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FreeSWITCH and Google SR

Usage Guide


Revision: 2

Created: June 23, 2017

Last updated: August 27, 2017

Author: Arsen Chaloyan


Table of Contents


1  Overview.. 3

1.1         Applicable Versions. 3

2  UniMRCP Module. 4

2.1         Overview.. 4

2.2         Configuration Steps. 4

2.3         Usage Examples. 5

Built-in Speech Context 5

Dynamic Speech Context 5



1       Overview

This guide describes how to utilize the Google Cloud Speech services with FreeSWITCH.




Note that the FreeSWITCH and the UniMRCP server typically reside on different hosts in a LAN, although both might be installed on the same host.


Installation of the FreeSWITCH and the UniMRCP server with the Google SR plugin is not covered in this document. Visit the corresponding web pages for more information.





1.1      Applicable Versions

Instructions provided in this guide are applicable to the following versions.


FreeSWITCH 1.4 and above

UniMRCP GSR Plugin 1.1.0 and above


2       UniMRCP Module

2.1      Overview

The module mod_unimrcp.so provides an implementation of the ASR and TTS interfaces of FreeSWITCH, based on the UniMRCP client library.

2.2      Configuration Steps

This section outlines major configuration steps required for use of the module mod_unimrcp.so with the UniMRCP server.


Create a new MRCP profile (or modify an existing one) in the configuration directory mrcp_profiles of FreeSWITCH. In the following example, the FreeSWITCH/UniMRCP client is located on and the UniMRCP server is on



  <!-- UniMRCP Server MRCPv2 -->

  <profile name="uni2" version="2">

    <!--param name="client-ext-ip" value="auto"-->

    <param name="client-ip" value=""/>

    <param name="client-port" value="16090"/>

    <param name="server-ip" value=""/>

    <param name="server-port" value="8060"/>

    <!--param name="force-destination" value="1"/-->

    <param name="sip-transport" value="udp"/>

    <!--param name="ua-name" value="FreeSWITCH"/-->

    <!--param name="sdp-origin" value="FreeSWITCH"/-->

    <!--param name="rtp-ext-ip" value="auto"/-->

    <param name="rtp-ip" value="auto"/>

    <param name="rtp-port-min" value="14000"/>

    <param name="rtp-port-max" value="15000"/>

    <!-- enable/disable rtcp support -->

    <param name="rtcp" value="0"/>

    <!-- rtcp bye policies (rtcp must be enabled first)

             0 - disable rtcp bye

             1 - send rtcp bye at the end of session

             2 - send rtcp bye also at the end of each talkspurt (input)


    <param name="rtcp-bye" value="2"/>

    <!-- rtcp transmission interval in msec (set 0 to disable) -->

    <param name="rtcp-tx-interval" value="5000"/>

    <!-- period (timeout) to check for new rtcp messages in msec (set 0 to disable) -->

    <param name="rtcp-rx-resolution" value="1000"/>

    <!--param name="playout-delay" value="50"/-->

    <!--param name="max-playout-delay" value="200"/-->

    <!--param name="ptime" value="20"/-->

    <param name="codecs" value="PCMU PCMA L16/96/8000"/>


    <!-- Add any default MRCP params for SPEAK requests here -->




    <!-- Add any default MRCP params for RECOGNIZE requests here -->


      <!--param name="start-input-timers" value="false"/-->





2.3      Usage Examples

Built-in Speech Context

Make use of a built-in speech grammar transcribe for recognition, having no speech contexts defined, by adding the following entry in the FreeSWITCH dialplan.


        <action application="play_and_detect_speech" data="ivr/ivr-welcome_to_freeswitch.wav detect:unimrcp:uni2 {start-input-timers=false}builtin:speech/transcribe"/>


Place a test call and make sure recognition works as expected.

Dynamic Speech Context

Dynamically load a speech context for recognition, by adding the following entries in the Asterisk configuration file extensions.conf.


        <action application="play_and_detect_speech" data="ivr/ivr-welcome_to_freeswitch.wav detect:unimrcp:uni2 {start-input-timers=false}/usr/local/unimrcp/data/directory.xml"/>


In this example, the speech context is loaded from a file located at /usr/local/unimrcp/directory.xml and having the following sample content.



  <phrase>call Steve</phrase>

  <phrase>call John</phrase>

  <phrase>dial 5</phrase>

  <phrase>dial 6</phrase>



Place a test call and make sure recognition works as expected.