rtc-phone
Version:
hl95 WebRTC phone
896 lines (820 loc) • 27.8 kB
JavaScript
import EventEmitter from "eventemitter3";
import utils from "./utils/utils";
import Log from "./utils/Log";
class RtcPhone extends EventEmitter {
/**
*
* @param {String} webRtcType Sets the default webrtc type. Supported values: native, w4a and erisson.
* @param {Integer} pfs Sets the video fps. Requires webrt4all plugin.
* @param {String} maxVideoSize maxVideoSize value. Supported values: "sqcif", "qcif" "qvga" "cif" "hvga", "vga", "4cif", "svga", "480p", "720p", "16cif", "1080p", "2160p".
* @param {Integer} maxBandwidthUp Sets the maximum bandwidth (upload). Requires webrt4all plugin. value (kbps).
* @param {Integer} maxBandwidthDown Sets the maximum bandwidth (down). Requires webrt4all plugin. value (kbps).
* @param {Boolean} zeroArtifacts Defines whether to enable "zero-artifacts" features. Requires webrt4all plugin.
* @param {Boolean} startNativeDebug Starts debugging the native (C/C++) code. Requires webrt4all plugin. On Windows, the output file should be at C:\Users\<YOUR LOGIN>\AppData\Local\Temp\Low\webrtc4all.log. Starting the native debug isn't recommended and must be done to track issues only.
*/
constructor({
debugLevel = "info",
webRtcType = "native",
pfs,
maxVideoSize,
maxBandwidthUp,
maxBandwidthDown,
zeroArtifacts,
startNativeDebug = false,
ringToneDom = null,
ringbackToneDom = null,
audioRemoteDom,
videoLocalDom = null,
videoRemoteDom = null,
onInitialized,
onStackStartFailed,
onConnected,
onDisconnected,
onRinging,
onDialing,
onTalking,
onReleased,
}) {
super();
if (document.readyState !== "complete") {
throw "Dom not ready !";
}
Log.info(`RtcPhone Constructor:location=${window.location}`);
// set debug level
this.debugLevel = debugLevel;
SIPml.setDebugLevel(this.debugLevel);
//
this.webRtcType = webRtcType;
//
this.pfs = pfs;
this.maxVideoSize = maxVideoSize;
this.maxBandwidthUp = maxBandwidthUp;
this.maxBandwidthDown = maxBandwidthDown;
this.zeroArtifacts = zeroArtifacts;
this.startNativeDebug = startNativeDebug;
// check dom
if (ringToneDom && !(ringToneDom instanceof HTMLAudioElement)) {
throw new Error(`Parameter 'ringToneDom' not found or not an audio element.`);
} else {
this.ringTone = ringToneDom;
}
if (ringbackToneDom && !(ringbackToneDom instanceof HTMLAudioElement)) {
throw new Error(`Parameter 'ringbackToneDom' not found or not an audio element.`);
} else {
this.ringbackTone = ringbackToneDom;
}
if (!(audioRemoteDom instanceof HTMLAudioElement)) {
throw new Error(`Parameter 'audioRemoteDom' not found or not an audio element.`);
} else {
this.audioRemote = audioRemoteDom;
}
if (videoLocalDom && !(videoLocalDom instanceof HTMLVideoElement)) {
throw new Error(`Parameter 'videoLocalDom' not found or not an video element.`);
} else {
this.videoLocal = videoLocalDom;
}
if (videoRemoteDom && !(videoRemoteDom instanceof HTMLVideoElement)) {
throw new Error(`Parameter 'videoRemoteDom' not found or not an video element.`);
} else {
this.videoRemote = videoRemoteDom;
}
this.configCall = {
audio_remote: this.audioRemote,
video_local: this.videoLocal,
video_remote: this.videoRemote,
screencast_window_id: 0x00000000, // entire desktop
bandwidth: { audio: undefined, video: undefined },
video_size: {
minWidth: undefined,
minHeight: undefined,
maxWidth: undefined,
maxHeight: undefined,
},
events_listener: {
events: "*",
listener: this.onSipEventSession.bind(this),
},
sip_caps: [
{ name: "+g.oma.sip-im" },
{ name: "language", value: '"en,fr"' },
],
};
// sip协议栈对象
this.sipStack = null;
// 注册对象
this.sipSessionRegister = null;
// 主会话
this.sipSessionCall = null;
// 转接呼叫会话
this.sipSessionTransferCallByCti = null;
this.status = "uninitialized";
// add event listener
utils.isFunction(onInitialized) && this.on("initialized", onInitialized);
utils.isFunction(onStackStartFailed) && this.on("stackStartFailed", onStackStartFailed);
utils.isFunction(onConnected) && this.on("connected", onConnected);
utils.isFunction(onDisconnected) && this.on("disconnected", onDisconnected);
utils.isFunction(onRinging) && this.on("ringing", onRinging);
utils.isFunction(onDialing) && this.on("dialing", onDialing);
utils.isFunction(onTalking) && this.on("talking", onTalking);
utils.isFunction(onReleased) && this.on("released", onReleased);
}
/**
* initialize SIPML5
*/
sipInit() {
if (SIPml.isInitialized()) {
this.postInit();
} else {
// set default webrtc type (before initialization)
SIPml.setWebRtcType(this.webRtcType);
// initialize SIPML5
SIPml.init(this.postInit.bind(this));
// set other options after initialization
if (this.pfs) SIPml.setFps(this.pfs);
if (this.maxVideoSize) SIPml.setMaxVideoSize(this.maxVideoSize);
if (this.maxBandwidthUp) SIPml.setMaxBandwidthUp(this.maxBandwidthUp);
if (this.maxBandwidthDown) SIPml.setMaxBandwidthDown(this.maxBandwidthDown);
if (this.zeroArtifacts) SIPml.setZeroArtifacts(this.zeroArtifacts === true);
if (this.startNativeDebug === true) {
SIPml.startNativeDebug();
}
}
}
/**
* 初始化完成
* @returns
*/
postInit() {
// 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;
} else {
if (
confirm(
"webrtc-everywhere extension is not installed. Do you want to install it?\nIMPORTANT: You must restart your browser after the installation."
)
) {
window.location = "https://github.com/sarandogou/webrtc-everywhere";
} else {
// Must do nothing: give the user the chance to accept the extension
// window.location = "index.html";
}
}
}
// checks for WebSocket support
if (!SIPml.isWebSocketSupported()) {
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;
}
if (!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/";
}
}
this.status = "initialized";
this.emit("initialized");
}
/**
* sends SIP REGISTER request to login
* @param {*} displayName
* @returns
*/
sipRegister({
displayName = "",
privateIdentity = "",
publicIdentity = "",
password = "",
realm = "",
websocketProxyUrl = "wss://rtc.95ykf.com:4443",
enableRtcwebBreaker = false,
}) {
if (this.status === "uninitialized") {
Log.error("RtcPhone uninitialized!");
return false;
}
if (this.sipSessionRegister) {
Log.error("RtcPhone Registered!");
return false;
}
// catch exception for IE (DOM not ready)
try {
let _impu = tsip_uri.prototype.Parse(publicIdentity);
if (!_impu || !_impu.s_user_name || !_impu.s_host) {
throw "Public Identity format error !";
}
this.realm = realm;
this.displayName = displayName;
this.privateIdentity = privateIdentity;
this.publicIdentity = publicIdentity;
this.websocketProxyUrl = websocketProxyUrl;
this.enableRtcwebBreaker = enableRtcwebBreaker;
// enable notifications if not already done
if (
window.webkitNotifications &&
window.webkitNotifications.checkPermission() != 0
) {
window.webkitNotifications.requestPermission();
}
// update debug level to be sure new values will be used if the user haven't updated the page
// TODO: 对外提供函数，不在单独设置
SIPml.setDebugLevel(
window.localStorage &&
window.localStorage.getItem("org.doubango.expert.disable_debug") ==
"true"
? "error"
: "info"
);
// create SIP stack
this.sipStack = new SIPml.Stack({
realm: this.realm,
impi: this.privateIdentity,
impu: this.publicIdentity,
password: password,
display_name: this.displayName,
websocket_proxy_url: this.websocketProxyUrl,
outbound_proxy_url: window.localStorage
? window.localStorage.getItem(
"org.doubango.expert.sip_outboundproxy_url"
)
: null,
ice_servers: [],
enable_rtcweb_breaker: this.enableRtcwebBreaker,
events_listener: {
events: "*",
listener: this.onSipEventStack.bind(this),
},
enable_early_ims: window.localStorage
? window.localStorage.getItem(
"org.doubango.expert.disable_early_ims"
) != "true"
: true, // Must be true unless you're using a real IMS network
enable_media_stream_cache: window.localStorage
? window.localStorage.getItem(
"org.doubango.expert.enable_media_caching"
) == "true"
: false,
bandwidth: window.localStorage
? tsk_string_to_object(
window.localStorage.getItem("org.doubango.expert.bandwidth")
)
: null, // could be redefined a session-level
video_size: window.localStorage
? tsk_string_to_object(
window.localStorage.getItem("org.doubango.expert.video_size")
)
: null, // could be redefined a session-level
sip_headers: [
{ name: "User-Agent", value: "HL-WebRTC" },
{ name: "Organization", value: "HL95" },
],
});
// Failed to start the SIP stack
if (this.sipStack.start() != 0) {
this.emit("stackStartFailed");
}
} catch (e) {
this.emit("stackStartFailed", e);
Log.error(e)
}
}
// sends SIP REGISTER (expires=0) to logout
sipUnRegister() {
if (this.sipStack) {
this.sipStack.stop(); // shutdown all sessions
}
}
/**
* makes a call (SIP INVITE)
* @param {String} type 'call-audio', 'call-audiovideo', 'call-video', 'call-screenshare'.
* @param {*} phoneNumber
* @returns
*/
sipCall(type, phoneNumber) {
if (!this.sipStack) {
utils.showMessage("UnRegister!");
return false;
}
if (!this.sipSessionCall) {
if (!utils.checkPhoneNumber(phoneNumber)) {
utils.showMessage("Phone number format error !");
return false;
}
if (type == "call-screenshare") {
if (!SIPml.isScreenShareSupported()) {
utils.showMessage(
"Screen sharing not supported. Are you using chrome 26+?"
);
return false;
}
if (!location.protocol.match("https")) {
utils.showMessage("Screen sharing requires https://.");
this.sipUnRegister();
return false;
}
}
if (window.localStorage) {
this.configCall.bandwidth = tsk_string_to_object(
window.localStorage.getItem("org.doubango.expert.bandwidth")
); // already defined at stack-level but redifined to use latest values
this.configCall.video_size = tsk_string_to_object(
window.localStorage.getItem("org.doubango.expert.video_size")
); // already defined at stack-level but redifined to use latest values
}
// create call session
this.sipSessionCall = this.sipStack.newSession(type, this.configCall);
// make call
if (this.sipSessionCall.call(phoneNumber) != 0) {
this.sipSessionCall = null;
this.emit("callFailed");
return;
}
} else {
// this.callStatus = 'Connecting';
this.sipSessionCall.accept(this.configCall);
}
}
sipAnswer() {
if (!this.sipStack) {
utils.showMessage("UnRegister!");
return false;
}
if (!this.sipSessionCall) {
utils.showMessage("no call!");
return false;
} else {
this.sipSessionCall.accept(this.configCall);
}
}
// Share entire desktop aor application using BFCP or WebRTC native implementation
sipShareScreen() {
if (SIPml.getWebRtcType() === "w4a") {
// Sharing using BFCP -> requires an active session
if (!this.sipSessionCall) {
utils.showMessage("No active session");
return false;
}
if (this.sipSessionCall.bfcpSharing) {
if (this.sipSessionCall.stopBfcpShare(this.configCall) != 0) {
txtCallStatus.value = "Failed to stop BFCP share";
} else {
this.sipSessionCall.bfcpSharing = false;
}
} else {
this.configCall.screencast_window_id = 0x00000000;
if (this.sipSessionCall.startBfcpShare(this.configCall) != 0) {
txtCallStatus.value = "Failed to start BFCP share";
} else {
this.sipSessionCall.bfcpSharing = true;
}
}
} else {
sipCall("call-screenshare");
}
}
/**
* transfers the call
* @param {String} phoneNumber
* @returns
*/
sipTransfer(phoneNumber) {
if (!this.sipSessionCall) {
return false;
}
if (!utils.checkPhoneNumber(phoneNumber)) {
return false;
}
let ret = this.sipSessionCall.transfer(phoneNumber);
if (ret != 0) {
Log.error('Call transfer failed');
return false;
}
Log.info('Transfering the call...');
return true;
}
// holds or resumes the call
sipToggleHoldResume() {
if (!this.sipSessionCall) {
return false;
}
let ret = this.sipSessionCall.bHeld
? this.sipSessionCall.resume()
: this.sipSessionCall.hold();
if (ret != 0) {
return false;
}
return true;
}
// Mute or Unmute the call
sipToggleMute() {
if (!this.sipSessionCall) {
return false;
}
let bMute = !this.sipSessionCall.bMute;
let ret = this.sipSessionCall.mute("audio" /*could be 'video'*/, bMute);
if (ret != 0) {
return false;
}
this.sipSessionCall.bMute = bMute;
return true;
}
// terminates the call (SIP BYE or CANCEL)
sipHangup() {
if (this.sipSessionCall) {
this.sipSessionCall.hangup({
events_listener: {
events: "*",
listener: this.onSipEventSession.bind(this),
},
});
}
}
/**
*
* @param {String} c 按键
*/
sipSendDTMF(c) {
if (this.sipSessionCall && c) {
if (this.sipSessionCall.dtmf(c) == 0) {
Log.info(`sipSendDTMF ${c}`);
}
}
}
/**
* 播放铃声
*/
startRingTone() {
try {
if (this.ringTone) {
this.ringTone.play();
}
} catch (e) {
Log.error("铃声播放失败", e)
}
}
stopRingTone() {
try {
if (this.ringTone) {
this.ringTone.pause();
}
} catch (e) {
Log.error("停止铃声播放失败", e)
}
}
/**
* 播放回铃音
*/
startRingbackTone() {
try {
if (this.ringbackTone) {
this.ringbackTone.play();
}
} catch (e) {
Log.error("播放回铃音失败", e)
}
}
stopRingbackTone() {
try {
if (this.ringbackTone) {
this.ringbackTone.pause();
}
} catch (e) {
Log.error("停止播放回铃音失败", e)
}
}
// Callback function for SIP Stacks
onSipEventStack(e /*SIPml.Stack.Event*/) {
Log.info("==stack event = " + e.type, e);
switch (e.type) {
case "started": {
// catch exception for IE (DOM not ready)
try {
// LogIn (REGISTER) as soon as the stack finish starting
this.sipSessionRegister = this.sipStack.newSession("register", {
expires: 200,
events_listener: {
events: "*",
listener: this.onSipEventSession.bind(this),
},
sip_caps: [
{ name: "+g.oma.sip-im", value: null },
//{ name: '+sip.ice' }, // rfc5768: FIXME doesn't work with Polycom TelePresence
{ name: "+audio", value: null },
{ name: "language", value: '"en,fr"' },
],
});
this.sipSessionRegister.register();
} catch (e) {
Log.error("注册失败", e);
}
break;
}
case "stopping":
case "stopped":
case "failed_to_start":
case "failed_to_stop": {
let bFailure =
e.type == "failed_to_start" || e.type == "failed_to_stop";
this.sipStack = null;
this.sipSessionRegister = null;
this.sipSessionCall = null;
this.sipSessionTransferCallByCti = null;
this.emit("disconnected");
this.stopRingbackTone();
this.stopRingTone();
// uiVideoDisplayShowHide(false); TODO: 是否加时间考虑
Log.info(bFailure ? "Disconnected: " + e.description : "Disconnected");
break;
}
case "i_new_call": {
Log.info("==i_new_call_log = ", this.sipSessionCall);
let ao_headers = e.o_event.get_message().ao_headers;
let callInfo = ao_headers.find((h) => h.s_name === "Call-Info");
if (this.sipSessionCall) {
// 如果已经存在通话，新增通话时为转接振铃，需自动接听
if (callInfo && callInfo.s_value === "answer-after=0") {
this.sipSessionTransferCallByCti = e.newSession;
Log.info("CTI转接呼叫自动接听");
this.sipSessionTransferCallByCti.accept(this.configCall);
} else {
// do not accept the incoming call if we're already 'in call'
e.newSession.hangup(); // comment this line for multi-line support
}
} else {
this.sipSessionCall = e.newSession;
// 自动接听
if (callInfo && callInfo.s_value === "answer-after=0") {
Log.info("CTI发起外呼自动接听");
this.sipSessionCall.accept(this.configCall);
} else {
// start listening for events
this.sipSessionCall.setConfiguration(this.configCall);
this.callStatus = "ringing";
this.startRingTone();
let remoteNumber = this.sipSessionCall.getRemoteFriendlyName() || "unknown";
Log.info("Incoming call from " + remoteNumber, this.sipSessionCall);
this.emit("ringing", remoteNumber);
}
}
break;
}
case "m_permission_requested": {
Log.log("=======m_permission_requested=====" + e);
break;
}
case "m_permission_accepted":
case "m_permission_refused": {
Log.log("=======m_permission_accepted=m_permission_refused11=====" + e);
break;
}
case "starting":
default:
break;
}
}
/**
*
* @param {SIPml.Session.Event} e SIPml.Session.Event
*/
onSipEventSession(e) {
Log.info("==session event = " + e.type, e);
switch (e.type) {
case "connecting": {
if (e.session == this.sipSessionRegister) {
this.status = e.type;
this.emit(e.type, e);
} else if (e.session == this.sipSessionCall) {
this.emit("session", e);
}
break;
} // 'connecting'
case "connected": {
if (e.session == this.sipSessionRegister) {
this.status = e.type;
this.emit(e.type, e);
} else if (e.session == this.sipSessionCall) {
this.stopRingbackTone();
this.stopRingTone();
this.callStatus = "talking";
this.emit("talking", e);
}
break;
} // 'connected'
case "terminating":
case "terminated": {
if (e.session == this.sipSessionRegister) {
this.status = e.type;
this.sipSessionCall = null;
this.sipSessionTransferCallByCti = null;
this.sipSessionRegister = null;
this.emit("disconnected");
} else if (e.session == this.sipSessionCall) {
this.sipSessionCall = null;
this.stopRingbackTone();
this.stopRingTone();
this.emit("released", e);
} else if (e.session == this.sipSessionTransferCallByCti) {
this.sipSessionTransferCallByCti = null;
}
break;
} // 'terminating' | 'terminated'
case "m_stream_video_local_added": {
if (e.session == this.sipSessionCall) {
this.emit("localVideoAdded");
}
break;
}
case "m_stream_video_local_removed": {
if (e.session == this.sipSessionCall) {
this.emit("localVideoRemoved");
}
break;
}
case "m_stream_video_remote_added": {
if (e.session == this.sipSessionCall) {
//uiVideoDisplayEvent(false, true);
}
break;
}
case "m_stream_video_remote_removed": {
if (e.session == this.sipSessionCall) {
//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 == this.sipSessionCall) {
let iSipResponseCode = e.getSipResponseCode();
if (iSipResponseCode == 180 || iSipResponseCode == 183) {
this.callStatus = "dialing";
// FIXME: 使用对方铃声，呼叫开始阶段会有一段没有铃声
// this.startRingbackTone();
this.emit("dialing", e);
Log.info("Remote ringing...");
}
}
break;
}
case "m_early_media": {
if (e.session == this.sipSessionCall) {
this.stopRingbackTone();
this.stopRingTone();
Log.info("Early media started");
}
break;
}
case "m_local_hold_ok": {
if (e.session == this.sipSessionCall) {
if (this.sipSessionCall.bTransfering) {
this.sipSessionCall.bTransfering = false;
// this.AVSession.TransferCall(this.transferUri);
}
this.emit("holdCompleted", e);
Log.info("Call placed on hold");
this.sipSessionCall.bHeld = true;
}
break;
}
case "m_local_hold_nok": {
if (e.session == this.sipSessionCall) {
this.sipSessionCall.bTransfering = false;
this.emit("holdFailed", e);
Log.info("Failed to place remote party on hold");
}
break;
}
case "m_local_resume_ok": {
if (e.session == this.sipSessionCall) {
this.sipSessionCall.bTransfering = false;
this.emit("resumeCompleted", e);
Log.info("Call taken off hold");
this.sipSessionCall.bHeld = false;
if (SIPml.isWebRtc4AllSupported()) {
// IE don't provide stream callback yet
//uiVideoDisplayEvent(false, true);
//uiVideoDisplayEvent(true, true);
}
}
break;
}
case "m_local_resume_nok": {
if (e.session == this.sipSessionCall) {
this.sipSessionCall.bTransfering = false;
this.emit("resumeFailed", e);
Log.info("Failed to unhold call");
}
break;
}
case "m_remote_hold": {
if (e.session == this.sipSessionCall) {
Log.info("Placed on hold by remote party");
}
break;
}
case "m_remote_resume": {
if (e.session == this.sipSessionCall) {
Log.info("Taken off hold by remote party");
}
break;
}
case "m_bfcp_info": {
if (e.session == this.sipSessionCall) {
txtCallStatus.innerHTML = "BFCP Info: <i>" + e.description + "</i>";
}
break;
}
case "o_ect_trying": {
if (e.session == this.sipSessionCall) {
Log.info("Call transfer in progress...");
}
break;
}
case "o_ect_accepted": {
if (e.session == this.sipSessionCall) {
Log.info("Call transfer accepted");
}
break;
}
case "o_ect_completed":
case "i_ect_completed": {
if (e.session == this.sipSessionCall) {
Log.info("Call transfer completed");
if (oSipSessionTransferCall) {
this.sipSessionCall = oSipSessionTransferCall;
}
oSipSessionTransferCall = null;
this.emit("transferCompleted", e);
}
break;
}
case "o_ect_failed":
case "i_ect_failed": {
if (e.session == this.sipSessionCall) {
Log.info("Call transfer failed");
this.emit("transferFailed", e);
}
break;
}
case "o_ect_notify":
case "i_ect_notify": {
if (e.session == this.sipSessionCall) {
txtCallStatus.innerHTML = "<i>Call Transfer: <b>" + e.getSipResponseCode() + " " + e.description + "</b></i>";
if (e.getSipResponseCode() >= 300) {
if (this.sipSessionCall.bHeld) {
this.sipSessionCall.resume();
}
btnTransfer.disabled = false;
}
}
break;
}
case "i_ect_requested": {
if (e.session == this.sipSessionCall) {
let s_message = "Do you accept call transfer to [" + e.getTransferDestinationFriendlyName() + "]?"; //FIXME:
if (confirm(s_message)) {
txtCallStatus.innerHTML = "<i>Call transfer in progress...</i>";
this.sipSessionCall.acceptTransfer();
break;
}
this.sipSessionCall.rejectTransfer();
}
break;
}
}
}
}
RtcPhone.utils = utils;
RtcPhone.Log = Log;
export default RtcPhone;