Hello,

We are currently investigating in how we can adapt our current DMCC recording application to Avaya's ACE platform.

Currenty, our recording app receives RTP streams through softphones which are connected to the recorded calls via SSC.

After browsing through the ACE FT developer docs, we unfortunately have not found any direct equivalent to redirecting RTP streams.

The only recording method described in the docs is the use of MediaService to store the audio to a file on the server.
Since we need a real time access to the audio data of a call, this approach does not fit our requirements.

A potential idea to retrieve a calls audio data is to register a SIP extension and to establish a conference from this 'softphone' to the call we want to record.

Unfortunately, it seems that conferencing to a third party is not sopported with sequenced apps. Is that correct?

Might it be possible to use a sequenced app to redirect the call to an endpoint app which finally establishes a conference between caller, callee and a recorder softphone?

Are there perhaps other methods available to fulfill our requirements?

Another topic that is not explained in the developer docs is the availability of additional call information (private data).
How can this data (agent id, VDN, universal call id, etc.) be retrieved with means of ACE FT?

Regards,
Claus Suffel

Honestly I have not forgotten about this thread. I have been kicking it around among a set of SMEs.

The consensus is that using ACE F/T (or Web) Services to do call recording with the constraints you impose is not viable. At the present time call recording utilizes the Avaya Media Server and that violates a requirement you have.

It is understood that there is a desire to move in this direction and there are outstanding requests to the product team to enhance the F/T and surrounding capabilities in order to provide a more tenable solution in the future. Unfortunately I can not talk about these as they are not committed for development.

It is possible to intercept the call in an application sequence and use proxy services proxy the call forward and to include a third endpoint (the recorder) in the call where the recording would occur. This solution though has a set of complexities that are undesirable in our view.

There is also the ability to modify SIP headers as the messaging flows through an application. Thus the SDP information could be modified so that it is directed through a recording server. I have been unable to ascertain if that is possible with all the impacted messages (INVITE, 200 OK and re-INVITE) which would need done before proceeding down this path. There are some issues in our minds with this solution as well (for one you are getting two streams that need merged creating extra work for the recorder, but that can also be viewed as useful).

In short I do not have a solution to this problem that meets the constraints you set forth using the current product capabilities.

Hi John

Thank you very much for your efforts in clearing up this topic.
We will discuss the described possibilities internally.

Please keep us informed in the case you will have more information about Avaya's future plans to enhance F/T's capabilities in the desired direction.

Please can you just add some additional information regarding our last question about the handling of private data with F/T.

Best regards
Claus

Q How can this data (agent id, VDN, universal call id, etc.) be retrieved with means of ACE FT?

A:As of Session Manager 6.2 and ACE 3.0, for the most part information such as agentID handling the call and the Hunt Group the call came through is not availalbe in the SIP headers (History-Info), and thus unavailable to ACE F/T applications. I was able to confirm that VDN information is available in the History-Info header, unfortunately in my environment I can only pass on third hand information related to hunt groups and agentIDs, as it would take some signification configuration work to have an environment where it could be validated.

You should note that the whole SIP Contact Center model with Avaya Aura Contact Center is quite different from that used with Contact Center Elite, and much of the traditional information would come from the APIs associated with Avaya Aura Contact Center and not from the F/T SDK.

UCID and UUI information can be sent over a properly configured SIP trunk to a service provider. At the moment (R6.2) the configuration for application sequenced endpoints precludes the SIP Trunk group configuration for sending User Info data with the call. Thus if you are seeking User Info date in a F/T sequenced application that is associated with a explicit endpoint, User Info is not available. If you are seeking it for Implicit Outbound calls, it can be made available. Depending on the external SIP Service Provider's ability to include User Information in SIP messaging, it may be available in an Inbound call sequence for a application sequence applied in an Implicit contenxt.

Explicit application sequences are part of a user's System Manager user profile (applied to a specific station). Implicit application sequences are provided to a range of SIP calls based on some form of pattern match.

Hi John,

Are there any news regarding the capabilities ACE F/T offers for implementing a call recording solution that works similar to SSC or SO call recording as used in AES environments?

Regards
Claus

Not that I can convey at this time. Sorry. Why the interest in ACE call recording as opposed to AE Services? Is it because of the other PBX vendor support by ACE or ???