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Community Member

SIP Reusing trunks and Proxy

Hi,

After some testing with my ITSP I found that outbound SIP is not working as I expect. In-bound works fine. I spoke to the ITSP and they said I didn`t re-register again on the outbound call. I checked the SIP-UA config, and I thought the `connection reuse`was suppose to keep the Trunk "up" and not having to re-register (let me know if I'm wrong.

So for testing I added (CLI) an outbound-proxy command to the dial-peer and got the following error:

Internal Error: unset return code 836A88E8

But the commands took. I've removed them and it added the "no voice-class sip out-bound proxy" to the config. I guess I need a little education here about what's a default and required specifically when using the connection resuse and outbound dial-peers.

I went from this:


dial-peer voice 1022 voip
corlist outgoing call-national
description **CCA*North American*Long Distance**
translation-profile outgoing PSTN_Outgoing
preference 6
destination-pattern 81[2-9]..[2-9]......
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
  session protocol sipv2
session target ipv4:xx.xx.xx.xx
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad

to this:

dial-peer voice 1022 voip
corlist outgoing call-national
description **CCA*North American*Long Distance**
translation-profile outgoing PSTN_Outgoing
preference 6
destination-pattern 81[2-9]..[2-9]......
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte

voice-class sip outbound-proxy ipv4:xx.xx.xx.xx
  session protocol sipv2
session target ipv4:xx.xx.xx.xx
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad

(Got the Internal Error), and then this:


dial-peer voice 1022 voip
corlist outgoing call-national
description **CCA*North American*Long Distance**
translation-profile outgoing PSTN_Outgoing
preference 6
destination-pattern 81[2-9]..[2-9]......
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
session protocol sipv2
session target ipv4:xx.xx.xx.xx
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad


Thanks in advance for any help,

Bob James

10 REPLIES

Re: SIP Reusing trunks and Proxy

connection-reuse is actually to make sure the same ports are used.  It can be used to help connections stay up, especially if your provider wants to keep using the same ports over again.

How often is the provider expecting the UC500 to reregister?  We should also try our credentials if they send a 401 in response to our invite.  We could probably look at changing some registration timers to register more often if that is the problem.

Community Member

Re: SIP Reusing trunks and Proxy

Steven,

So I see what part of the issue is; it's changing our invite from the user account in the proxy setup to the main number.

For example; our SIP username is 4035551212, and out main number is 4036667758, the sip invite is sending out "Bob James" <4036667758>

Where/how is it picking them up?

Thanks,

Bob james

Re: SIP Reusing trunks and Proxy

Would you mind posting your config and removing the passwords?  That would be useful.

Do you know how your SIP provider authenticates and what they want to see in the message?  Having the correct caller ID in the from Header isn't always incorrect.

Community Member

Re: SIP Reusing trunks and Proxy

Against policy, but I can put certain sections to you in PM..

I was wrong it's the reply header in the invite that's showning the main number. (I think it's the callerID being sent)

How do I PM you?

Re: SIP Reusing trunks and Proxy

We can do this over email and post the results of what fixes the problem here.

Community Member

Re: SIP Reusing trunks and Proxy

Working the issue with the ITSP will post result when I figure it out.

Bronze

Re: SIP Reusing trunks and Proxy

The "connection-reuse" CLI has nothing to do with Re-registration - all it ensures is that the UC500 uses the same source & destination port (UDP 5060) for all SIP requests. By default the source port is usually ephemeral (meaning random) inline with RFC3261.

"voice-class sip outbound-proxy" is meant to change the actual L3 destination for the SIP messages - instead of sending the SIP message to the SIP proxy server (defined in sip-server CLI). The reason this is useful is if the ITSP wants all SIP messages to go to their domain (say sip.itsp.com which may not be resolvable) but be fronted by an SBC (session border controller) as the "outbound-proxy". It is also used for NAT traversal. Again this has nothing to do with "re registration" per se

The caller ID issue you point out below is a setting on the UC500 / CCA - you can either choose to use the main number for caller ID for all outbound calls OR set it such that the individual DID for an extension is the caller ID for calls from that extension (for extensions that have no DID associated with them, main number is used as caller ID). You can set this in the Outgoing Dial Plan tab on CCA.

Community Member

Re: SIP Reusing trunks and Proxy

Yes thank you all. It appears the UC is using a different port as well as 5060. I am working on the NAT devices in line to see if I can correct this. Is there a way to lock udp5060 to both the source and destination ports on the UC and force the ITSP to do the same?

Bronze

Re: SIP Reusing trunks and Proxy

If you have:

sip-ua

   connection-reuse

The source UDP port for SIP messages sent by UC500 will always be 5060

If that is not the case, then we need a wireshark to see what other port is being used. Assume this is all SIP traffic (not RTP).

Re: SIP Reusing trunks and Proxy

As Maulik said, connection-reuse will force the UC to use 5060 as the source and destination port.  Having another NAT device in line may break this though.  The NAT device could change the source port, making the UC500 config worthless.

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