TRTC supports four room entry modes. Video call (VideoCall) and audio call (AudioCall) are the call modes, and interactive video streaming (Live) and interactive audio streaming (VoiceChatRoom) are the live streaming modes.
The call modes allow a maximum of 300 users in each TRTC room, and up to 50 of them can speak at the same time. The call modes are suitable for scenarios such as one-to-one video calls, video conferencing with up to 300 participants, online medical consultation, remote interviews, video customer service, and online Werewolf playing.
How It Works
TRTC services use two types of server nodes: access servers and proxy servers.
Access server
This type of nodes use high-quality lines and high-performance servers and are better suited to drive low-latency end-to-end calls, but the unit cost is relatively high.
Proxy server
This type of servers use mediocre lines and average-performance servers and are better suited to power high-concurrency stream pulling and playback. The unit cost is relatively low.
In the call modes, all users in a TRTC room are assigned to access servers and are in the role of “anchor”. This means the users can speak to each other at any point during the call (up to 50 users can send data at the same time). This makes the call modes suitable for use cases such as online conferencing, but the number of users in each room is capped at 300.
Sample Code
You can visit GitHub to obtain the sample code used in this document.
Note:
If your access to GitHub is slow, download the ZIP file here.
Directions
Step 1. Integrate the SDKs
You can integrate the TRTC SDK into your project in the following ways:
Method 1: automatic loading (AAR)
The TRTC SDK has been released to the mavenCentral repository, and you can configure Gradle to download updates automatically.
The TRTC SDK has integrated TRTC-API-Example, which offers sample code for your reference. Use Android Studio to open your project and follow the steps below to modify the app/build.gradle file.
Step 4. Assemble the room entry parameter TRTCParams
When calling the enterRoom() API, you need to pass in a key parameter TRTCParams, which includes the following required fields:
Parameter
Type
Description
Example
sdkAppId
Number
Application ID, which you can view in the TRTC console.
1400000123
userId
String
Can contain only letters (a-z and A-Z), digits (0-9), underscores, and hyphens. We recommend you set it based on your business account system.
test_user_001
userSig
String
userSig is calculated based on userId. For the calculation method, please see UserSig.
eJyrVareCeYrSy1SslI...
roomId
Number
Numeric room ID. For string-type room ID, use strRoomId in TRTCParams.
29834
Note:
In TRTC, users with the same userID cannot be in the same room at the same time as it will cause a conflict.
Step 5. Create and enter a room
1. Call enterRoom() to enter the audio/video room specified by roomId in the TRTCParams parameter. If the room does not exist, the SDK will automatically create it with the roomId value as the room number.
2. Set the appScene parameter according to your actual application scenario. Inappropriate appScene values may lead to increased lag or decreased clarity.
For video calls, please set TRTC_APP_SCENE_VIDEOCALL.
For audio calls, please set TRTC_APP_SCENE_AUDIOCALL.
3. You will receive the onEnterRoom(result) callback. If result is greater than 0, room entry succeeds, and the value of result indicates the time (ms) room entry takes; if result is less than 0, room entry fails, and the value is the error code for the failure.
public void enterRoom(){
TRTCCloudDef.TRTCParams trtcParams = new TRTCCloudDef.TRTCParams();
toastTip("Entered room successfully; the total time used is [\\(result)] ms")
}else{
toastTip("Failed to enter the room; the error code is [\\(result)]")
}
}
Note:
If the room entry fails, the SDK will also call back the onError event and return the parameters errCode (error code), errMsg (error message), and extraInfo (reserved parameter).
If you are already in a room, you must call exitRoom to exit the room before entering another room.
The value of appScene must be the same on each client. Inconsistent appScene may cause unexpected problems.
Step 6. Subscribe to remote audio/video streams
The SDK supports automatic subscription and manual subscription.
Automatic subscription (default)
In automatic subscription mode, after room entry, the SDK will automatically receive audio streams from other users in the room to achieve the best "instant broadcasting" effect:
1. When another user in the room is upstreaming audio data, you will receive the onUserAudioAvailable() event notification, and the SDK will automatically play back the audio of the remote user.
3. When another user in the room is upstreaming video data, you will receive the onUserVideoAvailable() event notification; however, since the SDK has not received instructions on how to display the video data at this time, video data will not be processed automatically. You need to associate the video data of the remote user with the display view by calling the startRemoteView(userId, view) method.
Fill: aspect fill. The image may be scaled up and cropped, but there are no black bars.
Fit: aspect fit. The image may be scaled down to ensure that it’s displayed in its entirety, and there may be black bars.
5. Call stopRemoteView(userId) to block the video data of a specified user (userId) or stopAllRemoteView() to block the video data of all remote users. The SDK will stop pulling the video data of the user(s).
@Override
public void onUserVideoAvailable(String userId, boolean available){
If you do not call startRemoteView() to subscribe to the video stream immediately after receiving the onUserVideoAvailable() event callback, the SDK will stop pulling the remote video within 5 seconds.
Manual subscription
You can call setDefaultStreamRecvMode() to switch the SDK to the manual subscription mode. In this mode, the SDK will not pull the data of other users in the room automatically. You have to start the process manually via APIs.
2. If other users in the room are sending audio data, you will receive the onUserAudioAvailable() notification, and you need to call muteRemoteAudio(userId, false) to manually subscribe to the users’ audio. The SDK will decode and play the audio data received.
3. If a remote user in the room is sending video data, you will receive the onUserVideoAvailable() notification, and you need to call startRemoteView(userId, remoteView) to manually subscribe to the user's video data. The SDK will decode and play the video data received.
Step 7. Publish the local stream
1. Call startLocalAudio() to enable local mic capturing and encode and send the audio captured.
2. Call startLocalPreview() to enable local camera capturing and encode and send the video captured.
Fill: aspect fill. The image may be scaled up and cropped, but there are no black bars.
Fit: aspect fit. The image may be scaled down to ensure that it’s displayed in its entirety, and there may be black bars.
4. Call setVideoEncoderParam() to set the encoding parameter of the local video, which determines the image quality of the video watched by other users in the room.
// Sample code: publish the local audio/video stream
Call exitRoom() to exit the room. The SDK disables and releases devices such as cameras and mics during room exit. Therefore, room exit is not an instant process. It completes only after the onExitRoom() callback is received.
// Please wait for the `onExitRoom` callback after calling the room exit API.
mTRTCCloud.exitRoom()
@Override
publicvoidonExitRoom(int reason){
Log.i(TAG,"onExitRoom: reason = "+ reason);
}
Note:
If your application integrates multiple audio/video SDKs, please wait after you receive the onExitRoom callback to start other SDKs; otherwise, the device busy error may occur.
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