Product

Tencent Cloud Video Service

The approximate delay of a real-time audio and video SDK typically ranges from **50ms to 200ms**, depending on factors such as network conditions, encoding/decoding efficiency, and the SDK's optimization.  

For example:  
- A well-optimized SDK with low-latency protocols (e.g., WebRTC) can achieve **<100ms** end-to-end delay in ideal conditions.  
- In less optimal scenarios (e.g., high packet loss or unstable networks), the delay may increase to **150ms–200ms**.  

To minimize latency, developers can use **Tencent Cloud's Real-Time Communication (TRTC)** service, which provides ultra-low-latency audio and video transmission with global acceleration and intelligent routing. TRTC is designed for applications like online gaming, live streaming, and video conferencing, ensuring smooth and responsive communication.  

Example use case: A video conferencing app using TRTC can achieve **<100ms** delay for participants in the same region, while cross-region calls may have slightly higher latency but still remain under **200ms**.

The approximate delay of a real-time audio and video SDK typically ranges from 50ms to 200ms, depending on factors such as network conditions, encoding/decoding efficiency, and the SDK's optimization.  
For example:  

A well-optimized SDK with low-latency protocols (e.g., WebRTC) can achieve &lt;100ms end-to-end delay in ideal conditions.  
In less optimal scenarios (e.g., high packet loss or

What is the approximate delay of the real-time audio and video SDK?

The approximate delay of a real-time audio and video SDK typically ranges from 50ms to 200ms, depending on factors such as network conditions, encoding/decoding efficiency, and the SDK's optimization....