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WebRTC: RTMP2RTC failure, RTC2RTMP failure, unable to achieve live streaming and WebRTC conversion, RTMP to WebRTC black screen, WebRTC to RTMP black screen. #2728
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English GuideFor the config Please use config Please use config For example: ./objs/srs -c rtmp2rtc.conf Chinese GuideDifferent configuration files must be used. Please refer to the content of the configuration for detailed differences. The main difference is that audio transcoding will not be automatically enabled due to high CPU consumption. Therefore, it must be explicitly enabled when converting between WebRTC and RTMP. Please refer to the official website documentation for updated configuration information.
For example: ./objs/srs -c rtmp2rtc.conf For example, RTMP streaming and FLV playback do not automatically enable RTC transcoding. If enabled, it can only support around 100 streams, but if not enabled, it can support around 3000 streams. Of course, enabling transcoding will have a significant impact on performance. RTC streaming and RTC playback are the same, they will not be transcoded to RTMP by default.
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So, have you solved your problem? I have also encountered such an issue.
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@xlzcjj It's resolved. The configuration files of the new version and the old version are not compatible. You can directly use the file "rtmp2rtc.conf" under the "conf" directory. When starting, use the command "-c conf/rtmp2rtc.conf".
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This problem is too common. We have optimized the error message. If the RTMP stream already exists but the
This allows errors to appear more quickly instead of having a black screen, as there are too many possibilities for a black screen. This obvious configuration issue should immediately result in an error to reduce troubleshooting time. From a business logic perspective, if an RTMP stream has already been pushed, it is not possible to push another RTC stream with the same URL. Therefore, it must belong to the RTMP to RTC conversion scenario.
Please refer to: 8bc2759
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Description
According to the official website documentation, I have installed SRS and used RTMP for streaming and WebRTC for playback. RTMP streaming is successful, and RTMP and FLV can be played back successfully. However, WebRTC cannot be played back. I have already modified the candidate to an external network address. Every time I click "Play Video", I get an error message saying "client disconnect peer. ret=1007"!
Previously, I followed the steps in the blog https://www.cnblogs.com/ziyue7575/p/13927894.html to install it four or five times without any issues. Please help me resolve this problem. Thank you.
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TRANS_BY_GPT3
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