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Merge pull request #784 from wazo-platform/ma_1501_normalize_sip_id_l…
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Fix SIP id length
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@manuquentin
manuquentin authored Aug 23, 2024
2 parents 8860d56 + fe5b6f4 commit 0de32cf
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Showing 2 changed files with 63 additions and 58 deletions.
63 changes: 34 additions & 29 deletions src/domain/Phone/WebRTCPhone.ts
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Original file line number Diff line number Diff line change
Expand Up @@ -159,7 +159,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
async sendReinvite(callSession?: CallSession, constraints: Record<string, any> | null | undefined = null, conference = false, audioOnly = false, iceRestart = false): Promise<OutgoingInviteRequest | void> {
const sipSession = this.findSipSession(callSession);
logger.info('WebRTC phone - send reinvite', {
sessionId: sipSession ? sipSession.id : null,
sessionId: this.getSipSessionId(sipSession),
constraints,
audioOnly,
conference,
Expand Down Expand Up @@ -380,7 +380,9 @@ export default class WebRTCPhone extends Emitter implements Phone {
const elsewhere = message.data.indexOf('cause=26') !== -1 && message.data.indexOf('completed elsewhere') !== -1;
const callSession = this._createCallSession(sipSession);

delete this.callSessions[sipSessionId];
if (sipSessionId) {
delete this.callSessions[sipSessionId];
}
this.eventEmitter.emit(ON_CALL_CANCELED, callSession, elsewhere);
};
}
Expand All @@ -389,15 +391,15 @@ export default class WebRTCPhone extends Emitter implements Phone {
switch (newState) {
case SessionState.Terminating:
logger.info('WebRTC phone - call terminating', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
});
this.eventEmitter.emit(ON_CALL_ENDING, this._createCallSession(sipSession));
break;

case SessionState.Terminated:
{
logger.info('WebRTC phone - call terminated', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
});

// Should be called before `_onCallTerminated` or the callCount will not decrement...
Expand Down Expand Up @@ -580,15 +582,15 @@ export default class WebRTCPhone extends Emitter implements Phone {

_onCallAccepted(sipSession: WazoSession, cameraEnabled: boolean): CallSession {
logger.info('WebRTC phone - on call accepted', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
cameraEnabled,
});

const callSession = this._createAcceptedCallSession(sipSession, cameraEnabled);

if (this.currentSipSession && !this.isCurrentCallSipSession(callSession)) {
logger.info('WebRTC phone - on call accepted with another ongoing call, holding it', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
cameraEnabled,
});

Expand Down Expand Up @@ -673,7 +675,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
const sipSessionId = this.getSipSessionId(sipSession);

logger.info('WebRTC phone - on call terminated', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
sipSessionId,
ids: Object.keys(this.callSessions),
});
Expand All @@ -697,9 +699,9 @@ export default class WebRTCPhone extends Emitter implements Phone {

if (sipSessionId) {
this.removeIncomingSessions(sipSessionId);
}

delete this.callSessions[sipSessionId];
delete this.callSessions[sipSessionId];
}

if (isCurrentSession) {
this.currentSipSession = undefined;
Expand All @@ -709,7 +711,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
const hasIncomingCallSession = this.hasIncomingCallSession();
// Re-trigger incoming call event for remaining incoming calls
logger.info('WebRTC phone - check to re-trigger incoming call', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
shouldRetrigger,
hasIncomingCallSession,
});
Expand Down Expand Up @@ -853,7 +855,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
const sipSession = this.client.getSipSession(callSession.getId()) as Invitation;

if (sipSession) {
logger.info('accept call, session found', { sipId: sipSession.id });
logger.info('accept call, session found', { sipId: this.getSipSessionId(sipSession) });

return this.client.answer(sipSession, this.allowVideo ? cameraEnabled : false).then(() => {
return callSession.sipCallId;
Expand Down Expand Up @@ -898,7 +900,7 @@ export default class WebRTCPhone extends Emitter implements Phone {

if (sipSession) {
logger.info('WebRTC rejecting', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
});
this.client.hangup(sipSession);
}
Expand All @@ -919,7 +921,7 @@ export default class WebRTCPhone extends Emitter implements Phone {

if (sipSession) {
logger.info('WebRTC atxfer', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
});
return this.client.atxfer(sipSession);
}
Expand Down Expand Up @@ -950,13 +952,13 @@ export default class WebRTCPhone extends Emitter implements Phone {
}

logger.info('WebRTC resuming', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
});

// Hold current session if different from the current one (we don't want 2 sessions active at the same time).
if (this.currentSipSession && this.currentSipSession.id !== sipSession.id) {
if (this.currentSipSession && this.getSipSessionId(this.currentSipSession) !== this.getSipSessionId(sipSession)) {
logger.info('WebRTC hold call after resume', {
id: this.currentSipSession.id,
id: this.getSipSessionId(this.currentSipSession),
});
this.holdSipSession(this.currentSipSession, this.currentCallSession, withEvent);
}
Expand All @@ -979,7 +981,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
const sessionId = this.getSipSessionId(sipSession);
const hasVideo = this.client.hasLocalVideo(sessionId);
logger.info('WebRTC hold sip session', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
hasVideo,
});

Expand Down Expand Up @@ -1076,7 +1078,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
}

logger.info('WebRTC muting', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
});
this.client.mute(sipSession);

Expand Down Expand Up @@ -1104,7 +1106,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
}

logger.info('WebRTC unmuting', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
});
this.client.unmute(sipSession);

Expand Down Expand Up @@ -1139,7 +1141,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
}

logger.info('WebRTC turn camera on', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
});
this.client.toggleCameraOn(sipSession);
this.eventEmitter.emit(ON_CAMERA_RESUMED, this._createCameraResumedCallSession(sipSession, callSession));
Expand All @@ -1153,7 +1155,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
}

logger.info('WebRTC turn camera off', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
});
this.client.toggleCameraOff(sipSession);
this.eventEmitter.emit(ON_CAMERA_DISABLED, this._createCameraDisabledCallSession(sipSession, callSession));
Expand All @@ -1167,7 +1169,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
}

logger.info('WebRTC send key', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
tone,
});
this.client.sendDTMF(sipSession, tone);
Expand Down Expand Up @@ -1245,7 +1247,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
}

logger.info('WebRTC transfer', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
target,
});
this.client.transfer(sipSession, target);
Expand All @@ -1255,8 +1257,8 @@ export default class WebRTCPhone extends Emitter implements Phone {
const sipSession = this.client.getSipSession(source.sipCallId);
const sipSessionTarget = this.client.getSipSession(destination.sipCallId);
logger.info('WebRTC indirect transfer', {
sipId: sipSession ? sipSession.id : null,
target: sipSessionTarget ? sipSessionTarget.id : null,
sipId: sipSession ? this.getSipSessionId(sipSession) : null,
target: sipSessionTarget ? this.getSipSessionId(sipSessionTarget) : null,
});

if (!sipSessionTarget) {
Expand Down Expand Up @@ -1295,7 +1297,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
const sipSessionId = this.getSipSessionId(sipSession);

logger.info('WebRTC hangup', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
sipSessionId,
});
await this.client.hangup(sipSession);
Expand Down Expand Up @@ -1484,7 +1486,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
const autoAnswer = sipSession.request.getHeader('Answer-Mode') === 'Auto';
const withVideo = this.allowVideo ? wantsToDoVideo : false;
logger.info('WebRTC invite received', {
sipId: sipSession.id,
sipId: this.getSipSessionId(sipSession),
withVideo,
autoAnswer,
});
Expand Down Expand Up @@ -1555,7 +1557,7 @@ export default class WebRTCPhone extends Emitter implements Phone {
this.eventEmitter.emit(ON_CALL_ERROR, e);
});
this.client.on(this.client.REJECTED, session => {
logger.info('WebRTC call rejected', session.id);
logger.info('WebRTC call rejected', { id: this.getSipSessionId(session) });

const callSession = this._createCallSession(session);

Expand Down Expand Up @@ -1687,7 +1689,10 @@ export default class WebRTCPhone extends Emitter implements Phone {
return this.callSessions[sipSessionId];
}

getSipSessionId(sipSession: WazoSession): string {
getSipSessionId(sipSession: WazoSession | null | undefined): string {
if (!sipSession) {
return '';
}
return this.client.getSipSessionId(sipSession);
}

Expand Down
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