[Asterisk-Users] Re: SIP and DSL Bandwidth queries.

Ratnakar rkolli at cisco.com
Thu Oct 2 16:55:04 MST 2003


yes i work for cisco. But playing around with asterisk is purely personal.
It is in no way related to my work at cisco.
I tried using another email id yesterday, but the post never showed up. Even 
though I got a mail from the news server that it was posted.

Thanks,
=ratnakar


Jon Pounder wrote:
> 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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>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>>
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