CISCO/CUBE SIPREC create two recording sessions for the same call

[miguelaustro]

Hi community,

I'm testing CUBE SIPREC feature.

The scenario is:

A--> CUBE -->B

         |

         |----> SIPREC third-party recorder (Rec).


When I make a call A->B, Rec receives two initial requests (INVITEs) rather than one, but with the metadata in opposite way.
For instance:

INVITE 1:
<recording xmlns="urn:ietf:params:xml:ns:recording:1">
.

<participant participant_id="6AgUdRjEEeeptbNuGDetvQ==">

<nameID aor="sip:A@IP2"> </nameID> 

</participant>

<participant participant_id="6AgUdRjEEeeptrNuGDetvQ==">

<nameID aor="sip:B@IP1"> </nameID>

 </participant>

INVITE 2:

<participant participant_id="6AgUdRjEEeepuLNuGDetvQ==">

<nameID aor="sip:B@IP1"></nameID>

</participant>

<participant participant_id="6AgUdRjEEeepubNuGDetvQ==">

<nameID aor="sip:A@IP2"> </nameID>

</participant>

As a result, two recordings are generated at the "Rec" side (only the 2nd INVITE is the right one)
I was expecting to receive just call one from the "recorded leg", if such thing exists in CUBE.

Any idea why CUBE creates two Recordings sessions for the same call, instead of just one?

I'll appreciate any orientation in this regards.

Thanks,
Miguel

PS: Checked http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-sip-recording.html
But not reference to an specific call leg is being recorded

[Dennis Mink]

how does your CUBE communicate to your recording server?

also, are you using the CUBE in flow through mode?  I am assuming yes, in other words does the CUBE terminate 2 RTP legs? one inbound from the caller, and one outbound to the callee?

Please rate if useful

[sunil.kumar]

Hello Everyone

I have simple vCube lab setup. with two CME on Right and Left. Each CME has 1 phone registered. 

The middle men is CSR1000v SIP Cube enabled with SIPREC config.

Call are working from right to left and Left to right.. and receiving SIP and Dialpeer debug messages. but SIPREC doesnt seems to be working.

Kindly advise on the missing part.

CSR 1000v - 16.8

config attached 

[R0g22]

What is the recording server ? When you say it is not working, what exactly is not working ?

[sunil.kumar]

Thanks for the reply!

Orecx is the recording Server located at 10.1.1.10
Dailpeer 559999

Orecx Server is also having wireshark where we are not able to see SIP message when we pickup the ringing Call and connect it to establish the rtp.

I have also put Debug voip dialpeer all .. in which i am not able to find any entry of recorder dialpeer.
Same goes for debug ccsip all .. Nothing found going towards 10.1.1.10 or initiating a SIP invite or Meta info

Regards

[R0g22]

Can you test with the following config instead -

media class 6677
recorder parameter siprec
media-recording XXX >>> XXX is the dial-peer tag pointing to recording server

On the dial-peer that needs to be forked, add "media-class 6677" command.

Test after the changes.

[taylor.averdick]

Did you ever resolve this problem? I'm experiencing a similar issue.