Creating a WebRTC receiver

Question

I am new to WebRTC and trying to figure out how to create a program outside a browser which receives a WebRTC audio stream and outputs it on speakers. Are there any WebRTC libraries for Java or C#? That receiver will be running on a linux machine.

--

I've been thinking about using getUserMedia() to access the microphone. But then: In what format will such a stream be transmitted? Let's say I use WebRTC2SIP and build a Java endpoint using JSIP; or I just use a socket and send the stream over http. What audio format will I get on the receiver side? So far I have read WebRTC does compress the stream somehow.

Answer 1

I guess there are two ways for you:

1. build the whole WebRTC voice engine for android/iOS or Mac etc., and just use the API provide by VOE.
2. build standalone NS/VAD/AECM/AGC modules and using it in your project. for example, you build standalone NS module for android mobile, you use AudioRecord(java layer, android things) to record sound from MIC, and do the noise suppression process on these data(jni layer, WebRTC things), and finally playback the processed data by using AudioTrack(java layer, android things).

EDIT:

1. for the 2nd situation, the format is PCM raw data.

Answer 2

Check out the working Audio demo and code at demo.easyrtc.com

The code is all open source and can be checked out at https://github.com/priologic/easyrtc

You can look for any known issues around easyRTC at our forum at https://groups.google.com/forum/#!forum/easyrtc

Also check out our main site at easyrtc.com