Many people believe that WebRTC is the only option regarding sub-second latency streaming due to its low latency and widespread browser support. However, other options may be better suited for specific use cases, especially when considering feedback from the open-source community and video cloud service customers.
Depending on the specific use case, WebRTC may be the best choice for web browsers, while SRT may be better suited for broadcasting devices. However, combining both may be necessary in some cases, and RTMP plus WebRTC can also provide sub-second latency.
In this discussion, I will explore the most effective solutions for these use cases using SRS technology to provide sub-second latency streaming services. Additionally, I will introduce the SRS project and community, which has existed for ten years and has over 20,000 stars on GitHub, making it a valuable resource for developers worldwide.
