[Asterisk-Users] SIP and DSL Bandwidth queries.

[Jon Pounder]

At 08:54 AM 10/3/2003 +1000, you wrote:
>Hi guys,
>
>Don't want to ruffle feathers, but did I see Ratnakar's email address as
>being @cisco.com.
>
>Is Cisco thinking of using Asterisk? Just a thought.

Well if I was a large hardware manufacturer I would certainly be testing 
compatibility of my hardware with other popular stuff, since only a fool 
would think people are going to buy open standards based equipment all from 
one manufacturer.

If cisco is doing some testing, great !, but I doubt they are actually 
planning to deploy asterisk corporate wide.




>Welcome Ratnakar
>
>Peter
>
>From: rkolli at cisco.com
>At 14:50 2/10/2003 -0700, you wrote:
> >Here is my setup
> >
> >7960(A)--Firewall/PAT--dsl---------Internet--------dsl--Firewall/NAT---7960
>(B)
> >                                     |     |
> >                                     |     |
> >7960(C)--NAT--cable-----------------      -----dsl -- Asterisk
> >
> >(A) can communicate with (C) only when C is configured with
>canreinvite=no. The
> >call gets dropped in few seconds if canreinvite is set to yes for C.
> >(A) and (B) can communicate fine when both sides have canreinvite=yes.
> >
> >Since (C) is not working with canreinvite, traffic goes thru Asterisk
>server.
> >This brings the Dsl connection to asterisk to a crawl. It is so bad that
>even a
> >idle ssh connection gets disconnected.
> >
> >Is it possible to configure C so that reinvite works. If not what kind of a
> >bandwidth should I have for Asterisk server. Currently it has a upload of
>128K.
> >
> >The codec currently getting used is ULAW. Even if I configure 7960's to use
> >g729, show sip channel reports as using ULAW.
> >
> >Thanks,
> >==ratnakar
> >
> >
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> >
>
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