The customer's UC and CCX environment has been configured to use Secure RTP everywhere for PCI compliance. Agents use both jabber and physical phones configured for SIP TLS/SRTP.
The CCX server is configured with a script that allows the caller to request a callback when the agents are busy or unavailable. The script collects the CLID of the caller, have them verify the number by selecting an option as well takes a voice message.
When an agent does become available, he/she is presented with a ghost call, and after playing the recorded message from the caller, gives an option to the agent for the system to dial back. This ghost call is secured noticed by the lock sign on the jabber.
The PSTN access is by means of a SIP trunk.
Based on some reading, this function is performed by the "place call" step within the CCX script. Unfortunately, this particular leg of the call has no media parameters (i.e. no SDP inserted in the initial invite) and hence a delayed offer over the SIP trunk. This had resulted in the call to get negotiated as normal RTP.
I tried enabling the SIP trunk with Early offer (Insert MTP if required). It now engages MTP of the call manager and creates SDP but only with RTP and not SRTP.
Wondering if anyone else has ran into a similar issue ?
Thinking
1. If there are additional parameters which could be defined for "placecall" step to have it start as an SRTP call?
2. If there was a possibility of not using "placecall" step and some other mechanism which uses agent's own line to make an outbound call to the party needed call back. When Jabber initiates a call, it is capable of starting the call with secure RTP and SIP messaging has the necessary SDP in it.
Was also looking into following bug but wasn't sure if this was somewhat related.
https://bst.cloudapps.cisco.com/bugsearch/bug/CSCvu03571
