IP-Phone Custom Rx Stream: Merging It Into Call's Tx Stream

[Bliss]

Hello!

I'm sending an RTPRx XSI command to a Cisco IP-Phone, in the following format:

<CiscoIPPhoneExecute><ExecuteItem URL="RTPRx:<Local-IP>:<Phone-Port>:100"/></CiscoIPPhoneExecute>

This successfully starts a custom Rx stream on this phone-device, so my app could stream audio to it.
This works well. The streamed audio is well-heard at device's end.

While the device is during a phone-call (regular, peer-to-peer), when streaming audio to its custom Rx stream, this audio is (still) well heard at device's end, but is not heard to the other call-party.

Is there a way to accomplish that - making the other call-party hear the audio streamed into device's custom Rx stream?
Could the custom Rx stream be "bridged" into the call, possibly merged into device's Tx stream for this call?

[upchaurasia]

RTPRx will enable phone to receive audio from defined IP address & port. This audio stream won't be heard by other party. Are you trying to achieve Agent Greeting feature?

Regards

Umesh

[Bliss]

Yes.

I would like my "custom" audio stream to be heard by the other party, to accomplish the same as the Agent Greeting feature, but without going through the CUCM, just by utilizing phone's internal audio-mixing capabilities.

So, in other words: I'm trying to accomplish an Agent Greeting feature, but directly through the phone (without using the CUCM).

After all, the phone does have a BiB (Built-in-Bridge), which is capable of mixing streams together.
So I thought that a way could be found to control it for local mixing of its own (call) Tx stream with the additional stream transmitted to it by a 3rd party (i.e. my app).

The IP Phone Services Application Development Notes document (for CUCM r9.1) implies this is possible (page #86):
"If the user initiates the Rx or MRx URI using an application, then the user likely is not concerned about having the audio mixed with the current call".

Could you shed some light over this?

[upchaurasia]

IP Phone Service RTPRx is not the way to implement this. You need to use Cisco JTAPI library to implement this. addMediaStream is the API it uses phone's BIB to mix the streams.

[DavidStaudt · Accepted solution]

My interpretation of the quoted passage is that the end user interacting with an IPPS/XSI application in the phone UI is taking a positive action to request receiving media streaming, and so expects/accepts that the incoming Rx/MRx stream will be mixed with the incoming call audio (which might otherwise be confusing/undesirable) and, assumedly via the same IPPS app UI, can terminate the media stream at will.
IMHO this passage in the same section precludes your use-case from working: "Tx or MTx URI stream requests received when a call is active are rejected"