feat(frontend): supervisor SIP client — JsSIP wrapper for barge sessions

Separate from agent sip-client.ts — different lifecycle (on-demand per
barge session, not persistent). Auto-answers incoming Ozonetel calls.
DTMF mode switching: 4=listen, 5=whisper, 6=barge. Event-driven with
registered/callConnected/callEnded events. Audio via hidden <audio> element.

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
This commit is contained in:
2026-04-12 16:08:28 +05:30
parent d730cda06d
commit 24b4e01292

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import JsSIP from 'jssip';
// Lightweight SIP client for supervisor barge sessions.
// Separate from the agent's sip-client.ts — different lifecycle.
// Modeled on Ozonetel's kSip utility (CA-Admin/.../utils/ksip.tsx).
//
// DTMF mode mapping (from Ozonetel CA-Admin BargeinDrawerSip.tsx):
// "4" → Listen (supervisor hears all, nobody hears supervisor)
// "5" → Whisper/Training (agent hears supervisor, patient doesn't)
// "6" → Barge (both hear supervisor)
type EventCallback = (...args: any[]) => void;
type SupervisorSipEvent =
| 'registered'
| 'registrationFailed'
| 'callReceived'
| 'callConnected'
| 'callEnded'
| 'callFailed';
type SupervisorSipConfig = {
domain: string;
port: string;
number: string;
password: string;
};
class SupervisorSipClient {
private ua: JsSIP.UA | null = null;
private session: JsSIP.RTCSession | null = null;
private config: SupervisorSipConfig | null = null;
private listeners = new Map<string, Set<EventCallback>>();
private audioElement: HTMLAudioElement | null = null;
init(config: SupervisorSipConfig): void {
this.config = config;
this.cleanup();
// Hidden audio element for remote call audio
this.audioElement = document.createElement('audio');
this.audioElement.id = 'supervisor-remote-audio';
this.audioElement.autoplay = true;
this.audioElement.setAttribute('playsinline', '');
document.body.appendChild(this.audioElement);
const socketUrl = `wss://${config.domain}:${config.port}`;
const socket = new JsSIP.WebSocketInterface(socketUrl);
this.ua = new JsSIP.UA({
sockets: [socket],
uri: `sip:${config.number}@${config.domain}`,
password: config.password,
registrar_server: `sip:${config.domain}`,
authorization_user: config.number,
session_timers: false,
register: false,
});
this.ua.on('registered', () => {
console.log('[SupervisorSIP] Registered');
this.emit('registered');
});
this.ua.on('registrationFailed', (e: any) => {
console.error('[SupervisorSIP] Registration failed:', e?.cause);
this.emit('registrationFailed', e?.cause);
});
this.ua.on('newRTCSession', (data: any) => {
const rtcSession = data.session as JsSIP.RTCSession;
if (rtcSession.direction !== 'incoming') return;
console.log('[SupervisorSIP] Incoming call — auto-answering');
this.session = rtcSession;
this.emit('callReceived');
rtcSession.on('accepted', () => {
console.log('[SupervisorSIP] Call accepted');
this.emit('callConnected');
});
rtcSession.on('confirmed', () => {
// Attach remote audio stream
const connection = rtcSession.connection;
if (connection && this.audioElement) {
// Modern browsers: track event
connection.addEventListener('track', (event: RTCTrackEvent) => {
if (event.streams[0] && this.audioElement) {
this.audioElement.srcObject = event.streams[0];
}
});
// Fallback: getRemoteStreams (older browsers/JsSIP versions)
const remoteStreams = (connection as any).getRemoteStreams?.();
if (remoteStreams?.[0] && this.audioElement) {
this.audioElement.srcObject = remoteStreams[0];
}
}
});
rtcSession.on('ended', () => {
console.log('[SupervisorSIP] Call ended');
this.session = null;
this.emit('callEnded');
});
rtcSession.on('failed', (e: any) => {
console.error('[SupervisorSIP] Call failed:', e?.cause);
this.session = null;
this.emit('callFailed', e?.cause);
});
// Auto-answer with audio
rtcSession.answer({
mediaConstraints: { audio: true, video: false },
});
});
this.ua.start();
}
register(): void {
this.ua?.register();
}
isRegistered(): boolean {
return this.ua?.isRegistered() ?? false;
}
isCallActive(): boolean {
return this.session?.isEstablished() ?? false;
}
sendDTMF(digit: string): void {
if (!this.session?.isEstablished()) {
console.warn('[SupervisorSIP] Cannot send DTMF — no active session');
return;
}
console.log(`[SupervisorSIP] Sending DTMF: ${digit}`);
this.session.sendDTMF(digit, {
duration: 160,
interToneGap: 1200,
});
}
hangup(): void {
if (this.session) {
try {
this.session.terminate();
} catch {
// Session may already be ended
}
this.session = null;
}
}
close(): void {
this.hangup();
if (this.ua) {
try {
this.ua.unregister({ all: true });
this.ua.stop();
} catch {
// UA may already be stopped
}
this.ua = null;
}
this.cleanup();
this.listeners.clear();
}
on(event: SupervisorSipEvent, callback: EventCallback): void {
if (!this.listeners.has(event)) {
this.listeners.set(event, new Set());
}
this.listeners.get(event)!.add(callback);
}
off(event: SupervisorSipEvent, callback: EventCallback): void {
this.listeners.get(event)?.delete(callback);
}
private emit(event: string, ...args: any[]): void {
this.listeners.get(event)?.forEach(cb => {
try { cb(...args); } catch (e) { console.error(`[SupervisorSIP] Event error [${event}]:`, e); }
});
}
private cleanup(): void {
if (this.audioElement) {
this.audioElement.srcObject = null;
this.audioElement.remove();
this.audioElement = null;
}
}
}
export const supervisorSip = new SupervisorSipClient();