Failed by RTMP writer error

[Andreas Ludwig]

Hello,
I am currently trying to test the WebRTC to RTMP functionality.

So whenever I start the Demo to stream webrtc to my webcall 5 server (running in a ubuntu vm on the same machine, bridged connection) the status is shown as "Publishing".

The status remains "Publishing" for ~10 seconds until it changes to "Failed".

The flashphoner_manager log shows two received Rest objects. First one is the message indicating that the stream is being published.
The second message indicated the status "FAILED" along with an info element:
"Failed by RTMP writer error"

Is it possible that my setup won't work due to the server being installed on a VM?

I can provide further information and logdata if needed for investigation.

 

[Max]

Hello
What is your stream target?
We had similar issue when publishing RTMP stream to Azure Live Streaming services

 

[Andreas Ludwig]

I was using Unreal Streaming media server for testing.
Streaming with OBS and the same parameters to that server worked fine.

 

[Max]

I have installed UMS. But I could not publish RTMP stream from Wirecast.
Could you check my screenshots and show yours?
Where can I find UMS logs?
1. Server properties.

2. Live broadcast.

3. Wirecast RTMP settings

4. Could not connect using Wirecast.

 

[Andreas Ludwig]

So I believe the "Live broadcast alias" should be what others call the stream name
rtmp://[rtmp-target-address]:[port]/[application-name]/[Live broadcast alias]

So you could try:
rtmp://192.168.88.254:5130/live/Test
I can send screenshots of my config when I'm back in the bureau tomorrow.

If this does not work out:
Is there a way to restream ingested WebRTC via rtmp with just the flashphoner server? I just picked ums since there is a free version and I wanted to give WebRTC -> RTMP restreaming a quick go.

 

[Max]

So you could try:
rtmp://192.168.88.254:5130/live/Test

That does not work for me too.
I have checked port 5130. It is listening

telnet 192.168.88.254 5130

So need to dig logs or some screen shots to move forward.

Is there a way to restream ingested WebRTC via rtmp with just the flashphoner server?

Yes. WCS supports both WebRTC and RTMP protocols.
Therefore you can publish WebRTC stream and just play this stream as RTMP.

Example:
1. Publish WebRTC stream.
https://wcs5-eu.flashphoner.com/demo2/two-way-streaming

2. Play WebRTC stream as RTMP.
https://wcs5-eu.flashphoner.com/demo2/flash-streaming

You can also try to pull RTMP stream from WCS server.
As I see from UMS docs, it can pull RTMP stream from another source.