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SIPml.js
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SIPml.js
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/*
* Copyright (C) 2012-2015 Doubango Telecom <http://www.doubango.org>
* License: BSD
* This file is part of Open Source sipML5 solution <http://www.sipml5.org>
*/
/**
@fileoverview This is SIPML5 "library" contains a lot of classes and functions.
@name sipML5 API
@author Doubango Telecom <http://www.doubango.org>
@version 1.5.222
*/
/**
@namespace
@description Root namesapce.
*/
SIPml = {};
/** @private */SIPml.b_initialized = false;
/** @private */SIPml.b_initializing = false;
/** @private */SIPml.s_navigator_friendly_name = 'unknown';
/** @private */SIPml.b_navigator_outdated = false;
/** @private */SIPml.s_navigator_version = 'unknown';
/** @private */SIPml.s_system_friendly_name = 'unknown';
/** @private */SIPml.b_webrtc4all_plugin_outdated = false;
/** @private */SIPml.b_webrtc4all_supported = false;
/** @private */SIPml.s_webrtc4all_version = 'unknown';
/** @private */SIPml.b_have_media_stream = false;
/** @private */SIPml.b_webrtc_supported = false;
/**
Sets the debug level.
@since version 1.3.203
@param {String} level The level. Supported values: <i>info</i>, <i>warn</i>, <i>error</i> and <i>fatal</i>.
*/
SIPml.setDebugLevel = function(level) {
tsk_utils_log_set_level(level === 'fatal' ? 1 : (level === 'error' ? 2 : (level === 'warn' ? 3 : 4)));
}
/**
Starts debugging the native (C/C++) code. Requires webrt4all plugin.
On Windows, the output file should be at <b>C:\Users\<YOUR LOGIN>\AppData\Local\Temp\Low\webrtc4all.log</b>.
Starting the native debug isn't recommended and must be done to track issues only.
@since version 1.5.222
*/
SIPml.startNativeDebug = function () {
WebRtc4all_GetPlugin().startDebug();
}
/**
Stops debugging the native (C/C++) code. Requires webrt4all plugin.
@since version 1.5.222
*/
SIPml.stopNativeDebug = function () {
WebRtc4all_GetPlugin().stopDebug();
}
/**
Gets the current WebRTC type being used.
@since version 1.4.217
@returns {String} the WebRTC type. Possible values: <i>native</i>, <i>w4a</i>, <i>erisson</i> or <i>unknown</i>.
*/
SIPml.getWebRtcType = function() {
switch (WebRtc4all_GetType()) {
case WebRtcType_e.W4A: case WebRtcType_e.IE: case WebRtcType_e.NPAPI: return "w4a";
case WebRtcType_e.ERICSSON: return "erisson";
case WebRtcType_e.NATIVE: return "native";
default: return "unknown";
}
}
/**
Sets the default webrtc type. Must be called before <a href="#.init">initializing</a> the engine.
@since version 1.5.222
@param {String} type The type. Supported values: <i>native</i>, <i>w4a</i> and <i>erisson</i>.
@returns {Boolean} <i>true</i> if succeed; otherwise <i>false</i>
*/
SIPml.setWebRtcType = function(type) {
if (SIPml.isInitialized()) {
throw new Error("ERR_ALREADY_INITIALIZED: Engine already initialized.");
}
return WebRtc4all_SetType(type);
}
/**
Gets the list of running apps. Requires webrt4all plugin.
@since version 1.5.222
@returns {String} the the list of running apps. Format: <i>(base64($$WindowID$$=str(...)$$Description$$=str(...)$$IconData$$=base64(...)$$IconType$$=str(...)))*</i>
@throws {ERR_NOT_READY | ERR_NOT_SUPPORTED} <font color="red">ERR_NOT_READY</font> | <font color="red">ERR_NOT_SUPPORTED</font>
*/
SIPml.getRunningApps = function() {
if (SIPml.getWebRtcType() != 'w4a') {
throw new Error("ERR_NOT_SUPPORTED: requires webrtc4all plugin");
}
return WebRtc4all_GetPlugin().runningApps();
}
/**
Sets the video fps. Requires webrt4all plugin.
@since version 1.5.222
@param {Integer} fps fps value.
@returns {Integer} 0 if successful; otherwise nonzero
@throws {ERR_NOT_READY | ERR_NOT_SUPPORTED} <font color="red">ERR_NOT_READY</font> | <font color="red">ERR_NOT_SUPPORTED</font>
*/
SIPml.setFps = function(fps) {
if (SIPml.getWebRtcType() != 'w4a') {
throw new Error("ERR_NOT_SUPPORTED: Setting maximum video size requires webrtc4all plugin");
}
WebRtc4all_GetPlugin().fps = fps;
return 0;
}
/**
Sets the maximum video size. Requires webrt4all plugin.
@since version 1.5.222
@param {String} maxVideoSize maxVideoSize value. Supported values: "sqcif", "qcif" "qvga" "cif" "hvga", "vga", "4cif", "svga", "480p", "720p", "16cif", "1080p", "2160p".
@returns {Integer} 0 if successful; otherwise nonzero
@throws {ERR_NOT_READY | ERR_NOT_SUPPORTED} <font color="red">ERR_NOT_READY</font> | <font color="red">ERR_NOT_SUPPORTED</font>
*/
SIPml.setMaxVideoSize = function(maxVideoSize) {
if (SIPml.getWebRtcType() != 'w4a') {
throw new Error("ERR_NOT_SUPPORTED: Setting FPS requires webrtc4all plugin");
}
WebRtc4all_GetPlugin().maxVideoSize = maxVideoSize;
return 0;
}
/**
Sets the maximum bandwidth (upload). Requires webrt4all plugin.
@since version 1.5.222
@param {Integer} maxBandwidthUp maxBandwidthUp value (kbps).
@returns {Integer} 0 if successful; otherwise nonzero
@throws {ERR_NOT_READY | ERR_NOT_SUPPORTED} <font color="red">ERR_NOT_READY</font> | <font color="red">ERR_NOT_SUPPORTED</font>
*/
SIPml.setMaxBandwidthUp = function(maxBandwidthUp) {
if (SIPml.getWebRtcType() != 'w4a') {
throw new Error("ERR_NOT_SUPPORTED: Setting maximum bandwidth requires webrtc4all plugin");
}
WebRtc4all_GetPlugin().maxBandwidthUp = maxBandwidthUp;
return 0;
}
/**
Sets the maximum bandwidth (down). Requires webrt4all plugin.
@since version 1.5.222
@param {Integer} maxBandwidthUp maxBandwidthUp value (kbps).
@returns {Integer} 0 if successful; otherwise nonzero
@throws {ERR_NOT_READY | ERR_NOT_SUPPORTED} <font color="red">ERR_NOT_READY</font> | <font color="red">ERR_NOT_SUPPORTED</font>
*/
SIPml.setMaxBandwidthDown = function(maxBandwidthDown) {
if (SIPml.getWebRtcType() != 'w4a') {
throw new Error("ERR_NOT_SUPPORTED: Setting maximum bandwidth requires webrtc4all plugin");
}
WebRtc4all_GetPlugin().maxBandwidthDown = maxBandwidthDown;
return 0;
}
/**
Defines whether to enable "zero-artifacts" features. Requires webrt4all plugin. <br />
More information about this option on Doubango's TelePresence wiki page: <a href="https://code.google.com/p/telepresence/wiki/Technical_Video_quality#Zero-artifacts">https://code.google.com/p/telepresence/wiki/Technical_Video_quality#Zero-artifacts</a>
@since version 1.5.222
@param {Boolean} zeroArtifacts New optional value.
@returns {Integer} 0 if successful; otherwise nonzero
@throws {ERR_NOT_READY | ERR_NOT_SUPPORTED} <font color="red">ERR_NOT_READY</font> | <font color="red">ERR_NOT_SUPPORTED</font>
*/
SIPml.setZeroArtifacts = function(zeroArtifacts) {
if (SIPml.getWebRtcType() != 'w4a') {
throw new Error("ERR_NOT_SUPPORTED: Setting maximum bandwidth requires webrtc4all plugin");
}
WebRtc4all_GetPlugin().zeroArtifacts = zeroArtifacts;
return 0;
}
/**
Gets the version name of the installed <a href="http://code.google.com/p/webrtc4all/">webrtc4all plugin</a>.
You must <a href="#.init">initialize</a> the engine before calling this function.
@static
@returns {String} Version name (e.g. '1.12.756')
@throws {ERR_NOT_INITIALIZED} <font color="red">ERR_NOT_INITIALIZED</font> if the engine is not <a href="#.init">initialized</a>.
*/
SIPml.getWebRtc4AllVersion = function() {
if(!SIPml.isInitialized()){
throw new Error("ERR_NOT_INITIALIZED: Engine not initialized yet. Please call 'SIPml.init()' first");
}
return SIPml.s_webrtc4all_version;
};
/**
Gets the web browser version (e.g. <i>'1.5.beta'</i>).
You must <a href="#.init">initialize</a> the engine before calling this function.
@static
@returns {String} The the web browser version.
@throws {ERR_NOT_INITIALIZED} <font color="red">ERR_NOT_INITIALIZED</font> if the engine is not <a href="#.init">initialized</a>.
*/
SIPml.getNavigatorVersion = function() {
if(!SIPml.isInitialized()){
throw new Error("ERR_NOT_INITIALIZED: Engine not initialized yet. Please call 'SIPml.init()' first");
}
return SIPml.s_navigator_version;
};
/**
Gets the web browser friendly name (e.g. <i>'chrome'</i>, <i>'firefox'</i>, <i>'safari'</i>, <i>'opera'</i>, <i>'ie'</i> or <i>'netscape'</i>).
You must <a href="#.init">initialize</a> the engine before calling this function.
@static
@returns {String} The web browser friendly name.
@throws {ERR_NOT_INITIALIZED} <font color="red">ERR_NOT_INITIALIZED</font> if the engine is not <a href="#.init">initialized</a>.
*/
SIPml.getNavigatorFriendlyName = function() {
if(!SIPml.isInitialized()){
throw new Error("ERR_NOT_INITIALIZED: Engine not initialized yet. Please call 'SIPml.init()' first");
}
return SIPml.s_navigator_friendly_name;
};
/**
Gets the Operating System friendly name (e.g. <i>'windows'</i>, <i>'mac'</i>, <i>'lunix'</i>, <i>'solaris'</i>, <i>'sunos'</i> or <i>'powerpc'</i>).
You must <a href="#.init">initialize</a> the engine before calling this function.
@static
@returns {String} The Operating System friendly name.
@throws {ERR_NOT_INITIALIZED} <font color="red">ERR_NOT_INITIALIZED</font> if the engine is not <a href="#.init">initialized</a>.
*/
SIPml.getSystemFriendlyName = function() {
if(!SIPml.isInitialized()){
throw new Error("ERR_NOT_INITIALIZED: Engine not initialized yet. Please call 'SIPml.init()' first");
}
return SIPml.s_system_friendly_name;
};
/**
Checks whether the web browser supports WebRTC but is outdated.
You must <a href="#.init">initialize</a> the engine before calling this function.
@static
@returns {Boolean} <i>true</i> if outdated; otherwise <i>false</i>
@throws {ERR_NOT_INITIALIZED} <font color="red">ERR_NOT_INITIALIZED</font> if the engine is not <a href="#.init">initialized</a>.
*/
SIPml.isNavigatorOutdated= function () {
if(!SIPml.isInitialized()){
throw new Error("ERR_NOT_INITIALIZED: Engine not initialized yet. Please call 'SIPml.init()' first");
}
return SIPml.b_navigator_outdated;
}
/**
Checks whether the <a href="http://code.google.com/p/webrtc4all/">webrtc4all plugin</a> is outdated or not.
You must <a href="#.init">initialize</a> the engine before calling this function.
@static
@returns {Boolean} <i>true</i> if outdated; otherwise <i>false</i>
@throws {ERR_NOT_INITIALIZED} <font color="red">ERR_NOT_INITIALIZED</font> if the engine is not <a href="#.init">initialized</a>.
*/
SIPml.isWebRtc4AllPluginOutdated = function(){
if(!SIPml.isInitialized()){
throw new Error("ERR_NOT_INITIALIZED: Engine not initialized yet. Please call 'SIPml.init()' first");
}
return SIPml.b_webrtc4all_plugin_outdated;
}
/**
Checks whether the <a href="http://code.google.com/p/webrtc4all/">webrtc4all plugin</a> is installed or not.
You must <a href="#.init">initialize</a> the engine before calling this function.
@static
@returns {Boolean} <i>true</i> if supported; otherwise <i>false</i>
@throws {ERR_NOT_INITIALIZED} <font color="red">ERR_NOT_INITIALIZED</font> if the engine is not <a href="#.init">initialized</a>.
*/
SIPml.isWebRtc4AllSupported = function(){
if(!SIPml.isInitialized()){
throw new Error("ERR_NOT_INITIALIZED: Engine not initialized yet. Please call 'SIPml.init()' first");
}
return SIPml.b_webrtc4all_supported;
}
/**
Checks whether Screen share is supported on this browser.
You must <a href="#.init">initialize</a> the engine before calling this function.
@since version 1.3.203
@static
@returns {Boolean} <i>true</i> if supported; otherwise <i>false</i>
@throws {ERR_NOT_INITIALIZED} <font color="red">ERR_NOT_INITIALIZED</font> if the engine is not <a href="#.init">initialized</a>.
*/
SIPml.isScreenShareSupported = function () {
if (!SIPml.isInitialized()) {
throw new Error("ERR_NOT_INITIALIZED: Engine not initialized yet. Please call 'SIPml.init()' first");
}
return (SIPml.getWebRtcType() === "w4a") || (navigator.userAgent.match('Chrome') && parseInt(navigator.userAgent.match(/Chrome\/(.*) /)[1]) >= 26);
}
/**
Checks whether WebRTC is supported or not.
You must <a href="#.init">initialize</a> the engine before calling this function.
@static
@returns {Boolean} <i>true</i> if supported; otherwise <i>false</i>
@throws {ERR_NOT_INITIALIZED} <font color="red">ERR_NOT_INITIALIZED</font> if the engine is not <a href="#.init">initialized</a>.
*/
SIPml.isWebRtcSupported = function () {
if(!SIPml.isInitialized()){
throw new Error("ERR_NOT_INITIALIZED: Engine not initialized yet. Please call 'SIPml.init()' first");
}
return SIPml.b_webrtc_supported;
}
/**
Checks whether WebSocket is supported or not.
@static
@returns {Boolean} <i>true</i> if supported; otherwise <i>false</i>
*/
SIPml.isWebSocketSupported = function () {
return tsk_utils_have_websocket();
}
/**
Checks whether <a href="https://developer.mozilla.org/en-US/docs/WebRTC/navigator.getUserMedia">getUserMedia</a> is supported or not. The engined must be initialized before calling this function.
@static
@returns {Boolean} <i>true</i> if <a href="https://developer.mozilla.org/en-US/docs/WebRTC/navigator.getUserMedia">getUserMedia</a> is supported; otherwise <i>false</i>
*/
SIPml.haveMediaStream = function () {
if(!SIPml.isInitialized()){
throw new Error("ERR_NOT_INITIALIZED: Engine not initialized yet. Please call 'SIPml.init()' first");
}
return SIPml.b_have_media_stream;
}
/**
Checks whether the engine is ready to make/receive calls or not. <br />
The engine is ready when:
<ul>
<li>engine is <a href="#.init">initialized</a></li>
<li>webrtc is supported</li>
<li>we got a valid media stream (from <a href="https://developer.mozilla.org/en-US/docs/WebRTC/navigator.getUserMedia">getUserMedia</a>)</li>
</ul>
@static
@returns {Boolean} <i>true</i> if the engine is ready; otherwise <i>false</i>
@throws {ERR_NOT_INITIALIZED} <font color="red">ERR_NOT_INITIALIZED</font> if the engine is not <a href="#.init">initialized</a>.
*/
SIPml.isReady = function () {
return (SIPml.isInitialized() && SIPml.isWebRtcSupported() && SIPml.haveMediaStream());
}
/**
Checks whether the engine is initialized or not. To initialize the stack you must call <a href="#.init">init()</a> function.
@static
@returns {Boolean} <i>true</i> if the engine is initialized; otherwise <i>false</i>
*/
SIPml.isInitialized = function () { return SIPml.b_initialized; }
/**
Initialize the engine. <b>You must call this function before any other.</b>.
@param {CallbackFunction} [readyCallback] Optional callback function to call when the stack finish initializing and become ready.
@param {CallbackFunction} [errorCallback] Optional callback function to call when initialization fails.
@example
SIPml.init(function(e){ console.info('engine is ready'); }, function(e){ console.info('Error: ' + e.message); });
@static
*/
SIPml.init = function (successCallback, errorCallback) {
if (!SIPml.b_initialized && !SIPml.b_initializing) {
SIPml.b_initializing = true;
tsk_utils_init_webrtc();
tsk_utils_log_info('User-Agent=' + (navigator.userAgent || "unknown"));
SIPml.b_have_media_stream = tsk_utils_have_stream();
SIPml.b_webrtc_supported = tsk_utils_have_webrtc();
SIPml.b_webrtc4all_supported = tsk_utils_have_webrtc4all();
SIPml.s_webrtc4all_version = tsk_utils_webrtc4all_get_version();
SIPml.s_navigator_friendly_name = tsk_utils_get_navigator_friendly_name();
SIPml.s_system_friendly_name = tsk_utils_get_system_friendly_name();
// prints whether WebSocket is supported
tsk_utils_log_info("WebSocket supported = " + (SIPml.isWebSocketSupported() ? "yes" : "no"));
// check webrtc4all version
if (tsk_utils_have_webrtc4all()) {
tsk_utils_log_info("WebRTC type = " + WebRtc4all_GetType() + " version = " + tsk_utils_webrtc4all_get_version());
if (SIPml.s_webrtc4all_version != '1.35.981') {
SIPml.b_webrtc4all_plugin_outdated = true;
}
}
// prints navigator friendly name
tsk_utils_log_info("Navigator friendly name = " + SIPml.s_navigator_friendly_name);
// gets navigator version
if(SIPml.s_navigator_friendly_name == 'ie'){
var re = new RegExp("MSIE ([0-9]{1,}[\.0-9]{0,})");
if (re.exec(navigator.userAgent) != null) {
SIPml.s_navigator_version = RegExp.$1;
}
}
// prints OS friendly name
tsk_utils_log_info("OS friendly name = " + SIPml.s_system_friendly_name);
// prints support for WebRTC (native or plugin)
tsk_utils_log_info("Have WebRTC = " + (tsk_utils_have_webrtc() ? "yes" : "false"));
// prints support for getUserMedia
tsk_utils_log_info("Have GUM = " + (tsk_utils_have_stream() ? "yes" : "false"));
// checks for WebRTC support
if (!tsk_utils_have_webrtc()) {
// is it chrome?
if (SIPml.s_navigator_friendly_name == "chrome") {
SIPml.b_navigator_outdated = true;
return;
}
// for now the plugins (WebRTC4all only works on Windows)
if (SIPml.s_system_friendly_name == 'win' || SIPml.s_system_friendly_name == 'windows') {
// Internet explorer
if (SIPml.s_navigator_friendly_name == 'ie') {
// Check for IE version
var rv = -1;
var ua = navigator.userAgent;
var re = new RegExp("MSIE ([0-9]{1,}[\.0-9]{0,})");
if (re.exec(ua) != null) {
rv = parseFloat(RegExp.$1);
}
if (rv < 9.0) {
SIPml.b_navigator_outdated = true;
return;
}
// break page loading ('window.location' won't stop JS execution)
if (!tsk_utils_have_webrtc4all()) {
return;
}
}
}
}
if(SIPml.b_webrtc_supported && SIPml.b_have_media_stream){
SIPml.b_initialized = true;
SIPml.b_initializing = false;
tsk_utils_log_info("Engine initialized");
if(successCallback){
successCallback({});
}
}
else{
if(errorCallback){
var s_description = !SIPml.b_webrtc_supported ? "WebRTC not supported" : (!SIPml.b_have_media_stream ? "getUserMedia not supported" : "Internal error");
errorCallback({description: s_description});
}
}
}
}
// ================================== SIPml.EventTarget ==========================================
/**
@constructor
Defines an event target. You sould never create an event target object.
*/
SIPml.EventTarget = function () {
this.ao_listeners = [];
}
/**
Adds an event listener to the target object. <br /><br />
<table border="1">
<tr>
<td><b>Target classes<b></td>
<td><b>Supported event types<b></td>
<td><b>Raised event object<b></td>
<td><b>Remarques<b></td>
</tr>
<tr>
<td><a href="SIPml.Stack.html" name="SIPml.EventTarget.Stack">SIPml.Stack</a></td>
<td>
<b>*</b><br/> starting<br/> started<br/> stopping<br/> stopped<br/> failed_to_start<br/> failed_to_stop<br/> i_new_call<br /> i_new_message<br />
m_permission_requested<br/> m_permission_accepted<br/> m_permission_refused
</td>
<td><a href="SIPml.Stack.Event.html">SIPml.Stack.Event</a></td>
<td>'*' is used to listen for all events</td>
</tr>
<tr>
<td>
<a href="SIPml.Session.html" name="SIPml.EventTarget.Session">SIPml.Session</a>
<ul>
<li><a href="SIPml.Session.Call.html">SIPml.Session.Call</a></li>
<li><a href="SIPml.Session.Message.html">SIPml.Session.Message</a></li>
<li><a href="SIPml.Session.Message.html">SIPml.Session.Registration</a></li>
<li><a href="SIPml.Session.Message.html">SIPml.Session.Subscribe</a></li>
<li><a href="SIPml.Session.Message.html">SIPml.Session.Publish</a></li>
<ul>
</td>
<td><b>*</b><br/> connecting<br /> connected<br /> terminating<br /> terminated<br />
i_ao_request<br />
media_added<br/> media_removed<br/>
i_request<br/> o_request<br/> cancelled_request<br/> sent_request<br/>
transport_error<br/> global_error<br/> message_error<br/> webrtc_error
</td>
<td><a href="SIPml.Session.Event.html">SIPml.Session.Event</a></td>
<td>'*' is used to listen for all events<br /></td>
</tr>
<tr>
<td><a href="SIPml.Session.Call.html" name="SIPml.EventTarget.Session.Call">SIPml.Session.Call</a></td>
<td>
m_early_media<br/> m_local_hold_ok<br/> m_local_hold_nok<br/> m_local_resume_ok<br/> m_local_resume_nok<br/> m_remote_hold<br/> m_remote_resume<br/>
m_stream_video_local_added<br /> m_stream_video_local_removed<br/> m_stream_video_remote_added<br/> m_stream_video_remote_removed <br />
m_stream_audio_local_added<br /> m_stream_audio_local_removed<br/> m_stream_audio_remote_added<br/> m_stream_audio_remote_removed <br />
i_ect_new_call<br/> o_ect_trying<br/> o_ect_accepted<br/> o_ect_completed<br/> i_ect_completed<br/> o_ect_failed<br/> i_ect_failed<br/> o_ect_notify<br/> i_ect_notify<br/> i_ect_requested <br />
m_bfcp_info<br />
i_info
</td>
<td><a href="SIPml.Session.Event.html">SIPml.Session.Event</a></td>
<td>borrows all events supported by <a href="SIPml.Session.html">SIPml.Session</a></td>
</tr>
<tr>
<td><a href="SIPml.Session.Subscribe.html" name="SIPml.EventTarget.Session.Subscribe">SIPml.Session.Subscribe</a></td>
<td>
i_notify
</td>
<td><a href="SIPml.Session.Event.html">SIPml.Session.Event</a></td>
<td>borrows all events supported by <a href="SIPml.Session.html">SIPml.Session</a></td>
</tr>
</table>
@param {String|Array} type The event type/identifier. Must not be null or empty. Use <b>'*'</b> to listen for all events.
@param {function} listener The object that receives a notification when an event of the specified type occurs. This must be an object implementing the <a href="http://www.w3.org/TR/DOM-Level-2-Events/events.html#Events-EventListener">EventListener</a> interface, or simply a JavaScript function.
@example
// listen for a single event
this.addEventListener('started', function(e){
console.info("'started' event fired");
});
// or listen for two or more events
this.addEventListener(['started', 'stopped'], function(e){
console.info("'"+e.type+"' event fired");
});
// or listen for all events
this.addEventListener('*', function(e){
console.info("'"+e.type+"' event fired");
});
@see <a href="#removeEventListener">removeEventListener</a>
@throws {ERR_INVALID_PARAMETER_VALUE|ERR_INVALID_PARAMETER_TYPE} <font color="red">ERR_INVALID_PARAMETER_VALUE</font> | <font color="red">ERR_INVALID_PARAMETER_TYPE</font>
*/
SIPml.EventTarget.prototype.addEventListener = function (o_type, o_listener) {
if (!o_listener) {
throw new Error("ERR_INVALID_PARAMETER_VALUE: 'listener' must not be null");
}
if (!o_type) {
throw new Error("ERR_INVALID_PARAMETER_VALUE: 'type' must not be null");
}
if(!(o_type instanceof String || typeof o_type == "string" || o_type instanceof Array)){
throw new Error("ERR_INVALID_PARAMETER_TYPE: 'type' must be a string or array");
}
if(o_type instanceof Array){
var This = this;
o_type.forEach(function (s_type) {
if (!tsk_string_is_null_or_empty(s_type) && tsk_string_is_string(s_type)) {
This.ao_listeners[s_type] = o_listener;
}
});
}
else{
this.ao_listeners[o_type] = o_listener;
}
}
/**
Removes an event listener from the target object.
@param {String} type The event type/identifier to stop listening for.
@see <a href="#addEventListener">addEventListener</a>
@exemple
this.removeEventListener('started');
*/
SIPml.EventTarget.prototype.removeEventListener = function (s_type) {
if (tsk_string_is_string(s_type) && !tsk_string_is_null_or_empty(s_type)) {
this.ao_listeners[s_type] = undefined;
}
}
/**
@ignore
@private
@param {Object} event
*/
SIPml.EventTarget.prototype.dispatchEvent = function (o_event) {
var o_listener = (this.ao_listeners[o_event.s_type] || this.ao_listeners['*']);
if (o_listener) {
o_listener.call(this, o_event.o_value);
}
}
// ================================== SIPml.Event ==========================================
/**
SIP event object. You should never create an instance of this class by yourself.
@constructor
@param {String} type The event type or identifier. Please check <a href="SIPml.EventTarget.html#SIPml.EventTarget.Session">this link</a> for more information about all supported session event types.
@param {tsip_event} [event] Private wrapped session object.
@property {String} type The event <a href="SIPml.EventTarget.html#SIPml.EventTarget.Session">type or identifier</a> (e.g. <i>'connected'</i>).
@property {String} description User-friendly description in english (e.g. <i>'Session is now connected'</i>).
*/
SIPml.Event = function (s_type, o_event) {
this.type = s_type;
this.description = o_event ? o_event.s_phrase : s_type;
this.o_event = o_event;
}
/**
Gets the SIP response code.
@returns {Integer} The SIP response code (e.g. 404).
*/
SIPml.Event.prototype.getSipResponseCode = function () {
var o_message = this.o_event ? this.o_event.get_message() : null;
if (o_message && o_message.is_response()) {
return o_message.get_response_code();
}
return -1;
}
/**
Gets the SIP content associated to this event. This function could be called to get the content of the incoming SIP message ('i_new_message' event).
@returns {Object} SIP content.
@see <a href="#getContentType">getContentType</a>, <a href="#getContentString">getContentString</a>
*/
SIPml.Event.prototype.getContent = function () {
var o_message = this.o_event ? this.o_event.get_message() : null;
if (o_message) {
return o_message.get_content();
}
return null;
}
/**
Gets the SIP content associated to this event. This function could be called to get the content of the incoming SIP message ('i_new_message' event).
@returns {String} SIP content.
@see <a href="#getContentType">getContentType</a>, <a href="#getContent">getContent</a>
*/
SIPml.Event.prototype.getContentString = function () {
var o_message = this.o_event ? this.o_event.get_message() : null;
if (o_message) {
return o_message.get_content_as_string();
}
return null;
}
/**
Gets the SIP content-type associated to this event. This function could be called to get the content-type of the incoming SIP message ('i_new_message' event).
@returns {Object} SIP content-type.
@see <a href="#getContent">getContent</a>
*/
SIPml.Event.prototype.getContentType = function () {
var o_message = this.o_event ? this.o_event.get_message() : null;
if (o_message) {
return o_message.get_content_type();
}
return null;
}
// ================================== SIPml.Stack ==========================================
/**
Anonymous SIP Stack configuration object.
@namespace SIPml.Stack.Configuration
@name SIPml.Stack.Configuration
@property {String} realm The domain name. Required for stack <a href="SIPml.Stack.html#constructor">constructor</a> but optional when used with <a href="SIPml.Stack.html#setConfiguration">setConfiguration</a>. <br />
Example: <i>example.org</i>
@property {String} impi The authentication name. Required for stack <a href="SIPml.Stack.html#constructor">constructor</a> but optional when used with <a href="SIPml.Stack.html#setConfiguration">setConfiguration</a>.<br />
Example: <i>+33600000000</i> or <i>bob</i>.
@property {string} impu The full SIP uri address. Required for stack <a href="SIPml.Stack.html#constructor">constructor</a> but optional when used with <a href="SIPml.Stack.html#setConfiguration">setConfiguration</a>.<br />
Example: <i>sip:+33600000000@example.com</i> or <i>tel:+33600000000</i> or <i>sip:bob@example.com</i>
@property {String} [password] The password to use for SIP authentication.<br />
Example: <i>mysecret</i>
@property {String} [display_name] The display name to use in SIP requests. This is the String displayed by the called party for incoming calls. <br />
Example: <i>I Am Legend</i>
@property {String} [websocket_proxy_url] The websocket proxy url to connect to (SIP server or gateway address). If unset the stack will use sipml5.org as host and a random port. You should not set this value unless you know what you're doing.<br />
Example: <i>ws://sipml5.org:5060</i>
@property {String} [outbound_proxy_url] The outbound Proxy URL is used to set the destination IP address and Port to use for all outgoing requests regardless the <i>domain name</i> (a.k.a <i>realm</i>). <br />
This is a good option for developers using a SIP domain name without valid DNS A/NAPTR/SRV records. You should not set this value unless you know what you're doing. <br />
Example: <i>udp://192.168.0.12:5060</i>
@property {Array} [ice_servers] The list of the STUN/TURN servers to use. The format must be as explained at <a target=_blank href="http://www.w3.org/TR/webrtc/#rtciceserver-type">http://www.w3.org/TR/webrtc/#rtciceserver-type</a>. <br />
To disable TURN/STUN to speedup ICE candidates gathering you can use an empty array. e.g. <i>[]</i>. <br />
Example: <i>[{ url: 'stun:stun.l.google.com:19302'}, { url:'turn:user@numb.viagenie.ca', credential:'myPassword'}]</i>
@property {Object} [bandwidth] Defines the maximum audio and video bandwidth to use. This will change the outhoing SDP to include a "b:AS=" attribute. Use <i>0</i> to let the browser negotiates the right value using RTCP-REMB and congestion control. Same property could be used at session level to override this value.<br />
<i>Available since version 1.3.203</i>. <br />
Example: <i>{ audio:64, video:512 }</i>
@property {Object} [video_size] Defines the maximum and minimum video size to be used. All values are optional. The browser will try to find the best video size between <i>max</i> and <i>min</i> based on the camera capabilities. Same property could be used at session level to override this value.<br />
<i>Available since version 1.3.203</i>. <br />
Example: <i>{ minWidth:640, minHeight:480, maxWidth:1920, maxHeight:1080 }</i>
@property {Boolean} [enable_rtcweb_breaker] Whether to enable the <a href="http://webrtc2sip.org/#aRTCWebBreaker" target=_blank>RTCWeb Breaker</a> module to allow calling SIP-legacy networks. <br />
Example: <i>true</i>
@property {Boolean} [enable_click2call] Whether to enable the <a href="http://click2dial.org" target=_blank>Click2Call / Click2Dial</a> service.
<i>Available since version 1.2.181</i>. <br />
Example: <i>true</i>
@property {Boolean} [enable_early_ims] Whether to enable 3GGP Early IMS as per <a href="http://www.arib.or.jp/english/html/overview/doc/STD-T63v9_60/5_Appendix/Rel6/33/33978-660.pdf" target=_blank>TR 33.978</a>. Should be 'true' unless you're using a real IMS network. <br />
<i>Available since version 1.3.203</i>. <br />
Example: <i>true</i>
@property {Boolean} [enable_media_stream_cache] Whether to reuse the same media stream for all calls. If your website is <b>not using https</b> then, the browser will request access to the camera (or microphone) every time you try to make a call. Caching the media stream will avoid getting these notifications for each call. <br />
<i>Available since version 1.3.203</i>. <br />
Example: <i>true</i>
@property {Object} [events_listener] Object to subscribe to some events.
Example:
<ul>
<li><i>{ events: '*', listener: function(e){} }</i> </li>
<li><i>{ events: 'started', listener: function(e){} }</i></li>
<li><i>{ events: ['started', 'stopped'], listener: function(e){} }</i></li>
</ul>
You can also use <a href="#addEventListener">addEventListener</a> to add listeners to the stack.
@property {Array} [sip_headers] Stack-level SIP headers to add to all outgoing requests. Each header is an object with a <i>name</i> and <i>value</i> fields. <br />
Example: <i>sip_headers: [{name: 'User-Agent', value: 'IM-client/OMA1.0 sipML5-v1.0.89.0'}, {name: 'Organization', value: 'Doubango Telecom'}]</i>
@example
var configuration = {
realm: 'example.org',
impi: 'bob',
impu: 'sip:bob@example.org',
password: 'mysecret', // optional
display_name: 'I Am Legend', // optional
websocket_proxy_url: 'ws://192.168.0.10:5060', // optional
outbound_proxy_url: 'udp://192.168.0.12:5060', // optional
ice_servers: [{ url: 'stun:stun.l.google.com:19302'}, { url:'turn:user@numb.viagenie.ca', credential:'myPassword'}], // optional
enable_rtcweb_breaker: true, // optional
enable_click2call: false, // optional
enable_early_ims: true, // optional
events_listener: { events: '*', listener: listenerFunc }, // optional
sip_headers: [ //optional
{name: 'User-Agent', value: 'IM-client/OMA1.0 sipML5-v1.0.89.0'},
{name: 'Organization', value: 'Doubango Telecom'}
]
};
*/
/**
This is the root object used by any other object to make/receive calls, messages or manage presence.
You have to create an instance of this class before anything else.
@extends SIPml.EventTarget
@constructor
@class
@param {SIPml.Stack.Configuration} configuration Configuration object. Could be updated later using <a href="#setConfiguration">setConfiguration</a>.
@throws {ERR_INVALID_PARAMETER_VALUE|ERR_INVALID_PARAMETER_TYPE} <font color="red">ERR_INVALID_PARAMETER_VALUE</font> | <font color="red">ERR_INVALID_PARAMETER_TYPE</font>
@example
var listenerFunc = function(e){
console.info('stack event = ' + e.type);
// Please check <a href="SIPml.EventTarget.html#SIPml.EventTarget.Stack">this link</a> for more information on all supported events.
}
var o_stack = new SIPml.Stack({
realm: 'example.org',
impi: 'bob',
impu: 'sip:bob@example.org',
password: 'mysecret', // optional
display_name: 'I Am Legend', // optional
websocket_proxy_url: 'ws://192.168.0.10:5060', // optional
outbound_proxy_url: 'udp://192.168.0.12:5060', // optional
ice_servers: [{ url: 'stun:stun.l.google.com:19302'}, { url:'turn:user@numb.viagenie.ca', credential:'myPassword'}], // optional
bandwidth: { audio:64, video:512 }, // optional
video_size: { minWidth:640, minHeight:480, maxWidth:1920, maxHeight:1080 }, // optional
enable_rtcweb_breaker: true, // optional
enable_click2call: false, // optional
events_listener: { events: '*', listener: listenerFunc }, //optional
sip_headers: [ //optional
{name: 'User-Agent', value: 'IM-client/OMA1.0 sipML5-v1.0.89.0'},
{name: 'Organization', value: 'Doubango Telecom'}
]
}
);
@see <a href="#setConfiguration">setConfiguration</a>
*/
SIPml.Stack = function (o_conf) {
SIPml.init();
SIPml.EventTarget.call(this);
/*
members:
- o_stack {tsip_stack}
*/
if (!o_conf) {
throw new Error("ERR_INVALID_PARAMETER_VALUE: null configuration value");
}
if (tsk_string_is_null_or_empty(o_conf.realm)) {
throw new Error("ERR_INVALID_PARAMETER_VALUE: '" + o_conf.realm + "' is not valid as realm value");
}
if (tsk_string_is_null_or_empty(o_conf.impi)) {
throw new Error("ERR_INVALID_PARAMETER_VALUE: '" + o_conf.impi + "' is not valid as impi value");
}
if (tsk_string_is_null_or_empty(o_conf.impu)) {
throw new Error("ERR_INVALID_PARAMETER_VALUE: '" + o_conf.impu + "' is not valid as impu value");
}
// check IMPU validity
var o_impu = tsip_uri.prototype.Parse(o_conf.impu);
if (!o_impu || !o_impu.s_user_name || !o_impu.s_host) {
throw new Error("ERR_INVALID_PARAMETER_VALUE: '" + o_conf.impu + "' is not valid as SIP Uri");
}
var i_port;
var s_proxy;
if (!SIPml.isWebSocketSupported()) {
// port and host will be updated using the result from DNS SRV(NAPTR(realm))
i_port = 5060;
s_proxy = o_conf.realm;
}
else {
// there are at least 5 servers running on the cloud.
// we will connect to one of them and let the balancer to choose the right one (less connected sockets)
// each port can accept up to 65K connections which means that the cloud can manage 325K active connections
// the number of port will be increased or decreased based on the current trafic
// webrtc2sip 2.2+ (Doubango):
// WS: 10060, 11060, 12060, 13060, 14060
// WSS: 10062, 11062, 12062, 13062, 14062
//
i_port = (o_conf.enable_rtcweb_breaker ? 10062 : 10060) + (((new Date().getTime()) % 5) * 1000);
s_proxy = "ns313841.ovh.net";
}
// create the stack
this.o_stack = new tsip_stack(o_conf.realm, o_conf.impi, o_conf.impu, s_proxy, i_port);
this.o_stack.oStack = this;
// set configurations
this.setConfiguration(o_conf);
// listen for stack events
this.o_stack.on_event_stack = function(e) {
var s_type;
switch (e.i_code) {
case tsip_event_code_e.STACK_STARTING: s_type = 'starting'; break;
case tsip_event_code_e.STACK_STARTED: s_type = 'started'; break;
case tsip_event_code_e.STACK_STOPPING: s_type = 'stopping'; break;
case tsip_event_code_e.STACK_STOPPED: s_type = 'stopped'; break;
case tsip_event_code_e.STACK_FAILED_TO_START: s_type = 'failed_to_start'; break;
case tsip_event_code_e.STACK_FAILED_TO_STOP: s_type = 'failed_to_stop'; break;
}
if(s_type){
e.o_stack.oStack.dispatchEvent({ s_type: s_type, o_value: new SIPml.Stack.Event(s_type, e) });
}
}
// listen for dialog events
this.o_stack.on_event_dialog = function (e) {
var s_type = null;
var i_session_id = e.o_session.i_id;
var oSession = e.o_session.o_stack.oStack.ao_sessions[i_session_id];
if (!oSession) {
tsk_utils_log_warn('Cannot find session with id = ' + i_session_id);
return;
}
switch (e.i_code) {
case tsip_event_code_e.DIALOG_TRANSPORT_ERROR: s_type = 'transport_error'; break;
case tsip_event_code_e.DIALOG_GLOBAL_ERROR: s_type = 'global_error'; break;
case tsip_event_code_e.DIALOG_MESSAGE_ERROR: s_type = 'message_error'; break;
case tsip_event_code_e.DIALOG_WEBRTC_ERROR: s_type = 'webrtc_error'; break;
case tsip_event_code_e.DIALOG_REQUEST_INCOMING: s_type = 'i_request'; break;
case tsip_event_code_e.DIALOG_REQUEST_OUTGOING: s_type = 'o_request'; break;
case tsip_event_code_e.DIALOG_REQUEST_CANCELLED: s_type = 'cancelled_request'; break;
case tsip_event_code_e.DIALOG_REQUEST_SENT: s_type = 'sent_request'; break;
case tsip_event_code_e.DIALOG_MEDIA_ADDED: s_type = 'media_added'; break;
case tsip_event_code_e.DIALOG_MEDIA_REMOVED: s_type = 'media_removed'; break;
case tsip_event_code_e.DIALOG_CONNECTING: s_type = 'connecting'; break;
case tsip_event_code_e.DIALOG_CONNECTED: s_type = 'connected'; break;
case tsip_event_code_e.DIALOG_BFCP_INFO: s_type = 'bfcp_info'; break;
case tsip_event_code_e.DIALOG_TERMINATING: s_type = 'terminating'; break;
case tsip_event_code_e.DIALOG_TERMINATED:
{
s_type = 'terminated';
e.o_session.o_stack.oStack.ao_sessions[i_session_id] = undefined;
break;
}
default: break;
}
if (s_type) {
oSession.dispatchEvent({ s_type: s_type, o_value: new SIPml.Session.Event(oSession, s_type, e) });
}
}
// listen for MESSAGE events
this.o_stack.on_event_message = function (e) {
var s_type = null;
var i_session_id = e.o_session.i_id;
var oSession = e.o_session.o_stack.oStack.ao_sessions[i_session_id];
switch (e.e_message_type) {
case tsip_event_message_type_e.I_MESSAGE: s_type = 'i_new_message'; break;
case tsip_event_message_type_e.AO_MESSAGE: s_type = 'i_ao_request'; break;
}
if (s_type) {
// 'i_new_call' is stack-level event
if (s_type == 'i_new_message') {
var oNewEvent = new SIPml.Stack.Event(s_type, e);
oNewEvent.newSession = new SIPml.Session.Message(e.o_session);
e.o_session.o_stack.oStack.ao_sessions[i_session_id] = oNewEvent.newSession; // save session
e.o_session.o_stack.oStack.dispatchEvent({ s_type: s_type, o_value: oNewEvent});
}
else {
if(oSession){
oSession.dispatchEvent({ s_type: s_type, o_value: new SIPml.Session.Event(oSession, s_type, e) });
}
else{
tsk_utils_log_warn('Cannot find session with id = ' + i_session_id + ' and event = ' + e.e_invite_type);
}
}
}
};
// listen for PUBLISH events
this.o_stack.on_event_publish = function (e) {
var s_type = null;
var i_session_id = e.o_session.i_id;
var oSession = e.o_session.o_stack.oStack.ao_sessions[i_session_id];
if(!oSession){
tsk_utils_log_warn('Cannot find session with id = ' + i_session_id + ' and event = ' + e.e_invite_type);
return;
}
switch(e.e_publish_type){
case tsip_event_publish_type_e.I_PUBLISH: break;
case tsip_event_publish_type_e.I_UNPUBLISH: break;
case tsip_event_publish_type_e.AO_PUBLISH:
case tsip_event_publish_type_e.AO_UNPUBLISH:
{
s_type = 'i_ao_request';
break;
}
}
if(s_type){
oSession.dispatchEvent({ s_type: s_type, o_value: new SIPml.Session.Event(oSession, s_type, e) });
}
}
// listen for SUBSCRIBE events
this.o_stack.on_event_subscribe = function (e) {
var s_type = null;
var i_session_id = e.o_session.i_id;
var oSession = e.o_session.o_stack.oStack.ao_sessions[i_session_id];
if(!oSession){
tsk_utils_log_warn('Cannot find session with id = ' + i_session_id + ' and event = ' + e.e_invite_type);
return;
}
switch(e.e_subscribe_type){
case tsip_event_subscribe_type_e.I_SUBSCRIBE: break;
case tsip_event_subscribe_type_e.I_UNSUBSRIBE: break;
case tsip_event_subscribe_type_e.AO_SUBSCRIBE:
case tsip_event_subscribe_type_e.AO_UNSUBSCRIBE:
case tsip_event_subscribe_type_e.AO_NOTIFY:
{
s_type = 'i_ao_request';
break;
}
case tsip_event_subscribe_type_e.I_NOTIFY:
{
s_type = 'i_notify';
break;
}
}
if(s_type){
oSession.dispatchEvent({ s_type: s_type, o_value: new SIPml.Session.Event(oSession, s_type, e) });
}
}
// listen for INVITE events
this.o_stack.on_event_invite = function (e) {
var s_type = null;
var i_session_id = e.o_session.i_id;
var oSession = e.o_session.o_stack.oStack.ao_sessions[i_session_id];
if (!oSession) {
switch (e.e_invite_type) {
case tsip_event_invite_type_e.I_NEW_CALL:
case tsip_event_invite_type_e.M_STREAM_LOCAL_REQUESTED:
case tsip_event_invite_type_e.M_STREAM_LOCAL_ACCEPTED:
case tsip_event_invite_type_e.M_STREAM_LOCAL_REFUSED:
case tsip_event_invite_type_e.M_BFCP_INFO:
break;
case tsip_event_invite_type_e.M_STREAM_LOCAL_ADDED:
case tsip_event_invite_type_e.M_STREAM_REMOTE_ADDED:
case tsip_event_invite_type_e.M_STREAM_LOCAL_REMOVED:
case tsip_event_invite_type_e.M_STREAM_REMOTE_REMOVED:
case tsip_event_invite_type_e.I_AO_REQUEST:
tsk_utils_log_info('Not notifying to session with id = ' + i_session_id + ' for event = ' + e.e_invite_type);
return;
default:
tsk_utils_log_warn('Cannot find session with id = ' + i_session_id + ' and event = ' + e.e_invite_type);
return;
}
}