[cisco-voip] Video Conferencing ? CM 4.1(3)

[Lelio Fulgenzi]

802.3af is the inline power standard...that might be one way, but not sure 
how.

802.1x is more likely the solution, which authenticates a user and allows 
them on a particular VLAN.

That would be a novel way around CDP.


--------------------------------------------------------------------------------
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"Life expectancy would grow by leaps and bounds if green vegetables smelled 
as good as bacon."
Doug Larson


----- Original Message ----- 
From: "Philip Walenta" <pwalenta at wi.rr.com>
To: "'Lelio Fulgenzi'" <lelio at uoguelph.ca>; "'Robert Kulagowski'" 
<rkulagow at gmail.com>; <cisco-voip at puck.nether.net>
Sent: Wednesday, January 30, 2008 7:18 PM
Subject: RE: [cisco-voip] Video Conferencing ? CM 4.1(3)


> I'm not sure if CDP is going to be licensed, but I know we're working on a
> way to do it via 802.x standards (the appropriate standard escapes me at 
> the
> moment - it might be 802.3af).
>
> -----Original Message-----
> From: Lelio Fulgenzi [mailto:lelio at uoguelph.ca]
> Sent: Wednesday, January 30, 2008 5:50 PM
> To: Philip Walenta; 'Robert Kulagowski'; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] Video Conferencing ? CM 4.1(3)
>
> Phil...any chance on Polycom licencing CDP from Cisco so I can slap the
> phone onto a port and have it configure voice VLANs appropriately?
>
>
> ----------------------------------------------------------------------------
> ----
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "Life expectancy would grow by leaps and bounds if green vegetables 
> smelled
> as good as bacon."
> Doug Larson
>
>
> ----- Original Message ----- 
> From: "Philip Walenta" <pwalenta at wi.rr.com>
> To: "'Robert Kulagowski'" <rkulagow at gmail.com>; 
> <cisco-voip at puck.nether.net>
> Sent: Wednesday, January 30, 2008 5:32 PM
> Subject: Re: [cisco-voip] Video Conferencing ? CM 4.1(3)
>
>
>> I'll start by saying that I work for Polycom...that being said..
>>
>> We do have a product that does allow control of the video unit from a
>> Cisco
>> IP phone (it can control our SCCP based devices or H.323/SIP devices).
>>
>> Link here (sorry for the product advertisment):
>>
> http://www.polycom.com/usa/en/products/video/video_conferencing_systems/desk
>> top/video_control_app.html
>>
>>
>> Now, as for what folks usually do for overflow:
>>
>> I've built several scenarios for customers:
>>
>> 1.  MCU, IP and ISDN attached, with GK control.  The GK knows that the 
>> MCU
>> can "dial out" as a backup path.
>> 2.  H.323/GK control, with Cisco IOS gateways (28XX or 38XX) and attached
>> PRI's using the new IOS ability to do H.323 to H.320 conversion.
>>
>> Both work pretty well, the MCU variant allows content, the Cisco one does
>> not (support is coming from what I understand).
>>
>> -----Original Message-----
>> From: cisco-voip-bounces at puck.nether.net
>> [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Robert 
>> Kulagowski
>> Sent: Wednesday, January 30, 2008 2:38 PM
>> To: cisco-voip at puck.nether.net
>> Subject: Re: [cisco-voip] Video Conferencing ? CM 4.1(3)
>>
>> Matthew Saskin wrote:
>>> VT Advantage is basically your only option if you want to provide
>>> video from the desktop IP phone.  If you want room-based solutions
>>> they can integrate with CallManager in multiple ways (as mentioned in
>>> previous
>>> emails) but they are going to be dedicated units in a room (or on a
>>> crash cart, etc.)
>>>
>>> As far as providing overflow from IP to ISDN, I can't even think how
>>> to do that.  The biggest thing standing in the way is that you would
>>> need to change the destination being dialed from an IP address to a
>>> phone number :-\
>>
>> Right; as I'm thinking about it now, the gatekeeper knows that there's 
>> "x"
>> amount of bandwidth between the two locations; if the bandwidth request 
>> is
>> rejected then there should be _some_ way to do this though, isn't there?
>> Or
>> do people just not place the call at that point?
>> That's how AAR in the voice world works...
>>
>> As far as the one-touch capability, I think I should be able to create a
>> service URL on the phone that does an AXL call to determine who I'm
>> currently talking to.  I can then have a widget that uses the video
>> conferencing equipment API to dial the call.  I've already gotten it to
>> work
>> manually using the Polycom telnet interface, so at this point it's just a
>> Small Matter of Programming. :)
>>
>>
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>
>