[asterisk-users] sip <> zap calls choppy, where to setup the jbuffer?

[Jay Wilton]

Hello all,

cell <-T1-> zap <-internet-very remote-> sip (ip430)

The audio is choppy ONLY to cell USER.  The polycom user
says the audio is fine.  SIP-SIP calls sound good for both
parties.

Where should I setup the jitterbuffer?  The zapata.conf
(recent * 1.2) and/or the polycom configs (fw 2.0.3)?  Any
tips with the zap or polycom settings below would rock. 

Packet loss - average 7%  --> ping test
Latency - average 300ms -> sip show peers
- latency ranges from 200-330 but stays within 10ms of
initial value on ping test

I tried to implement the jitterbuffer in zap.  
/etc/asterisk/zapata.conf 
jitterbuffers=16   
; covers the 300ms latency?  20ms each x16 = 320ms

On the Polycom IP430's, I setup the jitterbuffer.  The
audio was still poor to the cell phone user.
Jitter Buffer Minimum - 80
Jitter Buffer Shrink - 1000
Jitter Buffer Maximum - 220

I tried these values, but the phone stopped passing ALL
audio.
Jitter Buffer Minimum - 80
Jitter Buffer Shrink - 3000
Jitter Buffer Maximum - 340

Thanks, JJ


 
____________________________________________________________________________________
Food fight? Enjoy some healthy debate 
in the Yahoo! Answers Food & Drink Q&A.
http://answers.yahoo.com/dir/?link=list&sid=396545367