[Asterisk-Users] Choppy Sound on PSTN End

[Tim Connolly]

    I have the exact setup you describe, SJPhone -> * -> Zap/PRI. I think
you need to twiddle some settings. You might turn on qualify just to see if
the * is seeing network flaws. Keep in mind, if your using windows, anytime
the user starts clicking around, you can expect less than ideal audio. Also,
why disable GSM ?

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tim Chandler
Sent: Monday, May 02, 2005 11:23 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Choppy Sound on PSTN End



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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