@ariaflowagents/livekit-plugin-transport-sip 0.9.0

package/dist/jssip_signaling.js

@@ -0,0 +1,255 @@
+/**
+ * SIP Signaling implementation using JsSIP library for WebSocket transport.
+ *
+ * @experimental
+ *
+ * WARNING: JsSIP is a WebRTC-only library. Its `session.answer()` API is a
+ * WebRTC API and cannot bridge to the raw RTP pipeline used by `SIPAudioInput`
+ * and `SIPAudioOutput`. This file is kept for reference but the WebSocket SIP
+ * signaling path is architecturally incompatible with the RTP transport.
+ *
+ * For production use, use `sip_signaling.ts` (UDP-based, compatible with RTP).
+ *
+ * JsSIP supports:
+ * - SIP over WebSocket (secure WSS and non-secure WS)
+ * - WebRTC for media (NOT compatible with raw RTP)
+ * - Full SIP protocol compliance (INVITE, ACK, BYE, etc.)
+ */
+import { UA, WebSocketInterface } from 'jssip';
+/**
+ * SDP offer for G.711 μ-law audio.
+ */
+const G711_SDP_OFFER = (localAddress, rtpPort) => `v=0\r\n` +
+    `o=- ${Date.now()} ${Date.now()} IN IP4 ${localAddress}\r\n` +
+    `s=AriaFlow Voice Agent\r\n` +
+    `c=IN IP4 ${localAddress}\r\n` +
+    `t=0 0\r\n` +
+    `m=audio ${rtpPort} RTP/AVP 0 8\r\n` + // 0 = G.711 μ-law (PCMU), 8 = G.711 A-law (PCMA)
+    `a=rtpmap:0 PCMU/8000\r\n` +
+    `a=rtpmap:8 PCMA/8000\r\n` +
+    `a=fmtp:0\r\n` +
+    `a=fmtp:8\r\n` +
+    `a=ptime:20\r\n` +
+    `a=sendrecv\r\n`;
+/**
+ * SIP Signaling handler using JsSIP over WebSocket.
+ *
+ * This implementation uses the JsSIP library for WebSocket-based SIP
+ * signaling, commonly used with WebRTC and modern SIP providers.
+ */
+export class JsSIPSignaling {
+    options;
+    ua = null;
+    activeSessions = new Map();
+    nextRtpPort;
+    localAddress;
+    constructor(options) {
+        this.options = options;
+        this.localAddress = options.localAddress;
+        this.nextRtpPort = options.rtpPortStart ?? 10000;
+    }
+    /**
+     * Start the SIP user agent and begin listening for calls over WebSocket.
+     *
+     * @param onInvite - Callback when a new INVITE is received
+     * @param onBye - Callback when a BYE is received (remote hangup)
+     */
+    async start(onInvite, onBye) {
+        // Create WebSocket URL
+        const wsProtocol = this.options.secureWebSocket === false ? 'ws' : 'wss';
+        const wsUrl = `${wsProtocol}://${this.options.wsServerHost || this.localAddress}:${this.options.wsServerPort || 8080}/ws`;
+        // Create SIP URI
+        const sipUri = `sip:${this.options.sipUsername || 'agent'}@${this.options.sipDomain || this.localAddress}`;
+        // Create JsSIP User Agent
+        this.ua = new UA({
+            uri: sipUri,
+            sockets: [new WebSocketInterface(wsUrl)],
+            password: this.options.sipPassword,
+            register: this.options.shouldRegister !== false, // Register by default
+        });
+        // Set up event handlers for incoming calls
+        this.ua.on('newRTCSession', (data) => {
+            this.handleIncomingInvite(data.session, onInvite, onBye);
+        });
+        // Set up connection event handlers
+        this.ua.on('connected', () => {
+            console.log(`[JsSIPSignaling] Connected to SIP server: ${wsUrl}`);
+        });
+        this.ua.on('disconnected', () => {
+            console.warn('[JsSIPSignaling] Disconnected from SIP server');
+        });
+        this.ua.on('registered', () => {
+            console.log(`[JsSIPSignaling] Registered as: ${sipUri}`);
+        });
+        this.ua.on('registrationFailed', () => {
+            console.error('[JsSIPSignaling] Registration failed');
+        });
+        // Start the user agent
+        this.ua.start();
+        console.log(`[JsSIPSignaling] Started with WebSocket: ${wsUrl}`);
+    }
+    /**
+     * Handle an incoming INVITE request.
+     */
+    async handleIncomingInvite(session, onInvite, onBye) {
+        const callId = session.request?.callId || `call-${Date.now()}`;
+        const remoteSdp = session.request?.body || '';
+        console.log(`[JsSIPSignaling] Received INVITE for call: ${callId}`);
+        console.log(`[JsSIPSignaling] From: ${session.remote_identity?.uri?.toString() || session}`);
+        // Allocate RTP port for this call
+        const rtpPort = this.nextRtpPort;
+        this.nextRtpPort += 2; // RTP uses even ports
+        // Generate SDP answer with our RTP port
+        const sdpAnswer = G711_SDP_OFFER(this.localAddress, rtpPort);
+        // Set up session state handlers
+        session.on('progress', () => {
+            console.log(`[JsSIPSignaling] Call ${callId} in progress`);
+        });
+        session.on('confirmed', () => {
+            console.log(`[JsSIPSignaling] Call ${callId} established`);
+        });
+        session.on('ended', () => {
+            console.log(`[JsSIPSignaling] Call ${callId} ended`);
+            this.activeSessions.delete(callId);
+            if (onBye) {
+                onBye(callId);
+            }
+        });
+        session.on('failed', (error) => {
+            console.error(`[JsSIPSignaling] Call ${callId} failed:`, error);
+            this.activeSessions.delete(callId);
+        });
+        // Accept the invitation with our SDP
+        try {
+            await session.answer({
+                extraHeaders: ['Content-Type: application/sdp'],
+            });
+            // Track the active session
+            this.activeSessions.set(callId, { session, rtpPort });
+            // Notify the application about the new call
+            onInvite(callId, remoteSdp, rtpPort);
+        }
+        catch (error) {
+            console.error(`[JsSIPSignaling] Error accepting call ${callId}:`, error);
+            try {
+                session.terminate();
+            }
+            catch {
+                // Ignore terminate errors
+            }
+        }
+    }
+    /**
+     * Hang up an active call.
+     */
+    async hangup(callId) {
+        const activeSession = this.activeSessions.get(callId);
+        if (!activeSession) {
+            console.warn(`[JsSIPSignaling] No active session for call: ${callId}`);
+            return;
+        }
+        try {
+            await activeSession.session.terminate();
+            this.activeSessions.delete(callId);
+            console.log(`[JsSIPSignaling] Hung up call: ${callId}`);
+        }
+        catch (error) {
+            console.error(`[JsSIPSignaling] Error hanging up call ${callId}:`, error);
+        }
+    }
+    /**
+     * Make an outgoing call.
+     */
+    async makeCall(targetUri, onInvite, onBye) {
+        if (!this.ua) {
+            throw new Error('[JsSIPSignaling] User agent not started');
+        }
+        // Allocate RTP port for this call
+        const rtpPort = this.nextRtpPort;
+        this.nextRtpPort += 2;
+        // Generate SDP offer with our RTP port
+        const sdpOffer = G711_SDP_OFFER(this.localAddress, rtpPort);
+        // Create the call session
+        const session = this.ua.call(targetUri, {
+            extraHeaders: ['Content-Type: application/sdp'],
+            eventHandlers: {
+                progress: () => {
+                    const callId = session.request?.callId || `call-${Date.now()}`;
+                    console.log(`[JsSIPSignaling] Outgoing call ${callId} in progress`);
+                },
+                confirmed: () => {
+                    const callId = session.request?.callId || `call-${Date.now()}`;
+                    console.log(`[JsSIPSignaling] Outgoing call ${callId} accepted`);
+                    this.activeSessions.set(callId, { session, rtpPort });
+                    if (onInvite) {
+                        onInvite(callId, session.request?.body || '', rtpPort);
+                    }
+                },
+                ended: () => {
+                    const callId = session.request?.callId || `call-${Date.now()}`;
+                    console.log(`[JsSIPSignaling] Outgoing call ${callId} ended`);
+                    this.activeSessions.delete(callId);
+                    if (onBye) {
+                        onBye(callId);
+                    }
+                },
+                failed: (error) => {
+                    const callId = session.request?.callId || `call-${Date.now()}`;
+                    console.error(`[JsSIPSignaling] Outgoing call ${callId} failed:`, error);
+                    this.activeSessions.delete(callId);
+                },
+            },
+        });
+        return session.request?.callId || `call-${Date.now()}`;
+    }
+    /**
+     * Stop the SIP user agent and hang up all active calls.
+     */
+    async stop() {
+        if (!this.ua) {
+            return;
+        }
+        // Hang up all active calls
+        const hangupPromises = Array.from(this.activeSessions.entries()).map(async ([callId, { session }]) => {
+            try {
+                await session.terminate();
+            }
+            catch (error) {
+                console.error(`[JsSIPSignaling] Error hanging up call ${callId} during shutdown:`, error);
+            }
+        });
+        await Promise.allSettled(hangupPromises);
+        this.activeSessions.clear();
+        // Stop the user agent
+        this.ua.stop();
+        this.ua = null;
+        console.log('[JsSIPSignaling] Stopped');
+    }
+    /**
+     * Get the RTP port allocated for a specific call.
+     */
+    getRtpPort(callId) {
+        return this.activeSessions.get(callId)?.rtpPort;
+    }
+    /**
+     * Check if the user agent is registered.
+     */
+    get isRegistered() {
+        return this.ua?.isRegistered() ?? false;
+    }
+    /**
+     * Get the user agent status.
+     */
+    get status() {
+        if (!this.ua)
+            return 'disconnected';
+        const uaStatus = this.ua.status;
+        // JsSIP UA status: 0 = Init, 1 = Ready, 2 = UserRegistered, 3 = Unregistered
+        if (uaStatus === 0)
+            return 'connecting';
+        if (uaStatus === 1 || uaStatus === 2)
+            return 'connected';
+        return 'disconnected';
+    }
+}
+//# sourceMappingURL=jssip_signaling.js.map

package/dist/jssip_signaling.js.map

@@ -0,0 +1 @@
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package/dist/rtp/jitter_buffer.d.ts

@@ -0,0 +1,19 @@
+import type { RtpPacket } from './rtp_packet.js';
+/**
+ * Simple jitter buffer that reorders RTP packets by sequence number.
+ *
+ * Fixed depth (default 3 packets = 60ms at 20ms/packet).
+ * Late packets are dropped. Missing packets cause the buffer to skip ahead
+ * when it reaches max depth.
+ */
+export declare class JitterBuffer {
+    private buffer;
+    private lastDeliveredSeq;
+    private nextExpectedSeq;
+    private maxDepth;
+    constructor(maxDepth?: number);
+    push(packet: RtpPacket): void;
+    pull(): RtpPacket | null;
+    private isOlder;
+}
+//# sourceMappingURL=jitter_buffer.d.ts.map

package/dist/rtp/jitter_buffer.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"jitter_buffer.d.ts","sourceRoot":"","sources":["../../src/rtp/jitter_buffer.ts"],"names":[],"mappings":"AAAA,OAAO,KAAK,EAAE,SAAS,EAAE,MAAM,iBAAiB,CAAC;AAEjD;;;;;;GAMG;AACH,qBAAa,YAAY;IACvB,OAAO,CAAC,MAAM,CAAqC;IACnD,OAAO,CAAC,gBAAgB,CAAc;IACtC,OAAO,CAAC,eAAe,CAAc;IACrC,OAAO,CAAC,QAAQ,CAAS;gBAEb,QAAQ,GAAE,MAAU;IAIhC,IAAI,CAAC,MAAM,EAAE,SAAS,GAAG,IAAI;IAe7B,IAAI,IAAI,SAAS,GAAG,IAAI;IA0BxB,OAAO,CAAC,OAAO;CAIhB"}

package/dist/rtp/jitter_buffer.js

@@ -0,0 +1,51 @@
+/**
+ * Simple jitter buffer that reorders RTP packets by sequence number.
+ *
+ * Fixed depth (default 3 packets = 60ms at 20ms/packet).
+ * Late packets are dropped. Missing packets cause the buffer to skip ahead
+ * when it reaches max depth.
+ */
+export class JitterBuffer {
+    buffer = new Map();
+    lastDeliveredSeq = -1;
+    nextExpectedSeq = -1;
+    maxDepth;
+    constructor(maxDepth = 3) {
+        this.maxDepth = maxDepth;
+    }
+    push(packet) {
+        if (this.lastDeliveredSeq >= 0 &&
+            this.isOlder(packet.sequenceNumber, this.lastDeliveredSeq)) {
+            return; // Late packet, drop
+        }
+        this.buffer.set(packet.sequenceNumber, packet);
+        if (this.nextExpectedSeq < 0) {
+            this.nextExpectedSeq = packet.sequenceNumber;
+        }
+    }
+    pull() {
+        if (this.buffer.has(this.nextExpectedSeq)) {
+            const packet = this.buffer.get(this.nextExpectedSeq);
+            this.buffer.delete(this.nextExpectedSeq);
+            this.lastDeliveredSeq = this.nextExpectedSeq;
+            this.nextExpectedSeq = (this.nextExpectedSeq + 1) & 0xffff;
+            return packet;
+        }
+        // Buffer full -- skip to oldest available
+        if (this.buffer.size >= this.maxDepth) {
+            const sorted = Array.from(this.buffer.keys()).sort((a, b) => this.isOlder(a, b) ? -1 : 1);
+            const oldestSeq = sorted[0];
+            const packet = this.buffer.get(oldestSeq);
+            this.buffer.delete(oldestSeq);
+            this.lastDeliveredSeq = oldestSeq;
+            this.nextExpectedSeq = (oldestSeq + 1) & 0xffff;
+            return packet;
+        }
+        return null;
+    }
+    isOlder(a, b) {
+        const diff = (b - a) & 0xffff;
+        return diff > 0 && diff < 0x8000;
+    }
+}
+//# sourceMappingURL=jitter_buffer.js.map

package/dist/rtp/jitter_buffer.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"jitter_buffer.js","sourceRoot":"","sources":["../../src/rtp/jitter_buffer.ts"],"names":[],"mappings":"AAEA;;;;;;GAMG;AACH,MAAM,OAAO,YAAY;IACf,MAAM,GAA2B,IAAI,GAAG,EAAE,CAAC;IAC3C,gBAAgB,GAAW,CAAC,CAAC,CAAC;IAC9B,eAAe,GAAW,CAAC,CAAC,CAAC;IAC7B,QAAQ,CAAS;IAEzB,YAAY,WAAmB,CAAC;QAC9B,IAAI,CAAC,QAAQ,GAAG,QAAQ,CAAC;IAC3B,CAAC;IAED,IAAI,CAAC,MAAiB;QACpB,IACE,IAAI,CAAC,gBAAgB,IAAI,CAAC;YAC1B,IAAI,CAAC,OAAO,CAAC,MAAM,CAAC,cAAc,EAAE,IAAI,CAAC,gBAAgB,CAAC,EAC1D,CAAC;YACD,OAAO,CAAC,oBAAoB;QAC9B,CAAC;QAED,IAAI,CAAC,MAAM,CAAC,GAAG,CAAC,MAAM,CAAC,cAAc,EAAE,MAAM,CAAC,CAAC;QAE/C,IAAI,IAAI,CAAC,eAAe,GAAG,CAAC,EAAE,CAAC;YAC7B,IAAI,CAAC,eAAe,GAAG,MAAM,CAAC,cAAc,CAAC;QAC/C,CAAC;IACH,CAAC;IAED,IAAI;QACF,IAAI,IAAI,CAAC,MAAM,CAAC,GAAG,CAAC,IAAI,CAAC,eAAe,CAAC,EAAE,CAAC;YAC1C,MAAM,MAAM,GAAG,IAAI,CAAC,MAAM,CAAC,GAAG,CAAC,IAAI,CAAC,eAAe,CAAE,CAAC;YACtD,IAAI,CAAC,MAAM,CAAC,MAAM,CAAC,IAAI,CAAC,eAAe,CAAC,CAAC;YACzC,IAAI,CAAC,gBAAgB,GAAG,IAAI,CAAC,eAAe,CAAC;YAC7C,IAAI,CAAC,eAAe,GAAG,CAAC,IAAI,CAAC,eAAe,GAAG,CAAC,CAAC,GAAG,MAAM,CAAC;YAC3D,OAAO,MAAM,CAAC;QAChB,CAAC;QAED,0CAA0C;QAC1C,IAAI,IAAI,CAAC,MAAM,CAAC,IAAI,IAAI,IAAI,CAAC,QAAQ,EAAE,CAAC;YACtC,MAAM,MAAM,GAAG,KAAK,CAAC,IAAI,CAAC,IAAI,CAAC,MAAM,CAAC,IAAI,EAAE,CAAC,CAAC,IAAI,CAAC,CAAC,CAAC,EAAE,CAAC,EAAE,EAAE,CAC1D,IAAI,CAAC,OAAO,CAAC,CAAC,EAAE,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC,CAC5B,CAAC;YAEF,MAAM,SAAS,GAAG,MAAM,CAAC,CAAC,CAAC,CAAC;YAC5B,MAAM,MAAM,GAAG,IAAI,CAAC,MAAM,CAAC,GAAG,CAAC,SAAS,CAAE,CAAC;YAC3C,IAAI,CAAC,MAAM,CAAC,MAAM,CAAC,SAAS,CAAC,CAAC;YAC9B,IAAI,CAAC,gBAAgB,GAAG,SAAS,CAAC;YAClC,IAAI,CAAC,eAAe,GAAG,CAAC,SAAS,GAAG,CAAC,CAAC,GAAG,MAAM,CAAC;YAChD,OAAO,MAAM,CAAC;QAChB,CAAC;QAED,OAAO,IAAI,CAAC;IACd,CAAC;IAEO,OAAO,CAAC,CAAS,EAAE,CAAS;QAClC,MAAM,IAAI,GAAG,CAAC,CAAC,GAAG,CAAC,CAAC,GAAG,MAAM,CAAC;QAC9B,OAAO,IAAI,GAAG,CAAC,IAAI,IAAI,GAAG,MAAM,CAAC;IACnC,CAAC;CACF"}

package/dist/rtp/rtp_packet.d.ts

@@ -0,0 +1,23 @@
+/**
+ * RTP packet parser and builder (RFC 3550).
+ *
+ * For G.711 at 8000 Hz with 20ms packets:
+ *   Payload: 160 bytes (160 samples * 1 byte per G.711 sample)
+ *   Timestamp increment: 160 per packet
+ *   Total: 12 (header) + 160 (payload) = 172 bytes
+ */
+export interface RtpPacket {
+    version: number;
+    padding: boolean;
+    extension: boolean;
+    csrcCount: number;
+    marker: boolean;
+    payloadType: number;
+    sequenceNumber: number;
+    timestamp: number;
+    ssrc: number;
+    payload: Uint8Array;
+}
+export declare function parseRtpPacket(data: Buffer): RtpPacket | null;
+export declare function buildRtpPacket(payloadType: number, sequenceNumber: number, timestamp: number, ssrc: number, payload: Uint8Array, marker?: boolean): Buffer;
+//# sourceMappingURL=rtp_packet.d.ts.map

package/dist/rtp/rtp_packet.d.ts.map

@@ -0,0 +1 @@
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package/dist/rtp/rtp_packet.js

@@ -0,0 +1,63 @@
+/**
+ * RTP packet parser and builder (RFC 3550).
+ *
+ * For G.711 at 8000 Hz with 20ms packets:
+ *   Payload: 160 bytes (160 samples * 1 byte per G.711 sample)
+ *   Timestamp increment: 160 per packet
+ *   Total: 12 (header) + 160 (payload) = 172 bytes
+ */
+export function parseRtpPacket(data) {
+    if (data.length < 12)
+        return null;
+    const firstByte = data[0];
+    const version = (firstByte >> 6) & 0x03;
+    if (version !== 2)
+        return null;
+    const padding = ((firstByte >> 5) & 0x01) === 1;
+    const extension = ((firstByte >> 4) & 0x01) === 1;
+    const csrcCount = firstByte & 0x0f;
+    const secondByte = data[1];
+    const marker = ((secondByte >> 7) & 0x01) === 1;
+    const payloadType = secondByte & 0x7f;
+    const sequenceNumber = data.readUInt16BE(2);
+    const timestamp = data.readUInt32BE(4);
+    const ssrc = data.readUInt32BE(8);
+    const headerLength = 12 + csrcCount * 4;
+    let payloadOffset = headerLength;
+    if (extension && data.length > headerLength + 4) {
+        const extensionLength = data.readUInt16BE(headerLength + 2);
+        payloadOffset = headerLength + 4 + extensionLength * 4;
+    }
+    if (payloadOffset >= data.length)
+        return null;
+    let payloadLength = data.length - payloadOffset;
+    if (padding && payloadLength > 0) {
+        const paddingLength = data[data.length - 1];
+        payloadLength -= paddingLength;
+    }
+    if (payloadLength <= 0)
+        return null;
+    const payload = new Uint8Array(data.buffer, data.byteOffset + payloadOffset, payloadLength);
+    return {
+        version,
+        padding,
+        extension,
+        csrcCount,
+        marker,
+        payloadType,
+        sequenceNumber,
+        timestamp,
+        ssrc,
+        payload,
+    };
+}
+export function buildRtpPacket(payloadType, sequenceNumber, timestamp, ssrc, payload, marker = false) {
+    const header = Buffer.alloc(12);
+    header[0] = 0x80; // Version=2
+    header[1] = (marker ? 0x80 : 0x00) | (payloadType & 0x7f);
+    header.writeUInt16BE(sequenceNumber & 0xffff, 2);
+    header.writeUInt32BE(timestamp >>> 0, 4);
+    header.writeUInt32BE(ssrc >>> 0, 8);
+    return Buffer.concat([header, Buffer.from(payload)]);
+}
+//# sourceMappingURL=rtp_packet.js.map

package/dist/rtp/rtp_packet.js.map

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package/dist/rtp/rtp_session.d.ts

@@ -0,0 +1,54 @@
+import { EventEmitter } from 'node:events';
+import { type Codec } from '@ariaflow/livekit-plugin/codec/g711';
+export interface RtpSessionOptions {
+    localPort: number;
+    jitterDepth?: number;
+    packetDurationMs?: number;
+    /**
+     * When true, outbound RTP is clocked at `packetDurationMs` (default 20ms)
+     * regardless of application timing: queue drains one frame per tick; idle
+     * ticks send codec-encoded silence. Default false preserves immediate
+     * `sendAudio` behavior.
+     */
+    continuousPacing?: boolean;
+}
+/**
+ * Manages a bidirectional RTP stream over UDP.
+ *
+ * Inbound: receives UDP datagrams, parses RTP, passes through jitter buffer,
+ * decodes with codec selected by RTP payload type (PCMU=0, PCMA=8), emits
+ * 'audio' events with PCM Int16Array data.
+ *
+ * Outbound: accepts PCM Int16Array, encodes with the negotiated outbound
+ * codec, builds RTP packets, sends as UDP datagrams.
+ */
+export declare class RtpSession extends EventEmitter {
+    private socket;
+    private jitterBuffer;
+    private outboundCodec;
+    private sendSequenceNumber;
+    private sendTimestamp;
+    private sendSsrc;
+    private remoteAddress;
+    private remotePort;
+    private samplesPerPacket;
+    private packetDurationMs;
+    private continuousPacing;
+    private pacedQueue;
+    private pacePending;
+    private paceTimer;
+    private nextPaceTime;
+    constructor(codec: Codec, options: RtpSessionOptions);
+    private decodeByPayloadType;
+    setRemote(address: string, port: number): void;
+    private sendRtpPayload;
+    /**
+     * Append PCM to the paced buffer and queue full 20ms (or packetDurationMs) frames.
+     */
+    private enqueuePacedPcm;
+    private runPacedTick;
+    private schedulePaceTick;
+    sendAudio(pcm: Int16Array): void;
+    close(): void;
+}
+//# sourceMappingURL=rtp_session.d.ts.map

package/dist/rtp/rtp_session.d.ts.map

@@ -0,0 +1 @@
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package/dist/rtp/rtp_session.js

@@ -0,0 +1,162 @@
+import * as dgram from 'node:dgram';
+import { EventEmitter } from 'node:events';
+import { parseRtpPacket, buildRtpPacket, } from './rtp_packet.js';
+import { JitterBuffer } from './jitter_buffer.js';
+import { PCMU, PCMA, } from '@ariaflow/livekit-plugin/codec/g711';
+/**
+ * Manages a bidirectional RTP stream over UDP.
+ *
+ * Inbound: receives UDP datagrams, parses RTP, passes through jitter buffer,
+ * decodes with codec selected by RTP payload type (PCMU=0, PCMA=8), emits
+ * 'audio' events with PCM Int16Array data.
+ *
+ * Outbound: accepts PCM Int16Array, encodes with the negotiated outbound
+ * codec, builds RTP packets, sends as UDP datagrams.
+ */
+export class RtpSession extends EventEmitter {
+    socket;
+    jitterBuffer;
+    outboundCodec;
+    sendSequenceNumber = 0;
+    sendTimestamp = 0;
+    sendSsrc;
+    remoteAddress = '';
+    remotePort = 0;
+    samplesPerPacket;
+    packetDurationMs;
+    continuousPacing;
+    pacedQueue = [];
+    pacePending = new Int16Array(0);
+    paceTimer = null;
+    nextPaceTime = 0;
+    constructor(codec, options) {
+        super();
+        this.outboundCodec = codec;
+        this.jitterBuffer = new JitterBuffer(options.jitterDepth ?? 3);
+        this.sendSsrc = Math.floor(Math.random() * 0xffffffff);
+        this.packetDurationMs = options.packetDurationMs ?? 20;
+        this.samplesPerPacket =
+            (codec.sampleRate * this.packetDurationMs) / 1000;
+        this.continuousPacing = options.continuousPacing === true;
+        this.socket = dgram.createSocket('udp4');
+        this.socket.on('message', (msg, rinfo) => {
+            if (!this.remoteAddress) {
+                this.remoteAddress = rinfo.address;
+                this.remotePort = rinfo.port;
+            }
+            const packet = parseRtpPacket(msg);
+            if (!packet)
+                return;
+            this.jitterBuffer.push(packet);
+            let buffered;
+            while ((buffered = this.jitterBuffer.pull()) !== null) {
+                const pcm = this.decodeByPayloadType(buffered.payloadType, buffered.payload);
+                this.emit('audio', pcm);
+            }
+        });
+        this.socket.on('error', (err) => {
+            console.error('[RtpSession] UDP socket error:', err.message);
+        });
+        this.socket.bind(options.localPort);
+        if (this.continuousPacing) {
+            // First `schedulePaceTick` advances by one `packetDurationMs`; start from "now"
+            // so the initial delay is exactly one frame (not two).
+            this.nextPaceTime = performance.now();
+            this.schedulePaceTick();
+        }
+    }
+    decodeByPayloadType(payloadType, payload) {
+        if (payloadType === PCMU.payloadType) {
+            return PCMU.decode(payload);
+        }
+        if (payloadType === PCMA.payloadType) {
+            return PCMA.decode(payload);
+        }
+        return this.outboundCodec.decode(payload);
+    }
+    setRemote(address, port) {
+        this.remoteAddress = address;
+        this.remotePort = port;
+    }
+    sendRtpPayload(encoded, samplesThisPacket) {
+        if (!this.remoteAddress || !this.remotePort)
+            return;
+        const packet = buildRtpPacket(this.outboundCodec.payloadType, this.sendSequenceNumber, this.sendTimestamp, this.sendSsrc, encoded, this.sendSequenceNumber === 0);
+        this.socket.send(packet, this.remotePort, this.remoteAddress);
+        this.sendSequenceNumber = (this.sendSequenceNumber + 1) & 0xffff;
+        this.sendTimestamp =
+            (this.sendTimestamp + samplesThisPacket) >>> 0;
+    }
+    /**
+     * Append PCM to the paced buffer and queue full 20ms (or packetDurationMs) frames.
+     */
+    enqueuePacedPcm(pcm) {
+        const merged = this.pacePending.length === 0
+            ? pcm
+            : concatInt16(this.pacePending, pcm);
+        let offset = 0;
+        while (offset + this.samplesPerPacket <= merged.length) {
+            this.pacedQueue.push(merged.subarray(offset, offset + this.samplesPerPacket));
+            offset += this.samplesPerPacket;
+        }
+        this.pacePending =
+            offset === merged.length
+                ? new Int16Array(0)
+                : merged.subarray(offset);
+    }
+    runPacedTick() {
+        if (!this.remoteAddress || !this.remotePort) {
+            return;
+        }
+        const pcm = this.pacedQueue.length > 0
+            ? this.pacedQueue.shift()
+            : new Int16Array(this.samplesPerPacket);
+        const encoded = this.outboundCodec.encode(pcm);
+        this.sendRtpPayload(encoded, pcm.length);
+    }
+    schedulePaceTick() {
+        const now = performance.now();
+        const drift = now - this.nextPaceTime;
+        if (drift > 2 * this.packetDurationMs) {
+            this.nextPaceTime = now + this.packetDurationMs;
+        }
+        else {
+            this.nextPaceTime += this.packetDurationMs;
+        }
+        const delay = Math.max(0, this.nextPaceTime - performance.now());
+        this.paceTimer = setTimeout(() => {
+            this.paceTimer = null;
+            this.runPacedTick();
+            this.schedulePaceTick();
+        }, delay);
+    }
+    sendAudio(pcm) {
+        if (this.continuousPacing) {
+            this.enqueuePacedPcm(pcm);
+            return;
+        }
+        const encoded = this.outboundCodec.encode(pcm);
+        this.sendRtpPayload(encoded, this.samplesPerPacket);
+    }
+    close() {
+        if (this.paceTimer !== null) {
+            clearTimeout(this.paceTimer);
+            this.paceTimer = null;
+        }
+        this.pacedQueue = [];
+        this.pacePending = new Int16Array(0);
+        try {
+            this.socket.close();
+        }
+        catch {
+            // Already closed
+        }
+    }
+}
+function concatInt16(a, b) {
+    const out = new Int16Array(a.length + b.length);
+    out.set(a, 0);
+    out.set(b, a.length);
+    return out;
+}
+//# sourceMappingURL=rtp_session.js.map