Juergen Kunz
f3e1c96872
Full-featured SIP router with multi-provider trunking, browser softphone via WebRTC, real-time Opus/G.722/PCM transcoding in Rust, RNNoise ML noise suppression, Kokoro neural TTS announcements, and a Lit-based web dashboard with live call monitoring and REST API.
105 lines
3.4 KiB
TypeScript
105 lines
3.4 KiB
TypeScript
/**
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* ILeg interface — abstract connection from a Call hub to an endpoint.
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*
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* Concrete implementations: SipLeg (SIP devices + providers) and WebRtcLeg (browsers).
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* Shared RTP utilities (header building, clock rates) are also defined here.
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*/
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import { Buffer } from 'node:buffer';
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import type dgram from 'node:dgram';
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import type { IEndpoint } from '../sip/index.ts';
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import type { TLegState, TLegType, ILegStatus } from './types.ts';
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import type { IRtpTranscoder } from '../codec.ts';
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import type { SipDialog } from '../sip/index.ts';
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import type { SipMessage } from '../sip/index.ts';
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// ---------------------------------------------------------------------------
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// ILeg interface
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// ---------------------------------------------------------------------------
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export interface ILeg {
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readonly id: string;
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readonly type: TLegType;
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state: TLegState;
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/** The SIP Call-ID used by this leg (for CallManager routing). */
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readonly sipCallId: string;
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/** Where this leg sends/receives RTP. */
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readonly rtpPort: number | null;
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readonly rtpSock: dgram.Socket | null;
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remoteMedia: IEndpoint | null;
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/** Negotiated codec payload type (e.g. 9 = G.722, 111 = Opus). */
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codec: number | null;
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/** Transcoder for converting to this leg's codec (set by Call when codecs differ). */
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transcoder: IRtpTranscoder | null;
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/** Packet counters. */
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pktSent: number;
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pktReceived: number;
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/** SIP dialog (SipLegs only, null for WebRtcLegs). */
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readonly dialog: SipDialog | null;
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/**
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* Send an RTP packet toward this leg's remote endpoint.
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* If a transcoder is set, the Call should transcode before calling this.
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*/
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sendRtp(data: Buffer): void;
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/**
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* Callback set by the owning Call — invoked when this leg receives an RTP packet.
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* The Call uses this to forward to other legs.
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*/
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onRtpReceived: ((data: Buffer) => void) | null;
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/**
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* Handle an incoming SIP message routed to this leg (SipLegs only).
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* Returns a SipMessage response if one needs to be sent, or null.
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*/
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handleSipMessage(msg: SipMessage, rinfo: IEndpoint): void;
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/** Release all resources (sockets, peer connections, etc.). */
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teardown(): void;
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/** Status snapshot for the dashboard. */
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getStatus(): ILegStatus;
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}
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// ---------------------------------------------------------------------------
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// Shared RTP utilities
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// ---------------------------------------------------------------------------
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/** RTP clock increment per 20ms frame for each codec. */
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export function rtpClockIncrement(pt: number): number {
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if (pt === 111) return 960; // Opus: 48000 Hz x 0.02s
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if (pt === 9) return 160; // G.722: 8000 Hz x 0.02s (SDP clock rate quirk)
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return 160; // PCMU/PCMA: 8000 Hz x 0.02s
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}
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/** Build a fresh RTP header with correct PT, timestamp, seq, SSRC. */
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export function buildRtpHeader(pt: number, seq: number, ts: number, ssrc: number, marker: boolean): Buffer {
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const hdr = Buffer.alloc(12);
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hdr[0] = 0x80; // V=2
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hdr[1] = (marker ? 0x80 : 0) | (pt & 0x7f);
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hdr.writeUInt16BE(seq & 0xffff, 2);
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hdr.writeUInt32BE(ts >>> 0, 4);
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hdr.writeUInt32BE(ssrc >>> 0, 8);
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return hdr;
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}
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/** Codec name for status display. */
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export function codecDisplayName(pt: number | null): string | null {
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if (pt === null) return null;
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switch (pt) {
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case 0: return 'PCMU';
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case 8: return 'PCMA';
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case 9: return 'G.722';
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case 111: return 'Opus';
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case 101: return 'telephone-event';
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default: return `PT${pt}`;
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}
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}
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