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 - [EMAIL PROTECTED]
AIM - b2Cory
----- Original Message -----
From: "Julio Arruda" <[EMAIL PROTECTED]>
To: "Commercial and Business-Oriented Asterisk Discussion"
<[email protected]>
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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