SIP trunking without ITSP

Answered Question
Jun 9th, 2010

Hi all.

I am just confused as to when SIP trunking is required. Lets say i have an organization with 100 nodes and we want to deploy centralized call manager at hub. Now what i am confused about is, do i ever need a sip trunk between my own gateways even when i am not communicating with any ITSP ? Like in my case i have 100 routers in all 100 nodes. Call mades on voip will be forwarded by these routers, so if we focus on the calls that are made within an organization, do i still need sip trunks anywhere ?

Pls clear it for me

Correct Answer by Jonathan Schulenberg about 6 years 8 months ago

That's a true statement.

You have vastly underestimated the role of the call agent in VoIP and I strongly suggest reading the SRND document I included previously. You're missing several large concepts.

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Overall Rating: 5 (1 ratings)
Jonathan Schulenberg Wed, 06/09/2010 - 04:07

The gateways need to run a protocol to communicate with UCM. This can be H.323, MGCP, or SIP.  A SIP Trunk is a loose term and most people use it whenever speaking of a SIP connection between two devices that are not an end-user device (e.g. gateway and UCM cluster). So, if you configure the dial-peers on the gateway to use SIP, you are effectively building a SIP trunk to UCM from the gateway.

Jonn cos Wed, 06/09/2010 - 08:13

Dear Jonathan,

Suppose i have 3 sites with 20 users each. I have 2801 on each site running CME. Now, does in this case, i need sip between these 3 routers ? or i just need to define dial peers ?

Jonathan Schulenberg Wed, 06/09/2010 - 11:12

You need to define dial-peers as they are your logical connection between call agents. You may choose to specify that the dial-peers utilize SIP as the signaling protocol when communicating. By default they will use H.323.

Jonn cos Thu, 06/10/2010 - 02:53

Dear Jonathan,

I though dial peers are just like static routes. You define them manually and they just match the extension and forward the call. Can you pls refer me some doc or book that describes this behaviour ?

Jonn cos Thu, 06/10/2010 - 04:59

Dear Jonathan just 1 more thing and it will clear my confusion

Whenever i am using dial-peer, does it always involve h323/sip signalling ? is it possible for dial-peers to work without H323/SIP signaling ?

Jonathan Schulenberg Thu, 06/10/2010 - 05:10

I'm not sure how else to explain it:  Dial peers define the logical next hop to route the call to; the protocol (h.323 or SIP) is how the call agents communicate and process the call leg.

If I were to use an analogy: If I hand you an object and talk to you in a language that you do not speak you won't know anything about or what I want you to do with that object.

Jonn cos Thu, 06/10/2010 - 05:32

Dear Jonathan,

Actually what i assumed was, when someone picks up the phone, dial numbers etc all of this would be converted into IP packets that are simply forwarded using dial-peers !! but you are saying that routers(gateways) need to establish the call !.

So if i ask you to endorse the following statement, can you do it pls ? it will give comfort to my heart i guess

"Everytime we configure dial-peers, they use either H323 or SIP, and they can never work without any signaling protocol"

Kindly confirm it pls

Correct Answer
Jonathan Schulenberg Thu, 06/10/2010 - 05:35

That's a true statement.

You have vastly underestimated the role of the call agent in VoIP and I strongly suggest reading the SRND document I included previously. You're missing several large concepts.

Jonn cos Thu, 06/10/2010 - 09:24

Dear Jonathan

I am really depressed :-(. I was reading CCNA voice but this is not even mentioned there. I know this may not be a just question but i really want to get going in Voip but i dont know how to start. Even if i start with CCNA voice, this concept is no where to be seen !! Can you guide me as a senior as to how should i approach my voip studies ? do you think SRND are good enough for a starter like me. By now you are surely familiar with my level, so you still prefer SRND ? do you prefer any other books or docs also ?

Pls take out some time to answer this also.

rzanett Tue, 07/06/2010 - 09:01


The underlying structure for VoIP is IP.  The first step would be to gain a solid understanding of route/switch and then move into courses such as voice over.  By having a route/switch understanding solidifies how data is routed and why.   Voice is then just another application with a given set of requirements for dely/jitter/etc that must be met. 

It would be very difficult to just jump in to voice and try to understand how to terminate a SIP trunk or how redundancy works or CAC, etc.




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