I have been trying to transmit some high quality audio stream through WebRTC. Opus, the main advertised codec seems perfect since it can support up to 510kbit/s, way more than needed.
The problem is, setting up the Webrtc SDP is way less obvious than it seems. Thanks to Muaz Khan great work, I have been able to force it to 128kbit/s. 
Basically the code looks like that: 

```
 function setBandwidth(sdp) {
 var sdpLines = sdp.split('\r\n');

// Find opus payload.
var opusIndex = findLine(sdpLines, 'a=rtpmap', 'opus/48000');
var opusPayload;
if (opusIndex) {
    opusPayload = '109';
}
sdpLines[opusIndex]='a=rtpmap:'+opusPayload+' opus/48000/2';

   var mediaIndex = findLine(sdpLines, 'm=audio');
sdpLines[mediaIndex]=(sdpLines[mediaIndex].slice(0,(sdpLines[mediaIndex].indexOf("RTP/SAVPF")+10))).concat(opusPayload); 
 var abIndex = findLine(sdpLines, 'a=mid:');
  sdpLines[abIndex]='a=mid:audio\r\nb=AS:300000';

// Find the payload in fmtp line.
  var fmtpLineIndex = findLine(sdpLines, 'a=fmtp:' + opusPayload.toString());

if (fmtpLineIndex == null) {
    sdpLines[opusIndex] = sdpLines[opusIndex].concat('\r\n'+'a=fmtp:' + opusPayload.toString()+ ' minptime=10; useinbandfec=1; maxaveragebitrate='+128*1024+'; stereo=1; sprop-stereo=1 ; cbr=1');
     sdp = sdpLines.join('\r\n');

    return sdp;
}

// Append stereo=1 to fmtp line.
// added maxaveragebitrate here; about 50 kbits/s
// added stereo=1 here for stereo audio
// x-google-min-bitrate=50; x-google-max-bitrate=50
sdpLines[fmtpLineIndex] = sdpLines[fmtpLineIndex].concat('; maxaveragebitrate='+128*1024+'; stereo=1; sprop-stereo=1 ; cbr=1');

sdp = sdpLines.join('\r\n');
return sdp;
}

```

So now everything is set, both firefox and chrome display the right value for sender and receiver, the communication opens, the music is played!

```
adding answer-sdp v=0
o=mozilla...THIS_IS_SDPARTA-42.0 502631676322875352 0 IN IP4 0.0.0.0
s=-
t=0 0
a=fingerprint:sha-256.....
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 9 RTP/SAVPF 109 
c=IN IP4 0.0.0.0
a=recvonly
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=ice-pwd:c56d106030599efe08cfa2a4f9b3ad5a
a=ice-ufrag:93982a76
a=mid:audio
b=AS:300000
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=fmtp:109 minptime=10; useinbandfec=1; maxaveragebitrate=131072; stereo=1; sprop-stereo=1 ; cbr=1
a=setup:active
a=ssrc:1948755120 cname:{208483df-13c9-e347-ba4a-c71604df3ad9}

```

But the quality is terrible. Chrome shows about 30kbit/s on chrome://webrtc-internals/ and the sound is heavily distorted with variable volume...
Any leads on the issue?