Migrating your PRI gateways from h.323 to SIP

One of the recommendations in the CSR version 10 SRND is to use SIP trunks from CallManager to your IOS voice gateway.

This makes sense for multiple reasons:

  1. Avoid dealing with MGCP’s finicky nature when making changes.
  2. A big step towards moving your PSTN connections to SIP trunks and using CUBE as an SBC.
  3. Gateway call forking for call recording.

CUCM 10 includes the ability to fork calls at the gateway to the call recording server.  This is great news for large deployments where using the phone’s BIB for forking isn’t desirable.  Some information about gateway forking is here.

Migrating from h.323

The conversion process is pretty straightforward for anyone running h.323 as they’ll already have all of the necessary dial-peers built typically and will only need to deal with SIP DTMF issues and SIP SDP Early Offer issues for SCCP phones.

In a recent conversion that I did I did the following:

Modified the GW -> CUCM dial peers to look like this:

dial-peer voice 101 voip
preference 1
destination-pattern 5…$
session protocol sipv2
session target ipv4:
session transport tcp
voice-class codec 1
dtmf-relay sip-notify

There are several ways to deal with DTMF.  I prefer OOB to In-band (rtp-nte / RFC 2833) personally, so I go with sip-notify.

We must make adjustments to the SIP Trunk Security Profile to Accept unsolicited notification so that the DTMF information will be allowed into CUCM.  I copy the normal Non-secure SIP Trunk Security profile to a new name (e.g. Non-secure SIP Trunk Security Profile — GW) and change this option.  I then apply this security profile to the trunk I create from CUCM to the Gateway.

DTMF becomes a problem for SCCP phones unless you do either an Early Offer SIP trunk, or you provide an MTP to deal with it.  There’s an excellent document here – https://supportforums.cisco.com/blog/154706 that discusses all of the various options.

I don’t really like having to force MTP for every call, especially since SIP phones won’t need it for DTMF, so I opted to create a new SIP Profile that enabled Early Offer support for voice and video calls (insert MTP if needed) and apply it to the trunk.

Converting from MGCP

Converting from MGCP is a more significant undertaking.  The PRI configurations all move from CUCM to the Gateway as dial-peers.  I’ll add more to this post to include a sample build of dial-peers to accomplish this.