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.
41 lines
1.3 KiB
TypeScript
41 lines
1.3 KiB
TypeScript
/**
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* Audio codec translation layer for bridging between WebRTC and SIP.
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*
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* All actual codec work (Opus, G.722, PCMU, PCMA) is done in Rust via
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* the smartrust bridge. This module provides the RTP-level transcoding
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* interface used by the webrtcbridge.
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*/
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import { Buffer } from 'node:buffer';
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import { transcode, isCodecReady } from './opusbridge.ts';
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/** Opus dynamic payload type (standard WebRTC assignment). */
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export const OPUS_PT = 111;
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export interface IRtpTranscoder {
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/** Transcode an RTP payload. Always async (Rust IPC). */
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payload(data: Buffer): Promise<Buffer>;
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fromPT: number;
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toPT: number;
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}
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/**
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* Create a transcoder that converts RTP payloads between two codecs.
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* Returns null if the codecs are the same or the Rust bridge isn't ready.
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*
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* @param sessionId - optional Rust codec session for isolated state per call
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*/
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export function createTranscoder(fromPT: number, toPT: number, sessionId?: string, direction?: string): IRtpTranscoder | null {
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if (fromPT === toPT) return null;
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if (!isCodecReady()) return null;
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return {
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fromPT,
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toPT,
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async payload(data: Buffer): Promise<Buffer> {
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const result = await transcode(data, fromPT, toPT, sessionId, direction);
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return result || Buffer.alloc(0); // return empty on failure — never pass raw codec bytes
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},
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};
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}
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