ASK THE EXPERTS - Configuring and Troubleshooting SIP Trunks

[ciscomoderator]

with David Whiteford

Welcome to the Cisco Networking Professionals Ask the Expert conversation. This is an opportunity to get an update on configuration and troubleshooting Session Initiation Protocol (SIP) trunks with Cisco expert David Whiteford. David has been a customer support engineer in the Cisco Systems Multiservice  Voice team since March 2009. He has over 20 years of experience in Voice. David has previously performed both software development and  interoperability testing with SIP trunking. He has a bachelor’s degree in computer engineering from Tulane University and currently holds a  CCNA Voice certification.

Remember to use the rating system to let David know if you have received an adequate response.

David  might not be able to  answer each question due to the volume expected  during this event.  Remember that you can continue the conversation on  the Collaboration, Voice and Video discussion forums shortly after the event. This event lasts through February 11, 2011. Visit this forum often to view responses to your questions and the questions of other community members.

[JustForVoice_2]

Hi David ,

Is there any recommendation from Cisco to Integrate CUCM solutions with SIP provider?

shall we use SIP trunk to SIP Provider?

Shall we use CUBE?

One more thing is there any references to understand SIP in General and how to write programs using SIP protocol and what is the preferred programming language?

Regards,

[dwhitefo]

It is recommended to place a CUBE between CUCM and a SIP provider.  The advantage to this is the CUBE has the ability to modify SIP message to meet the requirements of different providers and it also provides address hiding so the customer presents a single interface to the provider and all of the customer network details are hidden.

There are plenty of good resources on the web to find out more about SIP, but start by looking at some of the IETF RFC's like RFC 3271 which is baseline RFC for SIP.  There is a good book by Darryl Sladden called "SIP Trunking" which can be found at http://www.ciscopress.com/title/1587059444

For programming languages, I have not seen a preferred language, but there are SIP protocol stacks available in C, C++ and other languages which would give a head start on developing a SIP application.

David Whiteford

[Packetonwire]

Hello,

In response to your first question,

Is there any recommendation from Cisco to Integrate CUCM solutions with SIP provider?

Just wanted to share this, its been a good doc as a guide line

[juan.salaiza]

Hi david, i have a question, can you tell me what it the best practice to integrate a CUCM, a SIP PBX and a Gatekeeper.

Tthe main idea is. CUCM is already integrate with a GK, but we need to add a 2 SIP PBX´s to a GK, is it possible?. i´ll aprecciate your help.

[dwhitefo]

You have a few choices for interworking between the SIP PBX and CUCM.  You could use a CUBE to connect to the SIP PBX's and then connect to the CUCM from the CUBE using a SIP trunk or H.323.  You could also use a SIP trunk directly from the CUCM to the PBX, but the CUBE would provide much more flexibility and could provide the ability to route between the PBX's without having to go through CUCM.

[mcecilia]

Hello David,

We are deploying an scenario where PBXs from different vendors are connected to CUCM using SIP trunks. The problem comes, because we have needed MTPs for all.

In CUCM 6.1 it is not allowed to use G729 with MTPs but it is with 7.1. Is there any way to use G729 for trunks with 6.1 when MTP is needed?

Also, how does CUCM Session Manager Edition, Cisco Unified Sip Proxy and CUBE play in the SIP trunking solution?

What does any one differs from the others?

Thanks

[dwhitefo]

As far as I know, you should be able to use G.729 with MTPs in CUCM 6.1.  However, the MTP's have to be on an IOS device.  It is not clear from your discription whether you have SIP trunks directly from CUCM or whether you are using a CUBE between CUCM and the SIP PBX's.  If you use a CUBE, then the CUBE will perform some of the functions provided by an MTP.  CUBE is a session border element, so it actually terminates one leg of VoIP calls and then makes a new VoIP call on the other side.  Therefore, CUBE can provide interworking between H.323 and SIP if required and can perform any needed protocol transformation using profiles on the CUBE.  The Cisco Unified SIP Proxy (CUSP) is true SIP proxy server.  It simply routes SIP messages to the correct destination and can be used as a way to do load sharing across multiple SIP devices.

[mapleasant]

First time using this and simply asking for confirmation.

[dwhitefo]

Confirmed

[m.trautes]

Hello David,

do you know any konfiguration between a UC540 and a MS Lync 2010 - because a new company - who we sell - has a Lync and we use a UC540.

So i´m looking for a sample configuration for a SIP Trunk between this Systems.

Thanks for Help

Michael

[dwhitefo]

Lync is a renamed version of OCS.  You can find a number of interability guides for OCS on Cisco.com.  Here is a link to the one which I believe is most relevent: http://www.cisco.com/en/US/solutions/collateral/ns340/ns414/ns728/ns829/799011.pdf

[Nathan Compton]

Hello David,

I'm coming up on my first CCIE Voice Lab attempt in March.  On a practice test, I was given a requirement to integrate a SIP trunk with CUCM, but CUBE was not possible due to no H323 gateways in the scenario.

Routers running CUBE have the possibility to debug sip messaging to see what is failing during a SIP call.  Is there any equivalent feature in the CUCM that would allow you to see the SIP message exchange between the CUCM and the other SIP endpoint?

[dwhitefo]

The only way I know of to see the SIP messages on CUCM is to get a SDI trace for a call.

Here is a link for gathering the traces: http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080094e89.shtml

[joe-vieira]

We're considering adding a SIP trunk to our hosted call center provider. Currently the signaling/data portion uses the Internet link and the voice traffic flows in the PRIs. Questions;

- Do we need just a router on our end to support this?

- Is it recommended to have separate links for the Voice and Data traffic

- Does it make a difference if it's a Lan Extension or MPLS link?

- Can you provide links to documents that detail what's involved with this installation and configuration?

Thanks