"Is it possible to extract audio from a WebRTC voice call and apply real-time effects to the audio stream?"

[mbedial]

"I was wondering if anyone has successfully extracted audio from a voice call in order to process the RTP stream. I'm not sure whether WebCallServer makes this easier compared to other solutions."

Thanks in advance

 

[Max]

Good day.
You can add a custom Java class to intercept a decoded audio frames at server side: Server audio processing. Note that this feature requires audio transcoding to be enabled on the server, that increases the server CPU load in its turn.

 

[mbedial]

Thanks a lot for your answer. That server audio proccesiing would also apply to SIP calls?

 

[Max]

The server audio processing may be applied to streaming only, not to SIP calls. WCS is just a SIP gateway, not a PBX, so it may redirect media streams from browser to SIP PBX and vice versa. We recommend to process SIP audio at SIP PBX side (Asterisk, OpenSIPS etc).