[asterisk-users] SIP hardware phones

Bryant Zimmerman BryantZ at zktech.com
Mon Feb 13 08:11:48 CST 2012


Jason

A standard SIP VOIP phone will use less than 100k per voice call.  For 
example I have several bussiness customers that have a dedicated DSL line 
and they do up to 6 lines very well on that 1.5x384 (we do g729 which is 
37k per call). If your networks drops can test solid at 10mb you should be 
in good shape if they do not run solid at 100mb you should force the switch 
port to negoitate to 10mb not 100mb. Make sure the POE switches you are 
looking at allow you to force the port speed this may save you in the long 
run. Also make sure that the POE switch can handle the load and run lengths 
you are looking to put on it. 

Bryant

----------------------------------------
 BrFrom: "Jason W. Parks" <jason.w.parks at gmail.com>
Sent: Monday, February 13, 2012 8:32 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] SIP hardware phones

Thanks for the info. As we move forward, we'll be testing and making a 
phone selections. No doubt we'll run into this. Are you saying if the 
phone is stated to be a 10/100 phone, it still may not work at 10?

On 2/13/2012 1:32 AM, Benny Amorsen wrote:
> "Jason W. Parks"<jason.w.parks at gmail.com>  writes:
>
>> I can move my voice infrastructure to an IP-based one running 10Mbps,
>> utilize existing wiring infrastructure, with the only cost outlay
>> being low cost PoE managed switches (48 ports for about a grand), and
>> it ends up a lot cheaper than upgrading the data network to support
>> the phones. ...and I can still stay within standard.
> You can, but not all phones will link up at 10Mbps.
>
>
> /Benny
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                 http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>     http://lists.digium.com/mailman/listinfo/asterisk-users
>

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120213/e269c667/attachment.htm>


More information about the asterisk-users mailing list