This document describes how to enter a TRTC room. Only after entering an audio/video room can a user subscribe to the audio/video streams of other users in the room or publish his or her own audio/video streams.

Call Guidelines

Step 1: Import SDK and Configure App Permissions

Please refer to the document Import SDK into the project to complete the SDK import.

Step 2: Create an SDK instance and set an event listener

Invoke various platform initialization interfaces to create the object instance of TRTC.
// Create a TRTCCloud singleton trtcCloud = (await TRTCCloud.sharedInstance())!; // Register TRTC event callback trtcCloud.registerListener(onRtcListener);

Step 3. Listen for SDK events

You can use callback APIs to listen for errors, warnings, traffic statistics, network quality, as well as various audio/video events of the SDK.
// We need to define a method to handle event callbacks, processing appropriately based on the type of the received event. // Taking onError as an example trtcCloud.registerListener(onRtcListener); onRtcListener(type, param) async { if (type == TRTCCloudListener.onError) { if (param['errCode'] == -1308) { MeetingTool.toast('Failed to initiate screen recording', context); } else { showErrordDialog(param['errMsg']); } } }

Step 4: Prepare the TRTCParams for entering the room

When calling the enterRoom interface, two key parameters need to be filled, namely TRTCParams and the application scene. A detailed introduction is as follows:

Parameter one: scene

This parameter refers to the specific application scene, whether it be video calls, interactive video broadcasting, audio calls, or interactive audio broadcasting:
Scenario Type

Scenario Introduction


Within the context of video calling, 720p and 1080p high-definition image quality is supported. A single room can accommodate up to 300 simultaneous online users, with a maximum of 50 users speaking at the same time.


In the context of interactive video broadcasting, the mic can be smoothly turned on/off without switching latency, with host latency as low as 300 milliseconds. Supports live streaming for hundreds of thousands of concurrent viewers, with playback delay reduced to 1000 milliseconds.
Note: In this scenario, you need to specify the current user's role using the 'role' field in TRTCParams.


In the audio call context, it supports 48 kHz duplex audio calls. A single room accommodates up to 300 concurrent online users, with a maximum of 50 people speaking at once.


In the context of interactive audio live streaming, microphones can be switched on and off smoothly without delay. The host experiences a low latency of fewer than 300 milliseconds. It accommodates hundreds of thousands concurrent viewer users, with the broadcast delay reduced to 1000 milliseconds.
Note: In this scenario, you need to specify the current user's role using the 'role' field in TRTCParams.

Parameter 2: TRTCParams

TRTCParams is composed of numerous parameters, but typically, your attention could be principally directed towards filling out the following parameters:
Parameter name

Field Description

Supplementary Information

Data Type



Application ID

You can locate the SDKAppID within the Tencent Real-Time Communication console, if not present, click on the "Create Application" button to institute a new application.




User ID

It can contain only letters, digits, underscores, and hyphens. In TRTC, a user cannot use the same user ID to enter the same room on two different devices at the same time.


"denny" or "123321"


The authentication ticket needed to enter a room

You can calculate userSig using the SDKAppID and userId. Please refer to Calculating and Using UserSig for the calculation method.




Room ID

Numeric type 'Room ID'. Be aware, if you wish to utilize a character sequence as the Room ID, please resort to the strRoomId field, rather than the roomId field, as strRoomId and roomId should not be used interchangeably.




Room ID

Room ID of string type. Note that `strRoomId` and `roomId` shouldn't be used interchangeably, as to the TRTC backend service, "123" and 123 are not the same room.





Divided into "Anchor" and "Audience" roles, this field only needs to be specified when the TRTCAppScene is designated as TRTCAppSceneLIVE or TRTCAppSceneVoiceChatRoom, these two live streaming scenarios.


TRTCRoleAnchor or TRTCRoleAudience

TRTC does not support the simultaneous entry of the same userId on two different devices. Doing so could lead to interference.
Each endpoint in the application scenario, appScene, must be unified to prevent unpredictable issues from cropping up.

Step 5: Enter the room (enterRoom)

After preparing the two parameters from Step 4 (application scenario and TRTCParams), you can call the enterRoom function to enter the room.
enterRoom() async { try { userInfo['userSig'] = await GenerateTestUserSig.genTestSig(userInfo['userId']); meetModel.setUserInfo(userInfo); } catch (err) { userInfo['userSig'] = ''; print(err); }
// If your scenario is "interactive video live broadcast", please set the scene to TRTC_APP_SCENE_LIVE, and set the appropriate value for the role field in TRTCParams. await trtcCloud.enterRoom( TRTCParams( sdkAppId: GenerateTestUserSig.sdkAppId, userId: userInfo['userId'], userSig: userInfo['userSig'], role: TRTCCloudDef.TRTCRoleAnchor, roomId: meetId!), TRTCCloudDef.TRTC_APP_SCENE_LIVE); }
Event Callback If room entry is successful, SDK will return the onEnterRoom(result) event where result is a positive number, indicating the time consumed to join the room in milliseconds (ms). If room entry fails, SDK will also call back the onEnterRoom(result) event, but the parameter result will be a negative number, representing the error code of room entry failure.
//Listen for SDK's onEnterRoom event to check if room entry is successful or not
onRtcListener(type, param) async {
if (type == TRTCCloudListener.onEnterRoom) { if (param > 0) { MeetingTool.toast('Enter room success', context); } }