[Asterisk-Users] ChanSpy and Sipura 2100 jitter.

Domjan Attila adomjan at tvnet.hu
Tue Sep 27 14:47:14 MST 2005


I have the same problem with monitor application to. The sjphone does
not send periodically the rtp packets when the soundcard buffer is 32ms.
I set the buffer 4ms and num of buffers 5, after everything is ok. I
think the problem is similar with the sipura.
Another workaround, try the sip jitterbuffer, however it's not stable
yet. The jitterbuffer had eliminated this problem, but I removed after
the first crash...
A


On Tue, 2005-09-27 at 16:32 -0500, Trey Blancher wrote:
> I get this with Sipura 2000/2002s, but only half of the conversation
> sounds distorted.  The actual SIP user  on our end sounds clear, it's
> the other end that's distorted, sometimes garbled unintelligibly.
> Some of our calls bridge an outside (Zap) user to another outside user
> (usually IAX2, will be Zap in the near future), and in that case both
> sides are distorted.  I'm pretty sure our LAN is 100Mbps (looking at
> the spec sheets for the Sipura boxes I can't tell if their network
> interfaces are 10Mbps or 100Mbps).  I know our network switches are
> 10/100Mbps.
> 
> On 8/12/05, Script Head <scripthead at gmail.com> wrote:
>         I have an analog phone connected to a Sipura 2100 which in
>         turn
>         connecteds to * over a 100mbps LAN. When I do ChanSpy on a
>         bridged 
>         call, it causes massive jitter. When I attempt ChanSpy with a
>         Grandstream GXP-2000 the monitored call is clear. Has anyone
>         had this
>         happen? Any suggestions?
>         
>         ScriptHead
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> 
> -- 
> Trey Blancher
> Systems Administrator, USA Debt Management LLC
> (800)621-7806 ext 8601 
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Domjan Attila <adomjan at tvnet.hu>




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