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Hello, I've ran into an issue where if I'm inputting a WebRTC stream (via WHIP or the custom signalling protocol) I'm unable to transcode that input stream, e.g. with the following encodes element. <Encodes>
<Video>
<Name>test</Name>
<Codec>h264</Codec>
<Bitrate>20000000</Bitrate>
<Width>1280</Width>
</Video>
</Encodes> Any player which attempts to watch the stream will fail and a "stream has created but not started" warning along with a "Cannot find stream" error will show in the server log
Interestingly if I switch from a WebRTC input to a RTMP input, everything works fine! Players can connect to the transcoded stream. If I remove that transcode and just do a video bypass instead then the WebRTC stream can be watched fine as well. In all cases the player is trying to connect to "ws://192.168.1.144:3333/app/stream" and the video stream is being input to "app/stream" via either the rtmp or webrtc input. Looking at the server log when the webrtc input stream is setup shows the following
and the rtmp input shows the following
From what I understand this shows that the transcoded stream is created with RTMP but is not with WebRTC input. In both cases we see the 1920x1080 input, but only the RTMP input actually creates the 1280x720 transcode stream. Any idea why this might be? If it matters this is using an image created from master, not the last official release. Thanks. |
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Replies: 1 comment 3 replies
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Try adding Framerate to your encoding settings. And it would be a good idea to also specify Height.
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Try adding Framerate to your encoding settings. And it would be a good idea to also specify Height.