@ariaflowagents/livekit-plugin-transport-sip 0.9.0

package/dist/sip_signaling.js

@@ -0,0 +1,553 @@
+/**
+ * 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 { createRequire } from 'node:module';
+import { buildG711SdpAnswer, negotiateG711FromRemoteOffer, } from './sdp_g711.js';
+const require = createRequire(import.meta.url);
+const VOIP = require('node.js-sip');
+const SIP = require('node.js-sip/SIP/index.js');
+const DEFAULT_SIP_PORT = 5060;
+const USER_AGENT = 'ariaflow-sip';
+const ALLOW_METHODS = 'INVITE, ACK, BYE, CANCEL, OPTIONS';
+export class SIPSignaling {
+    options;
+    voip = null;
+    activeCalls = new Map();
+    pendingInvites = new Map();
+    nextRtpPort;
+    localAddress;
+    localPort;
+    g711Preferred;
+    ready = false;
+    readyResolve = null;
+    transportMessageListener = null;
+    static MIN_RTP_PORT = 10000;
+    static MAX_RTP_PORT = 60000;
+    onInviteCallback = () => { };
+    onByeCallback = () => { };
+    constructor(options) {
+        this.options = options;
+        this.localAddress = options.localAddress;
+        this.localPort = options.sipPort ?? DEFAULT_SIP_PORT;
+        this.nextRtpPort = options.rtpPortStart ?? 10000;
+        this.g711Preferred = options.codec === 'PCMA' ? 'PCMA' : 'PCMU';
+        if (options.sipProtocol && options.sipProtocol !== 'udp') {
+            throw new Error(`[SIPSignaling] sipProtocol="${options.sipProtocol}" is not supported. ` +
+                'This transport currently supports UDP only.');
+        }
+    }
+    async start(onInvite, onBye) {
+        this.onInviteCallback = onInvite;
+        this.onByeCallback = onBye ?? (() => { });
+        return new Promise((resolve, reject) => {
+            const readyTimeout = setTimeout(() => {
+                reject(new Error(`[SIPSignaling] Server did not become ready within 10 seconds on ` +
+                    `${this.localAddress}:${this.localPort}`));
+            }, 10_000);
+            this.readyResolve = () => {
+                clearTimeout(readyTimeout);
+                resolve();
+            };
+            try {
+                this.voip = new VOIP({
+                    type: 'server',
+                    transport: {
+                        type: 'UDP',
+                        port: this.localPort,
+                    },
+                }, (data) => {
+                    if (data.type === 'UAS_READY') {
+                        console.log(`[SIPSignaling] UAS_READY - Server ready on ${this.localAddress}:${this.localPort}`);
+                    }
+                });
+                if (this.voip?.transport?.socket?.on) {
+                    const socket = this.voip.transport.socket;
+                    const markReady = () => {
+                        if (this.ready)
+                            return;
+                        this.ready = true;
+                        this.readyResolve?.();
+                        this.readyResolve = null;
+                    };
+                    if (this.isSocketBound(socket)) {
+                        markReady();
+                    }
+                    else {
+                        socket.once('listening', markReady);
+                    }
+                    this.transportMessageListener = (msg, rinfo) => {
+                        this.handleTransportMessage(msg.toString(), rinfo).catch((error) => {
+                            console.error('[SIPSignaling] Unhandled SIP transport error:', error);
+                        });
+                    };
+                    socket.on('message', this.transportMessageListener);
+                }
+            }
+            catch (error) {
+                clearTimeout(readyTimeout);
+                this.readyResolve = null;
+                console.error('[SIPSignaling] Error starting VOIP server:', error);
+                reject(error);
+            }
+        });
+    }
+    async handleTransportMessage(rawMessage, rinfo) {
+        const message = SIP.Parser.parse(rawMessage);
+        if (!message?.headers) {
+            return;
+        }
+        const callId = this.extractCallId(message);
+        const method = message.method?.toUpperCase();
+        switch (method) {
+            case 'INVITE':
+                await this.handleInviteRequest(message, callId, rinfo);
+                return;
+            case 'BYE':
+                await this.handleByeRequest(message, callId, rinfo);
+                return;
+            case 'CANCEL':
+                await this.handleCancelRequest(message, callId, rinfo);
+                return;
+            case 'OPTIONS':
+                this.handleOptionsRequest(message, rinfo);
+                return;
+            case 'ACK':
+                console.log(`[SIPSignaling] ACK received for call: ${callId}`);
+                return;
+            case 'REGISTER':
+                this.handleRegisterRequest(message, rinfo);
+                return;
+            default:
+                console.log(`[SIPSignaling] Event: ${method ?? 'UNKNOWN'}`);
+                return;
+        }
+    }
+    async handleInviteRequest(message, callId, rinfo) {
+        const parsedHeaders = this.parseHeaders(message.headers);
+        const responseTarget = this.getResponseTarget(parsedHeaders, rinfo);
+        if (!responseTarget) {
+            console.warn(`[SIPSignaling] INVITE ${callId} missing routable response target; rejecting`);
+            this.sendResponse(message, parsedHeaders, 400, 'Bad Request');
+            return;
+        }
+        if (this.activeCalls.has(callId) || this.pendingInvites.has(callId)) {
+            console.warn(`[SIPSignaling] Call ${callId} already exists, sending 486 Busy Here`);
+            this.sendResponse(message, parsedHeaders, 486, 'Busy Here');
+            return;
+        }
+        const remoteSdp = message.body || '';
+        const negotiatedCodec = negotiateG711FromRemoteOffer(remoteSdp, this.g711Preferred);
+        const rtpPort = this.allocateRtpPort();
+        const localTag = this.generateTag();
+        const localUri = this.buildLocalUri(parsedHeaders, message);
+        const localContactUri = this.buildLocalContactUri(parsedHeaders);
+        const remoteUri = this.extractRemoteUri(message.headers, parsedHeaders);
+        const remoteTargetUri = this.extractRemoteTargetUri(message.headers, parsedHeaders);
+        const inviteCseq = this.extractCseq(parsedHeaders);
+        const pending = {
+            callId,
+            rtpPort,
+            localTag,
+            localUri,
+            localContactUri,
+            remoteUri,
+            remoteTargetUri,
+            responseHost: responseTarget.host,
+            responsePort: responseTarget.port,
+            inviteCseq,
+            nextLocalCseq: 0,
+            rawHeaders: message.headers,
+            canceled: false,
+        };
+        this.pendingInvites.set(callId, pending);
+        const sdpAnswer = buildG711SdpAnswer(this.localAddress, rtpPort, negotiatedCodec).replace(/^[ \t]+/gm, '');
+        console.log(`[SIPSignaling] Received INVITE for call: ${callId}`);
+        this.sendResponse(message, parsedHeaders, 100, 'Trying');
+        this.sendResponse(message, parsedHeaders, 180, 'Ringing', {
+            toHeader: this.ensureTag(message.headers.To ?? '', localTag),
+        });
+        try {
+            await this.onInviteCallback(callId, remoteSdp, rtpPort, negotiatedCodec);
+        }
+        catch (error) {
+            this.pendingInvites.delete(callId);
+            this.sendResponse(message, parsedHeaders, 500, 'Server Internal Error', {
+                toHeader: this.ensureTag(message.headers.To ?? '', localTag),
+            });
+            throw error;
+        }
+        const current = this.pendingInvites.get(callId);
+        if (!current || current.canceled) {
+            if (!current) {
+                return;
+            }
+            this.pendingInvites.delete(callId);
+            return;
+        }
+        this.sendResponse(message, parsedHeaders, 200, 'OK', {
+            body: sdpAnswer,
+            toHeader: this.ensureTag(message.headers.To ?? '', localTag),
+            contactUri: localContactUri,
+        });
+        this.activeCalls.set(callId, {
+            callId,
+            rtpPort,
+            localTag,
+            remoteTag: parsedHeaders.From?.tag ?? '',
+            localUri,
+            localContactUri,
+            remoteUri,
+            remoteTargetUri,
+            responseHost: responseTarget.host,
+            responsePort: responseTarget.port,
+            nextLocalCseq: current.nextLocalCseq,
+        });
+        this.pendingInvites.delete(callId);
+    }
+    async handleByeRequest(message, callId, rinfo) {
+        const parsedHeaders = this.parseHeaders(message.headers);
+        console.log(`[SIPSignaling] Received BYE for call: ${callId}`);
+        const call = this.activeCalls.get(callId);
+        if (!call) {
+            console.warn(`[SIPSignaling] Received BYE for unknown call: ${callId}`);
+            this.sendResponse(message, parsedHeaders, 481, 'Call/Transaction Does Not Exist', {
+                targetHost: rinfo?.address,
+                targetPort: rinfo?.port,
+            });
+            return;
+        }
+        this.sendResponse(message, parsedHeaders, 200, 'OK', {
+            targetHost: rinfo?.address,
+            targetPort: rinfo?.port,
+        });
+        this.activeCalls.delete(callId);
+        await this.onByeCallback(callId);
+    }
+    async handleCancelRequest(message, callId, rinfo) {
+        const parsedHeaders = this.parseHeaders(message.headers);
+        const pending = this.pendingInvites.get(callId);
+        this.sendResponse(message, parsedHeaders, 200, 'OK', {
+            targetHost: rinfo?.address,
+            targetPort: rinfo?.port,
+        });
+        if (!pending) {
+            console.warn(`[SIPSignaling] CANCEL for unknown or established call: ${callId}`);
+            return;
+        }
+        pending.canceled = true;
+        this.pendingInvites.delete(callId);
+        const inviteParsed = this.parseHeaders(pending.rawHeaders);
+        this.sendResponse({
+            headers: pending.rawHeaders,
+            body: '',
+        }, inviteParsed, 487, 'Request Terminated', {
+            toHeader: this.ensureTag(pending.rawHeaders.To ?? '', pending.localTag),
+            targetHost: pending.responseHost,
+            targetPort: pending.responsePort,
+            contactUri: pending.localContactUri,
+        });
+        await this.onByeCallback(callId);
+    }
+    handleOptionsRequest(message, rinfo) {
+        const parsedHeaders = this.parseHeaders(message.headers);
+        this.sendResponse(message, parsedHeaders, 200, 'OK', {
+            headers: {
+                Allow: ALLOW_METHODS,
+                Accept: 'application/sdp',
+            },
+            targetHost: rinfo?.address,
+            targetPort: rinfo?.port,
+        });
+    }
+    handleRegisterRequest(message, rinfo) {
+        const parsedHeaders = this.parseHeaders(message.headers);
+        this.sendResponse(message, parsedHeaders, 405, 'Method Not Allowed', {
+            headers: {
+                Allow: ALLOW_METHODS,
+            },
+            targetHost: rinfo?.address,
+            targetPort: rinfo?.port,
+        });
+    }
+    async hangup(callId) {
+        const call = this.activeCalls.get(callId);
+        if (!call) {
+            console.warn(`[SIPSignaling] No active call: ${callId}`);
+            return;
+        }
+        call.nextLocalCseq += 1;
+        const branch = this.generateBranch();
+        const bye = this.buildRequestMessage({
+            method: 'BYE',
+            requestUri: call.remoteTargetUri,
+            headers: {
+                Via: `SIP/2.0/UDP ${this.localAddress}:${this.localPort};branch=${branch};rport`,
+                To: this.buildAddressHeader(call.remoteUri, call.remoteTag || undefined),
+                From: this.buildAddressHeader(call.localUri, call.localTag),
+                'Call-ID': callId,
+                CSeq: `${call.nextLocalCseq} BYE`,
+                Contact: `<${call.localContactUri}>`,
+                'Max-Forwards': '70',
+                'User-Agent': USER_AGENT,
+                Allow: ALLOW_METHODS,
+                'Content-Length': '0',
+            },
+            body: '',
+        });
+        this.sendToPeer(bye, call.responseHost, call.responsePort);
+        this.activeCalls.delete(callId);
+        console.log(`[SIPSignaling] Sent BYE for call: ${callId}`);
+    }
+    async stop() {
+        if (this.voip) {
+            for (const callId of this.activeCalls.keys()) {
+                try {
+                    await this.hangup(callId);
+                }
+                catch (error) {
+                    console.error(`[SIPSignaling] Error hanging up ${callId} during shutdown:`, error);
+                }
+            }
+            const transportSocket = this.voip.transport?.socket;
+            if (this.transportMessageListener &&
+                transportSocket &&
+                typeof transportSocket.off === 'function') {
+                transportSocket.off('message', this.transportMessageListener);
+                this.transportMessageListener = null;
+            }
+            if (transportSocket && typeof transportSocket.close === 'function') {
+                try {
+                    transportSocket.close();
+                }
+                catch (error) {
+                    console.error('[SIPSignaling] Error closing SIP transport socket:', error);
+                }
+            }
+            this.voip = null;
+        }
+        this.activeCalls.clear();
+        this.pendingInvites.clear();
+        this.ready = false;
+        console.log('[SIPSignaling] Stopped');
+    }
+    getRtpPort(callId) {
+        return this.activeCalls.get(callId)?.rtpPort ?? this.pendingInvites.get(callId)?.rtpPort;
+    }
+    sendResponse(message, parsedHeaders, statusCode, statusText, options) {
+        const responseTarget = options?.targetHost && options?.targetPort
+            ? { host: options.targetHost, port: options.targetPort }
+            : this.getResponseTarget(parsedHeaders);
+        if (!responseTarget) {
+            console.warn(`[SIPSignaling] Cannot send ${statusCode} ${statusText}; missing response target`);
+            return;
+        }
+        const body = options?.body ?? '';
+        const wire = this.buildResponseMessage({
+            statusCode,
+            statusText,
+            rawHeaders: message.headers,
+            parsedHeaders,
+            body,
+            toHeader: options?.toHeader ??
+                message.headers.To ??
+                this.buildAddressHeader(this.buildLocalUri(parsedHeaders, message)),
+            contactUri: options?.contactUri,
+            extraHeaders: options?.headers,
+        });
+        this.sendToPeer(wire, responseTarget.host, responseTarget.port);
+    }
+    buildResponseMessage(args) {
+        const headers = {
+            Via: args.rawHeaders.Via ?? '',
+            To: args.toHeader,
+            From: args.rawHeaders.From ?? '',
+            'Call-ID': args.rawHeaders['Call-ID'] ?? '',
+            CSeq: args.rawHeaders.CSeq ?? '',
+            Contact: `<${args.contactUri ?? this.buildLocalContactUri(args.parsedHeaders)}>`,
+            'Max-Forwards': '70',
+            'User-Agent': USER_AGENT,
+            Allow: ALLOW_METHODS,
+        };
+        if (args.extraHeaders) {
+            for (const [key, value] of Object.entries(args.extraHeaders)) {
+                headers[key] = value;
+            }
+        }
+        if (args.body) {
+            headers['Content-Type'] = 'application/sdp';
+            headers['Content-Length'] = String(args.body.length);
+        }
+        else {
+            headers['Content-Length'] = '0';
+        }
+        return {
+            isResponse: true,
+            protocol: 'SIP/2.0',
+            statusCode: args.statusCode,
+            statusText: args.statusText,
+            headers,
+            body: args.body,
+        };
+    }
+    buildRequestMessage(args) {
+        return {
+            isResponse: false,
+            protocol: 'SIP/2.0',
+            method: args.method,
+            requestUri: args.requestUri,
+            headers: args.headers,
+            body: args.body,
+        };
+    }
+    sendToPeer(message, host, port) {
+        if (!this.voip?.transport?.send) {
+            throw new Error('[SIPSignaling] transport.send is unavailable');
+        }
+        const built = SIP.Builder.Build(message);
+        this.voip.transport.send(built, host, port);
+    }
+    allocateRtpPort() {
+        const rtpPort = this.nextRtpPort;
+        this.nextRtpPort += 2;
+        if (this.nextRtpPort >= SIPSignaling.MAX_RTP_PORT) {
+            this.nextRtpPort = SIPSignaling.MIN_RTP_PORT;
+            console.warn('[SIPSignaling] RTP port range exhausted, wrapping around to', this.nextRtpPort);
+        }
+        return rtpPort;
+    }
+    parseHeaders(headers) {
+        return SIP.Parser.ParseHeaders(headers);
+    }
+    isSocketBound(socket) {
+        if (typeof socket.address !== 'function') {
+            return false;
+        }
+        try {
+            socket.address();
+            return true;
+        }
+        catch {
+            return false;
+        }
+    }
+    getResponseTarget(parsedHeaders, rinfo) {
+        if (rinfo?.address && rinfo?.port) {
+            return { host: rinfo.address, port: rinfo.port };
+        }
+        const contactHost = parsedHeaders.Contact?.contact?.ip;
+        const contactPort = this.toPort(parsedHeaders.Contact?.contact?.port);
+        if (contactHost && contactPort) {
+            return { host: contactHost, port: contactPort };
+        }
+        const viaHost = parsedHeaders.Via?.uri?.ip;
+        const viaPort = this.toPort(parsedHeaders.Via?.uri?.port) ?? DEFAULT_SIP_PORT;
+        if (viaHost) {
+            return { host: viaHost, port: viaPort };
+        }
+        return null;
+    }
+    extractCallId(message) {
+        if (message.callId)
+            return message.callId;
+        if (typeof message.headers['Call-ID'] === 'string')
+            return message.headers['Call-ID'];
+        const parsedHeaders = this.parseHeaders(message.headers);
+        if (parsedHeaders['Call-ID'])
+            return parsedHeaders['Call-ID'];
+        return `call-${Date.now()}`;
+    }
+    extractCseq(parsedHeaders) {
+        const count = parsedHeaders.CSeq?.count;
+        if (typeof count === 'number')
+            return count;
+        if (typeof count === 'string') {
+            const parsed = parseInt(count, 10);
+            if (!Number.isNaN(parsed))
+                return parsed;
+        }
+        return 0;
+    }
+    buildLocalUri(parsedHeaders, message) {
+        const requestUri = message?.requestUri ??
+            this.extractHeaderUri(message?.headers?.To) ??
+            undefined;
+        if (requestUri) {
+            return this.normalizeSipUri(requestUri);
+        }
+        const user = parsedHeaders.To?.contact?.username ?? 'agent';
+        return `sip:${user}@${this.localAddress}`;
+    }
+    buildLocalContactUri(parsedHeaders) {
+        const user = parsedHeaders.To?.contact?.username ?? 'agent';
+        return `sip:${user}@${this.localAddress}:${this.localPort}`;
+    }
+    extractRemoteUri(rawHeaders, parsedHeaders) {
+        return (this.normalizeSipUri(this.extractHeaderUri(rawHeaders.From)) ??
+            this.buildFallbackRemoteUri(parsedHeaders));
+    }
+    extractRemoteTargetUri(rawHeaders, parsedHeaders) {
+        return (this.normalizeSipUri(this.extractHeaderUri(rawHeaders.Contact)) ??
+            this.normalizeSipUri(this.extractHeaderUri(rawHeaders.From)) ??
+            this.buildFallbackRemoteUri(parsedHeaders));
+    }
+    buildFallbackRemoteUri(parsedHeaders) {
+        const username = parsedHeaders.From?.contact?.username ?? 'unknown';
+        const host = parsedHeaders.Contact?.contact?.ip ?? parsedHeaders.Via?.uri?.ip ?? 'unknown';
+        const port = this.toPort(parsedHeaders.Contact?.contact?.port);
+        return port ? `sip:${username}@${host}:${port}` : `sip:${username}@${host}`;
+    }
+    extractHeaderUri(value) {
+        if (!value)
+            return null;
+        const angleMatch = value.match(/<([^>]+)>/);
+        if (angleMatch?.[1])
+            return angleMatch[1].trim();
+        const sipMatch = value.match(/sip:[^;\s]+/i);
+        if (sipMatch?.[0])
+            return sipMatch[0].trim();
+        return null;
+    }
+    normalizeSipUri(uri) {
+        if (!uri) {
+            return `sip:unknown@${this.localAddress}`;
+        }
+        return uri.trim();
+    }
+    buildAddressHeader(uri, tag) {
+        return tag ? `<${uri}>;tag=${tag}` : `<${uri}>`;
+    }
+    ensureTag(headerValue, tag) {
+        if (!headerValue) {
+            return this.buildAddressHeader(`sip:agent@${this.localAddress}`, tag);
+        }
+        if (/(^|;)tag=/i.test(headerValue)) {
+            return headerValue;
+        }
+        return `${headerValue};tag=${tag}`;
+    }
+    toPort(value) {
+        if (typeof value === 'number' && Number.isFinite(value) && value > 0) {
+            return value;
+        }
+        if (typeof value === 'string') {
+            const parsed = parseInt(value, 10);
+            if (!Number.isNaN(parsed) && parsed > 0) {
+                return parsed;
+            }
+        }
+        return null;
+    }
+    generateTag() {
+        return Math.random().toString(36).slice(2, 12);
+    }
+    generateBranch() {
+        return `z9hG4bK${Math.random().toString(36).slice(2, 12)}`;
+    }
+}
+//# sourceMappingURL=sip_signaling.js.map

package/dist/sip_signaling.js.map

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

@@ -0,0 +1,16 @@
+import { TextOutput, type TimedString } from '@ariaflow/livekit-plugin';
+/**
+ * Text output for SIP sessions.
+ *
+ * SIP has no native text channel. This uses a callback pattern so the
+ * application can decide how to deliver transcription text (SIP INFO,
+ * webhook, parallel WebSocket, or simply discard).
+ */
+export declare class SIPTextOutput extends TextOutput {
+    private textHandler;
+    constructor(nextInChain?: TextOutput);
+    onText(handler: (text: string, isFinal: boolean) => void): void;
+    captureText(text: string | TimedString): Promise<void>;
+    flush(): void;
+}
+//# sourceMappingURL=text_output.d.ts.map

package/dist/text_output.d.ts.map

@@ -0,0 +1 @@
+{"version":3,"file":"text_output.d.ts","sourceRoot":"","sources":["../src/text_output.ts"],"names":[],"mappings":"AAAA,OAAO,EAAE,UAAU,EAAiB,KAAK,WAAW,EAAE,MAAM,0BAA0B,CAAC;AAEvF;;;;;;GAMG;AACH,qBAAa,aAAc,SAAQ,UAAU;IAC3C,OAAO,CAAC,WAAW,CAEH;gBAEJ,WAAW,CAAC,EAAE,UAAU;IAIpC,MAAM,CAAC,OAAO,EAAE,CAAC,IAAI,EAAE,MAAM,EAAE,OAAO,EAAE,OAAO,KAAK,IAAI,GAAG,IAAI;IAIzD,WAAW,CAAC,IAAI,EAAE,MAAM,GAAG,WAAW,GAAG,OAAO,CAAC,IAAI,CAAC;IAY5D,KAAK,IAAI,IAAI;CASd"}

package/dist/text_output.js

@@ -0,0 +1,35 @@
+import { TextOutput, isTimedString } from '@ariaflow/livekit-plugin';
+/**
+ * Text output for SIP sessions.
+ *
+ * SIP has no native text channel. This uses a callback pattern so the
+ * application can decide how to deliver transcription text (SIP INFO,
+ * webhook, parallel WebSocket, or simply discard).
+ */
+export class SIPTextOutput extends TextOutput {
+    textHandler = null;
+    constructor(nextInChain) {
+        super(nextInChain);
+    }
+    onText(handler) {
+        this.textHandler = handler;
+    }
+    async captureText(text) {
+        const textContent = isTimedString(text) ? text.text : text;
+        if (this.textHandler) {
+            this.textHandler(textContent, false);
+        }
+        if (this.nextInChain) {
+            await this.nextInChain.captureText(text);
+        }
+    }
+    flush() {
+        if (this.textHandler) {
+            this.textHandler('', true);
+        }
+        if (this.nextInChain) {
+            this.nextInChain.flush();
+        }
+    }
+}
+//# sourceMappingURL=text_output.js.map

package/dist/text_output.js.map

@@ -0,0 +1 @@
+{"version":3,"file":"text_output.js","sourceRoot":"","sources":["../src/text_output.ts"],"names":[],"mappings":"AAAA,OAAO,EAAE,UAAU,EAAE,aAAa,EAAoB,MAAM,0BAA0B,CAAC;AAEvF;;;;;;GAMG;AACH,MAAM,OAAO,aAAc,SAAQ,UAAU;IACnC,WAAW,GAER,IAAI,CAAC;IAEhB,YAAY,WAAwB;QAClC,KAAK,CAAC,WAAW,CAAC,CAAC;IACrB,CAAC;IAED,MAAM,CAAC,OAAiD;QACtD,IAAI,CAAC,WAAW,GAAG,OAAO,CAAC;IAC7B,CAAC;IAED,KAAK,CAAC,WAAW,CAAC,IAA0B;QAC1C,MAAM,WAAW,GAAG,aAAa,CAAC,IAAI,CAAC,CAAC,CAAC,CAAC,IAAI,CAAC,IAAI,CAAC,CAAC,CAAC,IAAI,CAAC;QAE3D,IAAI,IAAI,CAAC,WAAW,EAAE,CAAC;YACrB,IAAI,CAAC,WAAW,CAAC,WAAW,EAAE,KAAK,CAAC,CAAC;QACvC,CAAC;QAED,IAAI,IAAI,CAAC,WAAW,EAAE,CAAC;YACrB,MAAO,IAAI,CAAC,WAA0B,CAAC,WAAW,CAAC,IAAI,CAAC,CAAC;QAC3D,CAAC;IACH,CAAC;IAED,KAAK;QACH,IAAI,IAAI,CAAC,WAAW,EAAE,CAAC;YACrB,IAAI,CAAC,WAAW,CAAC,EAAE,EAAE,IAAI,CAAC,CAAC;QAC7B,CAAC;QAED,IAAI,IAAI,CAAC,WAAW,EAAE,CAAC;YACpB,IAAI,CAAC,WAA0B,CAAC,KAAK,EAAE,CAAC;QAC3C,CAAC;IACH,CAAC;CACF"}

package/dist/transport_adapter.d.ts

@@ -0,0 +1,22 @@
+import { TransportAdapter, type TransportAdapterConfig } from '@ariaflow/livekit-plugin';
+import { SIPAudioInput } from './audio_input.js';
+import { SIPAudioOutput } from './audio_output.js';
+import { SIPTextOutput } from './text_output.js';
+import type { RtpSession } from './rtp/rtp_session.js';
+import type { Codec } from '@ariaflow/livekit-plugin/codec/g711';
+export declare class SIPTransportAdapter extends TransportAdapter {
+    private rtpSession;
+    readonly id: string;
+    readonly audioInput: SIPAudioInput;
+    readonly audioOutput: SIPAudioOutput;
+    readonly textOutput: SIPTextOutput;
+    readonly config: TransportAdapterConfig;
+    private _isOpen;
+    constructor(rtpSession: RtpSession, codec: Codec, options: {
+        id: string;
+        packetDurationMs?: number;
+    });
+    get isOpen(): boolean;
+    close(): Promise<void>;
+}
+//# sourceMappingURL=transport_adapter.d.ts.map