No audio between Freeswitch and UniMRCP

[Stefano Favaro]

Hello,

I have installed the UniMRCP server on a different machine than FreeSwitch (version 1.6.19) but on the same network.

FS is on 10.4.1.151
UniMRCP on 10.4.1.152 with GSR

I always have a "no input" result and making a trace I don't see any RTP stream.

I've attached the log and the configurations.

Is there anything I can check to see if there is some problems with the configuration, please?

Thanks

Stefano F.

---

UniMRCP Server config:

<?xml version="1.0" encoding="UTF-8"?>
<!-- UniMRCP server document -->
<unimrcpserver xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
               xsi:noNamespaceSchemaLocation="unimrcpserver.xsd"
               version="1.0">
  <properties>
    <!--
      If the attribute "type" is set to "auto", IP address is determined implicitly by the hostname.
      This is the default setting.
    -->
    <ip type="auto"/>

    <!--
      If the attribute "type" is set to "iface", IP address is determined by the specified name of
      network interface/adapter.
    -->
    <!-- <ip type="iface">eth0</ip>-->

    <!--
      IP address can also be specified explicitly.
    -->
    <!-- <ip>10.10.0.1</ip> -->

    <!-- <ext-ip>a.b.c.d</ext-ip> -->
  </properties>

  <components>
    <!-- Factory of MRCP resources -->
    <resource-factory>
      <resource id="speechsynth" enable="true"/>
      <resource id="speechrecog" enable="true"/>
      <resource id="recorder" enable="true"/>
      <resource id="speakverify" enable="true"/>
    </resource-factory>

    <!-- SofiaSIP MRCPv2 signaling agent -->
    <sip-uas id="SIP-Agent-1" type="SofiaSIP">
      <!--
        By default, "ip" and "ext-ip" addresses, set in the properties, are used. These parameters can
        explicitly be specified per "sip-uas" by means of "sip-ip" and "sip-ext-ip" correspondingly.
      -->
      <!-- <sip-ip>10.10.0.1</sip-ip> -->
      <!-- <sip-ext-ip>a.b.c.d</sip-ext-ip> -->
      <sip-port>8060</sip-port>
      <sip-transport>udp,tcp</sip-transport>
      <!-- <force-destination>true</force-destination> -->
      <ua-name>UniMRCP SofiaSIP</ua-name>
      <sdp-origin>UniMRCPServer</sdp-origin>
      <!-- <sip-t1>500</sip-t1> -->
      <!-- <sip-t2>4000</sip-t2> -->
      <!-- <sip-t4>4000</sip-t4> -->
      <!-- <sip-t1x64>32000</sip-t1x64> -->
      <sip-session-expires>600</sip-session-expires>
      <sip-min-session-expires>120</sip-min-session-expires>
      <!-- <sip-message-output>true</sip-message-output> -->
      <!-- <sip-message-dump>sofia-sip-uas.log</sip-message-dump> -->
    </sip-uas>

    <!-- UniRTSP MRCPv1 signaling agent -->
    <rtsp-uas id="RTSP-Agent-1" type="UniRTSP">
      <!--
        By default, "ip" and "ext-ip" addresses, set in the properties, are used. These parameters can
        explicitly be specified per "rtsp-uas" by means of "rtsp-ip" and "rtsp-ext-ip" correspondingly.
      -->
      <!-- <rtsp-ip>10.10.0.1</rtsp-ip> -->
      <!-- <rtsp-ext-ip>a.b.c.d</rtsp-ext-ip> -->
      <rtsp-port>1554</rtsp-port>
      <!-- <force-destination>true</force-destination> -->
      <resource-map>
        <param name="speechsynth" value="speechsynthesizer"/>
        <param name="speechrecog" value="speechrecognizer"/>
      </resource-map>
      <max-connection-count>100</max-connection-count>
      <inactivity-timeout>600</inactivity-timeout>
      <sdp-origin>UniMRCPServer</sdp-origin>
    </rtsp-uas>

    <!-- MRCPv2 connection agent -->
    <mrcpv2-uas id="MRCPv2-Agent-1">
      <!--
        By default, "ip" address, set in the properties, is used. This parameter can explicitly be
        specified per "mrcpv2-uas" by means of "mrcp-ip".
      -->
      <!-- <mrcp-ip>10.10.0.1</mrcp-ip> -->
      <mrcp-port>1544</mrcp-port>
      <max-connection-count>100</max-connection-count>
      <max-shared-use-count>100</max-shared-use-count>
      <force-new-connection>false</force-new-connection>
      <rx-buffer-size>1024</rx-buffer-size>
      <tx-buffer-size>1024</tx-buffer-size>
      <inactivity-timeout>600</inactivity-timeout>
      <termination-timeout>3</termination-timeout>
    </mrcpv2-uas>

    <!-- Media processing engine -->
    <media-engine id="Media-Engine-1">
      <realtime-rate>1</realtime-rate>
    </media-engine>

    <!-- Factory of RTP terminations -->
    <rtp-factory id="RTP-Factory-1">
      <!--
        By default, "ip" and "ext-ip" addresses, set in the properties, are used. These parameters can
        explicitly be specified per "rtp-factory" by means of "rtp-ip" and "rtp-ext-ip" correspondingly.
      -->
      <!-- <rtp-ip>10.10.0.1</rtp-ip> -->
      <!-- <rtp-ext-ip>a.b.c.d</rtp-ext-ip> -->
      <rtp-port-min>5000</rtp-port-min>
      <rtp-port-max>6000</rtp-port-max>
    </rtp-factory>

    <!-- Factory of plugins (MRCP engines) -->
    <plugin-factory>
      <engine id="Demo-Synth-1" name="demosynth" enable="true"/>
      <engine id="Demo-Recog-1" name="demorecog" enable="false"/>
      <engine id="Demo-Verifier-1" name="demoverifier" enable="true"/>
      <engine id="Recorder-1" name="mrcprecorder" enable="true"/>

      <engine id="GSR-1" name="umsgsr" enable="true"/>

      <!--
        Engines may have additional named ("max-channel-count") and generic (name/value) parameters.
        For example:
      -->
      <!--
      <engine id="Your-Engine-1" name="yourengine" enable="false">
        <max-channel-count>100</max-channel-count>
        <param name="..." value="..."/>
      </engine>
      -->
    </plugin-factory>
  </components>

  <settings>
    <!-- RTP/RTCP settings -->
    <rtp-settings id="RTP-Settings-1">
      <jitter-buffer>
        <adaptive>1</adaptive>
        <playout-delay>50</playout-delay>
        <max-playout-delay>600</max-playout-delay>
        <time-skew-detection>1</time-skew-detection>
      </jitter-buffer>
      <ptime>20</ptime>
      <codecs own-preference="false">PCMU PCMA L16/96/8000 telephone-event/101/8000</codecs>
      <!-- <codecs own-preference="false">PCMU PCMA L16/96/8000 PCMU/97/16000 PCMA/98/16000 L16/99/16000</codecs> -->
      <!-- Enable/disable RTCP support -->
      <rtcp enable="false">
        <!--
          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)
        -->
        <rtcp-bye>1</rtcp-bye>
        <!-- RTCP transmission interval in msec (set 0 to disable) -->
        <tx-interval>5000</tx-interval>
        <!-- Period (timeout) to check for new RTCP messages in msec (set 0 to disable) -->
        <rx-resolution>1000</rx-resolution>
      </rtcp>
    </rtp-settings>
  </settings>

  <profiles>
    <!-- MRCPv2 default profile -->
    <mrcpv2-profile id="uni2">
      <sip-uas>SIP-Agent-1</sip-uas>
      <mrcpv2-uas>MRCPv2-Agent-1</mrcpv2-uas>
      <media-engine>Media-Engine-1</media-engine>
      <rtp-factory>RTP-Factory-1</rtp-factory>
      <rtp-settings>RTP-Settings-1</rtp-settings>

      <!--
        Profile-based association of engines to resources. For example:
      -->
      <!--
      <resource-engine-map>
        <param name="speechsynth" value="Flite-1"/>
        <param name="speechrecog" value="PocketSphinx-1"/>
      </resource-engine-map>
      -->
    </mrcpv2-profile>

    <!-- MRCPv1 default profile -->
    <mrcpv1-profile id="uni1">
      <rtsp-uas>RTSP-Agent-1</rtsp-uas>
      <media-engine>Media-Engine-1</media-engine>
      <rtp-factory>RTP-Factory-1</rtp-factory>
      <rtp-settings>RTP-Settings-1</rtp-settings>
    </mrcpv1-profile>

    <!-- More profiles may follow. -->
  </profiles>
</unimrcpserver>

FreeSwitch mrcp profile used:

<include>

  <!-- UniMRCP Server MRCPv2 -->

  <profile name="uni2" version="2">

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

    <param name="client-ip" value="10.4.1.151"/>

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

    <param name="server-ip" value="10.4.1.152"/>

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

    <param name="force-destination" value="true"/>

    <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="16384"/>

    <param name="rtp-port-max" value="32768"/>

    <!--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 -->

    <synthparams>

    </synthparams>

 

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

    <recogparams>

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

    </recogparams>

  </profile>

</include>

[Arsen Chaloyan]

Hello Stefano,

The following statement indicates there was no RTP packets received by the server, as you noted.

2017-12-15 11:53:31:755083 [INFO]   Close RTP Receiver 10.4.1.152:5008 <- 127.0.1.1:16384 [r:0 l:0 j:0 p:50 d:0 i:0]

The problem is in the RTP IP address used by by FreeSWITCH.

2017-12-15 11:53:23:545841 [INFO]   Remote SDP 0x7fee18001a48 <new>
v=0
o=FreeSWITCH 7101049191926696257 8524173389514529569 IN IP4 10.4.1.151
s=-
c=IN IP4 127.0.1.1
t=0 0
m=application 9 TCP/MRCPv2 1
a=setup:active
a=connection:new
a=resource:speechrecog
a=cmid:1
m=audio 16384 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 L16/8000
a=sendonly
a=mid:1

You can address the problem by setting the corresponding IP address in FS MRCP profile

Instead of

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

use

    <param name="rtp-ip" value="10.4.1.151"/>

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Arsen Chaloyan
Author of UniMRCP
http://www.unimrcp.org

[Stefano Favaro]

Thank you Arsen!

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