[asterisk-users] volume control in VoIP

[Shane Spencer]

I hate to think of the possible echo if you change the volume and a
sip device didn't know about it.  It wouldn't effectively use the echo
cancellation on board, I am not an expert with echo cancellation
however.

It should be possible to just multiply each byte of a waveform by a
percentage if the codec is slinear/ulaw/alaw or similar almost as
quickly as it passes from one network stream to the next.  I would
snag the source for like xmms if it can do software volume control and
peek into it.. could be valuable. Let us know how well the echo
cancellation works in speakerphone eh!