[Asterisk-Users] Choppy Sound on PSTN End

[Tim Chandler]

Hi all,

I recently set up Asterisk on a Dell PowerEdge 1850 Dual Xeon 2.8 Ghz
processor.  I am running the latest build of White Box Enterprise Linux.
Our call routing is like this:

SJPHONE on Windows -> QoS-enabled Switch -> Asterisk -> T1 Line ->
Broadvoice SIP account -> PSTN

Calls seem to work great from user to user.  However, calls from a SJPhone
user to the PSTN are not so great.  The SJPhone user hears the person on the
PSTN perfectly - I mean, completely flawless.  However, the user on the PSTN
end hears choppy / jittery, extraneous clicks, etc.

Here is the SJPhone config:

Audio Compression: G.711
Driver buffer size: 20 msec
Driver input queue length: 6
Driver output queue length: 4
RTP jitter queue length: 6
Silence Suppression: No
DTMF Sending: RFC 2833
Signal Duration (ms): 270
RTP Payload type: 101
Signal volume: 10
Pause duration (ms): 100


And the sip extension config (in Asterisk Management Portal):
Allow: blank
Canreinvite: no
Disallow: gsm
Dtmfmode: rfc2833
Host: dynamic
Nat: yes (some users are behind NAT)
Qualify: no


Any ideas on what to do to get rid of the choppiness?

Thanks!
Tim

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