Video and Telephony Integration: Enterprise UC with common Dial-plan

Question

Dear Expert,
I have asked the same question under Collaboration but thought to post it under this group too as this is mainly inclined towards Video Integration. I hope this is the right place to post this question.
The enclosed design shows the proposed UC setup which is based upon CUCM (VEr 8.5), ISR Routers(G2 3900/2900Series with PVDM3), Tandberg VCS and VCS expressway..
As always Thank you for all your input..
Regards
M
The expected solution out of this design should provide following
1.Video End point(h323) can call any IP Phone by dialling 5 digit extension and vice-versa.
2.Video end point can make outbound Local,National and International call by dialling the same way as it exist for IP Phone meaning for Local and Long Distance within US by dialling 9 then 10 digit number ,For International: 9011- Country Code- E164 number
3. Video end point will have the extension within 443xx range and can be reached by IP phone by simply dialling 5 digit extension. Route pattern with 443xx will be defined in the CUCM
4. Video End point can be reached by PSTN world by dialling full E164 number i.e 1712-564-43xx . In this call, CUCM will route the call to Video set-up.
5. Common Billing to be used for both Video and Phone setup towards charge-back.6. The most important aspect is Encryption which created all confusion and stopped my progress
Scenario:
Video Conference is going on and the participating users are :
- User on the Internet which have been joined to the conference by dialing their public IP and call established via Tandberg Expressway
-The 2nd user is an ISDN end point and call is routed through the CUCM to ISDN GW(R2)
- This is encrypted Video call
Encrypted Video Call is going on and all of a sudden there is a need to bring in an IP Phone Skinny end point(Audio only) and also a Mobile Phone user. It should be easy to bring these two user as part of the encrypted VC without dropping the ongoing VC.
I have gone through few Networkers Slide and also read that the 3900/2900 ISR supports media as well as signalling encryption and it is possible to achieve this set-up.
But because of involvement of integration with different product line (specific to Tandberg), needs clarification before starting with the lab set-up.
The interpretation of H323 RAS part when Tandberg VCS is acting as a Gatekeeper with respect to CUCM --is different than when Cisco Routers are being used as GK.
Tandberg Gatekeeper if integrated with CUCM comes into play for both Media and Signalling and always participate in the H323 call whereas if Cisco Router is acting as a Gatekeeper then once the RAS part is done Cisco GK goes out of the call.
So,
Please suggest whether anyone tried this setup? If yes then how to achieve this and what should be the link coming out of CUCM to Tandberg VCS-GK.( I think using SIP trunking will do this but not 100% sure about this). Feel free to suggest any design change or any hardware which is required...

Answer 1

Hi
As long as this is new instalation I recommend you to use SIP on all of the end points where possible and integrate with CUCM using sip trunk this will give you two main benefits
- the transformation of the called and calling number from and to CUCM will be easier
-if have end point using H323 and communicating with other end using sip the vcs will do internetworking to this call and you will need license for each internetworked call plus the media path will go through the vcs not direct between end points for internetworking
If you use sip make the end point name/sip usri as [email protected]  Calls from vcs to CUCM use search rules with trsformation so if end point dial 123456 only from vcs and the default call is sip vcs will send it to CUCM as 123456@sip domain.com you need to do transformation before sending it to CUCM and send it as 123456@cucmip. 
This is just in brief and also using the expersss way you can have your sip domain registered over the Internet and configure dns srv record point sip ton the vcs public ip and Internet calls can come to your end point sip name directly no need to publish ip to others to dial you
HTH
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