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

[Philip Walenta]

I should add one more thing to this...4.1.3 does not support H.264, only
H.263 so you will not get a good set of features.  4.2 and higher support
H.264 perfectly. 

-----Original Message-----
From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Philip Walenta
Sent: Wednesday, January 30, 2008 4:33 PM
To: 'Robert Kulagowski'; cisco-voip at puck.nether.net
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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