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i changed it to false, but cant even start a stream after.

so meaning if I changed rtmp_transponder_full_url=false, it shall work?

requested info sent.

yes, i have a script to republish edited RTMP streams to FB and server itself (created another RTMP stream) via ffmeg....that works well for 1 stream. I have even tried turning off my script, meaning no RTMP streaming forwarding....but even with that I could not start 2 streams concurrently.

both streams from browser via WebRTC

More error msg: 12:04:14,014 INFO RestClient - API-ASYNC-pool-13-thread-3 content ->...

I cant seem to start 2 streams (with different ID) to my webRTC server, seeing below msg: 08:12:22,762 INFO RestClient - API-ASYNC-pool-13-thread-10 content ->...

I hv configured LB in AWS to forward http/https (port 80, 8443 and 8888) to webcallserver sitting in a auto scaling group, but how about RTMP traffic? How would it figure which server (if more than 1 in auto scaling group) to forward the RTMP traffic to?

As I currently pointing to my single Flashphoner instance as the webrtc server in the exisitng JS file: var URL = "wss://52.221.208.40:8443" I just need to replace it with LB hostname right? Also to use port 8443 in the LB protocol setting?

yes...I'm using custom image.

I meant 'renewed on monthly basis'

Currently I hv a license that its renewly on monthly basis, can I apply that to all scalable instances?

I would like to find out whats the best way to create auto scaling group in AWS to host Flashphoner instances behind a load balancer, questions like: a) how to configure load balancer? Determine in flow request by port 8443? b) How to apply license on scaled instances?

noted, thanks

Is there a way to kill by stream name?

how do I tear down/end the session from webrtc server? Using below? /usr/bin/curl -X POST "http://127.0.0.1:8081/rest-api/pull/rtmp/terminate" -H "Content-Type:application/json" -d '{"uri":"rtmp://127.0.0.1/live/10003"}'

As I read more into https://docs.flashphoner.com/display/WCS52EN/From+another+server+via+RTMP, it is to stream from a RTMP server as source...however what I intend to achieve is following: i) Create a rtmp session in webrtc server with a stream name ii) stream from Zoom to above RTMP session...

I use the following command to start the RTMP pull: /usr/bin/curl -X POST "http://127.0.0.1:8081/rest-api/pull/rtmp/pull" -H "Content-Type:application/json" -d '{"uri":"rtmp://127.0.0.1/live/10003","localStreamName":"test","record":"false"}' Whats the command to find the session and terminate...

Don't I need to create a session with a designated stream ID in Webrtc server before I could stream to it? I believe I would hv to modify the following code to start the stream, but wonder how do i update it to just start a stream, but not taking local VIDEO as source: function...

I learn that Zoom allows user to stream to custom platform, wonder if i could configure it stream to my webrtc server, if so, how?