We want to implement the following method for call recording -

Setup packet mirroring on the agent phones for media RTP
C++ TSAPI application to get call metadata

My questions are as follows -

1. Will we be able to get the RTP information (ports and IPs) on the TSAPI call events ?

2. Do we require any licenses to use TSAPI events ?

3. If TSAPI feed requires additional licences we could decode the Sip packets to get the metadata as well as the RTP information. Will the Sip packet provide all call metadata such as ANI, DNIS, Extension etc. Are there any additional information in the TSAPI call event not available in the Sip packets ?

I thank you in advance for your answer.

Debendra Modi

1. Will we be able to get the RTP information (ports and IPs) on the TSAPI call events ?

No.

2. Do we require any licenses to use TSAPI events ?

Yes. You will need 1 TSAPI Basic license for each station that you monitor.

3. If TSAPI feed requires additional licences we could decode the Sip packets to get the metadata as well as the RTP information. Will the Sip packet provide all call metadata such as ANI, DNIS, Extension etc. Are there any additional information in the TSAPI call event not available in the Sip packets ?

I do not know exactly what information is available in the SIP messages. They may not include ANI & DNIS in a consistent format. They may also be missing useful information such as UUI, UCID and VDN number.

Devconnect support DMCC recorders based on Single-Step Conference, Service Observer or Multiple Registration.

Martin

SIPREC is available from the ASBC-E. SIP messaging does not necessarily contain information like UUI, UCID, VDN, Skill information that is available from TSAPI and DMCC APIs.

No CM API provides the station/and switch side IP/ports used for calls, however DMCC can connect a party into a call and get at the RTP stream for a party in the call, thus it is the mechanism that call recorders use as opposed to having IP switches (line taps) forward RTP information that the PBX is sending/receiving. If AE Services SMS service supports the status station operation then you could get at the switch and station side IP/port information for H.323 device, not sure if it is available for SIP devices - I am not sure SMS supports 'status station,' you would need to research that.

Thanks John and Martin.

Can you please confirm if ANI (Caller ID) of an outside caller and the DNIS (The number the caller dialed for an incoming call) would be available on the Sip packets ?

We have setup a test AvayaLab and have been able to make internal calls. We are trying to get external lines connected so that we would be able to get external calls as well. In our internal calls sip captures we do see a CallID and AV-Global-Session-ID on the sip packets. Can these be linked to the call UCIDs ?


P-Asserted-Identity: <sip:9001@avayalab.local>
P-Location: SM;origlocname="Core1";origmedialocname="Core1";orighomelocname="Core1";termlocname="Core1";termmedialocname="Core1";termhomelocname="Core1";smaccounting="true"
Av-Global-Session-ID: 8708b560-8ac7-11e8-828a-005056874e76
Av-Call-Appearance: <sip:9002@avayalab.local>;+avaya-cm-line=1
Endpoint-View: <sip:9001@avayalab.local;gr=ebae5d91dd2490dc2eb2825e9ac128f3>;local-tag=-12965a385b4fa03d60ed88fc_T900110.245.15.8;call-id=8713969c8ac741e897800505687ed2e;remote-tag=8713966a8ac741e8977f0505687ed2e
RSeq: 1
Require: 100rel
Record-Route: <sip:192.168.14.153:5061;transport=tls;lr>
Record-Route: <sip:sm1@192.168.14.152;av-asset-uid=rw-2d44d717;lr;transport=TLS>
Record-Route: <sip:127.0.0.2:15061;transport=tls;lr;ibmsid=local.1531442497112_222061_222477;ibmdrr>
Record-Route: <sip:127.0.0.2:15060;transport=tcp;lr;ibmsid=local.1531442497112_222061_222477;ibmdrr>
Record-Route: <sip:sm1@192.168.14.152;av-asset-uid=rw-2d44d717;lr;transport=TCP>
Accept-Language: en
Contact: <sip:192.168.14.153:5061;transport=tls;gsid=8708b560-8ac7-11e8-828a-005056874e76;epv=%3Csip:9001%40avayalab.local%3Bgr%3Debae5d91dd2490dc2eb2825e9ac128f3%3E>
User-Agent: Avaya one-X Communicator/6.2.12.20 (Engine GA-2.2.0.169; Windows NT 6.2, 64-bit)
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, SUBSCRIBE, NOTIFY, REFER, INFO, PRACK, PUBLISH, UPDATE
Supported: 100rel, histinfo, join, replaces, sdp-anat, timer
To: <sip:9001@avayalab.local>;tag=87a85668ac741e897720505687ed2e
From: <sip:9002@avayalab.local>;tag=6e39950e5b4fa03b5b605eb8_F900210.245.15.7
Server: Avaya CM/R017x.01.0.532.0 AVAYA-SM-7.1.3.0.713014
Call-ID: 5b_10b9fb-722e373d5b605a37_I@10.245.15.7
CSeq: 92 INVITE
Via: SIP/2.0/TCP 10.245.15.7:49849;branch=z9hG4bK5c_10bc6c31263bbd5b605fb0_I9002
Content-Type: application/sdp
Content-Length: 178


Thanks again.

Calling party number is available (when the service provider provides it), keep in mind everything is a service that there is some fee for between carriers, and it is possible that the carrier originating the call does not have an agreement with the destination carrier, but that is rare anymore, at least in north america.

DNIS - Dialed Number Identification Service, is different than called party number. Typically the originating carrier maps DNIS (e.g. 18005551212) to the called number (12125551234). DNIS is sometimes carried carrier to carrier via SS7 in a separate parameter from the routing number, however it is not an information element in most signaling to PBXs (e.g. Communication Manager). Thus you can see called number, but not necessarily DNIS.

If UCID is not present in the SIP signaling, then there is not a way to map a UCID value that is presented via CTI when the call terminates at a station to the SIP call id. UCID is made up of a switch node number, and a call id value from the system that created the UCID value along with a timestamp. These values are not drawn from the SIP signalling.