[asterisk-biz] UTStarCom F1000 and AP Roaming

[Cory Andrews]

The "endpoint" of SIP and RTP would not change mid-call.   If the phone 
stays on the same subnet, off the same access controller, and with the same 
IP address, the hand over is accomplished at Layer 2.  From the perspective 
of the SIP server, all RTP and SIP packets would continue to go to the same 
place - the access controller.  The phone just moves its physical connection 
from one AP to another.  This is why the solution works with APs all 
connected to one router and not with independent AP/Routers.  In the 
AP/Router case, the phone would need to get a new IP address and reregister, 
then re-invite the call - which would take far too long (like 20 to 30 
seconds) to be considered a "hand over".

Cory Andrews
Purchasing Manager
++++++++++++++++++
VOIPSupply.com
A Division of b2 Technologies
454 Sonwil Drive
Buffalo, NY 14225
++++++++++++++++++++
direct - 716.250.3402
mobile - 716.907.4054
email - Cory at VOIPSupply.com
AIM - b2Cory

----- Original Message ----- 
From: "Julio Arruda" <jarruda-asterisk at jarruda.com>
To: "Commercial and Business-Oriented Asterisk Discussion" 
<asterisk-biz at lists.digium.com>
Sent: Thursday, February 09, 2006 11:04 AM
Subject: Re: [asterisk-biz] UTStarCom F1000 and AP Roaming


> Cory Andrews wrote:
>> Have had a lot of people asking about "roaming"capabilities of current 
>> SIP Based WIFI phones.  Here's a clarification using the UTStarCom F1000 
>> as an example.
>>
>> Roaming is an ill defined term.  To tell  you what the phones do:
>>
>> We would define "ROAMING" in the context of multiple locations.  An F1000 
>> will "roam" from AP to AP with no problems.  If it goes out of range of 
>> an AP, it will scan for another one in its profiles. The other item often 
>> confused with roaming is hand over.  In that case, the phone will move 
>> from AP to AP even while on a call and will maintain the call.  The F1000 
>> will do that if all APs are served from the same access controller and 
>> are on the same subnet.  This will not work if an area is covered by 
>> multiple AP/Routers (like the Linksys WRT54G's) where each router has a 
>> different DHCP server to allocate addresses on its subnet.  The whole 
>> DHCP/Registration process takes far too long to manage a hand over.
>
>
> How would the call stay established if the end-point of the SIP and RTP 
> conversations changed midcall ? Would not require some 'inteligence' in 
> the UA and possible the SIP proxy ?
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