Hi,
I am a newbie starting development with DMCC .NET SDK, Is it possible to record voice and call information of multiple extensions say 50 selected extensions using DMCC. I need to record trunk calls as well as extension to extension calls also. Should i process the RTP streams seperately?

Best Regards
Rasheed

Hi Rasheed,

There is a C# call recording sample application for the .NET SDK that you can use as a starting point. The RTP streams can be processed on the server or by your client application. The latter approach is more widely used. Recording call information messages must be performed by your application.

Regards
Isaac

The sample application that Isaac mentions sets up a one recorder for a specific (monitored) station. To record 50 stations, you would need to modify the sample so it 'replicates' this behavior 50 times (all from within one software solution). The solution will work regardless of the type of call (trunk to station, or station to station) that the monitored station is involved in.

The solution uses single step conference to do its work. Some call recording solutions utilize service observing to monitor the station for call traffic. There is a FAQ in the DMCC area that provides more details on these two possibile approaches.

I ahve a follow up question on this. Can SIP device be recorded using singlestepconference method. From DMCC java reference guide
"
For Call Control, any Communication Manager supported endpoint can be controlled by the AE Services server

You cannot choose independent or dependent dependency mode with SIP endpoints, so youcannot exercise device monitoring/control over them."

Any restrictions on recording SIP endpoints?

Ilana,

Two thoughts,
1) please post this in a new thread, with luck someone who knows will see it
2) please add some call flow details.. like is it a sip to sip call? PSTN to SIP through CM trunks, DCP to SIP? All of the above? Something else? I think that as long as you are talking about a scenario where a DMCC station is being SSC'ed into a call involving some parties, there are no restrictions on what types of parties can be in the call the DMCC station can be SSCed into. However if you are asking if a SIP station can be SSCed into an existing call, then there may be restrictions on what is supported there..

John

Dear John,
Thanks a lot for the reply. How can i test it using AES simulator. Can i make some calls and will get real audio wave file if i try this sample. which soft phone should i use for this testing. where can i download the same.

Thanks and regards
Rasheed

Abdul,
The simulator (IPCoDE) does not have an audio mixer. Thus it can not support audio in IP calls involving more than two audio participants. All call recording scenarios with Communication Manager involved a minimum of three parties in the call. Thus while you can set up the control plane of a three or more party call, you do not receive audio with IPCoDE.

To test this I suggest you utilize a timeslot in the 'AES/CM Remote Lab' Read through the following page for details.
https://devconnect.avaya.com/public/dyn/d_dyn.jsp?fn=135

Hi John,
Is there any softphone available which can be used with AES simulator

Thanks
Rasheed

IP softphone
https://support.avaya.com/japple/css/japple?temp.documentID=251593&temp.productID=107767&temp.releaseID=227980&temp.bucketID=108025&PAGE=Document

SIP one-X Desktop Edition

https://support.avaya.com/japple/css/japple?temp.documentID=252145&temp.productID=286865&temp.bucketID=108025&PAGE=Document

The SIP one-X Desktop Edition does not support third party call control services invoked via TSAPI ON it. it may be the target of some operations (like makeCall).

Ilana,
SingleStepConference of a SIP station into an existing call, or on behalf of a SIP station is supported using AES 4.1 and CM 5.0 and SES 4.1 or later. Be careful of what SIP endpoint you use 96XX and 1616 terminals are the only SIP devices, and you need to use the current station firmware.

John

Thanks John
Ilana

Hi John,
i installed the softphone, if i install one more softphone in same system or another system, can we make a voice call from this one to other using AES simulator and record it using the simplerecord application,or shoud i take a time slot for testing it in the remote lab?

Thanks and Regards
Rasheed

To test audio in a 3 or more audio stream call, you will need to use the AES CM Remote Lab.

You will need to put the softphone on another system to test point to point calls since it requires exclusive access to the microphone (two on one system does not work).

John