Setting Default Video and Audio setting for WebRTC as RTMP Streaming

[jasonkc]

Hi,

What are the parameters I should used to ensure all RTMP streams are encoded with H.264 & AAC?

Best Rgrds,
Jason

 

[Max]

Hello

All push RTMP streams (WebRTC as RTMP) are encoded with H.264 AAC by default.

Audio encoding
H.264 + Opus 48000 (WebRTC) > H.264 + AAC 48000 (RTMP)

Audio + Video encoding
VP8 + Opus 48000 (WebRTC) > H.264 + AAC 48000 (RTMP)

You can check http://host:8081?action=stat and native_resources output. Here you can find all active encoders.

Load and resource usage information - Web Call Server 5.2 - EN - Flashphoner Documentation

docs.flashphoner.com

 

[jasonkc]

noted..so if if just want to forward the RTMP stream to FB using ffmpeg, could I just use options - c:v copy and -c:a copy?

 

[Max]

noted..so if if just want to forward the RTMP stream to FB using ffmpeg, could I just use options - c:v copy and -c:a copy?

Please clarify the case: do you want to publish a stream to WCS from webcam, then republish this stream to a third party server, then publish the stream from third party server to Facebook using ffmpeg?
If yes, the common response is yes, you can republish the stream to Facebook as is. But this depends on your third party server which accepts RTMP stream republished from WCS.
You can also republish WebRTC stream as RTMP directly from WCS to Facebook (if this is not locked in the country where server instance is placed of course).