GSAPI - SIP Forking

[mauro.novi]

Hello everybody,

i have a problem with use of GSAPI feature, in order to fork sip stream to a recording destination.

In our infrastructure we have multiple recorder that need to receive SIP stream + RTP Stream.Contact Center

We configured GSAPI on CUBE to fork only the calls that we need to recorder. But we notice that the fork stream include just the RTP and not the SIP communication.

How can I send SIP stream in addition to RTP, throught GSAPI feature?

Thanks in advance

[ptindall]

GSAPI allows your application to direct the forked RTP streams from the gateway to a destination of your choice but it will only be the RTP.  To get a SIP call to your recorder you'll have to build the SIP user agent into your application to send signalling to the recorders or use another approach such as dial-peer controlled media forking or CUCM-controlled gateway forking.

[mauro.novi]

Hi Ptindall,

thanks for your answer, I still have another dubt: if we implemented mechanism to send SIP signaling from application to recorders, as you recommended, how we can correlate RTP and SIP signaling on the recording server?

Thanks in advance

[ptindall]

Your application will initiate forking from the gateway to the ports specified in the SDP m lines the recorder sends back to the application in its OK/SDP response.

The recorder allocates the receive ports for the 2 streams in response to the SIP INVITE.  You tell the gateway to fork media to the recorder IP address and those two ports.