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WebRTC publish / play demos automatically close connections

Hi there I sent an email to the preview location. However I thought I would try here.

I have configured webrtc and I can see logs and messages. However when trying to subscribe to a stream with the play application I am publishing via RTMP it automatically closes with this log “webrtc.js:179 wsConnection.onclose”

Is publishing to webrtc only intended via inputs from the browser rather than consuming other stream sources ? Can it accept H264 and AAC audio as I’m not hearing audio on the local video in the publisher either.

If I try publishing via the browser, the publisher also closes upon trying to publish with the same log.

The client code may have issues or incomplete ?

I have done webrtc tests with simplewebrtc library and it works with the signalling server. Nothing yet with wowza . I see no visible error log. All SSL certs are loaded correctly and the websocket service is loading correctly.

Your form is broken. Don’t validate for topics as that is broken it clears after I enter something.

Hi,
Please open a ticket using the link below so we can investigate further.

In the ticket, please include a ZIP file containing the following directories:
[Wowza-Install]/conf/
[Wowza-Install]/logs/
[Wowza-Install]/htdocs/ (if being used)
[Wowza-Install]/transcoder/ (if being used)

Please make sure you provide logs which show Wowza Streaming Engine starting.

If you are not sure how to get this information please see the following tutorial. https://www.wowza.com/docs/how-to-create-a-compressed-zip-file-in-windows-os-x-and-linux

Regards,
Jason Hilton
Wowza Media Systems Support Team

Hi it seems I need Opus audio to publish properly. That is why it wasn’t working. As soon as I disabled audio it started working.

So AAC audio won’t work either and needs a transcoder for WebRTC to work ?

Hi,

I’m glad to hear that you’re making progress with your WebRTC workflow.
Please see the Known issues and limitations section of the WebRTC article which states the following:

  • Google Chrome, Microsoft Edge, and Mozilla Firefox don’t support AAC audio over WebRTC. Streams with AAC audio are played as video-only.

To publish with a RTMP based encoder and playback using a WebRTC, you may need to upgrade to the latest version of the Wowza Streaming Engine software (currently 4.7.1) and add the following Property to the WebRTC application if you’re having issues with the H.264 video.

Path: /Root/Application/RTP
Name: rtpForceH264Constraint
Type: Boolean
Value: True

To override the constraints value for H264 and should only be used when WebRTC is being used.

Regards,
Jason

H264 video is publishing I had to disable audio. So a transcoder is required to be setup or the encoder needs to publish with an Opus audio codec ?

What is that config for ?

My issue now is rolling all the custom code into a proper universal client. I tried to look at SimpleWebRTC but it seems alot of work to integrate. They are both doing different things.

I’ll get back to the browser based publishing test after, it definitely has issues and fails.