@wazo/sdk 0.38.13 → 0.38.16

package/lib/simple/room/RemoteParticipant.js

@@ -4616,6 +4616,8 @@ class CTIPhone extends Emitter                  {
   }
 
   useLocalVideoElement() {}
+
+  setMediaConstraints() {}
 }
 
 /*       */
@@ -4793,6 +4795,11 @@ var dirdMethods = (client              , baseUrl        ) => ({
   listPersonalContacts: (queryParams                           = null)                          =>
     client.get(`${baseUrl}/personal`, queryParams).then(response => Contact.parseManyPersonal(response.items)),
 
+  fetchPersonalContact: (contactUuid        )                   =>
+    client
+      .get(`${baseUrl}/personal/${contactUuid}`)
+      .then(Contact.parsePersonal),
+
   addContact: (contact            )                   =>
     client.post(`${baseUrl}/personal`, getContactPayload(contact)).then(Contact.parsePersonal),
 
@@ -7206,6 +7213,12 @@ class WebRTCClient extends Emitter {
     };
 
     const ua = new userAgent.UserAgent(webRTCConfiguration);
+    if (ua.transport && ua.transport.connectPromise) {
+      ua.transport.connectPromise.catch((e => {
+        logger$8.warn('Transport connect error', e);
+      }));
+    }
+
     ua.transport.onMessage = (rawMessage        ) => {
       const message = parser.Parser.parseMessage(rawMessage, ua.transport.logger);
 
@@ -7427,6 +7440,7 @@ class WebRTCClient extends Emitter {
         conference,
         audioOnly,
       },
+      earlyMedia: true,
     };
 
     if (audioOnly) {
@@ -12441,10 +12455,15 @@ class Participant extends Emitter {
     }
   }
 
-  broadcastStatus(inboundStatus         = null) {
+  broadcastStatus(inboundStatus         = null, sendReinvite          ) {
     const status = inboundStatus || this.getStatus();
     logger$2.info('broadcasting participant status', { callId: this.callId, status });
 
+    if (sendReinvite && !this.streams.length && Phone$1.phone) {
+      // eslint-disable-next-line no-underscore-dangle
+      Phone$1.phone._sendReinviteMessage(this.room.callSession, false);
+    }
+
     this.room.sendSignal({
       type: SIGNAL_TYPE_PARTICIPANT_UPDATE,
       origin: this.callId,
@@ -14494,7 +14513,7 @@ class Room extends Emitter {
         // callId is null, someone's requesting everyone's state;
         // or callId is set and matches ours;
         if (this.localParticipant && (!callId || callId === this.localParticipant.callId)) {
-          this.localParticipant.broadcastStatus();
+          this.localParticipant.broadcastStatus(null, true);
         }
 
         // might as well update the requester's status
@@ -14596,7 +14615,7 @@ class Room extends Emitter {
 
     this.connected = true;
 
-    localParticipant.broadcastStatus();
+    localParticipant.broadcastStatus(null, true);
 
     // we're in the room, now let's request everyone's status
     if (this.localParticipant) {

package/lib/simple/room/Room.js

@@ -4618,6 +4618,8 @@ class CTIPhone extends Emitter                  {
   }
 
   useLocalVideoElement() {}
+
+  setMediaConstraints() {}
 }
 
 /*       */
@@ -4795,6 +4797,11 @@ var dirdMethods = (client              , baseUrl        ) => ({
   listPersonalContacts: (queryParams                           = null)                          =>
     client.get(`${baseUrl}/personal`, queryParams).then(response => Contact.parseManyPersonal(response.items)),
 
+  fetchPersonalContact: (contactUuid        )                   =>
+    client
+      .get(`${baseUrl}/personal/${contactUuid}`)
+      .then(Contact.parsePersonal),
+
   addContact: (contact            )                   =>
     client.post(`${baseUrl}/personal`, getContactPayload(contact)).then(Contact.parsePersonal),
 
@@ -7208,6 +7215,12 @@ class WebRTCClient extends Emitter {
     };
 
     const ua = new userAgent.UserAgent(webRTCConfiguration);
+    if (ua.transport && ua.transport.connectPromise) {
+      ua.transport.connectPromise.catch((e => {
+        logger$8.warn('Transport connect error', e);
+      }));
+    }
+
     ua.transport.onMessage = (rawMessage        ) => {
       const message = parser.Parser.parseMessage(rawMessage, ua.transport.logger);
 
@@ -7429,6 +7442,7 @@ class WebRTCClient extends Emitter {
         conference,
         audioOnly,
       },
+      earlyMedia: true,
     };
 
     if (audioOnly) {
@@ -12443,10 +12457,15 @@ class Participant extends Emitter {
     }
   }
 
-  broadcastStatus(inboundStatus         = null) {
+  broadcastStatus(inboundStatus         = null, sendReinvite          ) {
     const status = inboundStatus || this.getStatus();
     logger$2.info('broadcasting participant status', { callId: this.callId, status });
 
+    if (sendReinvite && !this.streams.length && Phone$1.phone) {
+      // eslint-disable-next-line no-underscore-dangle
+      Phone$1.phone._sendReinviteMessage(this.room.callSession, false);
+    }
+
     this.room.sendSignal({
       type: SIGNAL_TYPE_PARTICIPANT_UPDATE,
       origin: this.callId,
@@ -14507,7 +14526,7 @@ class Room extends Emitter {
         // callId is null, someone's requesting everyone's state;
         // or callId is set and matches ours;
         if (this.localParticipant && (!callId || callId === this.localParticipant.callId)) {
-          this.localParticipant.broadcastStatus();
+          this.localParticipant.broadcastStatus(null, true);
         }
 
         // might as well update the requester's status
@@ -14609,7 +14628,7 @@ class Room extends Emitter {
 
     this.connected = true;
 
-    localParticipant.broadcastStatus();
+    localParticipant.broadcastStatus(null, true);
 
     // we're in the room, now let's request everyone's status
     if (this.localParticipant) {

package/lib/simple/room/SipRoom.js

@@ -4616,6 +4616,8 @@ class CTIPhone extends Emitter                  {
   }
 
   useLocalVideoElement() {}
+
+  setMediaConstraints() {}
 }
 
 /*       */
@@ -4793,6 +4795,11 @@ var dirdMethods = (client              , baseUrl        ) => ({
   listPersonalContacts: (queryParams                           = null)                          =>
     client.get(`${baseUrl}/personal`, queryParams).then(response => Contact.parseManyPersonal(response.items)),
 
+  fetchPersonalContact: (contactUuid        )                   =>
+    client
+      .get(`${baseUrl}/personal/${contactUuid}`)
+      .then(Contact.parsePersonal),
+
   addContact: (contact            )                   =>
     client.post(`${baseUrl}/personal`, getContactPayload(contact)).then(Contact.parsePersonal),
 
@@ -7206,6 +7213,12 @@ class WebRTCClient extends Emitter {
     };
 
     const ua = new userAgent.UserAgent(webRTCConfiguration);
+    if (ua.transport && ua.transport.connectPromise) {
+      ua.transport.connectPromise.catch((e => {
+        logger$8.warn('Transport connect error', e);
+      }));
+    }
+
     ua.transport.onMessage = (rawMessage        ) => {
       const message = parser.Parser.parseMessage(rawMessage, ua.transport.logger);
 
@@ -7427,6 +7440,7 @@ class WebRTCClient extends Emitter {
         conference,
         audioOnly,
       },
+      earlyMedia: true,
     };
 
     if (audioOnly) {
@@ -12302,10 +12316,15 @@ class Participant extends Emitter {
     }
   }
 
-  broadcastStatus(inboundStatus         = null) {
+  broadcastStatus(inboundStatus         = null, sendReinvite          ) {
     const status = inboundStatus || this.getStatus();
     logger$3.info('broadcasting participant status', { callId: this.callId, status });
 
+    if (sendReinvite && !this.streams.length && Phone$1.phone) {
+      // eslint-disable-next-line no-underscore-dangle
+      Phone$1.phone._sendReinviteMessage(this.room.callSession, false);
+    }
+
     this.room.sendSignal({
       type: SIGNAL_TYPE_PARTICIPANT_UPDATE,
       origin: this.callId,
@@ -12948,7 +12967,7 @@ class Room$1 extends Emitter {
         // callId is null, someone's requesting everyone's state;
         // or callId is set and matches ours;
         if (this.localParticipant && (!callId || callId === this.localParticipant.callId)) {
-          this.localParticipant.broadcastStatus();
+          this.localParticipant.broadcastStatus(null, true);
         }
 
         // might as well update the requester's status
@@ -13050,7 +13069,7 @@ class Room$1 extends Emitter {
 
     this.connected = true;
 
-    localParticipant.broadcastStatus();
+    localParticipant.broadcastStatus(null, true);
 
     // we're in the room, now let's request everyone's status
     if (this.localParticipant) {

package/lib/simple/room/Stream.js

@@ -4616,6 +4616,8 @@ class CTIPhone extends Emitter                  {
   }
 
   useLocalVideoElement() {}
+
+  setMediaConstraints() {}
 }
 
 /*       */
@@ -4793,6 +4795,11 @@ var dirdMethods = (client              , baseUrl        ) => ({
   listPersonalContacts: (queryParams                           = null)                          =>
     client.get(`${baseUrl}/personal`, queryParams).then(response => Contact.parseManyPersonal(response.items)),
 
+  fetchPersonalContact: (contactUuid        )                   =>
+    client
+      .get(`${baseUrl}/personal/${contactUuid}`)
+      .then(Contact.parsePersonal),
+
   addContact: (contact            )                   =>
     client.post(`${baseUrl}/personal`, getContactPayload(contact)).then(Contact.parsePersonal),
 
@@ -7206,6 +7213,12 @@ class WebRTCClient extends Emitter {
     };
 
     const ua = new userAgent.UserAgent(webRTCConfiguration);
+    if (ua.transport && ua.transport.connectPromise) {
+      ua.transport.connectPromise.catch((e => {
+        logger$8.warn('Transport connect error', e);
+      }));
+    }
+
     ua.transport.onMessage = (rawMessage        ) => {
       const message = parser.Parser.parseMessage(rawMessage, ua.transport.logger);
 
@@ -7427,6 +7440,7 @@ class WebRTCClient extends Emitter {
         conference,
         audioOnly,
       },
+      earlyMedia: true,
     };
 
     if (audioOnly) {
@@ -12302,10 +12316,15 @@ class Participant extends Emitter {
     }
   }
 
-  broadcastStatus(inboundStatus         = null) {
+  broadcastStatus(inboundStatus         = null, sendReinvite          ) {
     const status = inboundStatus || this.getStatus();
     logger$3.info('broadcasting participant status', { callId: this.callId, status });
 
+    if (sendReinvite && !this.streams.length && Phone$1.phone) {
+      // eslint-disable-next-line no-underscore-dangle
+      Phone$1.phone._sendReinviteMessage(this.room.callSession, false);
+    }
+
     this.room.sendSignal({
       type: SIGNAL_TYPE_PARTICIPANT_UPDATE,
       origin: this.callId,
@@ -12948,7 +12967,7 @@ class Room$1 extends Emitter {
         // callId is null, someone's requesting everyone's state;
         // or callId is set and matches ours;
         if (this.localParticipant && (!callId || callId === this.localParticipant.callId)) {
-          this.localParticipant.broadcastStatus();
+          this.localParticipant.broadcastStatus(null, true);
         }
 
         // might as well update the requester's status
@@ -13050,7 +13069,7 @@ class Room$1 extends Emitter {
 
     this.connected = true;
 
-    localParticipant.broadcastStatus();
+    localParticipant.broadcastStatus(null, true);
 
     // we're in the room, now let's request everyone's status
     if (this.localParticipant) {

package/lib/simple/room/__tests__/Participant.test.js

@@ -4616,6 +4616,8 @@ class CTIPhone extends Emitter                  {
   }
 
   useLocalVideoElement() {}
+
+  setMediaConstraints() {}
 }
 
 /*       */
@@ -4793,6 +4795,11 @@ var dirdMethods = (client              , baseUrl        ) => ({
   listPersonalContacts: (queryParams                           = null)                          =>
     client.get(`${baseUrl}/personal`, queryParams).then(response => Contact.parseManyPersonal(response.items)),
 
+  fetchPersonalContact: (contactUuid        )                   =>
+    client
+      .get(`${baseUrl}/personal/${contactUuid}`)
+      .then(Contact.parsePersonal),
+
   addContact: (contact            )                   =>
     client.post(`${baseUrl}/personal`, getContactPayload(contact)).then(Contact.parsePersonal),
 
@@ -7206,6 +7213,12 @@ class WebRTCClient extends Emitter {
     };
 
     const ua = new userAgent.UserAgent(webRTCConfiguration);
+    if (ua.transport && ua.transport.connectPromise) {
+      ua.transport.connectPromise.catch((e => {
+        logger$8.warn('Transport connect error', e);
+      }));
+    }
+
     ua.transport.onMessage = (rawMessage        ) => {
       const message = parser.Parser.parseMessage(rawMessage, ua.transport.logger);
 
@@ -7427,6 +7440,7 @@ class WebRTCClient extends Emitter {
         conference,
         audioOnly,
       },
+      earlyMedia: true,
     };
 
     if (audioOnly) {
@@ -12302,10 +12316,15 @@ class Participant extends Emitter {
     }
   }
 
-  broadcastStatus(inboundStatus         = null) {
+  broadcastStatus(inboundStatus         = null, sendReinvite          ) {
     const status = inboundStatus || this.getStatus();
     logger$3.info('broadcasting participant status', { callId: this.callId, status });
 
+    if (sendReinvite && !this.streams.length && Phone$1.phone) {
+      // eslint-disable-next-line no-underscore-dangle
+      Phone$1.phone._sendReinviteMessage(this.room.callSession, false);
+    }
+
     this.room.sendSignal({
       type: SIGNAL_TYPE_PARTICIPANT_UPDATE,
       origin: this.callId,
@@ -12948,7 +12967,7 @@ class Room$1 extends Emitter {
         // callId is null, someone's requesting everyone's state;
         // or callId is set and matches ours;
         if (this.localParticipant && (!callId || callId === this.localParticipant.callId)) {
-          this.localParticipant.broadcastStatus();
+          this.localParticipant.broadcastStatus(null, true);
         }
 
         // might as well update the requester's status
@@ -13050,7 +13069,7 @@ class Room$1 extends Emitter {
 
     this.connected = true;
 
-    localParticipant.broadcastStatus();
+    localParticipant.broadcastStatus(null, true);
 
     // we're in the room, now let's request everyone's status
     if (this.localParticipant) {

package/lib/simple/room/__tests__/RemoteParticipant.test.js

@@ -4616,6 +4616,8 @@ class CTIPhone extends Emitter                  {
   }
 
   useLocalVideoElement() {}
+
+  setMediaConstraints() {}
 }
 
 /*       */
@@ -4793,6 +4795,11 @@ var dirdMethods = (client              , baseUrl        ) => ({
   listPersonalContacts: (queryParams                           = null)                          =>
     client.get(`${baseUrl}/personal`, queryParams).then(response => Contact.parseManyPersonal(response.items)),
 
+  fetchPersonalContact: (contactUuid        )                   =>
+    client
+      .get(`${baseUrl}/personal/${contactUuid}`)
+      .then(Contact.parsePersonal),
+
   addContact: (contact            )                   =>
     client.post(`${baseUrl}/personal`, getContactPayload(contact)).then(Contact.parsePersonal),
 
@@ -7206,6 +7213,12 @@ class WebRTCClient extends Emitter {
     };
 
     const ua = new userAgent.UserAgent(webRTCConfiguration);
+    if (ua.transport && ua.transport.connectPromise) {
+      ua.transport.connectPromise.catch((e => {
+        logger$8.warn('Transport connect error', e);
+      }));
+    }
+
     ua.transport.onMessage = (rawMessage        ) => {
       const message = parser.Parser.parseMessage(rawMessage, ua.transport.logger);
 
@@ -7427,6 +7440,7 @@ class WebRTCClient extends Emitter {
         conference,
         audioOnly,
       },
+      earlyMedia: true,
     };
 
     if (audioOnly) {
@@ -12302,10 +12316,15 @@ class Participant extends Emitter {
     }
   }
 
-  broadcastStatus(inboundStatus         = null) {
+  broadcastStatus(inboundStatus         = null, sendReinvite          ) {
     const status = inboundStatus || this.getStatus();
     logger$3.info('broadcasting participant status', { callId: this.callId, status });
 
+    if (sendReinvite && !this.streams.length && Phone$1.phone) {
+      // eslint-disable-next-line no-underscore-dangle
+      Phone$1.phone._sendReinviteMessage(this.room.callSession, false);
+    }
+
     this.room.sendSignal({
       type: SIGNAL_TYPE_PARTICIPANT_UPDATE,
       origin: this.callId,
@@ -12948,7 +12967,7 @@ class Room$1 extends Emitter {
         // callId is null, someone's requesting everyone's state;
         // or callId is set and matches ours;
         if (this.localParticipant && (!callId || callId === this.localParticipant.callId)) {
-          this.localParticipant.broadcastStatus();
+          this.localParticipant.broadcastStatus(null, true);
         }
 
         // might as well update the requester's status
@@ -13050,7 +13069,7 @@ class Room$1 extends Emitter {
 
     this.connected = true;
 
-    localParticipant.broadcastStatus();
+    localParticipant.broadcastStatus(null, true);
 
     // we're in the room, now let's request everyone's status
     if (this.localParticipant) {

package/lib/simple/utils.js

@@ -4618,6 +4618,8 @@ class CTIPhone extends Emitter                  {
   }
 
   useLocalVideoElement() {}
+
+  setMediaConstraints() {}
 }
 
 /*       */
@@ -4795,6 +4797,11 @@ var dirdMethods = (client              , baseUrl        ) => ({
   listPersonalContacts: (queryParams                           = null)                          =>
     client.get(`${baseUrl}/personal`, queryParams).then(response => Contact.parseManyPersonal(response.items)),
 
+  fetchPersonalContact: (contactUuid        )                   =>
+    client
+      .get(`${baseUrl}/personal/${contactUuid}`)
+      .then(Contact.parsePersonal),
+
   addContact: (contact            )                   =>
     client.post(`${baseUrl}/personal`, getContactPayload(contact)).then(Contact.parsePersonal),
 
@@ -7208,6 +7215,12 @@ class WebRTCClient extends Emitter {
     };
 
     const ua = new userAgent.UserAgent(webRTCConfiguration);
+    if (ua.transport && ua.transport.connectPromise) {
+      ua.transport.connectPromise.catch((e => {
+        logger$8.warn('Transport connect error', e);
+      }));
+    }
+
     ua.transport.onMessage = (rawMessage        ) => {
       const message = parser.Parser.parseMessage(rawMessage, ua.transport.logger);
 
@@ -7429,6 +7442,7 @@ class WebRTCClient extends Emitter {
         conference,
         audioOnly,
       },
+      earlyMedia: true,
     };
 
     if (audioOnly) {
@@ -12304,10 +12318,15 @@ class Participant extends Emitter {
     }
   }
 
-  broadcastStatus(inboundStatus         = null) {
+  broadcastStatus(inboundStatus         = null, sendReinvite          ) {
     const status = inboundStatus || this.getStatus();
     logger$3.info('broadcasting participant status', { callId: this.callId, status });
 
+    if (sendReinvite && !this.streams.length && Phone$1.phone) {
+      // eslint-disable-next-line no-underscore-dangle
+      Phone$1.phone._sendReinviteMessage(this.room.callSession, false);
+    }
+
     this.room.sendSignal({
       type: SIGNAL_TYPE_PARTICIPANT_UPDATE,
       origin: this.callId,
@@ -12950,7 +12969,7 @@ class Room$1 extends Emitter {
         // callId is null, someone's requesting everyone's state;
         // or callId is set and matches ours;
         if (this.localParticipant && (!callId || callId === this.localParticipant.callId)) {
-          this.localParticipant.broadcastStatus();
+          this.localParticipant.broadcastStatus(null, true);
         }
 
         // might as well update the requester's status
@@ -13052,7 +13071,7 @@ class Room$1 extends Emitter {
 
     this.connected = true;
 
-    localParticipant.broadcastStatus();
+    localParticipant.broadcastStatus(null, true);
 
     // we're in the room, now let's request everyone's status
     if (this.localParticipant) {

package/lib/web-rtc-client.js

@@ -4540,6 +4540,11 @@ var dirdMethods = (client              , baseUrl        ) => ({
   listPersonalContacts: (queryParams                           = null)                          =>
     client.get(`${baseUrl}/personal`, queryParams).then(response => Contact.parseManyPersonal(response.items)),
 
+  fetchPersonalContact: (contactUuid        )                   =>
+    client
+      .get(`${baseUrl}/personal/${contactUuid}`)
+      .then(Contact.parsePersonal),
+
   addContact: (contact            )                   =>
     client.post(`${baseUrl}/personal`, getContactPayload(contact)).then(Contact.parsePersonal),
 
@@ -5797,6 +5802,12 @@ class WebRTCClient extends Emitter {
     };
 
     const ua = new userAgent.UserAgent(webRTCConfiguration);
+    if (ua.transport && ua.transport.connectPromise) {
+      ua.transport.connectPromise.catch((e => {
+        logger.warn('Transport connect error', e);
+      }));
+    }
+
     ua.transport.onMessage = (rawMessage        ) => {
       const message = parser.Parser.parseMessage(rawMessage, ua.transport.logger);
 
@@ -6018,6 +6029,7 @@ class WebRTCClient extends Emitter {
         conference,
         audioOnly,
       },
+      earlyMedia: true,
     };
 
     if (audioOnly) {

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@wazo/sdk",
-  "version": "0.38.13",
+  "version": "0.38.16",
   "description": "Wazo's JavaScript Software Development Kit.",
   "main": "dist/wazo-sdk.js",
   "author": "Wazo (http://wazo.io)",

package/src/api/dird.js

@@ -35,6 +35,11 @@ export default (client: ApiRequester, baseUrl: string) => ({
   listPersonalContacts: (queryParams: ContactSearchQueryParams = null): Promise<Array<Contact>> =>
     client.get(`${baseUrl}/personal`, queryParams).then(response => Contact.parseManyPersonal(response.items)),
 
+  fetchPersonalContact: (contactUuid: string): Promise<Contact> =>
+    client
+      .get(`${baseUrl}/personal/${contactUuid}`)
+      .then(Contact.parsePersonal),
+
   addContact: (contact: NewContact): Promise<Contact> =>
     client.post(`${baseUrl}/personal`, getContactPayload(contact)).then(Contact.parsePersonal),
 

package/src/domain/Phone/CTIPhone.js

@@ -341,4 +341,6 @@ export default class CTIPhone extends Emitter implements Phone {
   }
 
   useLocalVideoElement() {}
+
+  setMediaConstraints() {}
 }

package/src/domain/Phone/Phone.js

@@ -110,4 +110,6 @@ export interface Phone {
   isRegistered(): boolean;
 
   hasIncomingCallSession(): boolean;
+
+  setMediaConstraints(media: MediaStreamConstraints): PhoneVoid;
 }

package/src/simple/room/Participant.js

@@ -2,6 +2,7 @@
 import Emitter from '../../utils/Emitter';
 import IssueReporter from '../../service/IssueReporter';
 
+import Phone from '../Phone';
 import Room, { SIGNAL_TYPE_PARTICIPANT_UPDATE } from './Room';
 import Contact from '../../domain/Contact';
 import getApiClient from '../../service/getApiClient';
@@ -296,10 +297,15 @@ class Participant extends Emitter {
     }
   }
 
-  broadcastStatus(inboundStatus: Object = null) {
+  broadcastStatus(inboundStatus: Object = null, sendReinvite: ?boolean) {
     const status = inboundStatus || this.getStatus();
     logger.info('broadcasting participant status', { callId: this.callId, status });
 
+    if (sendReinvite && !this.streams.length && Phone.phone) {
+      // eslint-disable-next-line no-underscore-dangle
+      Phone.phone._sendReinviteMessage(this.room.callSession, false);
+    }
+
     this.room.sendSignal({
       type: SIGNAL_TYPE_PARTICIPANT_UPDATE,
       origin: this.callId,

package/src/simple/room/Room.js

@@ -602,7 +602,7 @@ class Room extends Emitter {
         // callId is null, someone's requesting everyone's state;
         // or callId is set and matches ours;
         if (this.localParticipant && (!callId || callId === this.localParticipant.callId)) {
-          this.localParticipant.broadcastStatus();
+          this.localParticipant.broadcastStatus(null, true);
         }
 
         // might as well update the requester's status
@@ -704,7 +704,7 @@ class Room extends Emitter {
 
     this.connected = true;
 
-    localParticipant.broadcastStatus();
+    localParticipant.broadcastStatus(null, true);
 
     // we're in the room, now let's request everyone's status
     if (this.localParticipant) {

package/src/web-rtc-client.js

@@ -251,6 +251,12 @@ export default class WebRTCClient extends Emitter {
     };
 
     const ua = new UserAgent(webRTCConfiguration);
+    if (ua.transport && ua.transport.connectPromise) {
+      ua.transport.connectPromise.catch((e => {
+        logger.warn('Transport connect error', e);
+      }));
+    }
+
     ua.transport.onMessage = (rawMessage: string) => {
       const message = Parser.parseMessage(rawMessage, ua.transport.logger);
 
@@ -472,6 +478,7 @@ export default class WebRTCClient extends Emitter {
         conference,
         audioOnly,
       },
+      earlyMedia: true,
     };
 
     if (audioOnly) {