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Requirements for SIP Gateway

clamasters
Level 1
Level 1

Hello all,

I'm pretty new to VoIP and have been given a little research to do. What do I need to configure on our router (Cisco 3845) to do the job that our AudioCodes device is doing (i.e. what do I need to configure to make the router a SIP/Voice gateway for Voice over Frame Relay connections)? We are using Interactive Intellegence software and Polycom phones for our phone system if that helps.

22 Replies 22

paolo bevilacqua
Hall of Fame
Hall of Fame

A little network diagram pretty please ? Where are you doing voice over Frame Relay? What is the other SIP party? Is there a PBX for the phones ?

I have posted a very simplified network diagram at http://www.curtis-lamasters.com/networking/vofr.jpg

I'm afraid I do not know what you mean by SIP Party? The other phone system is a ShoreTel if that helps. The SIP traffic from the corporate segment goes to an AudioCodes proxy/gateway which plugs into a TI going to our ISP. We had a second one that was working at the VoFR proxy/gateway that failed and instead of replacing it we decided to use the router for this service. Now I just need to konw what to setup on the router itself to do such a task.

Hello,

I do not see the failed device in the diagram. Into what it was plugged ? When doing VoFR, one side was the failed device, and the other side (branch) was what ? Do you use Frame Relay for data also, and if so do you have separate PVC ?

The thing is that the router can certainly do VoFR and a variety of other things, but unless one see the whole picture, is hard to tell what is the next best move.

Well, I believe at this point I do not fully understand how to answer your questions. The AudioCodes device was plugged into the switch and the phone system was setup to use it when a specific dial pattern was used, then route the traffic over a frame relay connection. On the same AudioCodes device, another port plugged into a TI csu/dsu on the router and used it as it's calling mechanism. I have updated the diagram to show this device.

Ok, so apparently the device was taking voip/SIP from softswitch via ethernet and converting to frame relay via a serial interface to the router.

Because this device has failed, you have lost "phone server" connectivity to sites a and b downstream.

You haven't specified what are the routers at these locations and how the call is supposed to get to the local "phone server".

Anyway, the router configuration will tell you how the router would be treating this data on T1. Quite possibly the router can talk SIP to the "phone server", but then what todo with the call must be seen at the light of what I was asking before.

How is the router configured ?

The router config is now located at http://www.curtis-lamasters.com/networking/corporate.txt

We still have connectivity to the other sites through the PSTN however, we would like to bypass that cost by routing the traffic over the WAN. The routers at the other locations are Cisco 2811's.

I can't access the page. You can also use attachments here.

All the matter is about how the router was treating that T1.

Possibly the "phone servers" should be able to talk to each other via SIP, you may or not may need the router to participate.

The config permissions have been changed now. Sorry about that.

You have an interesting scenario. They designed around the now dead gateway to convert voip/sip to tdm with E&M signalling to the router

(as a consequence that makes that you shouldn't have seen calling number when calling intra-site).

The router has multilink frame relay circuits to the SP with single pvs to branches.

VoFR evidently has been used for bandwidth saving and all appears to have been configured .

Now in practice I think you have few choices:

1) check if the "phone servers" are able to call each other over IP/SIP *and* making so the call is g729 compressed. If they can, just configure them to do. At this point they will use yor IP cloud for all means, and the only improvment you can do on the router is to configure rtp header compression. You could remove or leave the VoFR configuration that wouldn't be used anymore.

2) configure the 3845 to do the exact job of the dead gateway. You can do this as you have two t1 voice ports. Configure main "phone server" to use SIP/VOIP to the router for calls to remote sites. Any codec acceptable.

On the router, connect back-to-back the two voice T1 with a crossed T1 cable, config one interface with internal clock other line. Configure these as ISDN PRI (one end has to be network side).

On port communicates with SIP/VoIP, the other will do match to VoFR like before. The DP has to be set correctly but most are already in place.

As a varion to the above you cold place an additional smaller router to do the voip / pots conversion.

3) the most drastic alternative would be of redesign everything around pure VoIP, what telephones are you using? If SIP they can be controlled by a cisco "phone server" that runs with IOS in the router. That is called CCME as is very practical to use in all type of office but very large and sophisticated ones.

Hope this helps, if so please rate post!

Thank you for the info. I'm having another associate look into doing that with the phone system itself. However, how would one go about configureing option 2 on the router. I would also like to note that we do have a second AudioCodes device that routes through the PSTN if that matters.

For option two I gave you the configuration outline above, first you make the cable and T1 controllers go up, then deconfigure E&M, configure pri on both and the sip dialpeer, if is not the one already there.

You can ask for details again once you are going since you never did it. You can look on CCO for things like "configuring PRI network side", or "understanding dial-peer", etc.

Doing this should not change the functionality of the current phone system and do not interfere with the other device.

If this helps, would you consider rating using the box on the right side?

Ok, sorry for the seemingly dumb questions. The two ports are connected via T1 crossover and when doing "show controller t1" they both show as up. Now, the configuration is a complete mystery to me. What do you mean by configuring pri on the controllers? The command is really all I'm looking for. I read a few documents showing how to configure network side PRI but those commands don't translate to the controller interfaces.

Look for "pri-group ..." that goes in place of "ds0-group ..." under controller t1.

(actually you could even put the controller in E1 mode to have 30 channels instead of 23, it that matters to you)

After you do that, interface serial x/y/z:23 and voice-port will appear. Put the isdn configuration commands in there.

Doesn't matter which side is network and which is not. Pick any siwtch-type like, it does not make a difference. You will see that ISDN is OK with "show isdn status".

Then you configure a dial-peer pots to route IP/SIP calls from the "phone server" out to the chosen T1 port.

dial-peer 1 is already the pots dial peer for FR and must indicate the other port.

I know it sounds confusing but to go from VoFR to VoIP and viceversa you must cross a pots interface so try to think you are actually configuring two separate routers one for VoIP and one for VoFR and they are linked by the back-to-back cable.

When you place a call you will see it with "debug ccsip message", "debug isdn q931" and "term mon". To see where the call to a number would be sent, try "show dialplan number ".

Good luck.

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