@wazo/sdk 0.43.5 → 1.0.0-alpha.0

package/lib/web-rtc-client.js

@@ -1,28 +1,5 @@
 "use strict";
 /* eslint-disable class-methods-use-this, no-param-reassign, max-classes-per-file, no-underscore-dangle */
-var __createBinding = (this && this.__createBinding) || (Object.create ? (function(o, m, k, k2) {
-    if (k2 === undefined) k2 = k;
-    var desc = Object.getOwnPropertyDescriptor(m, k);
-    if (!desc || ("get" in desc ? !m.__esModule : desc.writable || desc.configurable)) {
-      desc = { enumerable: true, get: function() { return m[k]; } };
-    }
-    Object.defineProperty(o, k2, desc);
-}) : (function(o, m, k, k2) {
-    if (k2 === undefined) k2 = k;
-    o[k2] = m[k];
-}));
-var __setModuleDefault = (this && this.__setModuleDefault) || (Object.create ? (function(o, v) {
-    Object.defineProperty(o, "default", { enumerable: true, value: v });
-}) : function(o, v) {
-    o["default"] = v;
-});
-var __importStar = (this && this.__importStar) || function (mod) {
-    if (mod && mod.__esModule) return mod;
-    var result = {};
-    if (mod != null) for (var k in mod) if (k !== "default" && Object.prototype.hasOwnProperty.call(mod, k)) __createBinding(result, mod, k);
-    __setModuleDefault(result, mod);
-    return result;
-};
 var __awaiter = (this && this.__awaiter) || function (thisArg, _arguments, P, generator) {
     function adopt(value) { return value instanceof P ? value : new P(function (resolve) { resolve(value); }); }
     return new (P || (P = Promise))(function (resolve, reject) {
@@ -47,8 +24,8 @@ var __importDefault = (this && this.__importDefault) || function (mod) {
     return (mod && mod.__esModule) ? mod : { "default": mod };
 };
 Object.defineProperty(exports, "__esModule", { value: true });
-exports.CanceledCallError = exports.transportEvents = exports.events = exports.replaceLocalIpModifier = void 0;
-/* global window, document, navigator */
+exports.CanceledCallError = exports.transportEvents = exports.events = void 0;
+/* global document, navigator */
 require("webrtc-adapter");
 const user_agent_state_1 = require("sip.js/lib/api/user-agent-state");
 const parser_1 = require("sip.js/lib/core/messages/parser");
@@ -60,21 +37,14 @@ const messager_1 = require("sip.js/lib/api/messager");
 const registerer_state_1 = require("sip.js/lib/api/registerer-state");
 const session_state_1 = require("sip.js/lib/api/session-state");
 const transport_state_1 = require("sip.js/lib/api/transport-state");
-const peer_connection_configuration_default_1 = require("sip.js/lib/platform/web/session-description-handler/peer-connection-configuration-default");
 const getstats_1 = __importDefault(require("getstats"));
 const api_1 = require("sip.js/lib/api");
-const WazoSessionDescriptionHandler_1 = __importStar(require("./lib/WazoSessionDescriptionHandler"));
 const Emitter_1 = __importDefault(require("./utils/Emitter"));
 const api_client_1 = __importDefault(require("./api-client"));
 const IssueReporter_1 = __importDefault(require("./service/IssueReporter"));
 const Heartbeat_1 = __importDefault(require("./utils/Heartbeat"));
 const sdp_1 = require("./utils/sdp");
-const array_1 = require("./utils/array");
-// We need to replace 0.0.0.0 to 127.0.0.1 in the sdp to avoid MOH during a createOffer.
-const replaceLocalIpModifier = (description) => Promise.resolve(Object.assign(Object.assign({}, JSON.parse(JSON.stringify(description))), { sdp: description.sdp.replace('c=IN IP4 0.0.0.0', 'c=IN IP4 127.0.0.1') }));
-exports.replaceLocalIpModifier = replaceLocalIpModifier;
-const SIP_ID_LENGTH = 36;
-const DEFAULT_ICE_TIMEOUT = 3000;
+const webrtc_1 = require("./utils/webrtc");
 const SEND_STATS_DELAY = 5000;
 const states = ['STATUS_NULL', 'STATUS_NEW', 'STATUS_CONNECTING', 'STATUS_CONNECTED', 'STATUS_COMPLETED'];
 const logger = IssueReporter_1.default ? IssueReporter_1.default.loggerFor('webrtc-client') : console;
@@ -104,21 +74,7 @@ class CanceledCallError extends Error {
 }
 exports.CanceledCallError = CanceledCallError;
 const MAX_REGISTER_TRIES = 5;
-// setting a 24hr timeout and letting the backend define the actual value
-const NO_ANSWER_TIMEOUT = 60 * 60 * 24; // in seconds
 class WebRTCClient extends Emitter_1.default {
-    static isAPrivateIp(ip) {
-        const regex = /^(?:10|127|172\.(?:1[6-9]|2[0-9]|3[01])|192\.168)\..*/;
-        return regex.exec(ip) == null;
-    }
-    static getIceServers(ip) {
-        if (WebRTCClient.isAPrivateIp(ip)) {
-            return [{
-                    urls: ['stun:stun.l.google.com:19302', 'stun:stun4.l.google.com:19302'],
-                }];
-        }
-        return [];
-    }
     constructor(config, session, uaConfigOverrides) {
         super();
         // For debug purpose
@@ -183,7 +139,7 @@ class WebRTCClient extends Emitter_1.default {
     }
     createUserAgent(uaConfigOverrides) {
         var _a;
-        const uaOptions = this._createUaOptions(uaConfigOverrides);
+        const uaOptions = (0, webrtc_1.createUaOptions)(this.config, uaConfigOverrides, this.audio, this.video);
         logger.info('sdk webrtc, creating UA', {
             uaOptions: Object.assign(Object.assign({}, uaOptions), { authorizationPassword: `${(_a = uaOptions === null || uaOptions === void 0 ? void 0 : uaOptions.authorizationPassword) === null || _a === void 0 ? void 0 : _a.slice(0, 5)}xxxx` }),
             clientId: this.clientId,
@@ -200,7 +156,7 @@ class WebRTCClient extends Emitter_1.default {
                 });
                 this._setupSession(invitation);
                 const shouldAutoAnswer = !!invitation.request.getHeader('alert-info');
-                this.eventEmitter.emit(INVITE, invitation, this.sessionWantsToDoVideo(invitation), shouldAutoAnswer);
+                this.eventEmitter.emit(INVITE, invitation, (0, webrtc_1.sessionWantsToDoVideo)(invitation), shouldAutoAnswer);
             },
         };
         const ua = new user_agent_1.UserAgent(uaOptions);
@@ -299,7 +255,7 @@ class WebRTCClient extends Emitter_1.default {
                 logger.info('sdk webrtc registering aborted due to a registration in progress.', { clientId: this.clientId });
                 return Promise.resolve();
             }
-            const registerOptions = this._isWeb() ? {} : {
+            const registerOptions = (0, webrtc_1.isWeb)() ? {} : {
                 extraContactHeaderParams: ['mobility=mobile'],
             };
             const onRegisterFailed = () => {
@@ -456,7 +412,7 @@ class WebRTCClient extends Emitter_1.default {
         if (audioOnly) {
             inviterOptions.sessionDescriptionHandlerModifiersReInvite = [modifiers_1.stripVideo];
         }
-        const uri = this._makeURI(number);
+        const uri = (0, sdp_1.makeURI)(number, this.config.host);
         let session = null;
         if (uri) {
             session = this.userAgent ? new api_1.Inviter(this.userAgent, uri, inviterOptions) : null;
@@ -469,7 +425,7 @@ class WebRTCClient extends Emitter_1.default {
             this._setupSession(session);
         }
         if (conference) {
-            this.conferences[this.getSipSessionId(session)] = true;
+            this.conferences[(0, sdp_1.getSipSessionId)(session)] = true;
         }
         const inviteOptions = {
             requestDelegate: {
@@ -494,12 +450,12 @@ class WebRTCClient extends Emitter_1.default {
                     this.eventEmitter.emit(REJECTED, session, response);
                 },
             },
-            sessionDescriptionHandlerOptions: this.getMediaConfiguration(enableVideo || false, conference),
+            sessionDescriptionHandlerOptions: (0, webrtc_1.getMediaConfiguration)(enableVideo || false, conference),
         };
         if (inviteOptions.sessionDescriptionHandlerOptions) {
             inviteOptions.sessionDescriptionHandlerOptions.audioOnly = audioOnly;
         }
-        inviteOptions.sessionDescriptionHandlerModifiers = [exports.replaceLocalIpModifier];
+        inviteOptions.sessionDescriptionHandlerModifiers = [sdp_1.replaceLocalIpModifier];
         if (audioOnly) {
             inviteOptions.sessionDescriptionHandlerModifiers.push(modifiers_1.stripVideo);
         }
@@ -524,7 +480,7 @@ class WebRTCClient extends Emitter_1.default {
             return Promise.reject(new Error(error));
         }
         const options = {
-            sessionDescriptionHandlerOptions: this.getMediaConfiguration(enableVideo || false),
+            sessionDescriptionHandlerOptions: (0, webrtc_1.getMediaConfiguration)(enableVideo || false),
         };
         return this._accept(session, options).then(() => {
             // @ts-ignore: private
@@ -550,7 +506,7 @@ class WebRTCClient extends Emitter_1.default {
             try {
                 this._stopSendingStats(session);
                 this._cleanupMedia(session);
-                delete this.sipSessions[this.getSipSessionId(session)];
+                delete this.sipSessions[(0, sdp_1.getSipSessionId)(session)];
                 // Check if Invitation or Inviter (Invitation = incoming call)
                 const isInviter = session instanceof api_1.Inviter;
                 // @see github.com/onsip/SIP.js/blob/f11dfd584bc9788ccfc94e03034020672b738975/src/platform/web/simple-user/simple-user.ts#L1004
@@ -571,19 +527,9 @@ class WebRTCClient extends Emitter_1.default {
             return Promise.resolve(null);
         });
     }
-    getStats(session) {
-        return __awaiter(this, void 0, void 0, function* () {
-            var _a;
-            const pc = (_a = session.sessionDescriptionHandler) === null || _a === void 0 ? void 0 : _a.peerConnection;
-            if (!pc) {
-                return null;
-            }
-            return pc.getStats(null);
-        });
-    }
     // Fetch and emit an event at `interval` with session network stats
     startNetworkMonitoring(session, interval = 1000) {
-        const sessionId = this.getSipSessionId(session);
+        const sessionId = (0, sdp_1.getSipSessionId)(session);
         logger.info('starting network inspection', {
             id: sessionId,
             clientId: this.clientId,
@@ -592,7 +538,7 @@ class WebRTCClient extends Emitter_1.default {
         this.networkMonitoringInterval[sessionId] = setInterval(() => this._fetchNetworkStats(sessionId), interval);
     }
     stopNetworkMonitoring(session) {
-        const sessionId = this.getSipSessionId(session);
+        const sessionId = (0, sdp_1.getSipSessionId)(session);
         const exists = (sessionId in this.networkMonitoringInterval);
         logger.info('stopping network inspection', {
             clientId: this.clientId,
@@ -664,24 +610,17 @@ class WebRTCClient extends Emitter_1.default {
             logger.info('sdk webrtc client closed', { clientId: this.clientId });
         });
     }
-    getNumber(session) {
-        if (!session) {
-            return null;
-        }
-        // @ts-ignore: private
-        return session.remoteIdentity.uri._normal.user;
-    }
     mute(session) {
         logger.info('sdk webrtc mute', {
             id: session.id,
         });
-        this._toggleAudio(session, true);
+        (0, webrtc_1.toggleAudio)(session, true);
     }
     unmute(session) {
         logger.info('sdk webrtc unmute', {
             id: session.id,
         });
-        this._toggleAudio(session, false);
+        (0, webrtc_1.toggleAudio)(session, false);
     }
     isAudioMuted(session) {
         if (!session || !session.sessionDescriptionHandler) {
@@ -718,16 +657,16 @@ class WebRTCClient extends Emitter_1.default {
         logger.info('sdk webrtc toggle camera on', {
             id: session.id,
         });
-        this._toggleVideo(session, false);
+        (0, webrtc_1.toggleVideo)(session, false);
     }
     toggleCameraOff(session) {
         logger.info('sdk webrtc toggle camera off', {
             id: session.id,
         });
-        this._toggleVideo(session, true);
+        (0, webrtc_1.toggleVideo)(session, true);
     }
     hold(session, isConference = false, hadVideo = false) {
-        const sessionId = this.getSipSessionId(session);
+        const sessionId = (0, sdp_1.getSipSessionId)(session);
         const hasVideo = hadVideo || this.hasLocalVideo(sessionId);
         logger.info('sdk webrtc hold', {
             sessionId,
@@ -755,8 +694,8 @@ class WebRTCClient extends Emitter_1.default {
             hold: true,
             conference: isConference,
         };
-        const options = this.getMediaConfiguration(false, isConference);
-        if (!this._isWeb()) {
+        const options = (0, webrtc_1.getMediaConfiguration)(false, isConference);
+        if (!(0, webrtc_1.isWeb)()) {
             options.sessionDescriptionHandlerModifiers = [modifiers_1.holdModifier];
         }
         options.sessionDescriptionHandlerOptions = {
@@ -775,7 +714,7 @@ class WebRTCClient extends Emitter_1.default {
         });
     }
     unhold(session, isConference = false) {
-        const sessionId = this.getSipSessionId(session);
+        const sessionId = (0, sdp_1.getSipSessionId)(session);
         const hasVideo = sessionId in this.heldSessions && this.heldSessions[sessionId].hasVideo;
         logger.info('sdk webrtc unhold', {
             sessionId,
@@ -790,13 +729,13 @@ class WebRTCClient extends Emitter_1.default {
             return Promise.resolve();
         }
         this.unmute(session);
-        delete this.heldSessions[this.getSipSessionId(session)];
+        delete this.heldSessions[(0, sdp_1.getSipSessionId)(session)];
         session.sessionDescriptionHandlerOptionsReInvite = {
             hold: false,
             conference: isConference,
         };
-        const options = this.getMediaConfiguration(false, isConference);
-        if (!this._isWeb()) {
+        const options = (0, webrtc_1.getMediaConfiguration)(false, isConference);
+        if (!(0, webrtc_1.isWeb)()) {
             // We should sent an empty `sessionDescriptionHandlerModifiers` or sip.js will take the last sent modifiers
             // (eg: holdModifier)
             options.sessionDescriptionHandlerModifiers = [];
@@ -811,110 +750,27 @@ class WebRTCClient extends Emitter_1.default {
             logger.warn('sdk webrtc re-invite during resume, error', e);
         });
     }
-    // Returns true if a re-INVITE is required
-    upgradeToVideo(session, constraints, isConference) {
-        return __awaiter(this, void 0, void 0, function* () {
-            var _a, _b;
-            const pc = (_a = session.sessionDescriptionHandler) === null || _a === void 0 ? void 0 : _a.peerConnection;
-            // Check if a video sender already exists
-            let videoSender;
-            if (isConference) {
-                // We search for the last transceiver without `video-` in the mid (video- means remote transceiver)
-                const transceivers = (pc === null || pc === void 0 ? void 0 : pc.getTransceivers()) || [];
-                const transceiverIdx = (0, array_1.lastIndexOf)(transceivers, transceiver => transceiver.sender.track === null && transceiver.mid && transceiver.mid.indexOf('video') === -1);
-                videoSender = transceiverIdx !== -1 ? transceivers[transceiverIdx].sender : null;
-            }
-            else {
-                videoSender = pc && pc.getSenders && pc.getSenders().find((sender) => sender.track === null);
-            }
-            if (!videoSender) {
-                // When no video sender found, it means that we're in the first video upgrade in 1:1
-                return;
-            }
-            // Reuse bidirectional video stream
-            const newStream = yield this.getStreamFromConstraints(constraints);
-            if (!newStream) {
-                console.warn(`Can't create media stream with: ${JSON.stringify(constraints || {})}`);
-                return;
-            }
-            // Add previous local audio track
-            if (constraints && !constraints.audio) {
-                const localVideoStream = (_b = session.sessionDescriptionHandler) === null || _b === void 0 ? void 0 : _b.localMediaStream;
-                const localAudioTrack = localVideoStream.getTracks().find(track => track.kind === 'audio');
-                if (localAudioTrack) {
-                    newStream.addTrack(localAudioTrack);
-                }
-            }
-            const videoTrack = newStream.getVideoTracks()[0];
-            if (videoTrack) {
-                videoSender.replaceTrack(videoTrack);
-            }
-            this.setLocalMediaStream(this.getSipSessionId(session), newStream);
-            return newStream;
-        });
-    }
-    downgradeToAudio(session) {
-        // Release local video stream when downgrading to audio
-        const sessionDescriptionHandler = session.sessionDescriptionHandler;
-        const localStream = sessionDescriptionHandler === null || sessionDescriptionHandler === void 0 ? void 0 : sessionDescriptionHandler.localMediaStream;
-        const pc = sessionDescriptionHandler === null || sessionDescriptionHandler === void 0 ? void 0 : sessionDescriptionHandler.peerConnection;
-        const videoTracks = localStream.getVideoTracks();
-        // Remove video senders
-        if (pc === null || pc === void 0 ? void 0 : pc.getSenders) {
-            pc.getSenders().filter((sender) => sender.track && sender.track.kind === 'video').forEach((videoSender) => {
-                videoSender.replaceTrack(null);
-            });
-        }
-        videoTracks.forEach((videoTrack) => {
-            videoTrack.enabled = false;
-            videoTrack.stop();
-            localStream.removeTrack(videoTrack);
-        });
-    }
-    getStreamFromConstraints(constraints_1) {
-        return __awaiter(this, arguments, void 0, function* (constraints, conference = false) {
-            const video = constraints && constraints.video;
-            const { constraints: newConstraints, } = this.getMediaConfiguration(video, conference, constraints);
-            let newStream = null;
-            try {
-                newStream = yield (0, WazoSessionDescriptionHandler_1.wazoMediaStreamFactory)(newConstraints);
-            }
-            catch (e) { // Nothing to do when the user cancel the screensharing prompt
-            }
-            if (!newStream) {
-                return null;
-            }
-            newStream.local = true;
-            return newStream;
-        });
-    }
     getHeldSession(sessionId) {
         return this.heldSessions[sessionId];
     }
     isCallHeld(session) {
-        return this.getSipSessionId(session) in this.heldSessions;
+        return (0, sdp_1.getSipSessionId)(session) in this.heldSessions;
     }
     isVideoRemotelyHeld(sessionId) {
-        const pc = this.getPeerConnection(sessionId);
-        const sdp = pc && pc.remoteDescription ? pc.remoteDescription.sdp : null;
-        if (!sdp) {
-            return false;
-        }
-        const videoDirection = (0, sdp_1.getVideoDirection)(sdp);
-        return videoDirection === 'sendonly';
+        return (0, webrtc_1.isVideoRemotelyHeld)(this.heldSessions[sessionId]);
     }
     sendDTMF(session, tone) {
         if (!session.sessionDescriptionHandler) {
             return false;
         }
         logger.info('Sending DTMF', {
-            id: this.getSipSessionId(session),
+            id: (0, sdp_1.getSipSessionId)(session),
             tone,
         });
         return session.sessionDescriptionHandler.sendDtmf(tone);
     }
     message(destination, message) {
-        const uri = this._makeURI(destination);
+        const uri = (0, sdp_1.makeURI)(destination, this.config.host);
         if (!this.userAgent || !uri) {
             logger.warn('Null value on message', { uri, userAgent: this.userAgent });
             return;
@@ -925,7 +781,7 @@ class WebRTCClient extends Emitter_1.default {
     transfer(session, target) {
         this.hold(session);
         logger.info('Transfering a session', {
-            id: this.getSipSessionId(session),
+            id: (0, sdp_1.getSipSessionId)(session),
             target,
         });
         const options = {
@@ -936,7 +792,7 @@ class WebRTCClient extends Emitter_1.default {
             },
         };
         setTimeout(() => {
-            const uri = this._makeURI(target);
+            const uri = (0, sdp_1.makeURI)(target, this.config.host);
             if (!uri) {
                 logger.warn('transfer timeout: null URI');
                 return;
@@ -948,21 +804,21 @@ class WebRTCClient extends Emitter_1.default {
     atxfer(session) {
         this.hold(session);
         logger.info('webrtc transfer started', {
-            id: this.getSipSessionId(session),
+            id: (0, sdp_1.getSipSessionId)(session),
         });
         const result = {
             newSession: null,
             init: (target) => __awaiter(this, void 0, void 0, function* () {
                 logger.info('webrtc transfer initialized', {
-                    id: this.getSipSessionId(session),
+                    id: (0, sdp_1.getSipSessionId)(session),
                     target,
                 });
                 result.newSession = yield this.call(target);
             }),
             complete: () => {
                 logger.info('webrtc transfer completed', {
-                    id: this.getSipSessionId(session),
-                    referId: this.getSipSessionId(result.newSession),
+                    id: (0, sdp_1.getSipSessionId)(session),
+                    referId: (0, sdp_1.getSipSessionId)(result.newSession),
                 });
                 session.refer(result.newSession);
             },
@@ -1004,7 +860,7 @@ class WebRTCClient extends Emitter_1.default {
             return;
         }
         const core = (_a = this.userAgent) === null || _a === void 0 ? void 0 : _a.userAgentCore;
-        const fromURI = this._makeURI(this.config.authorizationUser || '');
+        const fromURI = (0, sdp_1.makeURI)(this.config.authorizationUser || '', this.config.host);
         const toURI = new uri_1.URI('sip', '', this.config.host);
         if (fromURI) {
             const message = core === null || core === void 0 ? void 0 : core.makeOutgoingRequestMessage('OPTIONS', toURI, fromURI, toURI, {});
@@ -1022,9 +878,6 @@ class WebRTCClient extends Emitter_1.default {
     getContactIdentifier() {
         return this.userAgent ? `${this.userAgent.configuration.contactName}/${this.userAgent.contact.uri}` : null;
     }
-    isFirefox() {
-        return this._isWeb() && navigator.userAgent.toLowerCase().indexOf('firefox') > -1;
-    }
     changeAudioOutputVolume(volume) {
         logger.info('Changing audio output volume', {
             volume,
@@ -1132,12 +985,12 @@ class WebRTCClient extends Emitter_1.default {
             } });
     }
     changeSessionVideoInputDevice(id, session) {
-        if (!this.sessionWantsToDoVideo(session)) {
+        if (!(0, webrtc_1.sessionWantsToDoVideo)(session)) {
             return Promise.resolve();
         }
         const sdh = session.sessionDescriptionHandler;
         const pc = sdh.peerConnection;
-        const sessionId = this.getSipSessionId(session);
+        const sessionId = (0, sdp_1.getSipSessionId)(session);
         const localStream = this.getLocalStream(sessionId);
         logger.info('changing video input device', {
             id,
@@ -1174,7 +1027,7 @@ class WebRTCClient extends Emitter_1.default {
             }
             // let's update the local stream
             this.eventEmitter.emit('onVideoInputChange', stream);
-            this.setLocalMediaStream(sessionId, stream);
+            (0, webrtc_1.setLocalMediaStream)(this.sipSessions[sessionId], stream);
             return stream;
         }));
     }
@@ -1195,12 +1048,12 @@ class WebRTCClient extends Emitter_1.default {
             return Promise.resolve();
         }
         const wasMuted = this.isAudioMuted(sipSession);
-        const shouldDoVideo = newConstraints ? newConstraints.video : this.sessionWantsToDoVideo(sipSession);
+        const shouldDoVideo = newConstraints ? newConstraints.video : (0, webrtc_1.sessionWantsToDoVideo)(sipSession);
         const shouldDoScreenSharing = newConstraints && newConstraints.screen;
         const desktop = newConstraints && newConstraints.desktop;
         logger.info('Sending reinvite', {
             clientId: this.clientId,
-            id: this.getSipSessionId(sipSession),
+            id: (0, sdp_1.getSipSessionId)(sipSession),
             newConstraints,
             conference,
             audioOnly,
@@ -1216,7 +1069,7 @@ class WebRTCClient extends Emitter_1.default {
         }
         sipSession.sessionDescriptionHandlerOptionsReInvite = Object.assign(Object.assign({}, sipSession.sessionDescriptionHandlerOptionsReInvite), { conference,
             audioOnly });
-        const { constraints, } = this.getMediaConfiguration(shouldDoVideo, conference, newConstraints);
+        const { constraints, } = (0, webrtc_1.getMediaConfiguration)(shouldDoVideo, conference, newConstraints);
         return sipSession.invite({
             requestDelegate: {
                 onAccept: (response) => {
@@ -1232,11 +1085,11 @@ class WebRTCClient extends Emitter_1.default {
                         sipSession.incomingInviteRequest.message.body = response.message.body;
                     }
                     logger.info('on re-INVITE accepted', {
-                        id: this.getSipSessionId(sipSession),
+                        id: (0, sdp_1.getSipSessionId)(sipSession),
                         wasMuted,
                         shouldDoScreenSharing,
                     });
-                    this.updateRemoteStream(this.getSipSessionId(sipSession), false);
+                    (0, webrtc_1.updateRemoteStream)(sipSession, false);
                     if (wasMuted) {
                         this.mute(sipSession);
                     }
@@ -1247,7 +1100,7 @@ class WebRTCClient extends Emitter_1.default {
                     return this.eventEmitter.emit(ON_REINVITE, sipSession, response);
                 },
             },
-            sessionDescriptionHandlerModifiers: [exports.replaceLocalIpModifier],
+            sessionDescriptionHandlerModifiers: [sdp_1.replaceLocalIpModifier],
             requestOptions: {
                 extraHeaders: [`Subject: ${shouldDoScreenSharing ? 'screenshare' : 'upgrade-video'}`],
             },
@@ -1271,107 +1124,57 @@ class WebRTCClient extends Emitter_1.default {
         });
     }
     getPeerConnection(sessionId) {
-        const sipSession = this.sipSessions[sessionId];
-        if (!sipSession) {
-            return null;
-        }
-        return sipSession.sessionDescriptionHandler ? sipSession.sessionDescriptionHandler.peerConnection : null;
+        return (0, webrtc_1.getPeerConnection)(this.sipSessions[sessionId]);
     }
     // Local streams
     getLocalStream(sessionId) {
-        var _a;
-        const sipSession = this.sipSessions[sessionId];
-        return ((_a = sipSession === null || sipSession === void 0 ? void 0 : sipSession.sessionDescriptionHandler) === null || _a === void 0 ? void 0 : _a.localMediaStream) || null;
+        return (0, webrtc_1.getLocalStream)(this.sipSessions[sessionId]);
     }
     getLocalTracks(sessionId) {
-        const localStream = this.getLocalStream(sessionId);
-        if (!localStream) {
-            return [];
-        }
-        return localStream.getTracks();
+        return (0, webrtc_1.getLocalTracks)(this.sipSessions[sessionId]);
     }
     hasLocalVideo(sessionId) {
-        return this.getLocalTracks(sessionId).some(this._isVideoTrack);
+        return (0, webrtc_1.hasLocalVideo)(this.sipSessions[sessionId]);
     }
     // Check if we have at least one video track, even muted or not live
     hasALocalVideoTrack(sessionId) {
-        return this.getLocalTracks(sessionId).some(track => track.kind === 'video');
+        return (0, webrtc_1.hasALocalVideoTrack)(this.sipSessions[sessionId]);
     }
     getLocalVideoStream(sessionId) {
-        return this.hasLocalVideo(sessionId) ? this.getLocalStream(sessionId) : null;
+        return (0, webrtc_1.getLocalVideoStream)(this.sipSessions[sessionId]);
     }
     // Remote streams
     getRemoteStream(sessionId) {
-        var _a;
-        const sipSession = this.sipSessions[sessionId];
-        return ((_a = sipSession === null || sipSession === void 0 ? void 0 : sipSession.sessionDescriptionHandler) === null || _a === void 0 ? void 0 : _a.remoteMediaStream) || null;
+        return (0, webrtc_1.getRemoteStream)(this.sipSessions[sessionId]);
     }
     getRemoteTracks(sessionId) {
-        const remoteStream = this.getRemoteStream(sessionId);
-        if (!remoteStream) {
-            return [];
-        }
-        return remoteStream.getTracks();
+        return (0, webrtc_1.getRemoteTracks)(this.sipSessions[sessionId]);
     }
     hasRemoteVideo(sessionId) {
-        return this.getRemoteTracks(sessionId).some(this._isVideoTrack);
+        return (0, webrtc_1.hasRemoteVideo)(this.sipSessions[sessionId]);
     }
     // Check if we have at least one video track, even muted or not live
     hasARemoteVideoTrack(sessionId) {
-        return this.getRemoteTracks(sessionId).some(track => track.kind === 'video');
+        return (0, webrtc_1.hasARemoteVideoTrack)(this.sipSessions[sessionId]);
     }
     getRemoteVideoStream(sessionId) {
-        if (this.isVideoRemotelyHeld(sessionId)) {
-            return null;
-        }
-        return this.hasRemoteVideo(sessionId) ? this.getRemoteStream(sessionId) : null;
+        return (0, webrtc_1.getRemoteVideoStream)(this.sipSessions[sessionId]);
     }
     //  Useful in a react-native environment when remoteMediaStream is not updated
     getRemoteVideoStreamFromPc(sessionId) {
-        const pc = this.getPeerConnection(sessionId);
-        if (!pc) {
-            return null;
-        }
-        return pc.getRemoteStreams().find((stream) => !!stream.getVideoTracks().length);
+        return (0, webrtc_1.getRemoteVideoStreamFromPc)(this.sipSessions[sessionId]);
     }
     hasVideo(sessionId) {
-        return this.hasLocalVideo(sessionId) || this.hasRemoteVideo(sessionId);
+        return (0, webrtc_1.hasVideoTrack)(this.sipSessions[sessionId]);
     }
     hasAVideoTrack(sessionId) {
-        return this.hasALocalVideoTrack(sessionId) || this.hasARemoteVideoTrack(sessionId);
-    }
-    getSipSessionId(sipSession) {
-        if (!sipSession) {
-            return '';
-        }
-        const message = sipSession.message;
-        if (message === null || message === void 0 ? void 0 : message.callId) {
-            return message.callId.substring(0, SIP_ID_LENGTH);
-        }
-        // For Inviter
-        // @ts-ignore: private
-        if (sipSession instanceof api_1.Inviter && sipSession.outgoingRequestMessage) {
-            // @ts-ignore: private
-            return sipSession.outgoingRequestMessage.callId.substring(0, SIP_ID_LENGTH);
-        }
-        // For Invitation
-        return (sipSession.id || '').substring(0, SIP_ID_LENGTH);
+        return (0, webrtc_1.hasAVideoTrack)(this.sipSessions[sessionId]);
     }
     waitForRegister() {
         return __awaiter(this, void 0, void 0, function* () {
             return new Promise(resolve => this.on(REGISTERED, resolve));
         });
     }
-    // eslint-disable-next-line no-unused-vars
-    sessionWantsToDoVideo(session) {
-        if (!session) {
-            return false;
-        }
-        const { body, } = session.request || session;
-        // Sometimes with InviteClientContext the body is in the body attribute ...
-        const sdp = typeof body === 'object' && body ? body.body : body;
-        return (0, sdp_1.hasAnActiveVideo)(sdp);
-    }
     hasHeartbeat() {
         return this.heartbeat.hasHeartbeat;
     }
@@ -1397,7 +1200,7 @@ class WebRTCClient extends Emitter_1.default {
     }
     onCallEnded(session) {
         this._cleanupMedia(session);
-        delete this.sipSessions[this.getSipSessionId(session)];
+        delete this.sipSessions[(0, sdp_1.getSipSessionId)(session)];
         this._stopSendingStats(session);
         this.stopNetworkMonitoring(session);
     }
@@ -1410,31 +1213,16 @@ class WebRTCClient extends Emitter_1.default {
         this.userAgent.attemptReconnection();
     }
     storeSipSession(session) {
-        const id = this.getSipSessionId(session);
+        const id = (0, sdp_1.getSipSessionId)(session);
         logger.info('storing sip session', { id, clientId: this.clientId });
         this.sipSessions[id] = session;
     }
     getSipSession(id) {
-        return id in this.sipSessions ? this.sipSessions[id.substring(0, SIP_ID_LENGTH)] : null;
+        return id in this.sipSessions ? this.sipSessions[id.substring(0, sdp_1.SIP_ID_LENGTH)] : null;
     }
     getSipSessionIds() {
         return Object.keys(this.sipSessions);
     }
-    setLocalMediaStream(sipSessionId, newStream) {
-        const sipSession = this.sipSessions[sipSessionId];
-        if (!sipSession) {
-            return;
-        }
-        logger.info('setting local media stream', {
-            sipSessionId,
-            tracks: newStream ? newStream.getTracks() : null,
-        });
-        if (sipSession.sessionDescriptionHandler) {
-            // eslint-disable-next-line no-underscore-dangle
-            // @ts-ignore: protected
-            sipSession.sessionDescriptionHandler._localMediaStream = newStream;
-        }
-    }
     updateLocalStream(sipSessionId, newStream) {
         return __awaiter(this, void 0, void 0, function* () {
             const sipSession = this.sipSessions[sipSessionId];
@@ -1448,74 +1236,13 @@ class WebRTCClient extends Emitter_1.default {
                 tracks: newStream.getTracks(),
             });
             if (oldStream) {
-                this._cleanupStream(oldStream);
+                (0, webrtc_1.cleanupStream)(oldStream);
             }
-            this.setLocalMediaStream(sipSessionId, newStream);
+            (0, webrtc_1.setLocalMediaStream)(sipSession, newStream);
             // Update the local video element
             yield this._setupMedias(sipSession, newStream);
         });
     }
-    updateRemoteStream(sessionId, allTracks = true) {
-        const remoteStream = this.getRemoteStream(sessionId);
-        const pc = this.getPeerConnection(sessionId);
-        logger.info('Updating remote stream', {
-            sessionId,
-            tracks: remoteStream ? remoteStream.getTracks() : null,
-            receiverTracks: pc && pc.getReceivers ? pc.getReceivers().map((receiver) => receiver.track) : null,
-        });
-        if (!pc || !remoteStream) {
-            return;
-        }
-        remoteStream.getTracks().forEach(track => {
-            if (allTracks || track.kind === 'video') {
-                remoteStream.removeTrack(track);
-            }
-        });
-        if (pc.getReceivers) {
-            pc.getReceivers().forEach((receiver) => {
-                if (allTracks || receiver.track.kind === 'video') {
-                    remoteStream.addTrack(receiver.track);
-                }
-            });
-        }
-    }
-    getMediaConfiguration(enableVideo, conference = false, constraints = null) {
-        const screenSharing = constraints && 'screen' in constraints ? constraints.screen : false;
-        const isDesktop = constraints && 'desktop' in constraints ? constraints.desktop : false;
-        const withAudio = constraints && 'audio' in constraints ? constraints.audio : true;
-        const mandatoryVideo = constraints && typeof constraints.video === 'object' ? constraints.video.mandatory : {};
-        logger.info('Retrieving media a configuration', {
-            enableVideo,
-            screenSharing,
-            isDesktop,
-            withAudio,
-            constraints,
-        });
-        return {
-            constraints: {
-                // Exact constraint are not supported with `getDisplayMedia` and we must have a video=false in desktop screenshare
-                audio: screenSharing ? !isDesktop : withAudio ? this._getAudioConstraints() : false,
-                video: screenSharing ? isDesktop ? {
-                    mandatory: Object.assign({ chromeMediaSource: 'desktop' }, (mandatoryVideo || {})),
-                } : {
-                    cursor: 'always',
-                } : this._getVideoConstraints(enableVideo),
-                screen: screenSharing,
-                desktop: isDesktop,
-                conference,
-            },
-            enableVideo,
-            conference,
-            offerOptions: {
-                OfferToReceiveAudio: this._hasAudio(),
-                OfferToReceiveVideo: enableVideo,
-                mandatory: {
-                    OfferToReceiveAudio: this._hasAudio(),
-                    OfferToReceiveVideo: enableVideo,
-                },
-            },
-        };
-    }
     isConference(sessionId) {
         return sessionId in this.conferences;
     }
@@ -1596,27 +1323,14 @@ class WebRTCClient extends Emitter_1.default {
             }
         });
     }
-    _isWeb() {
-        return typeof window === 'object' && typeof document === 'object';
-    }
-    _isVideoTrack(track) {
-        return track.kind === 'video' && track.readyState === 'live';
-    }
     _hasAudio() {
         return this.hasAudio;
     }
     _getAudioConstraints() {
-        var _a, _b;
-        // @ts-ignore: media constraints
-        return ((_b = (_a = this.audio) === null || _a === void 0 ? void 0 : _a.deviceId) === null || _b === void 0 ? void 0 : _b.exact) ? this.audio : true;
+        return (0, webrtc_1.getAudioConstraints)(this.audio);
     }
     _getVideoConstraints(video = false) {
-        var _a, _b;
-        if (!video) {
-            return false;
-        }
-        // @ts-ignore: media constraints
-        return ((_b = (_a = this.video) === null || _a === void 0 ? void 0 : _a.deviceId) === null || _b === void 0 ? void 0 : _b.exact) ? this.video : true;
+        return (0, webrtc_1.getVideoConstraints)(video && this.video);
     }
     _connectIfNeeded() {
         logger.info('connect if needed, checking', { connected: this.isConnected(), clientId: this.clientId });
@@ -1661,80 +1375,16 @@ class WebRTCClient extends Emitter_1.default {
         client.setRefreshToken(session.refreshToken);
         return client.confd.getUserLineSipFromToken(session.uuid).then(sipLine => (Object.assign({ authorizationUser: sipLine.username, password: sipLine.secret, uri: `${sipLine.username}@${config.host}` }, config)));
     }
-    _createUaOptions(uaOptionsOverrides = {}) {
-        let { host, } = this.config;
-        let port = this.config.port || 443;
-        if (this.config.websocketSip) {
-            const webSocketSip = this.config.websocketSip.split(':');
-            [host] = webSocketSip;
-            port = Number(webSocketSip[1]);
-        }
-        const { userUuid } = this.config;
-        const uaOptions = {
-            noAnswerTimeout: NO_ANSWER_TIMEOUT,
-            authorizationUsername: this.config.authorizationUser,
-            authorizationPassword: this.config.password,
-            displayName: this.config.displayName,
-            hackIpInContact: true,
-            contactParams: {
-                transport: 'wss',
-            },
-            logBuiltinEnabled: this.config.log ? this.config.log.builtinEnabled : null,
-            logLevel: this.config.log ? this.config.log.logLevel : null,
-            logConnector: this.config.log ? this.config.log.connector : null,
-            uri: this._makeURI(this.config.authorizationUser || ''),
-            userAgentString: this.config.userAgentString || 'wazo-sdk',
-            reconnectionAttempts: 10000,
-            reconnectionDelay: 5,
-            sessionDescriptionHandlerFactory: (session, options = {}) => {
-                const uaLogger = session.userAgent.getLogger('sip.WazoSessionDescriptionHandler');
-                const isWeb = this._isWeb();
-                const iceGatheringTimeout = 'peerConnectionOptions' in options ? options.peerConnectionOptions.iceGatheringTimeout || DEFAULT_ICE_TIMEOUT : DEFAULT_ICE_TIMEOUT;
-                const sdhOptions = Object.assign(Object.assign({}, options), { iceGatheringTimeout, peerConnectionConfiguration: Object.assign(Object.assign({}, (0, peer_connection_configuration_default_1.defaultPeerConnectionConfiguration)()), (options.peerConnectionConfiguration || {})) });
-                return new WazoSessionDescriptionHandler_1.default(uaLogger, WazoSessionDescriptionHandler_1.wazoMediaStreamFactory, sdhOptions, isWeb, session);
-            },
-            transportOptions: {
-                traceSip: (uaOptionsOverrides === null || uaOptionsOverrides === void 0 ? void 0 : uaOptionsOverrides.traceSip) || false,
-                wsServers: `wss://${host}:${port}/api/asterisk/ws${userUuid ? `?userUuid=${userUuid}` : ''}`,
-            },
-            sessionDescriptionHandlerFactoryOptions: {
-                modifiers: [exports.replaceLocalIpModifier],
-                alwaysAcquireMediaFirst: this.isFirefox(),
-                constraints: {
-                    audio: this._getAudioConstraints(),
-                    video: this._getVideoConstraints(),
-                },
-                peerConnectionOptions: {
-                    iceCheckingTimeout: this.config.iceCheckingTimeout || 1000,
-                    iceGatheringTimeout: this.config.iceCheckingTimeout || 1000,
-                    rtcConfiguration: Object.assign(Object.assign({ rtcpMuxPolicy: 'require', iceServers: WebRTCClient.getIceServers(this.config.host) }, this._getRtcOptions()), ((uaOptionsOverrides === null || uaOptionsOverrides === void 0 ? void 0 : uaOptionsOverrides.peerConnectionOptions) || {})),
-                },
-                // Configuration used in SDH to create the PeerConnection
-                peerConnectionConfiguration: Object.assign({ rtcpMuxPolicy: 'require', iceServers: WebRTCClient.getIceServers(this.config.host) }, ((uaOptionsOverrides === null || uaOptionsOverrides === void 0 ? void 0 : uaOptionsOverrides.peerConnectionOptions) || {})),
-            },
-        };
-        uaOptionsOverrides === null || uaOptionsOverrides === void 0 ? true : delete uaOptionsOverrides.traceSip;
-        return Object.assign(Object.assign({}, uaOptions), uaOptionsOverrides);
-    }
-    // eslint-disable-next-line no-unused-vars
-    _getRtcOptions() {
-        return {
-            mandatory: {
-                OfferToReceiveAudio: this._hasAudio(),
-                OfferToReceiveVideo: false,
-            },
-        };
-    }
     // Invitation and Inviter extends Session
     _setupSession(session) {
-        const sipSessionId = this.getSipSessionId(session);
+        const sipSessionId = (0, sdp_1.getSipSessionId)(session);
         // When receiving an Invitation, the delegate is not defined.
         if (!session.delegate) {
             session.delegate = {};
         }
         session.delegate.onSessionDescriptionHandler = (sdh) => {
             sdh.on('error', (e) => {
-                this.eventEmitter.emit(ON_ERROR, e);
+                this.eventEmitter.emit(ON_ERROR, e, session);
             });
             sdh.peerConnectionDelegate = {
                 onicecandidateerror: (error) => {
@@ -1783,7 +1433,7 @@ class WebRTCClient extends Emitter_1.default {
                 // @ts-ignore: private
                 session.outgoingInviteRequest.message.body.body = request.body;
             }
-            this.updateRemoteStream(sipSessionId, false);
+            (0, webrtc_1.updateRemoteStream)(this.sipSessions[sipSessionId], false);
             yield this._setupMedias(session);
             return this.eventEmitter.emit(ON_REINVITE, session, request, updatedCalleeName, updatedNumber, hadRemoteVideo);
         });
@@ -1791,8 +1441,8 @@ class WebRTCClient extends Emitter_1.default {
     _onProgress(session_1) {
         return __awaiter(this, arguments, void 0, function* (session, early = false) {
             yield this._setupMedias(session);
-            const sessionId = this.getSipSessionId(session).substr(0, 20);
-            this.updateRemoteStream(sessionId);
+            const sessionId = (0, sdp_1.getSipSessionId)(session).substr(0, 20);
+            (0, webrtc_1.updateRemoteStream)(session);
             logger.info('progress received', {
                 sessionId,
                 early,
@@ -1802,7 +1452,6 @@ class WebRTCClient extends Emitter_1.default {
     }
     _onAccepted(session_1, sessionDialog_1) {
         return __awaiter(this, arguments, void 0, function* (session, sessionDialog, withEvent = true, initAllTracks = true) {
-            var _a;
             logger.info('on call accepted', {
                 id: session.id,
                 clientId: this.clientId,
@@ -1810,21 +1459,14 @@ class WebRTCClient extends Emitter_1.default {
             });
             this.storeSipSession(session);
             yield this._setupMedias(session);
-            this.updateRemoteStream(this.getSipSessionId(session), initAllTracks);
-            const pc = (_a = session.sessionDescriptionHandler) === null || _a === void 0 ? void 0 : _a.peerConnection;
+            (0, webrtc_1.updateRemoteStream)(session, initAllTracks);
+            const pc = (0, webrtc_1.getPeerConnection)(session);
             const onTrack = (event) => {
-                const isAudioOnly = this._isAudioOnly(session);
-                const { kind, label, readyState, id, muted, } = event.track;
-                logger.info('on track event', {
-                    isAudioOnly,
-                    kind,
-                    label,
-                    readyState,
-                    id,
-                    muted,
-                });
+                const audioOnly = (0, webrtc_1.isAudioOnly)(session);
+                const { kind, label, readyState, id, muted } = event.track;
+                logger.info('on track event', { audioOnly, kind, label, readyState, id, muted });
                 // Stop video track in audio only mode
-                if (isAudioOnly && kind === 'video') {
+                if (audioOnly && kind === 'video') {
                     event.track.stop();
                 }
                 this.eventEmitter.emit(ON_TRACK, session, event);
@@ -1850,17 +1492,14 @@ class WebRTCClient extends Emitter_1.default {
             this._startSendingStats(session);
         });
     }
-    _isAudioOnly(session) {
-        return Boolean(session.sessionDescriptionHandlerModifiersReInvite.find(modifier => modifier === modifiers_1.stripVideo));
-    }
     _setupMedias(session_1) {
         return __awaiter(this, arguments, void 0, function* (session, newStream = null) {
-            if (!this._isWeb()) {
+            if (!(0, webrtc_1.isWeb)()) {
                 logger.info('Setup media on mobile, no need to setup html element, bailing');
                 return;
             }
             // Safari hack, because you cannot call .play() from a non user action
-            const sessionId = this.getSipSessionId(session);
+            const sessionId = (0, sdp_1.getSipSessionId)(session);
             const isConference = this.isConference(sessionId);
             if (sessionId in this.audioElements) {
                 logger.info('html element already exists for session', {
@@ -1868,7 +1507,7 @@ class WebRTCClient extends Emitter_1.default {
                 });
                 return;
             }
-            if (this._hasAudio() && this._isWeb() && !(sessionId in this.audioElements)) {
+            if (this._hasAudio() && (0, webrtc_1.isWeb)() && !(sessionId in this.audioElements)) {
                 yield this.createAudioElementFor(sessionId);
             }
             const audioElement = this.audioElements[sessionId];
@@ -1906,10 +1545,10 @@ class WebRTCClient extends Emitter_1.default {
         });
     }
     _cleanupMedia(session) {
-        const sessionId = this.getSipSessionId(session);
+        const sessionId = (0, sdp_1.getSipSessionId)(session);
         const localStream = this.getLocalStream(sessionId);
         if (localStream) {
-            this._cleanupStream(localStream);
+            (0, webrtc_1.cleanupStream)(localStream);
         }
         const cleanLocalElement = (id) => {
             const element = this.audioElements[id];
@@ -1923,80 +1562,15 @@ class WebRTCClient extends Emitter_1.default {
             element.srcObject = null;
             delete this.audioElements[id];
         };
-        if (this._hasAudio() && this._isWeb()) {
+        if (this._hasAudio() && (0, webrtc_1.isWeb)()) {
             if (session) {
-                cleanLocalElement(this.getSipSessionId(session));
+                cleanLocalElement((0, sdp_1.getSipSessionId)(session));
             }
             else {
                 Object.keys(this.audioElements).forEach(id => cleanLocalElement(id));
             }
         }
     }
-    _cleanupStream(stream) {
-        stream.getTracks().filter(track => track.enabled).forEach(track => track.stop());
-    }
-    _toggleAudio(session, muteAudio) {
-        const sdh = session.sessionDescriptionHandler;
-        const pc = ((sdh === null || sdh === void 0 ? void 0 : sdh.peerConnection) || null);
-        if (!pc) {
-            return;
-        }
-        if (pc.getSenders) {
-            pc.getSenders().forEach((sender) => {
-                if (sender && sender.track && sender.track.kind === 'audio') {
-                    // eslint-disable-next-line
-                    sender.track.enabled = !muteAudio;
-                }
-            });
-        }
-        else {
-            pc.getLocalStreams().forEach((stream) => {
-                stream.getAudioTracks().forEach(track => {
-                    // eslint-disable-next-line
-                    track.enabled = !muteAudio;
-                });
-            });
-        }
-    }
-    _toggleVideo(session, muteCamera) {
-        const sdh = session.sessionDescriptionHandler;
-        const pc = sdh === null || sdh === void 0 ? void 0 : sdh.peerConnection;
-        if (pc === null || pc === void 0 ? void 0 : pc.getSenders) {
-            pc.getSenders().forEach((sender) => {
-                if (sender && sender.track && sender.track.kind === 'video') {
-                    // eslint-disable-next-line
-                    sender.track.enabled = !muteCamera;
-                }
-            });
-        }
-        else {
-            pc.getLocalStreams().forEach((stream) => {
-                stream.getVideoTracks().forEach(track => {
-                    // eslint-disable-next-line
-                    track.enabled = !muteCamera;
-                });
-            });
-        }
-    }
-    /**
-     * @param pc RTCPeerConnection
-     */
-    _getRemoteStream(pc) {
-        let remoteStream = null;
-        if (pc && pc.getReceivers) {
-            remoteStream = typeof global !== 'undefined' && global.window && global.window.MediaStream ? new global.window.MediaStream() : new window.MediaStream();
-            pc.getReceivers().forEach((receiver) => {
-                const { track, } = receiver;
-                if (track && remoteStream) {
-                    remoteStream.addTrack(track);
-                }
-            });
-        }
-        else if (pc) {
-            [remoteStream] = pc.getRemoteStreams();
-        }
-        return remoteStream;
-    }
     _cleanupRegister() {
         if (this.registerer) {
             // @ts-ignore: removeAllListeners does not exist
@@ -2010,7 +1584,7 @@ class WebRTCClient extends Emitter_1.default {
         if (!pc) {
             return;
         }
-        const sessionId = this.getSipSessionId(session);
+        const sessionId = (0, sdp_1.getSipSessionId)(session);
         (0, getstats_1.default)(pc, (result) => {
             const { results, internal, nomore } = result, stats = __rest(result, ["results", "internal", "nomore"]);
             this.statsIntervals[sessionId] = nomore;
@@ -2018,7 +1592,7 @@ class WebRTCClient extends Emitter_1.default {
         }, SEND_STATS_DELAY);
     }
     _stopSendingStats(session) {
-        const sessionId = this.getSipSessionId(session);
+        const sessionId = (0, sdp_1.getSipSessionId)(session);
         logger.trace('Check for stopping stats', {
             sessionId,
             ids: Object.keys(this.statsIntervals),
@@ -2031,9 +1605,6 @@ class WebRTCClient extends Emitter_1.default {
             delete this.statsIntervals[sessionId];
         }
     }
-    _makeURI(target) {
-        return user_agent_1.UserAgent.makeURI(`sip:${target}@${this.config.host}`);
-    }
     _disconnectTransport() {
         return __awaiter(this, arguments, void 0, function* (force = false) {
             var _a;
@@ -2059,96 +1630,13 @@ class WebRTCClient extends Emitter_1.default {
     }
     _fetchNetworkStats(sessionId) {
         return __awaiter(this, void 0, void 0, function* () {
-            const session = this.getSipSession(sessionId);
-            const stats = session ? yield this.getStats(session) : null;
-            if (!stats || !(sessionId in this.sessionNetworkStats)) {
+            const session = this.sipSessions[sessionId];
+            if (!(sessionId in this.sessionNetworkStats)) {
                 return Promise.resolve(null);
             }
-            const networkStats = {
-                audioBytesSent: 0,
-                videoBytesSent: 0,
-                videoBytesReceived: 0,
-                audioBytesReceived: 0,
-            };
             const nbStats = this.sessionNetworkStats[sessionId].length;
             const lastNetworkStats = this.sessionNetworkStats[sessionId][nbStats - 1];
-            const lastAudioSent = lastNetworkStats ? lastNetworkStats.audioBytesSent : 0;
-            const lastAudioContentSent = lastNetworkStats ? lastNetworkStats.audioContentSent : 0;
-            const lastVideoSent = lastNetworkStats ? lastNetworkStats.videoBytesSent : 0;
-            const lastAudioReceived = lastNetworkStats ? lastNetworkStats.audioBytesReceived : 0;
-            const lastAudioContentReceived = lastNetworkStats ? lastNetworkStats.audioContentReceived : 0;
-            const lastVideoReceived = lastNetworkStats ? lastNetworkStats.videoBytesReceived : 0;
-            const lastTransportSent = lastNetworkStats ? lastNetworkStats.transportSent : 0;
-            const lastTransportReceived = lastNetworkStats ? lastNetworkStats.transportReceived : 0;
-            let audioBytesSent = 0; // content + header
-            let audioBytesReceived = 0; // content + header
-            let audioContentSent = 0; // useful to detect blank call
-            let audioContentReceived = 0;
-            let videoBytesSent = 0;
-            let videoBytesReceived = 0;
-            let transportSent = 0;
-            let transportReceived = 0;
-            let packetsLost = 0;
-            stats.forEach(report => {
-                if (report.type === 'outbound-rtp' && report.kind === 'audio') {
-                    audioBytesSent += Number(report.bytesSent) + Number(report.headerBytesSent);
-                    audioContentSent += Number(report.bytesSent);
-                }
-                if (report.type === 'remote-outbound-rtp' && report.kind === 'audio') {
-                    audioBytesSent += Number(report.bytesSent);
-                }
-                if (report.type === 'outbound-rtp' && report.kind === 'video') {
-                    videoBytesSent += Number(report.bytesSent) + Number(report.headerBytesSent);
-                }
-                if (report.type === 'inbound-rtp' && report.kind === 'audio') {
-                    packetsLost += Number(report.packetsLost);
-                    networkStats.packetsReceived = Number(report.packetsReceived);
-                    audioBytesReceived += Number(report.bytesReceived) + Number(report.headerBytesReceived);
-                    audioContentReceived += Number(report.bytesReceived);
-                }
-                if (report.type === 'inbound-rtp' && report.kind === 'video') {
-                    videoBytesReceived += Number(report.bytesReceived) + Number(report.headerBytesReceived);
-                }
-                if (report.type === 'outbound-rtp' && report.kind === 'video') {
-                    if ('framesPerSecond' in report) {
-                        networkStats.framesPerSecond = Number(report.framesPerSecond);
-                    }
-                    if ('framerateMean' in report) {
-                        // framerateMean is only available in FF
-                        networkStats.framesPerSecond = Math.round(report.framerateMean);
-                    }
-                }
-                if (report.type === 'remote-inbound-rtp' && report.kind === 'audio') {
-                    packetsLost += Number(report.packetsLost);
-                    networkStats.roundTripTime = Number(report.roundTripTime);
-                    networkStats.jitter = Number(report.jitter);
-                }
-                if (report.type === 'remote-inbound-rtp' && report.kind === 'video') {
-                    packetsLost += Number(report.packetsLost);
-                }
-                if (report.type === 'transport') {
-                    transportSent += Number(report.bytesSent);
-                    transportReceived += Number(report.bytesReceived);
-                }
-            });
-            networkStats.packetsLost = packetsLost;
-            networkStats.totalAudioBytesSent = audioBytesSent;
-            networkStats.audioBytesSent = audioBytesSent - lastAudioSent;
-            networkStats.totalAudioBytesReceived = audioBytesReceived - lastAudioReceived;
-            networkStats.audioBytesReceived = audioBytesReceived;
-            networkStats.totalAudioContentSent = audioContentSent;
-            networkStats.audioContentSent = audioContentSent - lastAudioContentSent;
-            networkStats.totalAudioBytesReceived = audioContentReceived - lastAudioContentReceived;
-            networkStats.audioContentReceived = audioContentReceived;
-            networkStats.totalVideoBytesSent = videoBytesSent;
-            networkStats.videoBytesSent = videoBytesSent - lastVideoSent;
-            networkStats.totalVideoBytesReceived = videoBytesReceived;
-            networkStats.videoBytesReceived = videoBytesReceived - lastVideoReceived;
-            networkStats.totalTransportSent = transportSent - lastTransportSent;
-            networkStats.transportSent = transportSent;
-            networkStats.totalTransportReceived = transportReceived;
-            networkStats.transportReceived = transportReceived - lastTransportReceived;
-            networkStats.bandwidth = networkStats.audioBytesSent + networkStats.audioBytesReceived + networkStats.videoBytesSent + networkStats.videoBytesReceived + networkStats.transportReceived + networkStats.transportSent;
+            const networkStats = yield (0, webrtc_1.fetchNetworkStats)(session, lastNetworkStats);
             if (this.sessionNetworkStats[sessionId]) {
                 this.eventEmitter.emit(ON_NETWORK_STATS, session, networkStats, this.sessionNetworkStats[sessionId]);
                 if (sessionId in this.sessionNetworkStats) {