the following VOIP problem is giving me headaches..:
3 machines connected via VPN A---Z---B
Z is a openvpn server and also runs asterisk.
The ping times from A to B are in the range of about 30 msec.
All calls are made via the asterisk server using ILBC.
I want to track down a wierd problem: when talking
from A to B (client software does not seem to matter, I
tried ekiga under linux, X-lite under windows XP)
then the sound starts to get distorted after about one minute
and totally unintelligible after about 2 minutes.
A redial gives again clear sound for about 1 minute (perhaps even less)
Then I made tcpdump traces on machine Z capturing all RTP traffic and
it seems that the timestamps of machine B are getting offset over the
time - the samplerate seems to be 8000.25 Hz or so - while machine
A keeps perfect time. B is a Laptop, a new Quanta machine,
exact name escapes me for the moment, with Intel ICH6
chipset and Realtek ALC880 sound system. Before I updated the sound
driver, the times were differing by 5 seconds after a 30 seconds call
and I could hardly understand the other party during a few seconds
at the beginning.
Now I want to listen to the captured RTP traffic to see if it is
already distorted before entering the asterisk server.
(preferably under Linux)
Hints about how to do this would be very much appreciated.
NB. today I bought a USB headset, perhaps it will cure this problem
but I am still curious.
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