Leveraging Tencent's 21 years of experience in network and audio/video technologies, Tencent Real-Time Communication (TRTC) offers multi-person audio/video calls and low-latency interactive live streaming solutions. With TRTC, you can quickly develop cost-effective, low-latency, and high-quality interactive audio/video services. Audio/Video Call Solution TRTC provides a client SDK that works across platforms and regions. This solution can be easily used on webpages to enable audio and video call capability for all applications. Low-Latency Interactive Live Streaming Solution TRTC leverages Tencent Cloud's high-quality node resources, industry-leading network and audio/video technologies to helps you build interactive live streaming services with minimal lags and a latency below 1 second, taking live streaming into the era of CDN 2.0.
TRTC allows you to start, answer, and end audio/video calls on mobile browsers through mobile webpages or WeChat Mini Programs. They can also be directly implemented on PC, Mac, or third-party apps through webpages or SDK integration.
Instant Messaging (IM)
TRTC provides secure, stable, and highly accessible instant messaging service, which is supported by Tencent's IM technologies that sustain hundreds of millions of users. You can permanently use the free tier of IM services with limited features and only pay for usage exceeding the free quota.
TRTC supports on-cloud recording. It can be used with Tencent Cloud VOD and its comprehensives APIs to achieve storage, transcoding, distribution and other features.
Relayed Live Streaming
TRTC supports relayed live streaming and can be used with Tencent Cloud LVB to achieve live streaming based on HLS, RTMP, and FLV.
HD Video Call
Backed by Tencent Cloud's media processing algorithms and TBS kernel compatibility, TRTC can optimize video definitions, reduce blurs, and support 720P HD video calls. Each video call supports up to 10 people by default. The user experience on a mobile browser is comparable to that of a dedicated application.
Stable Network Communication
TRTC offers a global end-to-end latency of less than 300 ms, packet loss prevention rate of over 40%, and network jitter prevention of over 1,000 ms. It guarantees high-quality, smooth and stable audio/video communication even in weak network environments.