Hi,
We have WebRTC solution built using flashphoner and would like to allow our users to stream using rtmp to WebRTC.
I am just testing this on live server using OBS to stream and playing in the player in the demo app shipped with flashphoner. The issue I am facing is stream in the player would freeze for 10-20 seconds and latency is also 30 plus seconds.
Other thing I want to check is how to get the webhooks working when someone start streaming using OBS. We already have webhook when you stream using sdk to authenticate.
We have implemented connect, publishStream,stopStream, StreamStatusEvents
Anything we can do to stop that happening?
We have WebRTC solution built using flashphoner and would like to allow our users to stream using rtmp to WebRTC.
I am just testing this on live server using OBS to stream and playing in the player in the demo app shipped with flashphoner. The issue I am facing is stream in the player would freeze for 10-20 seconds and latency is also 30 plus seconds.
Other thing I want to check is how to get the webhooks working when someone start streaming using OBS. We already have webhook when you stream using sdk to authenticate.
We have implemented connect, publishStream,stopStream, StreamStatusEvents
Anything we can do to stop that happening?
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