cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
3552
Views
40
Helpful
10
Replies

Cisco UCM 8.0 and new Tandberg VCS/ MCU

gmckay948vmb
Level 1
Level 1

I can find a lot of quite old interoperability documents on Tandberg and Cisco.

But if a customer already had Tandberg H.323 endpoints and wanted to add a Tandberg VCS and MCU for VC bridging, what is the best way to get the Cisco phones to be able to join a multi-party video call on the MCU?

I'm thinking the MCU could run as an end point to Call Manager (phones running SCCP)?

Or, can you have SIP trunk(s)from CUCM 8.0 to the VCS and use traversal calls?

Or even if the phones were running SIP, can they talk to the VCS via the CUCM. Do they even need to - I can't imagine they can act as SIP endpoints registered to the VCS while their "day job" is being registered to CUCM.

Any help appreciated....Even if it means telling my questions don't make sense - as long as you tell me why!

Regards

10 Replies 10

William Bell
VIP Alumni
VIP Alumni

Gordon,

The latest deployment guide for Cisco VCS and Unified Communications Manager (UCM) is available here:

http://www.cisco.com/en/US/docs/telepresence/infrastructure/vcs/config_guide/Cisco_VCS_Cisco_Unified_Communications_Manager_Deployment_Guide_CUCM_6-1_7_8_and_X6.pdf

Based on your post, I'd think you want to go this route:

1. Build a SIP trunk between VCS and CUCM.  You have to make some decisions as to how you are going to do this:

- Security: Are you going to use TLS for setup? SRTP for media?  If yes, then you will need to look at getting certificates aligned on both systems. The best approach is to use an internal CA to generate server certs for both clusters and then configured CUCM and VCS with the appropriate CA trust list.  There are several docs for both products that cover this.

- Call failover and/or load balancing:  If load balancing is required, you will need to leverage DNS SRV records that point to your CUCM cluster and a separate set of DNS SRV records for your VCS cluster.  The pros: you can get away with one SIP trunk on CUCM and a single DNS neighbor zone on VCS. The cons: Some folks get squeamish with DNS SRV.  An alternate approach is detailed in the deployment guide reference above.  You would createa "neighbor zone" on the VCS. One per CUCM call processing node. On the CUCM you would create a separate SIP trunk for each VCS (or, if using CUCM 8.6 you can get away with one trunk, multiple peers).

- SIP profiles:  This is easy. The deployment guide lays it out for you and the VCS includes a pre-canned profile so it knows how to talk to CUCM.

2. Dial plan. You will want to configure your dial plan so that you avoid overlaps/conflicts/call routing loops. As a simple example, assume you program all 7xxxx extensions to be video. Done. Tell CUCM to send all 7xxxx extensions to VCS. Tell VCS to send CUCM extensions to CUCM. Tricky parts come in when you want to enable features like SnR.  Both products have a facility for this. You have to learn both and pick one as an authority. There is more to it. But no need to get too fancy at this point.

3. SIP/H323 interworking.  The VCS Control can support up to 100 concurrent, interworked calls. So, the VCS can handle SIP to H323 interworking for you. It is good like that.

- MCU:  You can avoid interworking for your MCU if it supports SIP natively. If your MCU is one of the Codian boxes then it can do H323 and SIP simultaneously. A discussion on this can get quite involved. The basic idea is that you configure a SIP Neighbor Zone for the MCU. You configure membership rules on the VCS to put MCU SIP URI registrations into this neighbor zone. Finally, you configure search rules on the VCS so that H323 and SIP dial patterns for a MCU resource are given the same search priority. Say, you use 100 as your H323 prefix. Then you would have a "100\d*" H.323 rule and a "100\d*@mcu.uri.com" SIP rule. The first rule points to Local Zone and the 2nd points to the MCU SIP Neighbor Zone. They are given equal priority and are searched at the same time. VCS will prefer SIP if the original caller is SIP.  If this is too much info for you, just do the interworking until you get comfy and can optimize your design.

- VCS registered endpoints:  VCS registered endpoints that support SIP and H.323 should have both protocol stacks enabled.  This will minimize interworking when you call from VCS-to-CUCM. For CUCM-to-VCS you will probably still deal with interworking since your SIP URis probably are something more useful than an e164 number@yourdomain.

Phones on the CUCM can stay on the CUCM. Their calls will route via CUCM to VCS over the SIP trunk. You may need to dork around with CODECs, regions, and locations to ensure transcoding isn't needed. The bottom line is that if a phone is registered to CUCM, it doesn't need to have a separate registration for VCS:

phone-->CUCM-->VCS-->endpoint/mcu/etc.

Get things configured right, then you can think of this solution as a single system with multiple components rather than individual systems spliced together.

HTH.

Regards,

Bill

Please rate helpful posts.

HTH -Bill (b) http://ucguerrilla.com (t) @ucguerrilla

Please remember to rate helpful responses and identify

Nice write up!  I LOVE to dork around with Codecs.   

Interesting note on Bill's #2 above, dial plan.  I'm sure Bill knows this one already, but I didn't find this until I started digging.  I was surprised to learn that I can use URI routing from UCM via SNR to VCS.  In other words, if a PSTN call hits my phone at x1234, and I have a remote destination profile setup to hit a remote destination of billybob@mydomain.com, as well as a SIP route pattern, pointing mydomain.com to my VCS SIP trunk, then PSTN calls will hit my Movi and any other video billybob endpoints simultaneously.  On the VCS side, I am using FindMe, so incoming video calls will simultaneously ring my desk phone.  It is a pretty powerful solution, allowing us to live w/ E.164 on the UCM side and pure URI on the VCS side and have the best of both worlds in either direction.  I'm curious to see how FindMe and SNR will merge in the future, but what we have today is a good solution.

I was suprised that this wasn't covered in any of the VCS docs, but I did find a reference to UCM URI dialing here:

 

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fsmobmgr.html

Joshua

Joshua,

Nice add (+5). I was aware but it definitely is worth mentioning.

Regards,

Bill

HTH -Bill (b) http://ucguerrilla.com (t) @ucguerrilla

Please remember to rate helpful responses and identify

very nice posts 5+ Bill and Joshua

just to add one thing, which is regarding using H323 endpoints

it is better if you can change those end points to use SIP and have your signaling end to end SIP for 2 reasons

- less interworking/traversal licenses

- the media path will be directly between end points while if you gonna use the interworking of two signaling protocol types the media path will go via the VCS for every call

hope this help

marwanshawi,

I totally agree. When designing solutions for involving the VCS/Codian solution and CUCM, I provision the Codian for SIP+H.323 and then configure VCS to use equal-priority search rules so that I can leverage the VCS's preference for using the native signaling format of the call originator.

For endpoints, I always recommend customers enable both H.323 + SIP stacks for the same reason. I use pre-transforms to "normalize" (or globalize if you want to use the Cisco hot terms) first to keep it clean.

Regards,

Bill

HTH -Bill (b) http://ucguerrilla.com (t) @ucguerrilla

Please remember to rate helpful responses and identify

I have a new scenario.. (like always right?).

So lets say i have a VCS to CUCM 7.x

E20 Line1 is registered to VCS SIP

E20 Line2 is registered to CUCM SIP (third party)

VCS has a trunk to Polycom video world (h323)

E20 wants to make a video call to Polycom via h323.  VCS routes the call to Polycom, all good. (line1)

The end user wants to add a voice call to the existing call on Line1.  Can they use Line2 (connected to CUCM) and dial another phone on CUCM and hit "join" to add them to the call? 

First of all to make a multi point video call with the e20 you need multiway to be configured

Using a second line to join an active call on the first line I don't believe this can be done

But let's see if someone has another suggestion on this point

HTH

Line1 is Video Point to Point

Line 2 is audio only to join in….. IE MeetingPlace audio bridge. (or whatever conferencing bridge)

Supposedly, the Polycom vvx1500 can do this scenario.

That is pretty sweet, I never thought of single number reach and FindMe also...

Getting Started

Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: