[Asterisk-Users] Cisco 79XX Jitter Stats Question

[Andrew Herdman]

Actually I don't think it would matter if reinvite was on or not.  The
asterisk server in this case would simply be passing the RTP stream from and
to the LD provider to the phone.  Seeing the asterisk server is not
generating the stream, the jitter will be end to end, because asterisk is
simply acting as an RTP router.

End-to-End being defined as to where the RTP packets start, and end.

Andrew
 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
info at beprojects.com
Sent: Tuesday, July 12, 2005 9:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco 79XX Jitter Stats Question

It depends on whether you have re-invite enabled.  It is measuring the 
jitter of the rtp stream, so if re-invite is on, then it is measuring to 
the far end.  If re-invite is off, then it is measuring to *.

Matthew Boehm wrote:
> When on a call, you can press the middle round button and bring up some 
> RTP statistics.
> 
> Can anyone confirm my theory that the AvgJtr and MaxJtr are between this 
> phone and the far end?
> 
> Or is this jitter reading only between this phone and asterisk?
> 
> I'm guessing its the foremost, because when I make a local call to PRI, 
> the jitter is low/0 since the call would terminate at asterisk.
> 
> But when on LD call, which goes to carrier via sip, the jitter on the 
> phone bounces around.
> 
> Thanks,
> Matthew
> 
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