I am planning to implement AWS Lex bot integration with Cisco Call Studio (CVP).
* Need your suggestion on how to do this?
* Is there any connector needed like gRCP(Google Dialog flow) or Through RestAPI can this be achievable ?
* If some one already did this suggest me steps.
Current UCCE version 11.5
There's two different approaches you could take. One is to do transcription, bot intent processing and response TTS output as three separate operations using whatever mix of speech services you prefer. The other is to use a single speech service for voice input and voice bot processing with media or text output.
If you opt for an MRCP based approach then I believe you can get out-the-box Amazon integration using UniMRCP.
If you want to try something out in the lab and roll your own, then you could use the gateway media forking approach I published. I haven't pushed out an Amazon version yet as it's not quite finished but you could use it for transcription with Google or Microsoft and then use a backend HTTP API for the bot part (text in / text out).
Thank you, I have few queries.
One of the key reasons I didn't use the MRCP server approach was that it's only applicable to the IVR part of the call; I wanted a solution that would also allow various media stream processing scenarios between caller and agent. What I provided is custom code primarily to allow anyone interested to get started very quickly, explore some innovative speech scenarios and investigate their viability without having to commit significant investment to their own science project.
As it's custom, all the normal custom solution caveats apply with regard to lack of official support, level of testing, not being an official part of the product, etc. It's clearly not intended for production and any use for live call traffic is at own risk.
Regarding IOS device, I typically use vCUBE but have used it on 29xx/39xx hardware with SIP->SIP calls. It's just making use of the gateway media forking XMF interface, exactly the same thing that CUCM uses for network based recording. I think you might hit problems if you try to use ISDN->SIP.