

.label-align {

display: block;

padding-left: 15px;

text-indent: -15px;

}

.input-align {

width: 13px;

height: 13px;

padding: 0;

margin:0;

vertical-align: bottom;

position: relative;

top: -1px;

*overflow: hidden;

}

.glass-panel{

z-index: 99;

position: fixed;

width: 100%;

height: 100%;

margin: 0;

padding: 0;

top: 0;

left: 0;

opacity: 0.8;

background-color: Gray;

}

</style>

</head>

// to avoid caching

//if (window.location.href.indexOf("svn=") == -1) {

// window.location.href += (window.location.href.indexOf("?") == -1 ? "?svn=13" : "&svn=13");

//}

var sTransferNumber;

var oRingTone, oRingbackTone;

var oSipStack, oSipSessionRegister, oSipSessionCall, oSipSessionTransferCall;

var txtDisplayName, txtPrivateIdentity, txtPublicIdentity, txtPassword, txtRealm;

var txtPhoneNumber;

var btnCall, btnHangUp;

var txtRegStatus, txtCallStatus;

var btnRegister, btnUnRegister;

var btnFullScreen, btnHoldResume, btnTransfer, btnKeyPad;

var videoRemote, videoLocal, audioRemote;

var divVideo, divCallOptions;

var tdVideo;

var bFullScreen = false;

var oNotifICall;

var oReadyStateTimer;

var bDisableVideo = false;

var viewVideoLocal, viewVideoRemote; // <video> (webrtc) or <div> (webrtc4all)

window.onload = function () {

txtDisplayName = document.getElementById("txtDisplayName");

txtPrivateIdentity = document.getElementById("txtPrivateIdentity");

txtPublicIdentity = document.getElementById("txtPublicIdentity");

txtPassword = document.getElementById("txtPassword");

txtRealm = document.getElementById("txtRealm");

txtPhoneNumber = document.getElementById("txtPhoneNumber");

btnCall = document.getElementById("btnCall");

btnHangUp = document.getElementById("btnHangUp");

txtRegStatus = document.getElementById("txtRegStatus");

txtCallStatus = document.getElementById("txtCallStatus");

btnRegister = document.getElementById("btnRegister");

btnUnRegister = document.getElementById("btnUnRegister");

btnFullScreen = document.getElementById("btnFullScreen");

btnHoldResume = document.getElementById("btnHoldResume");

btnTransfer = document.getElementById("btnTransfer");

btnKeyPad = document.getElementById("btnKeyPad");

videoLocal = document.getElementById("video_local");

videoRemote = document.getElementById("video_remote");

audioRemote = document.getElementById("audio_remote");

divVideo = document.getElementById("divVideo");

divCallOptions = document.getElementById("divCallOptions");

//divVideo.style.height = '0px';

tdVideo = document.getElementById("tdVideo");

document.onkeyup = onKeyUp;

document.body.onkeyup = onKeyUp;

document.getElementById("divCallCtrl").onmousemove = onDivCallCtrlMouseMove;

loadCredentials();

loadCallOptions();

oReadyStateTimer = setInterval(function () {

if (document.readyState === "complete") {

clearInterval(oReadyStateTimer);

// initialize SIPML5

SIPml.init(postInit);

}

500);

};

function postInit() {

// check webrtc4all version

if (SIPml.isWebRtc4AllSupported() && SIPml.isWebRtc4AllPluginOutdated()) {

if (confirm("Your WebRtc4all extension is outdated. A new version(" +SIPml.getWebRtc4AllVersion()+") with critical bug fix is available. Do you want to install it?\nIMPORTANT: You must restart your browser after the installation.")) {

window.location = 'http://code.google.com/p/webrtc4all/downloads/list';

return;

}

// check for WebRTC support

if (!SIPml.isWebRtcSupported()) {

// is it chrome?

if (SIPml.getNavigatorFriendlyName() == 'chrome') {

if (confirm("You're using an old Chrome version or WebRTC is not enabled.\nDo you want to see how to enable WebRTC?")) {

window.location = 'http://www.webrtc.org/running-the-demos';

}

else {

window.location = "index.html";

}

return;

}

// for now the plugins (WebRTC4all only works on Windows)

if (SIPml.getSystemFriendlyName() == 'windows') {

// Internet explorer

if (SIPml.getNavigatorFriendlyName() == 'ie') {

// Check for IE version

if (parseFloat(SIPml.getNavigatorVersion()) < 9.0) {

if (confirm("You are using an old IE version. You need at least version 9. Would you like to update IE?")) {

window.location = 'http://windows.microsoft.com/en-us/internet-explorer/products/ie/home';

}

else {

window.location = "index.html";

}

// check for WebRTC4all extension

if (!SIPml.isWebRtc4AllSupported()) {

if (confirm("webrtc4all extension is not installed. Do you want to install it?\nIMPORTANT: You must restart your browser after the installation.")) {

window.location = 'http://code.google.com/p/webrtc4all/downloads/list';

}

else {

// Must do nothing: give the user the chance to accept the extension

// window.location = "index.html";

}

// break page loading ('window.location' won't stop JS execution)

if (!SIPml.isWebRtc4AllSupported()) {

return;

}

else if (SIPml.getNavigatorFriendlyName() == "safari" || SIPml.getNavigatorFriendlyName() == "firefox" || SIPml.getNavigatorFriendlyName() == "opera") {

if (confirm("Your browser don't support WebRTC.\nDo you want to install WebRTC4all extension to enjoy audio/video calls?\nIMPORTANT: You must restart your browser after the installation.")) {

window.location = 'http://code.google.com/p/webrtc4all/downloads/list';

}

else {

window.location = "index.html";

}

return;

}

// OSX, Unix, Android, iOS...

else {

if (confirm('WebRTC not supported on your browser.\nDo you want to download a WebRTC-capable browser?')) {

window.location = 'https://www.google.com/intl/en/chrome/browser/';

}

else {

window.location = "index.html";

}

return;

}

// checks for WebSocket support

if (!SIPml.isWebSocketSupported() && !SIPml.isWebRtc4AllSupported()) {

if (confirm('Your browser don\'t support WebSockets.\nDo you want to download a WebSocket-capable browser?')) {

window.location = 'https://www.google.com/intl/en/chrome/browser/';

}

else {

window.location = "index.html";

}

return;

}

// FIXME: displays must be per session

// attachs video displays

if (SIPml.isWebRtc4AllSupported()) {

viewVideoLocal = document.getElementById("divVideoLocal");

viewVideoRemote = document.getElementById("divVideoRemote");

WebRtc4all_SetDisplays(viewVideoLocal, viewVideoRemote); // FIXME: move to SIPml.* API

}

else{

viewVideoLocal = videoLocal;

viewVideoRemote = videoRemote;

}

if (!SIPml.isWebRtc4AllSupported() && !SIPml.isWebRtcSupported()) {

if (confirm('Your browser don\'t support WebRTC.\naudio/video calls will be disabled.\nDo you want to download a WebRTC-capable browser?')) {

window.location = 'https://www.google.com/intl/en/chrome/browser/';

}

btnRegister.disabled = false;

document.body.style.cursor = 'default';

}

function loadCallOptions() {

if (window.localStorage) {

var s_value;

if ((s_value = window.localStorage.getItem('org.doubango.call.phone_number'))) txtPhoneNumber.value = s_value;

bDisableVideo = (window.localStorage.getItem('org.doubango.expert.disable_video') == "true");

txtCallStatus.innerHTML = '<i>Video ' + (bDisableVideo ? 'disabled' : 'enabled') + '</i>';

}

function saveCallOptions() {

if (window.localStorage) {

window.localStorage.setItem('org.doubango.call.phone_number', txtPhoneNumber.value);

window.localStorage.setItem('org.doubango.expert.disable_video', bDisableVideo ? "true" : "false");

}

function loadCredentials() {

if (window.localStorage) {

// IE retuns 'null' if not defined

var s_value;

if ((s_value = window.localStorage.getItem('org.doubango.identity.display_name'))) txtDisplayName.value = s_value;

if ((s_value = window.localStorage.getItem('org.doubango.identity.impi'))) txtPrivateIdentity.value = s_value;

if ((s_value = window.localStorage.getItem('org.doubango.identity.impu'))) txtPublicIdentity.value = s_value;

if ((s_value = window.localStorage.getItem('org.doubango.identity.password'))) txtPassword.value = s_value;

if ((s_value = window.localStorage.getItem('org.doubango.identity.realm'))) txtRealm.value = s_value;

}

else {

txtDisplayName.value = "005";

txtPrivateIdentity.value = "005";

txtPublicIdentity.value = "sip:0...@192.168.0.42";

txtPassword.value = "005";

txtRealm.value = "192.168.0.42";

txtPhoneNumber.value = "005";

}

};

function saveCredentials() {

if (window.localStorage) {

window.localStorage.setItem('org.doubango.identity.display_name', txtDisplayName.value);

window.localStorage.setItem('org.doubango.identity.impi', txtPrivateIdentity.value);

window.localStorage.setItem('org.doubango.identity.impu', txtPublicIdentity.value);

window.localStorage.setItem('org.doubango.identity.password', txtPassword.value);

window.localStorage.setItem('org.doubango.identity.realm', txtRealm.value);

}

};

// sends SIP REGISTER request to login

function sipRegister() {

// catch exception for IE (DOM not ready)

try {

btnRegister.disabled = true;

if (!txtRealm.value || !txtPrivateIdentity.value || !txtPublicIdentity.value) {

txtRegStatus.innerHTML = '<b>Please fill madatory fields (*)</b>';

btnRegister.disabled = false;

return;

}

var o_impu = tsip_uri.prototype.Parse(txtPublicIdentity.value);

if (!o_impu || !o_impu.s_user_name || !o_impu.s_host) {

txtRegStatus.innerHTML = "<b>[" + txtPublicIdentity.value + "] is not a valid Public identity</b>";

btnRegister.disabled = false;

return;

}

// enable notifications if not already done

if (window.webkitNotifications && window.webkitNotifications.checkPermission() != 0) {

window.webkitNotifications.requestPermission();

}

// save credentials

saveCredentials();

// create SIP stack

oSipStack = new SIPml.Stack({

realm: txtRealm.value,

impi: txtPrivateIdentity.value,

impu: txtPublicIdentity.value,

password: txtPassword.value,

display_name: txtDisplayName.value,

websocket_proxy_url: (window.localStorage ? window.localStorage.getItem('org.doubango.expert.websocket_server_url') : null),

outbound_proxy_url: (window.localStorage ? window.localStorage.getItem('org.doubango.expert.sip_outboundproxy_url') : null),

enable_rtcweb_breaker: (window.localStorage ? window.localStorage.getItem('org.doubango.expert.enable_rtcweb_breaker') == "true" : false),

events_listener: { events: '*', listener: onSipEventStack },

sip_headers: [

{ name: 'User-Agent', value: 'IM-client/OMA1.0 sipML5-v1.2013.03.13' },

{ name: 'Organization', value: 'Doubango Telecom' }

]

}

);

if (oSipStack.start() != 0) {

txtRegStatus.innerHTML = '<b>Failed to start the SIP stack</b>';

}

else return;

}

catch (e) {

txtRegStatus.innerHTML = "<b>2:" + e + "</b>";

}

btnRegister.disabled = false;

}

// sends SIP REGISTER (expires=0) to logout

function sipUnRegister() {

if (oSipStack) {

oSipStack.stop(); // shutdown all sessions

}

// makes a call (SIP INVITE)

function sipCall() {

// call configuration

var oConf = {

audio_remote: audioRemote,

video_local: viewVideoLocal,

video_remote: viewVideoRemote,

events_listener: { events: '*', listener: onSipEventSession },

sip_caps: [

{ name: '+g.oma.sip-im' },

{ name: '+sip.ice' },

{ name: 'language', value: '\"en,fr\"' }

]

};

if (oSipStack && !oSipSessionCall && !tsk_string_is_null_or_empty(txtPhoneNumber.value)) {

btnCall.disabled = false;

btnHangUp.disabled = false;

// check whether video is disabled or not

bDisableVideo = (window.localStorage && window.localStorage.getItem('org.doubango.expert.disable_video') == "true");

// create call session

oSipSessionCall = oSipStack.newSession(bDisableVideo ? 'call-audio' : 'call-audiovideo', oConf);

// make call

if (oSipSessionCall.call(txtPhoneNumber.value) != 0) {

oSipSessionCall = null;

txtCallStatus.value = 'Failed to make call';

btnCall.disabled = false;

btnHangUp.disabled = true;

return;

}

saveCallOptions();

}

else if (oSipSessionCall) {

txtCallStatus.innerHTML = '<i>Connecting...</i>';

oSipSessionCall.accept(oConf);

}

// transfers the call

function sipTransfer() {

if (oSipSessionCall) {

var s_destination = prompt('Enter destination number', '');

if (!tsk_string_is_null_or_empty(s_destination)) {

btnTransfer.disabled = true;

if (oSipSessionCall.transfer(s_destination) != 0) {

txtCallStatus.innerHTML = '<i>Call transfer failed</i>';

btnTransfer.disabled = false;

return;

}

txtCallStatus.innerHTML = '<i>Transfering the call...</i>';

}

// holds or resumes the call

function sipToggleHoldResume() {

if (oSipSessionCall) {

var i_ret;

btnHoldResume.disabled = true;

txtCallStatus.innerHTML = oSipSessionCall.bHeld ? '<i>Resuming the call...</i>' : '<i>Holding the call...</i>';

i_ret = oSipSessionCall.bHeld ? oSipSessionCall.resume() : oSipSessionCall.hold();

if (i_ret != 0) {

txtCallStatus.innerHTML = '<i>Hold / Resume failed</i>';

btnHoldResume.disabled = false;

return;

}

// terminates the call (SIP BYE or CANCEL)

function sipHangUp() {

if (oSipSessionCall) {

txtCallStatus.innerHTML = '<i>Terminating the call...</i>';

oSipSessionCall.hangup({events_listener: { events: '*', listener: onSipEventSession }});

}

function startRingTone() {

try { ringtone.play(); }

catch (e) { }

}

function stopRingTone() {

try { ringtone.pause(); }

catch (e) { }

}

function startRingbackTone() {

try { ringbacktone.play(); }

catch (e) { }

}

function stopRingbackTone() {

try { ringbacktone.pause(); }

catch (e) { }

}

function toggleFullScreen() {

if (videoRemote.webkitSupportsFullscreen) {

fullScreen(!videoRemote.webkitDisplayingFullscreen);

}

else {

fullScreen(!bFullScreen);

}

function fullScreen(b_fs) {

bFullScreen = b_fs;

if (tsk_utils_have_webrtc4native() && bFullScreen && videoRemote.webkitSupportsFullscreen) {

if (bFullScreen) {

videoRemote.webkitEnterFullScreen();

}

else {

videoRemote.webkitExitFullscreen();

}

else {

if (tsk_utils_have_webrtc4npapi()) {

try { if(window.__o_display_remote) window.__o_display_remote.setFullScreen(b_fs); }

catch (e) { divVideo.setAttribute("class", b_fs ? "full-screen" : "normal-screen"); }

}

else {

divVideo.setAttribute("class", b_fs ? "full-screen" : "normal-screen");

}

function showNotifICall(s_number) {

// permission already asked when we registered

if (window.webkitNotifications && window.webkitNotifications.checkPermission() == 0) {

if (oNotifICall) {

oNotifICall.cancel();

}

oNotifICall = window.webkitNotifications.createNotification('images/sipml-34x39.png', 'Incaming call', 'Incoming call from ' + s_number);

oNotifICall.onclose = function () { oNotifICall = null; };

oNotifICall.show();

}

function onKeyUp(evt) {

evt = (evt || window.event);

if (evt.keyCode == 27) {

fullScreen(false);

}

else if (evt.ctrlKey && evt.shiftKey) { // CTRL + SHIFT

if (evt.keyCode == 65 || evt.keyCode == 86) { // A (65) or V (86)

bDisableVideo = (evt.keyCode == 65);

txtCallStatus.innerHTML = '<i>Video ' + (bDisableVideo ? 'disabled' : 'enabled') + '</i>';

window.localStorage.setItem('org.doubango.expert.disable_video', bDisableVideo);

}

function onDivCallCtrlMouseMove(evt) {

try { // IE: DOM not ready

if (tsk_utils_have_stream()) {

btnCall.disabled = (!tsk_utils_have_stream() || !oSipSessionRegister || !oSipSessionRegister.is_connected());

document.getElementById("divCallCtrl").onmousemove = null; // unsubscribe

}

catch (e) { }

}

function uiOnConnectionEvent(b_connected, b_connecting) { // should be enum: connecting, connected, terminating, terminated

btnRegister.disabled = b_connected || b_connecting;

btnUnRegister.disabled = !b_connected && !b_connecting;

btnCall.disabled = !(b_connected && tsk_utils_have_webrtc() && tsk_utils_have_stream());

btnHangUp.disabled = !oSipSessionCall;

}

function uiVideoDisplayEvent(b_local, b_added) {

//if (!bDisableVideo) {

var o_elt_video = b_local ? videoLocal : videoRemote;

if (b_added) {

if (SIPml.isWebRtc4AllSupported()) {

if (b_local){ if(window.__o_display_local) window.__o_display_local.style.visibility = "visible"; }

else { if(window.__o_display_remote) window.__o_display_remote.style.visibility = "visible"; }

}

else {

o_elt_video.style.opacity = 1;

}

uiVideoDisplayShowHide(true);

}

else {

if (SIPml.isWebRtc4AllSupported()) {

if (b_local){ if(window.__o_display_local) window.__o_display_local.style.visibility = "hidden"; }

else { if(window.__o_display_remote) window.__o_display_remote.style.visibility = "hidden"; }

}

else{

o_elt_video.style.opacity = 0;

}

fullScreen(false);

}

//}

}

function uiVideoDisplayShowHide(b_show) {

if (b_show) {

tdVideo.style.height = '340px';

divVideo.style.height = navigator.appName == 'Microsoft Internet Explorer' ? '100%' : '340px';

}

else {

tdVideo.style.height = '0px';

divVideo.style.height = '0px';

}

btnFullScreen.disabled = !b_show;

}

function uiCallTerminated(s_description){

btnCall.value = 'Call';

btnHangUp.value = 'HangUp';

btnHoldResume.value = 'hold';

btnCall.disabled = false;

btnHangUp.disabled = true;

oSipSessionCall = null;

stopRingbackTone();

stopRingTone();

txtCallStatus.innerHTML = "<i>" + s_description + "</i>";

uiVideoDisplayShowHide(false);

divCallOptions.style.opacity = 0;

if (oNotifICall) {

oNotifICall.cancel();

oNotifICall = null;

}

uiVideoDisplayEvent(true, false);

uiVideoDisplayEvent(false, false);

setTimeout(function () { if (!oSipSessionCall) txtCallStatus.innerHTML = ''; }, 2500);

}

// Callback function for SIP Stacks

function onSipEventStack(e /*SIPml.Stack.Event*/) {

tsk_utils_log_info('==stack event = ' + e.type);

switch (e.type) {

case 'started':

{

// catch exception for IE (DOM not ready)

try {

// LogIn (REGISTER) as soon as the stack finish starting

oSipSessionRegister = this.newSession('register', {

expires: 200,

events_listener: { events: '*', listener: onSipEventSession },

sip_caps: [

{ name: '+g.oma.sip-im', value: null },

{ name: '+audio', value: null },

{ name: 'language', value: '\"en,fr\"' }

]

});

oSipSessionRegister.register();

}

catch (e) {

txtRegStatus.value = txtRegStatus.innerHTML = "<b>1:" + e + "</b>";

btnRegister.disabled = false;

}

break;

}

case 'stopping': case 'stopped': case 'failed_to_start': case 'failed_to_stop':

{

var bFailure = (e.type == 'failed_to_start') || (e.type == 'failed_to_stop');

oSipStack = null;

oSipSessionRegister = null;

oSipSessionCall = null;

uiOnConnectionEvent(false, false);

stopRingbackTone();

stopRingTone();

uiVideoDisplayShowHide(false);

divCallOptions.style.opacity = 0;

txtCallStatus.innerHTML = '';

txtRegStatus.innerHTML = bFailure ? "<i>Disconnected: <b>" + e.description + "</b></i>" : "<i>Disconnected</i>";

break;

}

case 'i_new_call':

{

if (oSipSessionCall) {

// do not accept the incoming call if we're already 'in call'

e.newSession.hangup(); // comment this line for multi-line support

}

else {

oSipSessionCall = e.newSession;

btnCall.value = 'Answer';

btnHangUp.value = 'Reject';

btnCall.disabled = false;

btnHangUp.disabled = false;

startRingTone();

var sRemoteNumber = (oSipSessionCall.getRemoteFriendlyName() || 'unknown');

txtCallStatus.innerHTML = "<i>Incoming call from [<b>" + sRemoteNumber + "</b>]</i>";

showNotifICall(sRemoteNumber);

}

break;

}

case 'm_permission_requested':

{

divGlassPanel.style.visibility = 'visible';

break;

}

case 'm_permission_accepted':

case 'm_permission_refused':

{

divGlassPanel.style.visibility = 'hidden';

if(e.type == 'm_permission_refused'){

uiCallTerminated('Media stream permission denied');

}

break;

}

case 'starting': default: break;

}

};

// Callback function for SIP sessions (INVITE, REGISTER, MESSAGE...)

function onSipEventSession(e /* SIPml.Session.Event */) {

tsk_utils_log_info('==session event = ' + e.type);

switch (e.type) {

case 'connecting': case 'connected':

{

var bConnected = (e.type == 'connected');

if (e.session == oSipSessionRegister) {

uiOnConnectionEvent(bConnected, !bConnected);

txtRegStatus.innerHTML = "<i>" + e.description + "</i>";

}

else if (e.session == oSipSessionCall) {

btnHangUp.value = 'HangUp';

btnCall.disabled = true;

btnHangUp.disabled = false;

btnTransfer.disabled = false;

if (bConnected) {

stopRingbackTone();

stopRingTone();

if (oNotifICall) {

oNotifICall.cancel();

oNotifICall = null;

}

txtCallStatus.innerHTML = "<i>" + e.description + "</i>";

divCallOptions.style.opacity = bConnected ? 1 : 0;

if (SIPml.isWebRtc4AllSupported()) { // IE don't provide stream callback

uiVideoDisplayEvent(true, true);

uiVideoDisplayEvent(false, true);

}

break;

} // 'connecting' | 'connected'

case 'terminating': case 'terminated':

{

if (e.session == oSipSessionRegister) {

uiOnConnectionEvent(false, false);

oSipSessionCall = null;

oSipSessionRegister = null;

txtRegStatus.innerHTML = "<i>" + e.description + "</i>";

}

else if (e.session == oSipSessionCall) {

uiCallTerminated(e.description);

}

break;

} // 'terminating' | 'terminated'

case 'm_stream_video_local_added':

{

if (e.session == oSipSessionCall) {

uiVideoDisplayEvent(true, true);

}

break;

}

case 'm_stream_video_local_removed':

{

if (e.session == oSipSessionCall) {

uiVideoDisplayEvent(true, false);

}

break;

}

case 'm_stream_video_remote_added':

{

if (e.session == oSipSessionCall) {

uiVideoDisplayEvent(false, true);

}

break;

}

case 'm_stream_video_remote_removed':

{

if (e.session == oSipSessionCall) {

uiVideoDisplayEvent(false, false);

}

break;

}

case 'm_stream_audio_local_added':

case 'm_stream_audio_local_removed':

case 'm_stream_audio_remote_added':

case 'm_stream_audio_remote_removed':

{

break;

}

case 'i_ect_new_call':

{

oSipSessionTransferCall = e.session;

break;

}

case 'i_ao_request':

{

if(e.session == oSipSessionCall){

var iSipResponseCode = e.getSipResponseCode();

if (iSipResponseCode == 180 || iSipResponseCode == 183) {

startRingbackTone();

txtCallStatus.innerHTML = '<i>Remote ringing...</i>';

}

break;

}

case 'm_early_media':

{

if(e.session == oSipSessionCall){

stopRingbackTone();

stopRingTone();

txtCallStatus.innerHTML = '<i>Early media started</i>';

}

break;

}

case 'm_local_hold_ok':

{

if(e.session == oSipSessionCall){

if (oSipSessionCall.bTransfering) {

oSipSessionCall.bTransfering = false;

// this.AVSession.TransferCall(this.transferUri);

}

btnHoldResume.value = 'Resume';

btnHoldResume.disabled = false;

txtCallStatus.innerHTML = '<i>Call placed on hold</i>';

oSipSessionCall.bHeld = true;

}

break;

}

case 'm_local_hold_nok':

{

if(e.session == oSipSessionCall){

oSipSessionCall.bTransfering = false;

btnHoldResume.value = 'Hold';

btnHoldResume.disabled = false;

txtCallStatus.innerHTML = '<i>Failed to place remote party on hold</i>';

}

break;

}

case 'm_local_resume_ok':

{

if(e.session == oSipSessionCall){

oSipSessionCall.bTransfering = false;

btnHoldResume.value = 'Hold';

btnHoldResume.disabled = false;

txtCallStatus.innerHTML = '<i>Call taken off hold</i>';

oSipSessionCall.bHeld = false;

if (SIPml.isWebRtc4AllSupported()) { // IE don't provide stream callback yet

uiVideoDisplayEvent(true, true);

uiVideoDisplayEvent(false, true);

}

break;

}

case 'm_local_resume_nok':

{

if(e.session == oSipSessionCall){

oSipSessionCall.bTransfering = false;

btnHoldResume.disabled = false;

txtCallStatus.innerHTML = '<i>Failed to unhold call</i>';

}

break;

}

case 'm_remote_hold':

{

if(e.session == oSipSessionCall){

txtCallStatus.innerHTML = '<i>Placed on hold by remote party</i>';

}

break;

}

case 'm_remote_resume':

{

if(e.session == oSipSessionCall){

txtCallStatus.innerHTML = '<i>Taken off hold by remote party</i>';

}

break;

}

case 'o_ect_trying':

{

if(e.session == oSipSessionCall){

txtCallStatus.innerHTML = '<i>Call transfer in progress...</i>';

}

break;

}

case 'o_ect_accepted':

{

if(e.session == oSipSessionCall){

txtCallStatus.innerHTML = '<i>Call transfer accepted</i>';

}

break;

}

case 'o_ect_completed':

case 'i_ect_completed':

{

if(e.session == oSipSessionCall){

txtCallStatus.innerHTML = '<i>Call transfer completed</i>';

btnTransfer.disabled = false;

if (oSipSessionTransferCall) {

oSipSessionCall = oSipSessionTransferCall;

}

oSipSessionTransferCall = null;

}

break;

}

case 'o_ect_failed':

case 'i_ect_failed':

{

if(e.session == oSipSessionCall){

txtCallStatus.innerHTML = '<i>Call transfer failed</i>';

btnTransfer.disabled = false;

}

break;

}

case 'o_ect_notify':

case 'i_ect_notify':

{

if(e.session == oSipSessionCall){

txtCallStatus.innerHTML = "<i>Call Transfer: <b>" + e.getSipResponseCode() + " " + e.description + "</b></i>";

if (e.getSipResponseCode() >= 300) {

if (oSipSessionCall.bHeld) {

oSipSessionCall.resume();

}

btnTransfer.disabled = false;

}

break;

}

case 'i_ect_requested':

{

if(e.session == oSipSessionCall){

var s_message = "Do you accept call transfer to [" + e.getTransferDestinationFriendlyName() + "]?";//FIXME

if (confirm(s_message)) {

txtCallStatus.innerHTML = "<i>Call transfer in progress...</i>";

oSipSessionCall.acceptTransfer();

break;

}

oSipSessionCall.rejectTransfer();

}

break;

}

</script>

<a class="btn btn-navbar" data-toggle="collapse" data-target=".nav-collapse"><span

class="icon-bar"></span><span class="icon-bar"></span><span class="icon-bar"></span>

</a>

</ul>

</div>

</div>

</label>

<h2>

Registration</h2>

<br />

<tr>

<td>

Display Name:</label>

</td>

<td>

<input type="text" style="width: 100%; height: 100%" id="txtDisplayName" value=""

placeholder="e.g. John Doe" />

</td>

</tr>

<tr>

<td>

Private Identity<sup>*</sup>:</label>

</td>

<td>

<input type="text" style="width: 100%; height: 100%" id="txtPrivateIdentity" value=""

placeholder="e.g. +33600000000" />

</td>

</tr>

<tr>

<td>

Public Identity<sup>*</sup>:</label>

</td>

<td>

<input type="text" style="width: 100%; height: 100%" id="txtPublicIdentity" value=""

placeholder="e.g. sip:+33600...@doubango.org" />

</td>

</tr>

<tr>

<td>

<label style="height: 100%">Password:</label>

</td>

<td>

</td>

</tr>

<tr>

<td>

<label style="height: 100%">Realm<sup>*</sup>:</label>

</td>

<td>

</td>

</tr>

<tr>

</td>

</tr>

<tr>

<p class="small"><sup>*</sup> <i>Mandatory Field</i></p>

</td>

</tr>

<tr>

<a class="btn" href="http://code.google.com/p/sipml5/wiki/Public_SIP_Servers" target="_blank">Need SIP account?</a>

</td>

</tr>

<tr>

<a class="btn" href="./expert.htm" target="_blank">Expert mode?</a>

</td>

</tr>

</table>

</div>

</label>

<h2>

Call control

</h2>

<br />

<tr>

</td>

</tr>

<tr>

</td>

</tr>

<tr>

<video class="video" width="100%" height="100%" id="video_remote" autoplay="autoplay" style="opacity: 0;

background-color: #000000; -webkit-transition-property: opacity; -webkit-transition-duration: 2s;">

</video>

</div>

<video class="video" width="88px" height="72px" id="video_local" autoplay="autoplay" style="opacity: 0;

margin-top: -80px; margin-left: 5px; background-color: #000000; -webkit-transition-property: opacity;

-webkit-transition-duration: 2s;">

</video>

</div>

</td>

</tr>

<tr>

</div>

</td>

</tr>

</table>

</div>

<br />

<i>Inspiring the future</i>

</p>

<!-- Creates all ATL/COM objects right now

Will open confirmation dialogs if not already done

-->

<!--

NPAPI browsers: Safari, Opera and Firefox

-->

</footer>

</div>

<!-- Le javascript

================================================== -->

<!--

Microsoft Internet Explorer extension

For now we use msi installer as we don't have trusted certificate yet :(

-->

var gaJsHost = (("https:" == document.location.protocol) ? "https://ssl." : "http://www.");

document.write(unescape("%3Cscript src='" + gaJsHost + "google-analytics.com/ga.js' type='text/javascript'%3E%3C/script%3E"));

</script>

try {

var pageTracker = _gat._getTracker("UA-6868621-19");

pageTracker._trackPageview();

} catch (err) { }

</script>

</body>

</html>

========================================================

Mamadou DIOP

unread,

Mar 29, 2013, 1:43:41 PM3/29/13

to doub...@googlegroups.com

My first response was really very clear on how to get started.

I'd add:

- from the expert view, set the web socket url (your webrtc2sip server address on port 10060 or 10062) and SIP outbound proxy url (your Polycom MCU or SIP server address)

Ujjval Koul

unread,

Mar 29, 2013, 2:03:37 PM3/29/13

to doub...@googlegroups.com

I have the expoert.htm with info populated and saved.

When I open the call.htm now (one I pasted before), and enter a Number; the call button is not enabled.

Mamadou DIOP

unread,

Mar 29, 2013, 2:08:29 PM3/29/13

to doub...@googlegroups.com

You don't need to use your own call.htm, just open http://sipml5.or/call.htm.

The call button is only enabled if you're registered and webrtc is supported on your browser. If both conditions are ok and the button is still not enabled then, please share with us the javascript console logs.

Ujjval Koul

unread,

Mar 29, 2013, 2:14:36 PM3/29/13

to doub...@googlegroups.com

OK, My webrtc2sip is on a NAT'ed machine so maybe an issue trying from sipml5.org/call.htm

if I use my own - I know webrtc is supported as I am using latest stable Chrome. Whats the process to register?

Mamadou DIOP

unread,

Mar 29, 2013, 2:21:03 PM3/29/13

to doub...@googlegroups.com

I guess you're very very newbie.

- webrtc supports doesn't depend on the server used but on the browser (chrome, firefox...)

- no relation between NAT and using call.htm on local or not. This is just an HTML file running on your browser.

Chrome. Whats the process to register?

Fill the fields and press Login.

Ujjval Koul

unread,

Mar 29, 2013, 2:32:05 PM3/29/13

to doub...@googlegroups.com

Totally newbie...looking to intergrate webrtc into our Video Conf Solution

however, how will call.htm from sipml5.org be able to connect to my webrtc2sip aplication running on my laptop (192.168.0.129) in a private network w/o any firewall NAT rules put in?

As far as registration is concerned, I will try sip2sip info

Thx

Mamadou DIOP

unread,

Mar 29, 2013, 2:37:32 PM3/29/13

to doub...@googlegroups.com

No need to create an account on sip2sip.info.

The webpage is running on your PC even if it's hosted on sipml5.org. The web page will use websocket protocol to connect to your webrtc2sip server. Will be up to webrtc2sip to connect to your SIP server using well-know protocol (UDP, TCP or TLS).

I highly recommend reading the technical guide: http://webrtc2sip.org/technical-guide-1.0.pdf

Ujjval Koul

unread,

Mar 29, 2013, 3:32:14 PM3/29/13

to doub...@googlegroups.com

Your 2 statements confused me , please clarify

The call button is only enabled if you're registered and webrtc is supported on your browser. If both conditions are ok and the button is still not enabled then, please share with us the javascript console logs

Then you said

No need to create an account on sip2sip.info.

So how do I enable the call button on sipml5.org to dial out?

Ujjval Koul

unread,

Mar 30, 2013, 1:06:36 AM3/30/13

to doub...@googlegroups.com

Made some progress...here is an update

webrtc2sip not responding to WS socket request from Chrome Browser (on 192.168.0.131 - which):

Connecting to 'ws://192.168.0.135:10062' SIPml-api.js:1

==stack event = starting SIPml-api.js:1

__tsip_transport_ws_onclose SIPml-api.js:1

==stack event = failed_to_start

Result on the webrtc2sip server 192.168.0.135:

21:51:05.833244 IP 192.168.19.1.59720 > 192.168.19.129.10062: Flags [S], seq 899155698, win 65535, options [mss 1460,nop,wscale 1,nop,nop,sackOK], length 0
21:51:08.832475 IP 192.168.19.1.59720 > 192.168.19.129.10062: Flags [S], seq 899155698, win 65535, options [mss 1460,nop,wscale 1,nop,nop,sackOK], length 0
21:51:14.832042 IP 192.168.19.1.59720 > 192.168.19.129.10062: Flags [S], seq 899155698, win 65535, options [mss 1460,nop,nop,sackOK], length 0

ping 192.168.0.131
PING 192.168.0.131 (192.168.0.131) 56(84) bytes of data.
64 bytes from 192.168.0.131: icmp_seq=1 ttl=128 time=0.940 ms
64 bytes from 192.168.0.131: icmp_seq=2 ttl=128 time=0.748 ms
64 bytes from 192.168.0.131: icmp_seq=3 ttl=128 time=0.747 ms
^C
--- 192.168.0.131 ping statistics ---
3 packets transmitted, 3 received, 0% packet loss, time 2944ms
rtt min/avg/max/mdev = 0.747/0.811/0.940/0.096 ms

Ujjval Koul

unread,

Mar 30, 2013, 1:14:15 AM3/30/13

to doub...@googlegroups.com

tcpdump output is actually below:

22:00:27.006292 IP 192.168.0.131.59765 > 192.168.0.135.10062: Flags [S], seq 1152513202, win 65535, options [mss 1460,nop,wscale 2,nop,nop,sackOK], length 0
22:00:29.999758 IP 192.168.0.131.59765 > 192.168.0.135.10062: Flags [S], seq 1152513202, win 65535, options [mss 1460,nop,wscale 2,nop,nop,sackOK], length 0
22:00:35.998177 IP 192.168.0.131.59765 > 192.168.0.135.10062: Flags [S], seq 1152513202, win 65535, options [mss 1460,nop,nop,sackOK], length 0

Mamadou

unread,

Mar 30, 2013, 1:22:03 AM3/30/13

to doub...@googlegroups.com, Ujjval Koul

10062 is not a port for WebSocket but it's for Secure WebSocket (wss://).
You have to use ws://192.168.0.135:10060 unless you have valid SSL certificates.
In short:
valid SSL certificates -> wss://192.168.0.135:10062
otherwise -> ws://192.168.0.135:10060

See your config.xml

<transport>ws;*;10060</transport>
<transport>wss;*;10062</transport>

--
Mamadou DIOP - Technology Evangelist
Doubango Telecom - Paris, France
http://www.doubango.org
Click here to call me!

Ujjval Koul

unread,

Mar 30, 2013, 1:28:42 AM3/30/13

to Mamadou, doub...@googlegroups.com

corrected that...I see the WS connect now..and its sending a REGISTER msg to the my server (Broadsoft)

Will update you with latest tomorow.

This would be big in the Broadsoft community if we are able to get it to work with Broadsoft based softswitches

Ujjval Koul

unread,

Mar 30, 2013, 1:44:45 AM3/30/13

to Mamadou, doub...@googlegroups.com

Registered and placing a call however, my broadsoft switch challenges every INVITE for same credentials...

Can that be achieved?

webrtc2sip console logs:

***ERROR: function: "tsip_dialog_update_challenges()"

file: "src/dialogs/tsip_dialog.c"

line: "904"

MSG: Failed to handle new challenge

On Fri, Mar 29, 2013 at 11:22 PM, Mamadou <diopm...@doubango.org> wrote:

UK

unread,

Mar 31, 2013, 12:26:57 AM3/31/13

to doub...@googlegroups.com

I see the INVITE sent to Broadsoft switch and a challenge 401 back from Broadsoft.

www.Auth header with username (thats all I can see obviously as pw is not in the clear) is sent from the webrtc2sip, but for some reason Broadsoft thinks the password is not same as that used for registering...I saw a similar issue posted before and was wondering if that had a resolution