Recording Remote Audio Streams / RecordRTC

issue: unable to record remote audio streams using RecordRTC.

"opus" is the default codec on chrome;

Chrome uses two modes for opus....

1. "voip" (i.e. voice mode) which is used if mono audio stream is provided by the sender
2. "audio" which is used if stereo audio stream is provided by the sender

There is no parameter in sdp allows us control voip/audio mode for outgoing/incoming audio streams. This job can be done by sender ( I don't know how, well, still experimenting ).

They're using code like this:

// Default to VoIP application for mono, and AUDIO for stereo.
int application = (channels == 1) ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO;

state->encoder = opus_encoder_create(48000, channels, application, &error);

You can see that if channels equals 1; i.e. if mono audio; they're using "voip" mode. Otherwise "audio" mode.

In the session description; you can see this line:

a=rtpmap:111 opus/48000/2

It is mono section with 48000 Hz (i.e. highest) clock rate.

Bitrate is 48-64 kb/s .... i.e. FB mono music

For stereo; we need:

64-128 kb/s .... i.e. FB stereo music