Realtime audio over UDP - Low quality

[David Ventura]

Hi. I'm trying to get audio (voice only) over udp on local network.

First: Listening to the mic directly through alsa gives me great quality;
there's no overdrive when shouting/talking WAY too close to the mic.

Now, when using the following pipeline the audio is .. ok. The quality is
not 'good'. It's good enough, but there's a clear quality drop from
directly listening to the mic. There's overdrive when there's shouting, etc.


Producer:

gst-launch-1.0  alsasrc slave-method=resample do-timestamp=true !
audioconvert ! audioresample ! mulawenc ! rtppcmupay ! multiudpsink clients=
192.168.2.120:5001

Consumer:


gst-launch-1.0 -q udpsrc port=5001 caps="application/x-rtp"
do-timestamp=true ! rtppcmudepay ! mulawdec
! audioconvert ! audioresample ! audio/x-raw,format=S16LE,layout=interleaved,
rate=44100, channels=2 ! fdsink fd=1 sync=true

What's happening? Is this the recommended way to do audio over udp?
I have gstreamer 1.4.4 on the producer and 1.9.90 on the consumer.

-- 
*Stack* is the new term for "I have no idea what I'm actually using".
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