Cisco CUCM SIP CUBE calls fail early media

This fix is documented in various places on Cisco's site, but a common point of confusion.

See Update below for symptoms found after first resolution

I've run into two different scenarios where a users on a CUCM deployment with CUBE routers as the demarcation point to the PSTN via SIP trunks has had problems dialing numbers terminating on IVR or automated attendants.  One, frighteningly, was calls to 911, the emergency number in the US.  The callers would either experience continuous ringing, or no audio although the call did not appear to be dropped.

You don't want emergency calls going to dead air.

CUCM -> SIP trunk -> CUBE -> SIP trunk -> PSTN

The short story is that "early media" cut-through needed to be enabled.

This is enabled by default on the CUBE routers but could be potentially disabled or modified on the router using the "disable-early-media" command.  After reviewing the configuration, I found no variations of the command applied to the CUBE configuration.

The answer then is to send a PRACK via the CUCM configuration.  This is disabled by default in CUCM.

In CUCM, navigate to Device | Device Settings | SIP Profile.
Find the profile associated with the SIP trunk between CUCM and CUBE.
Modify the SIP Rel1XX Options. I've found Send PRACK for all 1XX messages to be universally successful.

Before:

After:

From CUCM help:

This field configures SIP Rel1XX, which determines whether all SIP provisional responses (other than 100 Trying messages) get sent reliably to the remote SIP endpoint. Valid values follow:

Disabled (default) — Disables SIP Rel1XX.
Send PRACK if 1XX contains SDP — Acknowledges a 1XX message with PRACK, only if the 1XX message contains SDP.
Send PRACK for all 1XX messages — Acknowledges all1XX messages with PRACK.

Update:

Found with this configuration above, users had issues with transferring incoming PSTN calls to external PSTN calls.  Incoming PSTN call could not be transferred or conference with another off premise call until until the callee answered.

I found disabling the SIP Rel1XX options and simply checking Early Offer support for voice and video calls (insert MTP if needed) allowed both successful calls to the network based IVRs and transfer / conferencing in a timely basis.

After: