快速入門：管理會議室通話

簡介

在 Azure 通訊服務 （ACS） 會議室通話期間，您可以使用通話 SDK 或通話自動化 SDK 或兩者來管理通話。 在會議室通話中，您可以使用指派給會議室中參與者和屬性的角色來控制通話內動作。 每個參與者允許參與者的角色控制功能，而會議室屬性則套用至整個會議室通話。

呼叫 SDK

通話 SDK 是用戶端通話連結庫，可讓會議室通話中的參與者執行數個通話內作業，例如螢幕共用、開啟/關閉視訊、靜音/取消靜音等等。 如需功能的完整清單，請參閱 呼叫 SDK 概觀。

您可以根據指派給通話參與者的角色來控制功能。 例如，只有演示者可以共享螢幕。 如需參與者角色和許可權，請參閱 會議室概念。

呼叫自動化 SDK

通話自動化 SDK 是一個伺服器端連結庫，可讓系統管理員在中央和控制的環境中管理進行中的會議室通話。 不同於呼叫 SDK，通話自動化 SDK 作業與角色無關。 因此，通話系統管理員可以代表會議室通話參與者執行數個通話內作業。

下列清單描述會議室通話中可用的常見通話內動作。

聯機到會議室通話

通話自動化必須先連線到現有的會議室通話，才能執行任何通話內作業。 CallConnected或 ConnectFailed 事件會使用回呼機制引發，以指出連接作業是否分別成功或失敗。

csharp

Uri callbackUri = new Uri("https://<myendpoint>/Events"); //the callback endpoint where you want to receive subsequent events
CallLocator roomCallLocator = new RoomCallLocator("<RoomId>");
ConnectCallResult response = await client.ConnectAsync(roomCallLocator, callbackUri);

Java

String callbackUri = "https://<myendpoint>/Events"; //the callback endpoint where you want to receive subsequent events
CallLocator roomCallLocator =  new RoomCallLocator("<RoomId>");
ConnectCallResult response = client.connectCall(roomCallLocator, callbackUri).block();

JavaScript

const roomCallLocator = { kind: "roomCallLocator", id: "<RoomId>" };
const callbackUri = "https://<myendpoint>/Events"; // the callback endpoint where you want to receive subsequent events 
const response = await client.connectCall(roomCallLocator, callbackUri);

Python

callback_uri = "https://<myendpoint>/Events"  # the callback endpoint where you want to receive subsequent events
room_call_locator = RoomCallLocator("<room_id>")
call_connection_properties = client.connect_call(call_locator=room_call_locator, callback_url=callback_uri)

成功連線到會議室通話之後， CallConnect 就會透過回呼 URI 通知事件。 您可以使用 callConnectionId ，視需要擷取會議室通話上的通話連線。 下列範例代碼段會使用 callConnectionId 來示範此函式。

新增 PSTN 參與者

使用通話自動化，您可以撥出 PSTN 號碼，並將參與者新增至會議室通話。 不過，您必須設定空間以啟用 PSTN 撥出選項（EnabledPSTNDialout設定為 true），且 Azure 通訊服務 資源必須布建有效的電話號碼。

如需詳細資訊，請參閱 會議室快速入門。

csharp

var callerIdNumber = new PhoneNumberIdentifier("+16044561234"); // This is the ACS-provisioned phone number for the caller  
var callThisPerson = new CallInvite(new PhoneNumberIdentifier("+16041234567"), callerIdNumber); // The target phone number to dial out to
CreateCallResult response = await client.GetCallConnection(callConnectionId).AddParticipantAsync(callThisPerson);

Java

PhoneNumberIdentifier callerIdNumber = new PhoneNumberIdentifier("+16044561234"); // This is the ACS-provisioned phone number for the caller
CallInvite callInvite = new CallInvite(new PhoneNumberIdentifier("+16041234567"), callerIdNumber); // The phone number participant to dial out to
AddParticipantOptions addParticipantOptions = new AddParticipantOptions(callInvite);
Response<AddParticipantResult> addParticipantResultResponse = client.getCallConnectionAsync(callConnectionId)
    .addParticipantWithResponse(addParticipantOptions).block();

JavaScript

const callInvite = {
    targetParticipant: { phoneNumber: "+18008008800" }, // The phone number participant to dial out to
    sourceCallIdNumber: { phoneNumber: "+18888888888" } // This is the ACS-provisioned phone number for the caller
};
const response = await client.getCallConnection(callConnectionId).addParticipant(callInvite);

Python

caller_id_number = PhoneNumberIdentifier(
    "+18888888888"
) # TThis is the ACS-provisioned phone number for the caller
target = PhoneNumberIdentifier("+18008008800"), # The phone number participant to dial out to

call_connection_client = call_automation_client.get_call_connection(
    "call_connection_id"
)
result = call_connection_client.add_participant(
    target,
    opration_context="Your context",
    operationCallbackUrl="<url_endpoint>"
)

拿掉 PSTN 參與者

csharp

var removeThisUser = new PhoneNumberIdentifier("+16044561234");

// Remove a participant from the call with optional parameters
var removeParticipantOptions = new RemoveParticipantOptions(removeThisUser)
{
    OperationContext = "operationContext",
    OperationCallbackUri = new Uri("uri_endpoint"); // Sending event to a non-default endpoint
}

RemoveParticipantsResult result = await client.GetCallConnection(callConnectionId).RemoveParticipantAsync(removeParticipantOptions);

Java

CommunicationIdentifier removeThisUser = new PhoneNumberIdentifier("+16044561234");
RemoveParticipantOptions removeParticipantOptions = new RemoveParticipantOptions(removeThisUser)
                .setOperationContext("<operation_context>")
                .setOperationCallbackUrl("<url_endpoint>");
Response<RemoveParticipantResult> removeParticipantResultResponse = client.getCallConnectionAsync(callConnectionId)
    .removeParticipantWithResponse(removeParticipantOptions);

JavaScript

const removeThisUser = { phoneNumber: "+16044561234" };
const removeParticipantResult = await client.getCallConnection(callConnectionId).removeParticipant(removeThisUser);

Python

remove_this_user = PhoneNumberIdentifier("+16044561234")
call_connection_client = call_automation_client.get_call_connection(
    "call_connection_id"
)
result = call_connection_client.remove_participant(remove_this_user, opration_context="Your context", operationCallbackUrl="<url_endpoint>")

傳送 DTMF

將 DTMF 音調清單傳送給外部參與者。

csharp

var tones = new DtmfTone[] { DtmfTone.One, DtmfTone.Two, DtmfTone.Three, DtmfTone.Pound }; 
var sendDtmfTonesOptions = new SendDtmfTonesOptions(tones, new PhoneNumberIdentifier(calleePhonenumber))
{ 
	OperationContext = "dtmfs-to-ivr" 
}; 

var sendDtmfAsyncResult = await callAutomationClient.GetCallConnection(callConnectionId).GetCallMedia().SendDtmfTonesAsync(sendDtmfTonesOptions);

Java

List<DtmfTone> tones = Arrays.asList(DtmfTone.ONE, DtmfTone.TWO, DtmfTone.THREE, DtmfTone.POUND); 
SendDtmfTonesOptions options = new SendDtmfTonesOptions(tones, new PhoneNumberIdentifier(c2Target)); 
options.setOperationContext("dtmfs-to-ivr"); 
client.getCallConnectionAsync(callConnectionId)
	.getCallMediaAsync() 
	.sendDtmfTonesWithResponse(options) 
	.block(); 

JavaScript

const tones = [DtmfTone.One, DtmfTone.Two, DtmfTone.Three];
const sendDtmfTonesOptions: SendDtmfTonesOptions = {
	operationContext: "dtmfs-to-ivr"
};
const result: SendDtmfTonesResult = await client.getCallConnection(callConnectionId)
	.getCallMedia()
	.sendDtmfTones(tones, {
		phoneNumber: c2Target
	}, sendDtmfTonesOptions);
console.log("sendDtmfTones, result=%s", result);

Python

tones = [DtmfTone.ONE, DtmfTone.TWO, DtmfTone.THREE]
call_connection_client = call_automation_client.get_call_connection(
    "call_connection_id"
)

result = call_connection_client.send_dtmf_tones(
	tones = tones,
	target_participant = PhoneNumberIdentifier(c2_target),
	operation_context = "dtmfs-to-ivr")

通話錄音

Azure 通訊服務 會議室支援通話自動化所提供的錄製功能，包括 start、pausestop、、 resume等。 請參閱下列代碼段，以在會議室通話中啟動/停止/暫停/繼續錄製。 如需動作的完整清單，請參閱 通話自動化錄製。

csharp

// Start recording
StartRecordingOptions recordingOptions = new StartRecordingOptions(new ServerCallLocator("<ServerCallId>"))
{
   RecordingContent = RecordingContent.Audio,
   RecordingChannel = RecordingChannel.Unmixed,
   RecordingFormat = RecordingFormat.Wav,
   RecordingStateCallbackUri = new Uri("<CallbackUri>"),
   RecordingStorage = RecordingStorage.CreateAzureBlobContainerRecordingStorage(new Uri("<YOUR_STORAGE_CONTAINER_URL>"))
};
Response<RecordingStateResult> response = await callAutomationClient.GetCallRecording()
.StartAsync(recordingOptions);

// Pause recording using recordingId received in response of start recording.
var pauseRecording = await callAutomationClient.GetCallRecording ().PauseAsync(recordingId);

// Resume recording using recordingId received in response of start recording.
var resumeRecording = await callAutomationClient.GetCallRecording().ResumeAsync(recordingId);

// Stop recording using recordingId received in response of start recording.
var stopRecording = await callAutomationClient.GetCallRecording().StopAsync(recordingId);

Java

// Start recording
StartRecordingOptions recordingOptions = new StartRecordingOptions(new ServerCallLocator("<serverCallId>"))
                    .setRecordingChannel(RecordingChannel.UNMIXED)
                    .setRecordingFormat(RecordingFormat.WAV)
                    .setRecordingContent(RecordingContent.AUDIO)
                    .setRecordingStateCallbackUrl("<recordingStateCallbackUrl>");

Response<RecordingStateResult> response = callAutomationClient.getCallRecording()
.startWithResponse(recordingOptions, null);

// Pause recording using recordingId received in response of start recording
Response<Void> response = callAutomationClient.getCallRecording()
              .pauseWithResponse(recordingId, null);

// Resume recording using recordingId received in response of start recording
Response<Void> response = callAutomationClient.getCallRecording()
               .resumeWithResponse(recordingId, null);

// Stop recording using recordingId received in response of start recording
Response<Void> response = callAutomationClient.getCallRecording()
               .stopWithResponse(recordingId, null);

JavaScript

// Start recording
var locator: CallLocator = { id: "<ServerCallId>", kind: "serverCallLocator" };

var options: StartRecordingOptions =
{
  callLocator: locator,
  recordingContent: "audio",
  recordingChannel:"unmixed",
  recordingFormat: "wav",
  recordingStateCallbackEndpointUrl: "<CallbackUri>"
};
var response = await callAutomationClient.getCallRecording().start(options);

// Pause recording using recordingId received in response of start recording
var pauseRecording = await callAutomationClient.getCallRecording().pause(recordingId);

// Resume recording using recordingId received in response of start recording.
var resumeRecording = await callAutomationClient.getCallRecording().resume(recordingId);

// Stop recording using recordingId received in response of start recording
var stopRecording = await callAutomationClient.getCallRecording().stop(recordingId);

Python

# Start recording
response = call_automation_client.start_recording(call_locator=ServerCallLocator(server_call_id),
            recording_content_type = RecordingContent.Audio,
            recording_channel_type = RecordingChannel.Unmixed,
            recording_format_type = RecordingFormat.Wav,
            recording_state_callback_url = "<CallbackUri>")

# Pause recording using recording_id received in response of start recording
pause_recording = call_automation_client.pause_recording(recording_id = recording_id)

# Resume recording using recording_id received in response of start recording
resume_recording = call_automation_client.resume_recording(recording_id = recording_id)

# Stop recording using recording_id received in response of start recording
stop_recording = call_automation_client.stop_recording(recording_id = recording_id)

終止呼叫

您可以使用通話自動化 SDK 掛斷動作來終止通話。 當掛斷動作完成時，SDK 會 CallDisconnected 發佈事件。

csharp

_ = await client.GetCallConnection(callConnectionId).HangUpAsync(forEveryone: true); 

Java

Response<Void> response = client.getCallConnectionAsync(callConnectionId).hangUpWithResponse(true).block();

JavaScript

await callConnection.hangUp(true);

Python

call_connection_client = call_automation_client.get_call_connection(
    "call_connection_id"
)

call_connection_client.hang_up(is_for_everyone=True)

其他動作

會議室通話中也支援下列通話內動作。

1. 新增參與者 （ACS 識別碼）
2. 移除參與者 （ACS 識別碼）
3. 取消新增參與者 （ACS 識別碼和 PSTN 號碼）
4. 掛接通話
5. 取得參與者 （ACS 識別碼和 PSTN 號碼）
6. 取得多個參與者（ACS 識別碼和 PSTN 號碼）
7. 取得通話的最新資訊
8. 播放音訊檔案和文字
9. 播放所有音訊檔案和文字
10. 同時辨識 DTMF 和語音
11. 辨識連續 DTMF

如需詳細資訊，請參閱 呼叫動作 和 媒體動作。

下一步

在本節中，您已了解如何：

• 從應用程式加入會議室通話
• 使用通話 SDK 將通話內動作新增至會議室通話
• 使用通話自動化 SDK 將通話內動作新增至會議室通話

您可能也會想要：

• 瞭解 會議室概念
• 了解 通話 SDK 功能
• 了解 通話自動化概念