{"version":3,"file":"index.mjs","names":[],"sources":["../../src/call-session/index.ts"],"sourcesContent":["import sdpTransform from \"sdp-transform\";\n\nimport EventEmitter from \"../event-emitter.js\";\nimport type WebPhone from \"../index.js\";\nimport type InboundMessage from \"../sip-message/inbound.js\";\nimport RequestMessage from \"../sip-message/outbound/request.js\";\nimport ResponseMessage from \"../sip-message/outbound/response.js\";\nimport {\n branch,\n extractAddress,\n extractNumber,\n extractTag,\n fakeDomain,\n uuid,\n} from \"../utils.js\";\nimport type OutboundCallSession from \"./outbound.js\";\n\ninterface CommandResult {\n code: number;\n description: string;\n}\ntype ParkResult = CommandResult & {\n \"park extension\": string;\n};\ntype FlipResult = CommandResult & {\n number: string;\n target: string;\n};\nconst DEFAULT_TRANSFER_TIMEOUT_MS = 10000;\n\nclass CallSession extends EventEmitter {\n public webPhone: WebPhone;\n public sipMessage!: InboundMessage;\n public localPeer!: string;\n public remotePeer!: string;\n public rtcPeerConnection!: RTCPeerConnection;\n public _mediaStream?: MediaStream;\n public audioElement!: HTMLAudioElement;\n public state: \"init\" | \"ringing\" | \"answered\" | \"disposed\" | \"failed\" =\n \"init\";\n public direction!: \"inbound\" | \"outbound\";\n public inputDeviceId!: string;\n public outputDeviceId: string | undefined;\n\n private reqid = 1;\n private sdpVersion = 1;\n\n public constructor(webPhone: WebPhone) {\n super();\n this.webPhone = webPhone;\n }\n\n public get mediaStream(): MediaStream | undefined {\n return this._mediaStream;\n }\n public set mediaStream(stream: MediaStream) {\n this._mediaStream = stream;\n this.emit(\"mediaStreamSet\", stream);\n }\n\n // for inbound call, this.sipMessage?.headers[\"Call-Id\"] will be the call id\n // for outbound call, this._callId will be the call id. Once the call session is out of \"init\" state, this.sipMessage will be set\n private _callId = uuid();\n public get callId() {\n return this.sipMessage?.headers[\"Call-Id\"] ?? this._callId;\n }\n\n public get sessionId() {\n return this.sipMessage?.headers[\"p-rc-api-ids\"].match(\n /session-id=(s-[0-9a-fz]+?)$/,\n )?.[1];\n }\n\n public get partyId() {\n return this.sipMessage?.headers[\"p-rc-api-ids\"].match(\n /party-id=(p-[0-9a-fz]+?-\\d);/,\n )?.[1];\n }\n\n public get remoteNumber(): string {\n return extractNumber(this.remotePeer);\n }\n\n public get localNumber(): string {\n return this.localPeer\n ? extractNumber(this.localPeer)\n : this.webPhone.sipInfo.username;\n }\n\n public get remoteTag() {\n return extractTag(this.remotePeer);\n }\n\n public get localTag() {\n return extractTag(this.localPeer);\n }\n\n public get isConference() {\n return this.remotePeer\n ? extractNumber(this.remotePeer).startsWith(\"conf_\")\n : false;\n }\n\n public async init() {\n this.rtcPeerConnection = new RTCPeerConnection({\n iceServers:\n this.webPhone.sipInfo.stunServers?.map((url) => ({\n urls: `stun:${url}`,\n })) ?? [],\n });\n\n // line below is to make sure that you have the permission to access the microphone\n const tempStream = await navigator.mediaDevices.getUserMedia({\n audio: true,\n video: false,\n });\n tempStream.getTracks().forEach((track) => track.stop()); // 🔥 Stop immediately!\n\n this.inputDeviceId = await this.webPhone.deviceManager.getInputDeviceId();\n this.mediaStream = await navigator.mediaDevices.getUserMedia({\n video: false,\n audio: { deviceId: { exact: this.inputDeviceId } },\n });\n this.mediaStream.getAudioTracks().forEach((track) => {\n const rtcRtpSender = this.rtcPeerConnection.addTrack(track);\n\n // ref: https://github.com/ringcentral/ringcentral-web-phone/issues/257\n const params = rtcRtpSender.getParameters();\n if (!params.encodings || params.encodings.length === 0) {\n params.encodings = [{}];\n }\n params.encodings.forEach((encoding) => {\n encoding.priority = \"high\";\n });\n rtcRtpSender.setParameters(params);\n });\n this.rtcPeerConnection.ontrack = async (event) => {\n const remoteStream = event.streams[0];\n this.audioElement = document.createElement(\"audio\") as HTMLAudioElement;\n this.audioElement.hidden = true;\n this.audioElement.autoplay = true;\n this.audioElement.srcObject = remoteStream;\n\n // this code should be run last\n this.outputDeviceId =\n await this.webPhone.deviceManager.getOutputDeviceId();\n if (this.outputDeviceId) {\n this.audioElement.setSinkId(this.outputDeviceId);\n }\n };\n }\n\n public async changeInputDevice(deviceId: string) {\n this.inputDeviceId = deviceId;\n this.mediaStream?.getAudioTracks().forEach((track) => track.stop());\n this.mediaStream = await navigator.mediaDevices.getUserMedia({\n video: false,\n audio: { deviceId: { exact: deviceId } },\n });\n const newAudioTrack = this.mediaStream.getAudioTracks()[0];\n const sender = this.rtcPeerConnection\n .getSenders()\n .find((sender) => sender.track?.kind === \"audio\");\n if (sender) {\n sender.replaceTrack(newAudioTrack);\n }\n }\n\n public async changeOutputDevice(deviceId: string) {\n this.outputDeviceId = deviceId;\n if (deviceId) {\n await this.audioElement.setSinkId(deviceId);\n }\n }\n\n public async transfer(target: string, timeout = DEFAULT_TRANSFER_TIMEOUT_MS) {\n return await this._transfer(`sip:${target}@sip.ringcentral.com`, timeout);\n }\n\n public async warmTransfer(\n target: string,\n options?: { callerId?: string; timeout?: number },\n ): Promise<{\n complete: () => Promise;\n cancel: () => Promise;\n newSession: OutboundCallSession;\n }> {\n await this.hold();\n // create a new session and user needs to talk to the target before transfer\n const newSession = await this.webPhone.call(target, options?.callerId);\n return {\n // complete the transfer\n complete: async () => {\n await this.completeWarmTransfer(\n newSession,\n options?.timeout ?? DEFAULT_TRANSFER_TIMEOUT_MS,\n );\n },\n // cancel the transfer\n cancel: async () => {\n await newSession.hangup();\n await this.unhold();\n },\n newSession,\n };\n }\n\n public async completeWarmTransfer(\n existingSession: CallSession,\n timeout = DEFAULT_TRANSFER_TIMEOUT_MS,\n ) {\n const target = existingSession.remoteNumber;\n await this._transfer(\n `\"${target}@sip.ringcentral.com\" `,\n timeout,\n );\n }\n\n public async hangup() {\n const requestMessage = new RequestMessage(\n `BYE sip:${this.webPhone.sipInfo.domain} SIP/2.0`,\n {\n \"Call-Id\": this.callId,\n From: this.localPeer,\n To: this.remotePeer,\n Via: `SIP/2.0/WSS ${fakeDomain};branch=${branch()}`,\n },\n );\n await this.webPhone.sipClient.request(requestMessage);\n }\n\n public async startRecording(): Promise {\n return await this.sendJsonMessage(\"startcallrecord\");\n }\n\n public async stopRecording(): Promise {\n return await this.sendJsonMessage(\"stopcallrecord\");\n }\n\n public async flip(target: string): Promise {\n const flipResult = await this.sendJsonMessage(\"callflip\", {\n target,\n });\n // note: we can't dispose the call session here\n // otherwise the caller will not be able to talk to the flip target\n // after the flip target answers the call, manually dispose the call session\n // todo: review this part\n return flipResult;\n }\n\n public async park(): Promise {\n const parkResult = await this.sendJsonMessage(\"callpark\");\n if (parkResult.code === 0) {\n await this.hangup();\n }\n return parkResult;\n }\n\n public async hold() {\n await this.toggleReceive(false);\n }\n public async unhold() {\n await this.toggleReceive(true);\n }\n\n public mute() {\n this.toggleTrack(false);\n }\n public unmute() {\n this.toggleTrack(true);\n }\n\n public sendDtmf(tones: string, duration?: number, interToneGap?: number) {\n for (const sender of this.rtcPeerConnection.getSenders()) {\n if (sender.dtmf?.canInsertDTMF) {\n sender.dtmf?.insertDTMF(tones, duration, interToneGap);\n }\n }\n }\n\n public dispose() {\n this.rtcPeerConnection?.close();\n this.mediaStream?.getTracks().forEach((track) => track.stop());\n if (this.audioElement) {\n this.audioElement.srcObject = null;\n }\n this.state = \"disposed\";\n this.emit(\"disposed\");\n this.removeAllListeners();\n }\n\n // for mute/unmute\n protected toggleTrack(enabled: boolean) {\n this.rtcPeerConnection.getSenders().forEach((sender) => {\n if (sender.track) {\n sender.track.enabled = enabled;\n }\n });\n }\n\n protected async waitForIceGatheringComplete(timeoutMs = 2000) {\n if (this.rtcPeerConnection.iceGatheringState === \"complete\") {\n return;\n }\n await new Promise((resolve) => {\n const timeout = setTimeout(() => {\n cleanup();\n resolve();\n }, timeoutMs);\n const onIceCandidate = (event: RTCPeerConnectionIceEvent) => {\n if (event.candidate === null) {\n cleanup();\n resolve();\n }\n };\n const cleanup = () => {\n clearTimeout(timeout);\n this.rtcPeerConnection.removeEventListener(\n \"icecandidate\",\n onIceCandidate,\n );\n };\n this.rtcPeerConnection.addEventListener(\"icecandidate\", onIceCandidate);\n });\n }\n\n // send re-INVITE.\n // If the call is on hold and you don't want to unhold it, set toReceive to false\n public async reInvite(toReceive: boolean = true) {\n const offer = await this.rtcPeerConnection.createOffer({\n iceRestart: true,\n });\n await this.rtcPeerConnection.setLocalDescription(offer);\n await this.waitForIceGatheringComplete();\n let sdp = this.rtcPeerConnection.localDescription!.sdp;\n // default value is `a=sendrecv`\n if (!toReceive) {\n sdp = sdp.replace(/a=sendrecv/g, \"a=sendonly\");\n }\n const requestMessage = new RequestMessage(\n `INVITE ${extractAddress(this.remotePeer)} SIP/2.0`,\n {\n \"Call-Id\": this.callId,\n From: this.localPeer,\n To: this.remotePeer,\n Via: `SIP/2.0/WSS ${fakeDomain};branch=${branch()}`,\n \"Content-Type\": \"application/sdp\",\n },\n sdp,\n );\n const replyMessage = await this.webPhone.sipClient.request(requestMessage);\n await this.rtcPeerConnection.setRemoteDescription({\n type: \"answer\",\n sdp: replyMessage.body,\n });\n const ackMessage = new RequestMessage(\n `ACK ${extractAddress(this.remotePeer)} SIP/2.0`,\n {\n \"Call-Id\": this.callId,\n From: this.localPeer,\n To: this.remotePeer,\n Via: replyMessage.headers.Via,\n CSeq: replyMessage.headers.CSeq.replace(\" INVITE\", \" ACK\"),\n },\n );\n await this.webPhone.sipClient.reply(ackMessage);\n }\n\n // handle re-INVITE from SIP server\n public async handleReInvite(reInviteMessage: InboundMessage) {\n this.sipMessage = reInviteMessage;\n await this.rtcPeerConnection.setRemoteDescription({\n type: \"offer\",\n sdp: reInviteMessage.body,\n });\n const answer = await this.rtcPeerConnection.createAnswer();\n await this.rtcPeerConnection.setLocalDescription(answer);\n await this.waitForIceGatheringComplete();\n\n const newMessage = new ResponseMessage(this.sipMessage, {\n responseCode: 200,\n headers: {\n \"Content-Type\": \"application/sdp\",\n },\n body: this.rtcPeerConnection.localDescription!.sdp,\n });\n await this.webPhone.sipClient.reply(newMessage);\n\n // note: no need to wait for the final SIP message (refer to inbound call answer function)\n // because nobody is supposed to proactively invoke this function.\n }\n\n // for hold/unhold\n // toggle between a=sendrecv and a=sendonly\n protected async toggleReceive(toReceive: boolean) {\n if (!this.rtcPeerConnection?.localDescription) {\n return;\n }\n let sdp = this.rtcPeerConnection.localDescription!.sdp;\n // default value is `a=sendrecv`\n if (!toReceive) {\n sdp = sdp.replace(/a=sendrecv/g, \"a=sendonly\");\n }\n // increase the sdp version\n const res = sdpTransform.parse(sdp);\n this.sdpVersion = Math.max(this.sdpVersion, res.origin!.sessionVersion + 1);\n res.origin!.sessionVersion = this.sdpVersion++;\n sdp = sdpTransform.write(res);\n const requestMessage = new RequestMessage(\n `INVITE ${extractAddress(this.remotePeer)} SIP/2.0`,\n {\n \"Call-Id\": this.callId,\n From: this.localPeer,\n To: this.remotePeer,\n Via: `SIP/2.0/WSS ${fakeDomain};branch=${branch()}`,\n \"Content-Type\": \"application/sdp\",\n },\n sdp,\n );\n const replyMessage = await this.webPhone.sipClient.request(requestMessage);\n const ackMessage = new RequestMessage(\n `ACK ${extractAddress(this.remotePeer)} SIP/2.0`,\n {\n \"Call-Id\": this.callId,\n From: this.localPeer,\n To: this.remotePeer,\n Via: replyMessage.headers.Via,\n CSeq: replyMessage.headers.CSeq.replace(\" INVITE\", \" ACK\"),\n },\n );\n await this.webPhone.sipClient.reply(ackMessage);\n }\n\n protected async sendJsonMessage(\n command: \"callpark\" | \"callflip\" | \"startcallrecord\" | \"stopcallrecord\",\n args: { [key: string]: string } = {},\n ) {\n const reqid = this.reqid++;\n const jsonBody = JSON.stringify({ request: { reqid, command, ...args } });\n const requestMessage = new RequestMessage(\n `INFO sip:${this.webPhone.sipInfo.domain} SIP/2.0`,\n {\n \"Call-Id\": this.callId,\n From: this.localPeer,\n To: this.remotePeer,\n Via: `SIP/2.0/WSS ${fakeDomain};branch=${branch()}`,\n \"Content-Type\": \"application/json;charset=utf-8\",\n },\n jsonBody,\n );\n await this.webPhone.sipClient.request(requestMessage);\n return new Promise((resolve) => {\n const resultHandler = (inboundMessage: InboundMessage) => {\n if (!inboundMessage.subject.startsWith(\"INFO sip:\")) {\n return;\n }\n const response = JSON.parse(inboundMessage.body).response;\n if (\n !response ||\n response.reqid !== reqid ||\n response.command !== command\n ) {\n return;\n }\n this.webPhone.sipClient.off(\"inboundMessage\", resultHandler);\n resolve(response.result);\n };\n this.webPhone.sipClient.on(\"inboundMessage\", resultHandler);\n });\n }\n\n protected async _transfer(\n uri: string,\n timeout = DEFAULT_TRANSFER_TIMEOUT_MS,\n ) {\n const requestMessage = new RequestMessage(\n `REFER ${extractAddress(this.remotePeer)} SIP/2.0`,\n {\n \"Call-Id\": this.callId,\n From: this.localPeer,\n To: this.remotePeer,\n Via: `SIP/2.0/WSS ${fakeDomain};branch=${branch()}`,\n \"Refer-To\": uri,\n \"Referred-By\": `<${extractAddress(this.localPeer)}>`,\n },\n );\n await this.webPhone.sipClient.request(requestMessage);\n\n // wait for the final SIP message\n let timeoutId: ReturnType;\n return new Promise((resolve, reject) => {\n const handler = (inboundMessage: InboundMessage) => {\n if (\n inboundMessage.subject.startsWith(\"BYE sip:\") &&\n inboundMessage.headers[\"Call-Id\"] === this.callId\n ) {\n clearTimeout(timeoutId);\n this.webPhone.sipClient.off(\"inboundMessage\", handler);\n resolve();\n }\n };\n timeoutId = setTimeout(() => {\n this.webPhone.sipClient.off(\"inboundMessage\", handler);\n reject(\n new Error(\n `\"REFER ${extractAddress(\n this.remotePeer,\n )} SIP/2.0\" request timed out. It often means either you don't have permission or the call is not in a correct state.`,\n ),\n );\n }, timeout);\n this.webPhone.sipClient.on(\"inboundMessage\", handler);\n });\n }\n}\n\nexport default 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