1. Basics
What browsers does the TRTC Web SDK support?
How do I test my audio/video devices before making a call?
How do I test my current network quality?
I can use the TRTC web SDK in a local development environment but not after it is deployed online. What should I do?
To ensure data security and protect users’ privacy, browsers allow access to the mic and camera only in secure environments, for example, when https, localhost, or file:// is used for access. Because HTTP is not secure, browsers block access to media devices when HTTP is used.
If you can use the SDK in a local development environment but cannot capture data from the camera or mic after deploying it to the web, check whether your webpage is deployed using HTTP. If so, use HTTPS instead and make sure you have a valid HTTPS certificate.
Does the web SDK support the dual-stream mode, watermarking?
What are the known issues of WebRTC?
What is the main/sub video stream?
1. TRTC has a main video stream (main stream) and an sub video stream (sub stream).
2. The camera is published through the main stream, and the screen sharing is published through the sub stream.
2. Capturing and Publishing
Why can I turn on the camera or microphone?
Why am I unable to publish or play streams on some mobile browsers?
If your browser is not listed in the above document, you can open the TRTC compatibility check page with the browser to test whether it fully supports WebRTC. Can I change the video profile(resolution, frameRate, bitrate)?
Why does the camera support capturing 1080p, but the captured resolution is less than 1080p?
This situation is generally due to the fact that the camera is occupied by low-resolution ahead of time, resulting in the SDK not being able to capture the target resolution.
Solution: Check the business side logic, if the camera is captured in advance, please release the camera in time when it is no longer in use.
Can I modify the style of screen sharing pop-up window in the TRTC Web SDK?
No. The style of screen sharing is controlled by browsers and cannot be modified.
How do I capture system audio during screen sharing in the TRTC Web SDK?
How do I switch the camera/mic?
How to detect camera added and removed?
What should I do if the error "Permission denied" occurs when I use the TRTC web SDK in iframes?
To use WebRTC in iframes, you need to add the following attribute to the iframe tag to obtain the permissions needed.
Mic, camera, and screen sharing permissions:
<iframe allow="microphone; camera; display-capture;">
3. Playing
Why is there video but no audio during a call?
It may be because of the browser’s autoplay policy, which causes the “PLAY_NOT_ALLOWED” error. To solve the problem, please see Handle Autoplay Restriction. Why does the sound output change from speaker to handset after microphone being captured?
On Android, there are usually multiple microphones, and the label list is: ['default', 'Speakerphone', 'Headset earpiece'].
If you do not specify the microphone in trtc.startLocalAudio, the browser default microphone may be the Headset earpiece and the sound will come out of the handset.
If you need to play out through the speaker, you need to specify the microphone with the label 'Speakerphone'.
4. Others
How to get current volume of microphone?
What triggers the trtc.on(TRTC.EVENT.KICKED_OUT) event?
The event is triggered when a user is removed from a room, for example, when the same user ID is used to log in from different devices or when the RESTful API RemoveUser is called to remove a user. For more information, see KICKED_OUT。 Note:
Repeated login is not allowed by the SDK (it may cause call exceptions) and should be avoided at the business layer.
Causion for Using Vue 3
When the Vue3 framework is used, it is necessary to use the markRaw interface to mark the trtc instance to avoid the conversion of trtc into the Proxy object by Vue, which may cause some unexpected problems.
import { markRaw } from 'vue';
const trtc = markRaw(TRTC.create());
Why mobile phone get hot when make a multi-person video call
