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Cool.....would Flashphoner server be able to receive the stream from browser with an ID (similar to RTMP, eg. rtmp://<ip>:<port>/live/<rtmp_id>)?

Does WebRTC support video streaming via SRT protocol?

Do I need a high CPU instance as edge since it will just serve as WRTC server to collect all streams?

Yes, the bottleneck is indeed ffmpeg as it uses up a lot of CPU resources especially when there are too many concurrent sessions (we are offering SaaS to our clients to live stream to multiple social platforms) running, however as there is long silent period especially from 12am - 12pm (as...

Could you explain the above using similar diagram? FYI, the reason that we want to put auto scaling group of instances behind LB is that ffmpeg (used to edit then forward streams to FB etc) used up a lot of CPU resources when launched; due to the unpredictable usage of our services by our...

Pls refer to attached for scenario i described above.

Not republishing....what i meant is the WebRTC stream triggered from the browser seems to be forwarded to wrong EC2 instance behind LBS when there are 2 instances running behind. For example, when a stream is started from web browser, and the websocket connection is to instance 1, but when I...

From browser, I used the modified 'webrtc-as-rtmp-republishing.js' to connect to WebRTC server in instances behind the load balancer, then stream RTMP traffic to it; so with 2 and above instances behind LB, I notice the session would fail (presume the RTMP stream was send to wrong WebRTC server).

I constantly encountering as issue when there are 2 EC2 instances sit behind LB - when I start a stream thru webRTC, its ended with stream not found; I suspect it is due to the fact that the websocket was established with instance 1, but the RTMP traffic (to port 1935) was sent to instance 2...

noted

I'm seeing below errors (Flashphoner is installed on AWS Linux ARM instance): 08:42:09,475 ERROR FFAudioGenericCodec - main Can not load lib: fpaj java.lang.UnsatisfiedLinkError: /usr/local/FlashphonerWebCallServer-5.2.1031-aa61467f4f8b95d229f3faaa04f6c6e8d43c388d/lib/so/libfpaj.so...

but what about the next month when i renew the license? Its impractical to update the snapshot and template on monthly basis.

If I were to renew my license on monthly basis, whats the most effective way to ensure the newly launched instance (2nd and above) would start with the new license?

ignore above....i got it working now after changing the right JS file

I changed to following: function startStreaming(session) { var streamName = urlParams.get('user_id'); var rtmpUrl = "rtmp://localhost:1935/live/" + "rtmp_" + streamName; session.createStream({ name: streamName, display: localVideo, cacheLocalResources: true...

anyway, I just tried with yr suggested method to add 'rtmp_' in front the of stream name, yet I could not start 2 streams in parallel.

function startStreaming(session) { var streamName = urlParams.get('user_id'); var rtmpUrl = "rtmp://localhost:1935/live"; session.createStream({ name: streamName, display: localVideo, cacheLocalResources: true, receiveVideo: false...

if you looked into the log file (server flashphoner log), i did specify 2 different RTMP stream name when I started 2 streams in parallel - dc78 & 100000000000003/100000000000008), but yet the 2nd stream failed. In below portion of the JS code, rtmpStreamPrefix is actually added into the URL...

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the issue does not seem to be on forwarded destinations....even after i disabled my forwarding script, I still could not start 2 RTMP streams in parallel from 2 different browsers to WebRTC server.