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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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Linux GSO can delay the segmentation of multiple UDP packets to improve performance, refer to UDP GSO principle and application.
rtc-plaintext-linux4-gso-ok.pcapng.zip rtc-plaintext-multiple-slices-as-one-NALU.pcapng.zip
rtc-plaintext-linux3-gso-invalid.pcapng.zip SRS has added an API for packet performance analysis: http://localhost:1985/api/v1/perf. You can use the tool to analyze it: No GSO, Fragment at SourceWithout turning on GSO, when receiving packets from the source, the messages are divided into RTP packets, and the statistics are as follows:
No GSO, Fragment at ConnectionWithout enabling GSO, when sending packets (Connection), the messages are divided into RTP packets. The statistics are as follows:
GSO, Fragment at ConnectionWhen GSO is enabled, the messages are divided into RTP packets during packet transmission (Connection), and the statistical data is as follows:
GSO, Larger FU-PayloadPreviously, the length of FU Payload was 1200, and it was changed to 1300, referring to bfc70d64 and b91e07f4. After the modification, the maximum size of IP packets is 1356 bytes, which is smaller than the 1500 bytes MTU. From the results, it can be seen that the RTP packet size has decreased from 1.56 times to 1.49 times, and the GSO fragmentation is not affected. GSO, Padding PacketsThere are usually more audio packets, and sometimes the difference is not significant. For example, there are three packets: From a data perspective, by enabling padding (127), you can lower the GSO packet multiplier from 0.74 to 0.67, and improve the efficiency from 0.67 to 0.74.
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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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