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New Member

Requirements for SIP Gateway

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
Hall of Fame Super Gold

Re: Requirements for SIP Gateway

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 ?

New Member

Re: Requirements for SIP Gateway

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.

Re: Requirements for SIP Gateway

Hall of Fame Super Gold

Re: Requirements for SIP Gateway

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.

New Member

Re: Requirements for SIP Gateway

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.

Hall of Fame Super Gold

Re: Requirements for SIP Gateway

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 ?

New Member

Re: Requirements for SIP Gateway

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.

Hall of Fame Super Gold

Re: Requirements for SIP Gateway

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.

New Member

Re: Requirements for SIP Gateway

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

Hall of Fame Super Gold

Re: Requirements for SIP Gateway

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!

New Member

Re: Requirements for SIP Gateway

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.

Hall of Fame Super Gold

Re: Requirements for SIP Gateway

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?

New Member

Re: Requirements for SIP Gateway

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.

Hall of Fame Super Gold

Re: Requirements for SIP Gateway

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.

New Member

Re: Requirements for SIP Gateway

Hello,

I am with the same team/organization of the original poster and may be able to shed some additional light on our current setup and what we are trying to do.

Old Setup:

Hub and spoke setup with frame relay links to each site. Head end site has a 3845 each remote site has a 2811 connected via Frame-Relay and is configured for voice over frame with specific QoS rules. Our head end site additionally has a softswitch phone system which has polycom sip phones registered to it. Each remote site has an access code assigned to it so when you dial the access code only the end result is the ringing of one or more of the FXS ports on the remote router. The 3845 matches the access code sends to the remote site DLCI, remote router matches the same access code with POTS dial-peers and rings the line. The call is then delivered by each remote sites regular plain jane phone system (not supported by us). Any site can dial another sites access code and ring another remote site most often by getting access to the remote analog trunk on the router via dialing 9 or hitting a special key on their keysystem and then dialling another site's access code. The head end site with the softswitch used to participate in this also however it gained access to the 3845's dialplan by first traversing the AUDIO CODES SIP to Analog gateway. They would dial 88XX with the X's representing the remote sites access code. The softswitch would key on the 88 and send the call to the AUDIO CODES / which would then pick up a trunk on the T1 directly connected from the AUDIO CODES to the router

Present Time:

The AUDIO CODES device has failed at the head end site, and we would like to not replace it and have the softswitch be able to send and receive SIP calls with the 3845 and then "route" the calls based on our old dial plan via Voice Over Frame if possible. We have tested this by trying to dial site codes defined in the 3845's dial plan directly from a SIP phoned configured to use the 3845 as a proxy however 3845 is returning destination unknown. I can provide the SIP traces from these tests. Do we need to some how "accept" these SIP calls with an inbound dial-peer first? Can I have the 3845 take an SIP Invite and then deliver the call via already in place Voice over Frame Dial-Peers?

Hall of Fame Super Gold

Re: Requirements for SIP Gateway

Hello,

As I indicated in my previous posts, you cannot call from a VoIP device to a VoFR directly - this is not supported by Cisco IOS.

The workaround is convert to TDM/POTS in between VoIP / VoFR. Your colleague was about to do that based on my previous configuration indications, and I understand he had already connected the required crossover cable between the two T1 ports.

Once you complete that configuration, is not much, you will be able to call from SIP softphone or softswitch to the remote sites.

New Member

Re: Requirements for SIP Gateway

p.bevilacqua,

Thank you for all your help so far. I'm still looking into how to correclty configure the dial peers and a test phone to go through the PRI then get routed throught the IP network. Do you have any configuration examples that would help in this respect? Thank you.

Curtis

Hall of Fame Super Gold

Re: Requirements for SIP Gateway

Hi,

After the two PRI interfaces are up, you only need three more dial-peers, one voip and two pots. The vofr DPs should be allright already.

dial-peer voice 10 voip

incoming called-number

destination-pattern

session-protocol sipv2

session-target ipv4:

dial-peer voice 20 pots

incoming called-number

destination-pattern

direct-inward-dial

port 0/1/1:23

dial-peer voice 30 pots

incoming called-number

destination-pattern

direct-inward-dial

port 0/1/0:23

New Member

Re: Requirements for SIP Gateway

Ok, with your guidance a little bit of reading and some time on the phone with a tech, this is working. However, now our phone system lets us only make 1 concurent call at time over the FR interface. At this point I don't know if its a phone server problem or still a router configuration problem. What kind of show commands or debug commands can I do to view the current stat before, during and after a call is made. Thanks

Curtis

Hall of Fame Super Gold

Re: Requirements for SIP Gateway

Glad to know things are starting working, I will love to see a resolved checkmark on this thred.

Now, the phone system doesn't know that you are using voFR or whatever, it just uses SIP, so it's trange no more than one calls goes through. Unless for some reason the harpinned PRI has only one channel working, we could look at "term mon" and "debug isdn q931" for this.

Then, for the call setup you have to look at "term mon" and "debug ccsip message" and this will tell how the calls is made and whay kind of error is returned. Also, try "debug voice dialpeer", but this one is a bit verbose and harder to interpret. To see the calls that are in place already, there is a bunch of commands, one is "show call active voice compact".

New Member

Re: Requirements for SIP Gateway

Well, here is the debug output I'm getting from "debug isdn q931":

Apr 20 14:52:28.332: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:88210000@192.168.100.1:5060 SIP/2.0

To: <88210000>

From: "Turner_Associates" <>88@csystems.com>;tag=20112

Via: SIP/2.0/UDP 192.168.100.50:5061

Call-ID: cbf746c098686f92bec8a981d0d2ebd9@192.168.100.50

CSeq: 1 INVITE

Contact: <88>

User-Agent: ININ-cssphone-87256560

Max-Forwards: 70

Allow: INVITE, BYE, ACK, CANCEL, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO

Accept: application/sdp

Accept-Encoding: identity

Content-Type: application/sdp

Content-Length: 205

v=0

o=ININ 649460000 649460000 IN IP4 192.168.100.50

s=Interaction

c=IN IP4 192.168.100.111

t=0 0

m=audio 2254 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

Apr 20 14:52:28.332: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.100.50:5061

From: "Turner_Associates" <>88@csystems.com>;tag=20112

To: <88210000>;tag=28838A2C-261C

Date: Fri, 20 Apr 2007 14:52:28 GMT

Call-ID: cbf746c098686f92bec8a981d0d2ebd9@192.168.100.50

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 1 INVITE

Allow-Events: telephone-event

Content-Length: 0

Apr 20 14:52:28.332: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 192.168.100.50:5061

From: "Turner_Associates" <>88@csystems.com>;tag=20112

To: <88210000>;tag=28838A2C-261C

Date: Fri, 20 Apr 2007 14:52:28 GMT

Call-ID: cbf746c098686f92bec8a981d0d2ebd9@192.168.100.50

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 1 INVITE

Allow-Events: telephone-event

Content-Length: 0

Apr 20 14:52:28.336: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:88210000@192.168.100.1:5060 SIP/2.0

To: <88210000>;tag=28838A2C-261C

From: "Turner_Associates" <>88@csystems.com>;tag=20112

Via: SIP/2.0/UDP 192.168.100.50:5061

CSeq: 1 ACK

Call-ID: cbf746c098686f92bec8a981d0d2ebd9@192.168.100.50

User-Agent: ININ-cssphone-87256560

Max-Forwards: 70

Content-Length: 0

Basically I need to find out why I'm getting the "503 Service Unavailable" error and why it only happens sometimes. 4/5 calls I get this error. The rest of the time it goes through.

Curtis

Hall of Fame Super Gold

Re: Requirements for SIP Gateway

Hi, this is "debug ccsip message". Can you also actually enable also "debug isdn q931" and take the call again ? This way we will be able to correlate the call flow.

Also please send output of "show controller t1" , "show isdn status", and "show dial-peer voice summary".

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