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WebRTC: Support WebRTC, for video meeting, SFU server, video converence. #307
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candidate is the address that RTC provides services to the outside. In certain deployment scenarios, we need to configure it in multiple ways. This is also the only item that RTC must confirm the configuration for, while others can be left at their default settings. For detailed configuration instructions, please refer to: https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate
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MacPro information:
Docker information:
BaselineBaseline without any optimization, the data is as follows:
SRTP-NASMSRTP enabled openssl, and openssl enabled assembly, which means that SRTP encryption and decryption are optimized with assembly. Refer to 5e06a256. The data is as follows:
UDP sendmmsg + mw_latencyAfter enabling sendmmsg, multiple UDP packets can be sent at once. The data is as follows:
Although the optimization of queues over 100ms may not be significant, I think there may be other bottlenecks. From the data above, it can be seen that sendmmsg is very useful for optimizing system performance. Currently, the server queue is set by increasing the delay, also known as mw_latency. This allows the server to send more data at once, which is useful for TCP optimization. However, for UDP, especially in low-latency RTC scenarios, it is not advisable to increase the delay to improve performance. In fact, sendmmsg allows for sending data to multiple clients at once, and future optimization will be aimed in this direction.
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WebRTC is now quite mature, with stable playback and the protocol already being an RFC. There are also quite a few corresponding open-source projects. However, I believe that WebRTC still lacks a high-performance, simple and easy-to-use server. I have analyzed the existing servers before and found various issues. SRS has a great opportunity to solve these problems.
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