MRCP for call handling

[Ahmad Eldaly]

Hi

I'm newbie in MRCP and still try to understand its basics and purpose.

Previously i have used SIP for handling calls in IVR service or Call Centers. Therefor, i want to know if MRCP can do the same and handle the call or it is used for other purpose?

What i understood from my online searches that MRCP is used in TTS and STT and those can be integrated with telephony systems or other system that need to do TTS or STT BUT not responsible for handling the call itself like SIP.

Then hypothetically, If I need to have TTS feature in Mobile application. I can use HTTP or MRCP to integrate with third party TTS service. But if i need TTS in telephony system I must have system like Avaya or asterisk to handle the call itself but they can integrate with TTS (like API) during the call to do the function without losing control of the call itself.

Did i understand it correct?

[Michael Levy]

Here is a drawing I often use to explain basic VXML IVR systems - https://imgur.com/a/uYQPIgk

Of course, this is not the only way to build a solution, but it is a very common approach.

I think your understanding is correct. MRCP is not a protocol for handling calls. Calls arrive via SIP or similar and are handled by something. Typically a system like Freeswitch or Asterisk, or in many systems VXML voice browsers, are used to answer the call. MRCP is a protocol used by these devices to control media. The media may be Speech sent for recognition, speech sent for recording, speech sent for verification, or text sent for speech synthesis. The most common cases are recognition (ASR, STT - https://www.rfc-editor.org/rfc/rfc6787.html#section-9) and synthesis (TTS - https://www.rfc-editor.org/rfc/rfc6787.html#section-8).

VXML is one way people build these applications. Many VXML applications still use SRGS+XML (.grxml) grammars to control recognition. Often outbound prompts are synthesized by TTS or pre-recorded audio files.

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[Ahmad Eldaly]

Thank you Michael for the details.