ringcentral-softphone 1.3.4 → 1.3.5

package/dist/call-session/index.cjs

@@ -1,274 +1,213 @@
-var __create = Object.create;
-var __defProp = Object.defineProperty;
-var __getOwnPropDesc = Object.getOwnPropertyDescriptor;
-var __getOwnPropNames = Object.getOwnPropertyNames;
-var __getProtoOf = Object.getPrototypeOf;
-var __hasOwnProp = Object.prototype.hasOwnProperty;
-var __export = (target, all) => {
-  for (var name in all)
-    __defProp(target, name, { get: all[name], enumerable: true });
+const require_runtime = require("../_virtual/_rolldown/runtime.cjs");
+const require_dtmf = require("../dtmf.cjs");
+const require_utils = require("../utils.cjs");
+const require_sip_message_outbound_request = require("../sip-message/outbound/request.cjs");
+const require_sip_message_outbound_response = require("../sip-message/outbound/response.cjs");
+require("../sip-message/index.cjs");
+const require_call_session_streamer = require("./streamer.cjs");
+let node_buffer = require("node:buffer");
+let node_events = require("node:events");
+node_events = require_runtime.__toESM(node_events, 1);
+let wait_for_async = require("wait-for-async");
+wait_for_async = require_runtime.__toESM(wait_for_async, 1);
+let node_dgram = require("node:dgram");
+node_dgram = require_runtime.__toESM(node_dgram, 1);
+let werift_rtp = require("werift-rtp");
+//#region src/call-session/index.ts
+const isDtmfChar = (value) => require_dtmf.phoneChars.includes(value);
+var CallSession = class extends node_events.default {
+	softphone;
+	sipMessage;
+	socket;
+	localPeer;
+	remotePeer;
+	remoteIP;
+	remotePort;
+	disposed = false;
+	srtpSession;
+	encoder;
+	decoder;
+	sdp;
+	ssrc = require_utils.randomInt();
+	sequenceNumber = require_utils.randomInt();
+	timestamp = require_utils.randomInt();
+	constructor(softphone, sipMessage) {
+		super();
+		this.softphone = softphone;
+		this.encoder = softphone.codec.createEncoder();
+		this.decoder = softphone.codec.createDecoder();
+		this.sipMessage = sipMessage;
+		if (this.sipMessage.body.length > 0) {
+			this.remoteIP = this.sipMessage.body.match(/c=IN IP4 ([\d.]+)/)[1];
+			this.remotePort = parseInt(this.sipMessage.body.match(/m=audio (\d+) /)[1], 10);
+		}
+	}
+	set remoteKey(key) {
+		const localKeyBuffer = node_buffer.Buffer.from(require_utils.localKey, "base64");
+		const remoteKeyBuffer = node_buffer.Buffer.from(key, "base64");
+		this.srtpSession = new werift_rtp.SrtpSession({
+			profile: 1,
+			keys: {
+				localMasterKey: localKeyBuffer.subarray(0, 16),
+				localMasterSalt: localKeyBuffer.subarray(16, 30),
+				remoteMasterKey: remoteKeyBuffer.subarray(0, 16),
+				remoteMasterSalt: remoteKeyBuffer.subarray(16, 30)
+			}
+		});
+	}
+	get callId() {
+		return this.sipMessage.getHeader("Call-ID");
+	}
+	send(data) {
+		this.socket.send(data, this.remotePort, this.remoteIP);
+	}
+	async hangup() {
+		const requestMessage = new require_sip_message_outbound_request(`BYE sip:${this.softphone.sipInfo.domain} SIP/2.0`, {
+			"Call-ID": this.callId,
+			From: this.localPeer,
+			To: this.remotePeer,
+			Via: `SIP/2.0/TLS ${this.softphone.fakeDomain};branch=${require_utils.branch()}`
+		});
+		await this.softphone.send(requestMessage);
+	}
+	sendDTMF(char) {
+		const payloads = require_dtmf.charToPayloads(char);
+		const timestamp = this.timestamp;
+		let first = true;
+		for (const payload of payloads) {
+			const rtpPacket = new werift_rtp.RtpPacket(new werift_rtp.RtpHeader({
+				version: 2,
+				padding: false,
+				paddingSize: 0,
+				extension: false,
+				marker: first,
+				payloadOffset: 12,
+				payloadType: 101,
+				sequenceNumber: this.sequenceNumber,
+				timestamp,
+				ssrc: this.ssrc,
+				csrcLength: 0,
+				csrc: [],
+				extensionProfile: 48862,
+				extensionLength: void 0,
+				extensions: []
+			}), payload);
+			this.send(this.srtpSession.encrypt(rtpPacket.payload, rtpPacket.header));
+			this.sequenceNumber = (this.sequenceNumber + 1) % 65536;
+			first = false;
+		}
+		this.timestamp += 800;
+	}
+	async sendDTMFs(s, delay = 500) {
+		for (const c of s) {
+			if (!isDtmfChar(c)) throw new Error(`invalid phone char: ${c}`);
+			this.sendDTMF(c);
+			await (0, wait_for_async.default)({ interval: delay });
+		}
+	}
+	streamAudio(input) {
+		const streamer = new require_call_session_streamer(this, input);
+		streamer.start();
+		return streamer;
+	}
+	sendPacket(rtpPacket) {
+		if (this.disposed) return;
+		this.send(this.srtpSession.encrypt(rtpPacket.payload, rtpPacket.header));
+	}
+	startLocalServices() {
+		this.socket = node_dgram.default.createSocket("udp4");
+		this.socket.on("message", (message) => {
+			const rtpPacket = werift_rtp.RtpPacket.deSerialize(this.srtpSession.decrypt(message));
+			this.emit("rtpPacket", rtpPacket);
+			if (rtpPacket.header.payloadType === 101) {
+				this.emit("dtmfPacket", rtpPacket);
+				const char = require_dtmf.payloadToChar(rtpPacket.payload);
+				if (char) this.emit("dtmf", char);
+			} else if (rtpPacket.header.payloadType === this.softphone.codec.id) {
+				if (rtpPacket.payload.length === 4 && rtpPacket.payload[0] >= 0 && rtpPacket.payload[0] < 12 && rtpPacket.payload[1] === 138 && rtpPacket.payload[2] === 3 && rtpPacket.payload[3] === 192) return;
+				try {
+					rtpPacket.payload = this.decoder.decode(rtpPacket.payload);
+					this.emit("audioPacket", rtpPacket);
+				} catch {
+					console.error("Audio packet decode failed", rtpPacket);
+				}
+			}
+		});
+		this.socket.bind();
+		this.send("hello");
+		const byeHandler = (inboundMessage) => {
+			if (inboundMessage.getHeader("Call-ID") !== this.callId) return;
+			if (inboundMessage.headers.CSeq.endsWith(" BYE")) {
+				this.softphone.off("message", byeHandler);
+				this.dispose();
+			}
+		};
+		this.softphone.on("message", byeHandler);
+	}
+	dispose() {
+		this.disposed = true;
+		this.emit("disposed");
+		this.removeAllListeners();
+		this.socket?.removeAllListeners();
+		this.socket?.close();
+	}
+	async transfer(transferTo) {
+		const requestMessage = new require_sip_message_outbound_request(`REFER sip:${this.softphone.sipInfo.username}@${this.softphone.sipInfo.outboundProxy};transport=tls SIP/2.0`, {
+			Via: `SIP/2.0/TLS ${this.softphone.client.localAddress}:${this.softphone.client.localPort};rport;branch=${require_utils.branch()};alias`,
+			"Max-Forwards": 70,
+			From: this.localPeer,
+			To: this.remotePeer,
+			Contact: `<sip:${this.softphone.sipInfo.username}@${this.softphone.client.localAddress}:${this.softphone.client.localPort};transport=TLS;ob>`,
+			"Call-ID": this.callId,
+			Event: "refer",
+			Expires: 600,
+			Supported: "replaces, 100rel, timer, norefersub",
+			Accept: "message/sipfrag;version=2.0",
+			"Allow-Events": "presence, message-summary, refer",
+			"Refer-To": `sip:${transferTo}@${this.softphone.sipInfo.domain}`,
+			"Referred-By": `<sip:${this.softphone.sipInfo.username}@${this.softphone.sipInfo.domain}>`
+		});
+		await this.softphone.send(requestMessage);
+		return new Promise((resolve) => {
+			const notifyHandler = (inboundMessage) => {
+				if (!inboundMessage.subject.startsWith("NOTIFY ")) return;
+				const responseMessage = new require_sip_message_outbound_response(inboundMessage, 200);
+				this.softphone.send(responseMessage);
+				if (inboundMessage.body.trim() === "SIP/2.0 200 OK") {
+					this.softphone.off("message", notifyHandler);
+					resolve();
+				}
+			};
+			this.softphone.on("message", notifyHandler);
+		});
+	}
+	async toggleReceive(toReceive) {
+		let newSDP = this.sdp;
+		if (!toReceive) newSDP = newSDP.replace(/a=sendrecv/, "a=sendonly");
+		const requestMessage = new require_sip_message_outbound_request(`INVITE ${require_utils.extractAddress(this.remotePeer)} SIP/2.0`, {
+			"Call-Id": this.callId,
+			From: this.localPeer,
+			To: this.remotePeer,
+			Via: `SIP/2.0/TLS ${this.softphone.client.localAddress}:${this.softphone.client.localPort};rport;branch=${require_utils.branch()};alias`,
+			"Content-Type": "application/sdp",
+			Contact: ` <sip:${this.softphone.sipInfo.username}@${this.softphone.client.localAddress}:${this.softphone.client.localPort};transport=TLS;ob>`
+		}, newSDP);
+		const replyMessage = await this.softphone.send(requestMessage, true);
+		const ackMessage = new require_sip_message_outbound_request(`ACK ${require_utils.extractAddress(this.remotePeer)} SIP/2.0`, {
+			"Call-Id": this.callId,
+			From: this.localPeer,
+			To: this.remotePeer,
+			Via: replyMessage.headers.Via,
+			CSeq: replyMessage.headers.CSeq.replace(" INVITE", " ACK")
+		});
+		await this.softphone.send(ackMessage);
+	}
+	async hold() {
+		return this.toggleReceive(false);
+	}
+	async unhold() {
+		return this.toggleReceive(true);
+	}
 };
-var __copyProps = (to, from, except, desc) => {
-  if (from && typeof from === "object" || typeof from === "function") {
-    for (let key of __getOwnPropNames(from))
-      if (!__hasOwnProp.call(to, key) && key !== except)
-        __defProp(to, key, { get: () => from[key], enumerable: !(desc = __getOwnPropDesc(from, key)) || desc.enumerable });
-  }
-  return to;
-};
-var __toESM = (mod, isNodeMode, target) => (target = mod != null ? __create(__getProtoOf(mod)) : {}, __copyProps(
-  // If the importer is in node compatibility mode or this is not an ESM
-  // file that has been converted to a CommonJS file using a Babel-
-  // compatible transform (i.e. "__esModule" has not been set), then set
-  // "default" to the CommonJS "module.exports" for node compatibility.
-  isNodeMode || !mod || !mod.__esModule ? __defProp(target, "default", { value: mod, enumerable: true }) : target,
-  mod
-));
-var __toCommonJS = (mod) => __copyProps(__defProp({}, "__esModule", { value: true }), mod);
-var call_session_exports = {};
-__export(call_session_exports, {
-  default: () => call_session_default
-});
-module.exports = __toCommonJS(call_session_exports);
-var import_node_buffer = require("node:buffer");
-var import_node_dgram = __toESM(require("node:dgram"), 1);
-var import_node_events = __toESM(require("node:events"), 1);
-var import_wait_for_async = __toESM(require("wait-for-async"), 1);
-var import_werift_rtp = require("werift-rtp");
-var import_dtmf = __toESM(require("../dtmf.js"), 1);
-var import_sip_message = require("../sip-message/index.js");
-var import_utils = require("../utils.js");
-var import_streamer = __toESM(require("./streamer.js"), 1);
-const isDtmfChar = (value) => import_dtmf.default.phoneChars.includes(value);
-class CallSession extends import_node_events.default {
-  softphone;
-  sipMessage;
-  socket;
-  localPeer;
-  remotePeer;
-  remoteIP;
-  remotePort;
-  disposed = false;
-  srtpSession;
-  encoder;
-  decoder;
-  sdp;
-  // for audio streaming
-  ssrc = (0, import_utils.randomInt)();
-  sequenceNumber = (0, import_utils.randomInt)();
-  timestamp = (0, import_utils.randomInt)();
-  constructor(softphone, sipMessage) {
-    super();
-    this.softphone = softphone;
-    this.encoder = softphone.codec.createEncoder();
-    this.decoder = softphone.codec.createDecoder();
-    this.sipMessage = sipMessage;
-    if (this.sipMessage.body.length > 0) {
-      this.remoteIP = this.sipMessage.body.match(/c=IN IP4 ([\d.]+)/)[1];
-      this.remotePort = parseInt(
-        this.sipMessage.body.match(/m=audio (\d+) /)[1],
-        10
-      );
-    }
-  }
-  set remoteKey(key) {
-    const localKeyBuffer = import_node_buffer.Buffer.from(import_utils.localKey, "base64");
-    const remoteKeyBuffer = import_node_buffer.Buffer.from(key, "base64");
-    this.srtpSession = new import_werift_rtp.SrtpSession({
-      profile: 1,
-      keys: {
-        localMasterKey: localKeyBuffer.subarray(0, 16),
-        localMasterSalt: localKeyBuffer.subarray(16, 30),
-        remoteMasterKey: remoteKeyBuffer.subarray(0, 16),
-        remoteMasterSalt: remoteKeyBuffer.subarray(16, 30)
-      }
-    });
-  }
-  get callId() {
-    return this.sipMessage.getHeader("Call-ID");
-  }
-  send(data) {
-    this.socket.send(data, this.remotePort, this.remoteIP);
-  }
-  async hangup() {
-    const requestMessage = new import_sip_message.RequestMessage(
-      `BYE sip:${this.softphone.sipInfo.domain} SIP/2.0`,
-      {
-        "Call-ID": this.callId,
-        From: this.localPeer,
-        To: this.remotePeer,
-        Via: `SIP/2.0/TLS ${this.softphone.fakeDomain};branch=${(0, import_utils.branch)()}`
-      }
-    );
-    await this.softphone.send(requestMessage);
-  }
-  sendDTMF(char) {
-    const payloads = import_dtmf.default.charToPayloads(char);
-    const timestamp = this.timestamp;
-    let first = true;
-    for (const payload of payloads) {
-      const rtpHeader = new import_werift_rtp.RtpHeader({
-        version: 2,
-        padding: false,
-        paddingSize: 0,
-        extension: false,
-        marker: first,
-        payloadOffset: 12,
-        payloadType: 101,
-        sequenceNumber: this.sequenceNumber,
-        timestamp,
-        ssrc: this.ssrc,
-        csrcLength: 0,
-        csrc: [],
-        extensionProfile: 48862,
-        extensionLength: void 0,
-        extensions: []
-      });
-      const rtpPacket = new import_werift_rtp.RtpPacket(rtpHeader, payload);
-      this.send(this.srtpSession.encrypt(rtpPacket.payload, rtpPacket.header));
-      this.sequenceNumber = (this.sequenceNumber + 1) % 65536;
-      first = false;
-    }
-    this.timestamp += 800;
-  }
-  async sendDTMFs(s, delay = 500) {
-    for (const c of s) {
-      if (!isDtmfChar(c)) {
-        throw new Error(`invalid phone char: ${c}`);
-      }
-      this.sendDTMF(c);
-      await (0, import_wait_for_async.default)({ interval: delay });
-    }
-  }
-  // buffer is the content of a audio file, it is supposed to be uncompressed PCM data
-  // The audio should be playable by command: play -t raw -b 16 -r 16000 -e signed-integer test.wav
-  streamAudio(input) {
-    const streamer = new import_streamer.default(this, input);
-    streamer.start();
-    return streamer;
-  }
-  // send a single rtp packet
-  sendPacket(rtpPacket) {
-    if (this.disposed) {
-      return;
-    }
-    this.send(this.srtpSession.encrypt(rtpPacket.payload, rtpPacket.header));
-  }
-  startLocalServices() {
-    this.socket = import_node_dgram.default.createSocket("udp4");
-    this.socket.on("message", (message) => {
-      const rtpPacket = import_werift_rtp.RtpPacket.deSerialize(
-        this.srtpSession.decrypt(message)
-      );
-      this.emit("rtpPacket", rtpPacket);
-      if (rtpPacket.header.payloadType === 101) {
-        this.emit("dtmfPacket", rtpPacket);
-        const char = import_dtmf.default.payloadToChar(rtpPacket.payload);
-        if (char) {
-          this.emit("dtmf", char);
-        }
-      } else if (rtpPacket.header.payloadType === this.softphone.codec.id) {
-        if (rtpPacket.payload.length === 4 && rtpPacket.payload[0] >= 0 && rtpPacket.payload[0] < 12 && rtpPacket.payload[1] === 138 && rtpPacket.payload[2] === 3 && rtpPacket.payload[3] === 192) {
-          return;
-        }
-        try {
-          rtpPacket.payload = this.decoder.decode(rtpPacket.payload);
-          this.emit("audioPacket", rtpPacket);
-        } catch {
-          console.error("Audio packet decode failed", rtpPacket);
-        }
-      }
-    });
-    this.socket.bind();
-    this.send("hello");
-    const byeHandler = (inboundMessage) => {
-      if (inboundMessage.getHeader("Call-ID") !== this.callId) {
-        return;
-      }
-      if (inboundMessage.headers.CSeq.endsWith(" BYE")) {
-        this.softphone.off("message", byeHandler);
-        this.dispose();
-      }
-    };
-    this.softphone.on("message", byeHandler);
-  }
-  dispose() {
-    this.disposed = true;
-    this.emit("disposed");
-    this.removeAllListeners();
-    this.socket?.removeAllListeners();
-    this.socket?.close();
-  }
-  async transfer(transferTo) {
-    const requestMessage = new import_sip_message.RequestMessage(
-      `REFER sip:${this.softphone.sipInfo.username}@${this.softphone.sipInfo.outboundProxy};transport=tls SIP/2.0`,
-      {
-        Via: `SIP/2.0/TLS ${this.softphone.client.localAddress}:${this.softphone.client.localPort};rport;branch=${(0, import_utils.branch)()};alias`,
-        "Max-Forwards": 70,
-        From: this.localPeer,
-        To: this.remotePeer,
-        Contact: `<sip:${this.softphone.sipInfo.username}@${this.softphone.client.localAddress}:${this.softphone.client.localPort};transport=TLS;ob>`,
-        "Call-ID": this.callId,
-        Event: "refer",
-        Expires: 600,
-        Supported: "replaces, 100rel, timer, norefersub",
-        Accept: "message/sipfrag;version=2.0",
-        "Allow-Events": "presence, message-summary, refer",
-        "Refer-To": `sip:${transferTo}@${this.softphone.sipInfo.domain}`,
-        "Referred-By": `<sip:${this.softphone.sipInfo.username}@${this.softphone.sipInfo.domain}>`
-      }
-    );
-    await this.softphone.send(requestMessage);
-    return new Promise((resolve) => {
-      const notifyHandler = (inboundMessage) => {
-        if (!inboundMessage.subject.startsWith("NOTIFY ")) {
-          return;
-        }
-        const responseMessage = new import_sip_message.ResponseMessage(inboundMessage, 200);
-        this.softphone.send(responseMessage);
-        if (inboundMessage.body.trim() === "SIP/2.0 200 OK") {
-          this.softphone.off("message", notifyHandler);
-          resolve();
-        }
-      };
-      this.softphone.on("message", notifyHandler);
-    });
-  }
-  async toggleReceive(toReceive) {
-    let newSDP = this.sdp;
-    if (!toReceive) {
-      newSDP = newSDP.replace(/a=sendrecv/, "a=sendonly");
-    }
-    const requestMessage = new import_sip_message.RequestMessage(
-      `INVITE ${(0, import_utils.extractAddress)(this.remotePeer)} SIP/2.0`,
-      {
-        "Call-Id": this.callId,
-        From: this.localPeer,
-        To: this.remotePeer,
-        Via: `SIP/2.0/TLS ${this.softphone.client.localAddress}:${this.softphone.client.localPort};rport;branch=${(0, import_utils.branch)()};alias`,
-        "Content-Type": "application/sdp",
-        Contact: ` <sip:${this.softphone.sipInfo.username}@${this.softphone.client.localAddress}:${this.softphone.client.localPort};transport=TLS;ob>`
-      },
-      newSDP
-    );
-    const replyMessage = await this.softphone.send(requestMessage, true);
-    const ackMessage = new import_sip_message.RequestMessage(
-      `ACK ${(0, import_utils.extractAddress)(this.remotePeer)} SIP/2.0`,
-      {
-        "Call-Id": this.callId,
-        From: this.localPeer,
-        To: this.remotePeer,
-        Via: replyMessage.headers.Via,
-        CSeq: replyMessage.headers.CSeq.replace(" INVITE", " ACK")
-      }
-    );
-    await this.softphone.send(ackMessage);
-  }
-  async hold() {
-    return this.toggleReceive(false);
-  }
-  async unhold() {
-    return this.toggleReceive(true);
-  }
-}
-var call_session_default = CallSession;
+//#endregion
+module.exports = CallSession;

package/dist/call-session/index.d.cts

@@ -0,0 +1,48 @@
+import InboundMessage from "../sip-message/inbound/index.cjs";
+import DTMF from "../dtmf.cjs";
+import Streamer from "./streamer.cjs";
+import Softphone from "../index.cjs";
+import EventEmitter from "node:events";
+import { Buffer } from "node:buffer";
+import dgram from "node:dgram";
+import { RtpPacket, SrtpSession } from "werift-rtp";
+
+//#region src/call-session/index.d.ts
+type DtmfChar = (typeof DTMF.phoneChars)[number];
+declare abstract class CallSession extends EventEmitter {
+  softphone: Softphone;
+  sipMessage: InboundMessage;
+  socket: dgram.Socket;
+  localPeer: string;
+  remotePeer: string;
+  remoteIP: string;
+  remotePort: number;
+  disposed: boolean;
+  srtpSession: SrtpSession;
+  encoder: {
+    encode: (pcm: Buffer) => Buffer;
+  };
+  decoder: {
+    decode: (audio: Buffer) => Buffer;
+  };
+  sdp: string;
+  ssrc: number;
+  sequenceNumber: number;
+  timestamp: number;
+  constructor(softphone: Softphone, sipMessage: InboundMessage);
+  set remoteKey(key: string);
+  get callId(): string | undefined;
+  send(data: string | Buffer): void;
+  hangup(): Promise<void>;
+  sendDTMF(char: DtmfChar): void;
+  sendDTMFs(s: string, delay?: number): Promise<void>;
+  streamAudio(input: Buffer): Streamer;
+  sendPacket(rtpPacket: RtpPacket): void;
+  protected startLocalServices(): void;
+  protected dispose(): void;
+  transfer(transferTo: string): Promise<void>;
+  toggleReceive(toReceive: boolean): Promise<void>;
+  hold(): Promise<void>;
+  unhold(): Promise<void>;
+}
+export = CallSession;

package/dist/call-session/index.d.ts

@@ -1,46 +1,49 @@
+import InboundMessage from "../sip-message/inbound/index.js";
+import DTMF from "../dtmf.js";
+import Streamer from "./streamer.js";
+import Softphone from "../index.js";
 import { Buffer } from "node:buffer";
-import dgram from "node:dgram";
 import EventEmitter from "node:events";
+import dgram from "node:dgram";
 import { RtpPacket, SrtpSession } from "werift-rtp";
-import DTMF from "../dtmf.js";
-import type Softphone from "../index.js";
-import { type InboundMessage } from "../sip-message/index.js";
-import Streamer from "./streamer.js";
+
+//#region src/call-session/index.d.ts
 type DtmfChar = (typeof DTMF.phoneChars)[number];
 declare abstract class CallSession extends EventEmitter {
-    softphone: Softphone;
-    sipMessage: InboundMessage;
-    socket: dgram.Socket;
-    localPeer: string;
-    remotePeer: string;
-    remoteIP: string;
-    remotePort: number;
-    disposed: boolean;
-    srtpSession: SrtpSession;
-    encoder: {
-        encode: (pcm: Buffer) => Buffer;
-    };
-    decoder: {
-        decode: (audio: Buffer) => Buffer;
-    };
-    sdp: string;
-    ssrc: number;
-    sequenceNumber: number;
-    timestamp: number;
-    constructor(softphone: Softphone, sipMessage: InboundMessage);
-    set remoteKey(key: string);
-    get callId(): string | undefined;
-    send(data: string | Buffer): void;
-    hangup(): Promise<void>;
-    sendDTMF(char: DtmfChar): void;
-    sendDTMFs(s: string, delay?: number): Promise<void>;
-    streamAudio(input: Buffer): Streamer;
-    sendPacket(rtpPacket: RtpPacket): void;
-    protected startLocalServices(): void;
-    protected dispose(): void;
-    transfer(transferTo: string): Promise<void>;
-    toggleReceive(toReceive: boolean): Promise<void>;
-    hold(): Promise<void>;
-    unhold(): Promise<void>;
+  softphone: Softphone;
+  sipMessage: InboundMessage;
+  socket: dgram.Socket;
+  localPeer: string;
+  remotePeer: string;
+  remoteIP: string;
+  remotePort: number;
+  disposed: boolean;
+  srtpSession: SrtpSession;
+  encoder: {
+    encode: (pcm: Buffer) => Buffer;
+  };
+  decoder: {
+    decode: (audio: Buffer) => Buffer;
+  };
+  sdp: string;
+  ssrc: number;
+  sequenceNumber: number;
+  timestamp: number;
+  constructor(softphone: Softphone, sipMessage: InboundMessage);
+  set remoteKey(key: string);
+  get callId(): string | undefined;
+  send(data: string | Buffer): void;
+  hangup(): Promise<void>;
+  sendDTMF(char: DtmfChar): void;
+  sendDTMFs(s: string, delay?: number): Promise<void>;
+  streamAudio(input: Buffer): Streamer;
+  sendPacket(rtpPacket: RtpPacket): void;
+  protected startLocalServices(): void;
+  protected dispose(): void;
+  transfer(transferTo: string): Promise<void>;
+  toggleReceive(toReceive: boolean): Promise<void>;
+  hold(): Promise<void>;
+  unhold(): Promise<void>;
 }
-export default CallSession;
+//#endregion
+export { CallSession as default };