@ariaflowagents/livekit-plugin-transport-sip 0.9.0

package/dist/rtp/rtp_session.js.map

@@ -0,0 +1 @@
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package/dist/sdp_g711.d.ts

@@ -0,0 +1,18 @@
+/**
+ * G.711 (PCMU/PCMA) SDP negotiation for SIP UAS answers.
+ *
+ * Parses the remote offer’s m=audio payload type list (preference order).
+ * PT 0 and PT 8 are the static assignments for PCMU and PCMA (RFC 3551).
+ */
+import { type Codec } from '@ariaflow/livekit-plugin/codec/g711';
+/**
+ * Choose G.711 codec for this call from the remote SDP offer.
+ * Respects remote m= line preference order; if both PCMU and PCMA are offered,
+ * uses `fallback` (server preferred codec) when that codec appears in the offer.
+ */
+export declare function negotiateG711FromRemoteOffer(remoteSdp: string, fallback: 'PCMU' | 'PCMA'): Codec;
+/**
+ * SDP answer body advertising a single negotiated G.711 codec (RFC 3264).
+ */
+export declare function buildG711SdpAnswer(localAddress: string, rtpPort: number, codec: Codec): string;
+//# sourceMappingURL=sdp_g711.d.ts.map

package/dist/sdp_g711.d.ts.map

@@ -0,0 +1 @@
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package/dist/sdp_g711.js

@@ -0,0 +1,74 @@
+/**
+ * G.711 (PCMU/PCMA) SDP negotiation for SIP UAS answers.
+ *
+ * Parses the remote offer’s m=audio payload type list (preference order).
+ * PT 0 and PT 8 are the static assignments for PCMU and PCMA (RFC 3551).
+ */
+import { PCMU, PCMA } from '@ariaflow/livekit-plugin/codec/g711';
+function parseAudioPayloadTypesFromMLine(remoteSdp) {
+    const lines = remoteSdp.split(/\r?\n/);
+    for (const line of lines) {
+        const m = line.match(/^m=audio\s+\d+\s+\S+\s+(.+)$/i);
+        if (!m)
+            continue;
+        return m[1]
+            .trim()
+            .split(/\s+/)
+            .map((t) => parseInt(t, 10))
+            .filter((n) => !Number.isNaN(n));
+    }
+    return [];
+}
+function codecFromStaticPt(pt) {
+    if (pt === 0)
+        return PCMU;
+    if (pt === 8)
+        return PCMA;
+    return null;
+}
+/**
+ * Choose G.711 codec for this call from the remote SDP offer.
+ * Respects remote m= line preference order; if both PCMU and PCMA are offered,
+ * uses `fallback` (server preferred codec) when that codec appears in the offer.
+ */
+export function negotiateG711FromRemoteOffer(remoteSdp, fallback) {
+    fallback = fallback === 'PCMA' ? 'PCMA' : 'PCMU';
+    const mPts = parseAudioPayloadTypesFromMLine(remoteSdp);
+    const fallbackCodec = fallback === 'PCMA' ? PCMA : PCMU;
+    const g711InOrder = [];
+    for (const pt of mPts) {
+        const c = codecFromStaticPt(pt);
+        if (c && !g711InOrder.some((x) => x.payloadType === c.payloadType)) {
+            g711InOrder.push(c);
+        }
+    }
+    if (g711InOrder.length === 0)
+        return fallbackCodec;
+    if (g711InOrder.length === 1)
+        return g711InOrder[0];
+    const preferred = fallbackCodec;
+    if (g711InOrder.some((c) => c.payloadType === preferred.payloadType)) {
+        return preferred;
+    }
+    return g711InOrder[0];
+}
+/**
+ * SDP answer body advertising a single negotiated G.711 codec (RFC 3264).
+ */
+export function buildG711SdpAnswer(localAddress, rtpPort, codec) {
+    const pt = codec.payloadType;
+    const rtpmapLine = codec.payloadType === 0
+        ? `a=rtpmap:0 PCMU/8000\r\n`
+        : `a=rtpmap:8 PCMA/8000\r\n`;
+    return (`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 ${pt}\r\n` +
+        rtpmapLine +
+        `a=fmtp:${pt}\r\n` +
+        `a=ptime:20\r\n` +
+        `a=sendrecv\r\n`);
+}
+//# sourceMappingURL=sdp_g711.js.map

package/dist/sdp_g711.js.map

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

@@ -0,0 +1,76 @@
+/**
+ * SIP Agent Server for AriaFlow Voice Agents.
+ *
+ * Production SIP server for SIP trunking and RTP telephony.
+ *
+ * UDP Transport:
+ * - Direct SIP signaling over UDP (port 5060)
+ * - Works with PBX systems and SIP gateways
+ * - G.711 codec negotiation
+ *
+ * For WebSocket/WebRTC SIP signaling, use
+ * `@ariaflow/livekit-plugin-transport-sip-jssip`.
+ */
+import { SessionManager, type AriaFlowVoiceSession } from '@ariaflow/livekit-plugin';
+import type { voice } from '@livekit/agents';
+import { SIPTransportAdapter } from './transport_adapter.js';
+import type { SIPServerOptions } from './types.js';
+/**
+ * A SIP server that accepts incoming calls and creates agent sessions.
+ *
+ * This server handles SIP signaling (via UDP or WebSocket) and manages
+ * RTP sessions for audio transmission.
+ *
+ * UDP Example:
+ *   const server = new SIPAgentServer({
+ *     transport: 'udp',
+ *     localAddress: '10.0.0.5',
+ *     sipPort: 5060,
+ *     rtpPortStart: 10000,
+ *   });
+ *
+ */
+export declare class SIPAgentServer {
+    private options;
+    private sessionManager;
+    private callHandler;
+    private signaling;
+    private nextRtpPort;
+    private activeTransports;
+    private transportType;
+    private activeVoiceSessions;
+    constructor(options: SIPServerOptions);
+    onCall(handler: (adapter: SIPTransportAdapter, callId: string) => void | Promise<void>): void;
+    startSession(callId: string, voiceSession: AriaFlowVoiceSession): Promise<voice.AgentSession>;
+    /**
+     * Allocate a local RTP port and create a transport adapter.
+     * This is called internally when a SIP INVITE is received.
+     */
+    private createTransport;
+    /**
+     * Start listening for SIP INVITE requests.
+     *
+     * For UDP transport: Binds UDP socket and listens for SIP messages.
+     * For WebSocket transport: Connects to WebSocket server and listens for SIP messages.
+     */
+    listen(): Promise<void>;
+    /**
+     * Hang up an active call.
+     */
+    hangup(callId: string): Promise<void>;
+    get sessions(): SessionManager;
+    /**
+     * Get the RTP port for a specific call.
+     */
+    getRtpPort(callId: string): number | undefined;
+    /**
+     * SIP trunk server is connectionless once listening on UDP.
+     */
+    get isRegistered(): boolean;
+    /**
+     * SIP trunk server status.
+     */
+    get status(): 'disconnected' | 'connecting' | 'connected';
+    close(): Promise<void>;
+}
+//# sourceMappingURL=server.d.ts.map

package/dist/server.d.ts.map

@@ -0,0 +1 @@
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package/dist/server.js

@@ -0,0 +1,197 @@
+/**
+ * SIP Agent Server for AriaFlow Voice Agents.
+ *
+ * Production SIP server for SIP trunking and RTP telephony.
+ *
+ * UDP Transport:
+ * - Direct SIP signaling over UDP (port 5060)
+ * - Works with PBX systems and SIP gateways
+ * - G.711 codec negotiation
+ *
+ * For WebSocket/WebRTC SIP signaling, use
+ * `@ariaflow/livekit-plugin-transport-sip-jssip`.
+ */
+import { SessionManager } from '@ariaflow/livekit-plugin';
+import { SIPTransportAdapter } from './transport_adapter.js';
+import { RtpSession } from './rtp/rtp_session.js';
+import { SIPSignaling } from './sip_signaling.js';
+/**
+ * Parse the remote RTP endpoint from an SDP body.
+ * Extracts the connection address (c= line) and audio port (m= line).
+ */
+function parseSdpRemoteEndpoint(sdp) {
+    const connectionMatch = sdp.match(/c=IN\s+IP[46]\s+(\S+)/);
+    if (!connectionMatch)
+        return null;
+    const host = connectionMatch[1];
+    const mediaMatch = sdp.match(/m=audio\s+(\d+)/);
+    if (!mediaMatch)
+        return null;
+    const port = parseInt(mediaMatch[1], 10);
+    if (isNaN(port) || port <= 0 || port > 65535)
+        return null;
+    return { host, port };
+}
+/**
+ * A SIP server that accepts incoming calls and creates agent sessions.
+ *
+ * This server handles SIP signaling (via UDP or WebSocket) and manages
+ * RTP sessions for audio transmission.
+ *
+ * UDP Example:
+ *   const server = new SIPAgentServer({
+ *     transport: 'udp',
+ *     localAddress: '10.0.0.5',
+ *     sipPort: 5060,
+ *     rtpPortStart: 10000,
+ *   });
+ *
+ */
+export class SIPAgentServer {
+    options;
+    sessionManager = new SessionManager();
+    callHandler = null;
+    signaling;
+    nextRtpPort;
+    activeTransports = new Map();
+    transportType;
+    activeVoiceSessions = new Map();
+    constructor(options) {
+        this.options = options;
+        this.transportType = options.transport ?? 'udp';
+        this.nextRtpPort = options.rtpPortStart ?? 10000;
+        if (this.transportType === 'websocket') {
+            throw new Error('[SIPAgentServer] WebSocket SIP signaling was moved to @ariaflow/livekit-plugin-transport-sip-jssip. ' +
+                'Use transport: "udp" in this package for RTP telephony.');
+        }
+        this.signaling = new SIPSignaling(options);
+    }
+    onCall(handler) {
+        this.callHandler = handler;
+    }
+    async startSession(callId, voiceSession) {
+        const transport = this.activeTransports.get(callId);
+        if (!transport) {
+            throw new Error(`Transport not found for call: ${callId}`);
+        }
+        const agentSession = await this.sessionManager.startSession(transport, voiceSession);
+        this.activeVoiceSessions.set(callId, voiceSession);
+        return agentSession;
+    }
+    /**
+     * Allocate a local RTP port and create a transport adapter.
+     * This is called internally when a SIP INVITE is received.
+     */
+    createTransport(callId, rtpPort, remoteSdp, negotiatedCodec) {
+        const rtpSession = new RtpSession(negotiatedCodec, {
+            localPort: rtpPort,
+            continuousPacing: this.options.continuousPacing === true,
+        });
+        // Parse the remote SDP to extract RTP endpoint so the agent can
+        // send audio before the remote sends first (e.g., greeting).
+        const remoteEndpoint = parseSdpRemoteEndpoint(remoteSdp);
+        if (remoteEndpoint) {
+            rtpSession.setRemote(remoteEndpoint.host, remoteEndpoint.port);
+        }
+        else {
+            console.warn(`[SIPAgentServer] Could not parse remote RTP endpoint from SDP for call ${callId}. ` +
+                'Agent-initiated audio will not work until the remote sends first.');
+        }
+        return new SIPTransportAdapter(rtpSession, negotiatedCodec, {
+            id: callId,
+        });
+    }
+    /**
+     * Start listening for SIP INVITE requests.
+     *
+     * For UDP transport: Binds UDP socket and listens for SIP messages.
+     * For WebSocket transport: Connects to WebSocket server and listens for SIP messages.
+     */
+    async listen() {
+        const transportName = this.transportType === 'websocket' ? 'WebSocket' : 'UDP';
+        console.log(`[SIPAgentServer] Starting SIP server using ${transportName} transport`);
+        await this.signaling.start(async (callId, remoteSdp, rtpPort, negotiatedCodec) => {
+            console.log(`[SIPAgentServer] Creating transport for call: ${callId} on RTP port ${rtpPort}`);
+            // Create transport with the allocated RTP port and remote SDP
+            const transport = this.createTransport(callId, rtpPort, remoteSdp, negotiatedCodec);
+            this.activeTransports.set(callId, transport);
+            // Invoke the call handler
+            if (this.callHandler) {
+                try {
+                    await this.callHandler(transport, callId);
+                }
+                catch (error) {
+                    console.error(`[SIPAgentServer] Error in call handler for ${callId}:`, error);
+                    this.activeTransports.delete(callId);
+                    await transport.close();
+                    this.activeVoiceSessions.delete(callId);
+                    throw error;
+                }
+            }
+        }, async (callId) => {
+            // Handle BYE from remote party
+            console.log(`[SIPAgentServer] Remote party hung up call: ${callId}`);
+            const transport = this.activeTransports.get(callId);
+            if (transport) {
+                await transport.close();
+                this.activeTransports.delete(callId);
+            }
+            const voiceSession = this.activeVoiceSessions.get(callId);
+            if (voiceSession) {
+                await voiceSession.close();
+                this.activeVoiceSessions.delete(callId);
+            }
+        });
+        console.log(`[SIPAgentServer] Listening for calls (${transportName})`);
+    }
+    /**
+     * Hang up an active call.
+     */
+    async hangup(callId) {
+        const transport = this.activeTransports.get(callId);
+        if (transport) {
+            await transport.close();
+            this.activeTransports.delete(callId);
+        }
+        const voiceSession = this.activeVoiceSessions.get(callId);
+        if (voiceSession) {
+            await voiceSession.close();
+            this.activeVoiceSessions.delete(callId);
+        }
+        await this.signaling.hangup(callId);
+    }
+    get sessions() {
+        return this.sessionManager;
+    }
+    /**
+     * Get the RTP port for a specific call.
+     */
+    getRtpPort(callId) {
+        return this.signaling.getRtpPort?.(callId);
+    }
+    /**
+     * SIP trunk server is connectionless once listening on UDP.
+     */
+    get isRegistered() {
+        return true;
+    }
+    /**
+     * SIP trunk server status.
+     */
+    get status() {
+        return 'connected';
+    }
+    async close() {
+        // Close all active voice sessions
+        const voiceSessionPromises = Array.from(this.activeVoiceSessions.values()).map((s) => s.close());
+        await Promise.allSettled(voiceSessionPromises);
+        this.activeVoiceSessions.clear();
+        // Close all active transports
+        const closePromises = Array.from(this.activeTransports.values()).map((t) => t.close());
+        await Promise.allSettled(closePromises);
+        this.activeTransports.clear();
+        await this.sessionManager.closeAll();
+        await this.signaling.stop();
+    }
+}
+//# sourceMappingURL=server.js.map

package/dist/server.js.map

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package/dist/sip_signaling.d.ts

@@ -0,0 +1,73 @@
+/**
+ * SIP signaling implementation using node.js-sip as a parser/transport shim.
+ *
+ * We intentionally construct server-side SIP responses and in-dialog requests
+ * ourselves rather than relying on node.js-sip's higher-level UAS helpers.
+ * Those helpers assume a different server/auth flow than this package uses,
+ * while this transport needs explicit control over provisional responses,
+ * pending INVITE cancellation, and dialog-correct BYE generation.
+ */
+import type { Codec } from '@ariaflow/livekit-plugin/codec/g711';
+import type { SIPServerOptions } from './types.js';
+/**
+ * Callback for incoming INVITE requests.
+ *
+ * The callback must finish transport/session bootstrap before the final 200 OK
+ * is emitted. Throwing rejects the INVITE with a 500 response.
+ */
+export type OnInviteCallback = (callId: string, remoteSdp: string, rtpPort: number, negotiatedCodec: Codec) => void | Promise<void>;
+/**
+ * Callback for call termination (remote BYE, local CANCEL unwind, etc.).
+ */
+export type OnByeCallback = (callId: string) => void | Promise<void>;
+export declare class SIPSignaling {
+    private options;
+    private voip;
+    private activeCalls;
+    private pendingInvites;
+    private nextRtpPort;
+    private localAddress;
+    private localPort;
+    private g711Preferred;
+    private ready;
+    private readyResolve;
+    private transportMessageListener;
+    private static readonly MIN_RTP_PORT;
+    private static readonly MAX_RTP_PORT;
+    private onInviteCallback;
+    private onByeCallback;
+    constructor(options: SIPServerOptions);
+    start(onInvite: OnInviteCallback, onBye?: OnByeCallback): Promise<void>;
+    private handleTransportMessage;
+    private handleInviteRequest;
+    private handleByeRequest;
+    private handleCancelRequest;
+    private handleOptionsRequest;
+    private handleRegisterRequest;
+    hangup(callId: string): Promise<void>;
+    stop(): Promise<void>;
+    getRtpPort(callId: string): number | undefined;
+    private sendResponse;
+    private buildResponseMessage;
+    private buildRequestMessage;
+    private sendToPeer;
+    private allocateRtpPort;
+    private parseHeaders;
+    private isSocketBound;
+    private getResponseTarget;
+    private extractCallId;
+    private extractCseq;
+    private buildLocalUri;
+    private buildLocalContactUri;
+    private extractRemoteUri;
+    private extractRemoteTargetUri;
+    private buildFallbackRemoteUri;
+    private extractHeaderUri;
+    private normalizeSipUri;
+    private buildAddressHeader;
+    private ensureTag;
+    private toPort;
+    private generateTag;
+    private generateBranch;
+}
+//# sourceMappingURL=sip_signaling.d.ts.map

package/dist/sip_signaling.d.ts.map

@@ -0,0 +1 @@
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