The SIP world

[Rolando Valenzuela]

I'm trying to move from a full T1 environment to a full SIP one, and while reading multiple guides there are commands that keeps showing and I would like to know more about it from people with real world experience and I was wondering if you can help me, most of them are from the global voip configuration mode.

What is the main purpose of the highlighted commands and if they are recommended for most of deployments.

voice service voip
 address-hiding
 mode border-element
 allow-connections sip to sip
 no supplementary-service sip moved-temporarily
 no supplementary-service sip refer
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
  bind control source-interface Port-channel33
  bind media source-interface Port-channel33
  header-passing
  asserted-id pai
  privacy pstn
  early-offer forced
  midcall-signaling passthru

Lastly this is the configuration I envision for incoming and outgoing dial peers, what you guys thing?

dial-peer voice 1 voip
 description ** INCOMING TFN **
 translation-profile outgoing PSTN_INCOMING
 preference 5
 destination-pattern 800.......$
 session protocol sipv2
 session server-group 1
 voice-class codec 1  
 voice-class sip early-offer forced
 dtmf-relay rtp-nte sip-notify
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
!
dial-peer voice 11 voip
 description ** 10 DIGIT DIALING **
 translation-profile outgoing PSTN_OUTGOING
 preference 5
 destination-pattern [2-9]..[2-9]......$
 session protocol sipv2
 session server-group 2
 voice-class codec 1
 voice-class sip early-offer forced
 dtmf-relay rtp-nte sip-notify
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad

Thank you so much in advance for all the suggestions.

Rolando A. Valenzuela.

[Dennis Mink]

Rolanda,

are you using this in conjunction with CUCM? in which case you will need to point to your CUCMs in terms of dial peers as well. 

address hiding is pretty much like flow through, where the CUBE will signal the ITSP its own IP address only and not that of the calling/called endpoint.  This is done purely for security purposes.

Please rate if useful

[Rolando Valenzuela]

Hi Dennis!

Yes, I am using CUCM with this... in fact server-group 1 has all the CUCM IPs while server-group 2 has TISP IPs.

Do you have more details about the other commands?

Thanks.

Rolando A. Valenzuela.

[Ayodeji Okanlawon · Accepted solution]

Here you go :)

no supplementary-service sip moved-temporarily: This is mostly used to handle call forwarding on CCME and you dont need it on your CUBE gateway. It is sometimes a good idea to disable this on CME because some ITSP do not support it.

More details on this here..

http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-express/91535-cme-sip-trunking-config.html

no supplementary-service sip refer: This is used to disable The SIP REFER method and to rather use an INVITE to complete supplementary services such as call transfer. Typically when an event such a transfer occurs within a dialog, CUCM/endpoint will send a REFER to the CUBE. CUBE will then pass this REFER to the other end. If you dont want to use this in scenarios where the ITSP cube is talking to doesnt support this, then you can disable it or do waht is called REFER consumption and then allow CUBE to re-originate the transfer using a normal INVITE.

More details here

http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cub-sip-dyn-refer-handling-pdf.html

 Header-Passing:

Header-passing enables CUBE to pass its SIP headers to other parties involved in the call. It is very critical in contact center environment where SIP headers are needed to be passed to other voice applications such bootstrap vxml applications required to invoke interactions with other applications such as CVP. Without header-passing these applications will not know thats in the ISP headers such as ANI, DNIS etc

asserted-id pai

This is telling the CUBE to user the P-Asserted-Identity as the preferred method for privacy or CLI. There are different methods to display CLI within the gateway.

PAI

PPI

Remote-Party-ID

From

To explain the last two commands we need to explain about privacy and CLI screening

asserted-id pai
 privacy pstn

++ Using the Privacy header ++

If the Privacy header is set to None, the calling number is delivered to the called party. If the Privacy header is set to a Privacy:id value, the calling number is not delivered to the called party.

++ Using Privacy values from the peer call leg ++

If the incoming INVITE has a Privacy header or a RPID with privacy on, the outgoing INVITE can be set to Privacy: id. This behavior is enabled by configuring privacy pstn command globally or voice-class sip privacy pstn command on the selected dial-per.

So in your scenario when a call comes in to your gateway with privacy set on it, then the outbound dial-peer to either CUCM and ITSP depending on the direction of the call will automatically be set to use privacy:id ( which will then honor the privacy of the caller)

HTH

[Rolando Valenzuela]

Amazing!

Thank you so much for all the comments, I read the command reference for all those commands but still I was kind of lost.

Since neither you or Dennis mentioned something about the dial-peers I will assume they are good and give them a try.

Thanks again for the help guys.

Rolando A. Valenzuela.