Design Question Sip Trunk

[Pieter de jong]

Hi,

I have a requirement to move from a h323 environment to a SIP environment. I am looking for best practises especially around security. I have 2 CUCM servers (8.5) located in separate cities for redundancy. I also have 2 voice gateways which at the moment are h323 to the PSTN, each located at different cities.

My  requirements are:

1. Creat a sip trunk to the provider instaed of using PRI.

2. If the Wan link fails on one gateway to provider, the alternate router in the other location should be able to receive the setup messages and if a user logs on via extension mobility, should be able to answer the call.

Are there any simplified design docos about for this? I am hesitant to create a SIP trunk straight to the provider for security, so thinking of terminating the call on the voice routers with CUBE. I'm pretty sure this is run of the mill and would appreciate some input.

Cheers!

Pieter

[Chris Deren · Accepted solution]

Simple answer ALWAYS use CUBE.  With IOS 15.1T and greater you have additional toll-fraud security that you can use to restrict what IP addresses can communicated with CUBE, this is all you should need.

HTH,

Chris

[Ayodeji Okanlawon · Accepted solution]

+5 to Chris..Always use CUBE...

Here are more ideas..

1. Create two sip trunks, first one to Cube 1, second to CUBE 2

    on your sip trunk set DTMF as no preference

2. Assign the trunks to CUCM group with your two servers in it

3. Configure route groups with Circular algorithm distribution (this way you have  load balancing on your two cubes)

4. Configure dial-peers on your CUBE gateway to point in preferential order to your cucm servers

5. Use dtmf-relay rtp-nte on your dial-peer (ensure you have sip as the protocol on your dial-peers)

6. configure your codec selection properly on your dial-peers

7. configure your region settings properly between your sip trunk and phones.

8. Evaluate if you need xcoders, provide one if you do and ensure you set your region correctly between your xcoder and CUBE

9. Is there voicemail involved? Ensure you set your region settings between cube and voicemail correctly, otherwise calls to voicemail may invoke xcoder (from experience)

10. Provision adequate bandwidth for your calls. Once you move to sip, you loose the luxury of e1 channels. You are solely relying on Bandwidth. Ensure you have adequate bandwidth for your concurrent calls

11. Provision QoS for your calls

12. Is there fax involved? You need to think carefully on this one. What fax method do you want to use. T.38/pass through. Does your provider support T.38?

13. Have a thorough test plan. Test call transfers, call on hold, call forward etc

Just a few pointers..Careful planning and implementation is required for a successful SIP implementation

[Pieter de jong]

Thanks for the input guys, much appreciated!