TRTC is a cross-platform solution compatible with more than 5,000 device models. It provides client SDKs and TencentCloud APIs for mobile and desktop platforms including iOS, Android, Windows, macOS, and web. End users can also use its services via the Mini Program of WeChat, QQ, and WeCom.
As the provider of WeChat’s built-in SDK, TRTC offers user experience in WeChat Mini Program comparable to that of native applications.
You can run the TRTC demo and integrate basic TRTC features into your project with simple code. In as short as 1 minute, you can build from scratch a real-time audio/video communication product featuring low latency, low stutter rate, and high-quality. For detailed directions, please see Quick Demo Start and SDK Quick Integration.
TRTC provides a rich set of scenario-specific components to help you quickly and easily implement different features, including audio chat, conferencing, interactive live streaming, and interactive teaching. For detailed directions, please see Scenario-specific Practice.
TRTC offers reliable and secure network connection across the globe. It uses Tencent Cloud’s proprietary multi-level addressing algorithm and can connect to nodes across the entire network. Abundant high-bandwidth resources and globally-distributed edge servers allow it to keep the average global end-to-end latency below 300 ms.
TRTC reduces stutter through intelligent QoS control and encoding optimization. It can ensure high-quality, smooth, and stable audio/video communication even under poor network conditions (packet loss over 80% and network jitter over 1,000 ms).
TRTC allows 720p and 1080p video calls and guarantees smooth calling experience at 70% packet loss. It allows 48 kHz audio calls at a bitrate of 128 Kbps and guarantees smooth calling experience at 80% packet loss. Furthermore, it leverages industry-leading 3A (i.e., AEC, ANS, and AGC) processing technologies to remove echo and prevent howling, delivering a lossless audio quality comparable to that of CDs.