SIP UA

Unanswered Question
Jun 24th, 2007

Hello,

I'm wondering how I can configure a Cisco router in a lab to integrate into an existing voip infrastructure running SIP. It's a 2610 XM with fxs ports as well as running CCME sitting behind a NAT router into the corporate network.

The only details i've been provided are the SIP Domain, SIP proxy server as well as a username and password for authentication.

I've tried different combinations of sip server and registrar within SIP-UA config and i've tried binding the media and setup packets to the inside global address of the NAT but i'm not having any luck registering or initiating a call.

Any help would be greatly appreciated!

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Paolo Bevilacqua Sun, 06/24/2007 - 06:00

Yes, you should be abe to.

However, please ask if the NAT device do support SIP, as this may be the reason is not working.

What error messages are you receiving? "please check with "debug ccsip message"

richard-o Tue, 06/26/2007 - 14:59

I've tried debug ccsip messages already and i'm not receiving any error messages. I'm only seeing it constantly send Register and Invite messages (depending on if i'm trying to use a registrar server or establish a call)

I've several softphone products which can detect the global address themselves and register accordingly, is there anything I can do on the router to achieve the same? I created a loopback interface with the global address and bound sip messages to it but it made no discernible difference.

Unsure if the NAT supports SIP, i'm assuming no.

Paolo Bevilacqua Tue, 06/26/2007 - 15:18

Hello,

I'm afraid that if don't see replies back it means the SIP device is not correctly handling the SIP. Even if you set the address via bind, the NAT device will not have a translation set and not let the messages come back.

Unfortunately cisco does not support STUN and similar mechanisms so either the NAT device supports NAT, or you are out of luck.

richard-o Fri, 06/29/2007 - 18:49

Just to revive this thread

I've finally got the router to receive replies from the sip proxy. I'm sending out register messages every 10 seconds or so and receiving a 200 OK in reply.

I've got a dial-peer setup to catch local pstn numbers and forward them to the outgoing proxy but i get a 604 Does not Exist message.

I'm taking a guess that the

To: [email protected]>

part of the invite message is supposed to be

To: [email protected]>

I can change the domain with the session target in the dial-peer but then its obviously not sending to the proxy server but to the domain. How can I change the domain for the requested address but not change the session target?

I'm wondering if that question makes sense!

richard-o Fri, 06/29/2007 - 19:34

Another step along to a fully working SIP-UA.

I've successfully received incoming calls now with full duplex voice, so everything is sorted out as far as that. There was a minor issue with codecs (Cisco defaulting to G.729 annex A and the networking running G.729b, G.711a, G.711u)

The incoming messages are as I suspected

To: [email protected]>

So the question remains - how do i change the domain i'm operating in without changing the destination pattern from the proxy?

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