UniMRCP with Flite and Asterisk

[viewb...@gmail.com]

I have installed Asterisk 13.9.1 with Flite (and pocketsphinx) and UniMRCP with the Asterisk installs. The computer on which UniMRCP is installed has ip address 192.168.187 and 192.168.188. Asterisk is binded to 192.168.1.188. With module show like res_speech_unimrcp.so and module show like app_unimrcp.so both modules are loaded properly in Asterisk. When I put MRCPSynth(this is a test,p=tts-flite&l=en-US) in the dialplan, I get the following messages:
app_mrcpsynth.c:217 speech_on_channel_add: (TTS-0) channel error status=2, response code-503

THIS IS RES-SPEECH-UNIMRCP.CONF IN ASTERISK

[general]
unimrcp-profile = uni2 ; UniMRCP MRCPv2 Server
log-level = DEBUG ;unimrcp loggong level

[grammars]
; grammar-name = path-to-grammar-file
; MRCPv2 properties (recognizer and generic header fields)
; http://tools.ietf.org/html/draft-ietf-speechsc-mrcpv2-20#section-9.4

[mrcpv2-properties]
Recognition-Timeout = 20000
No-Input-Timeout = 15000
; MRCPv1 properties (recognizer and generic header fields)
; http://tools.ietf.org/html/rfc4463#section-8.4

[mrcpv1-properties]
Recognition-Timeout = 20000
No-Input-Timeout = 15000

THIS IS MRCP.CONF IN ASTERISK

[general]
default-asr-profile = asr-pocketsphinx
default-tts-profile = tts-flite
log-level = DEBUG lUniMRCP to appear in Asterisk logs
max-connection-count = 100
offer-new-connection = 1
; rx-buffer-size = 1024
; tx-buffer-size = 1024
; request-timeout = 5000

[asr-pocketsphinx]
version = 2 ;mrcp server
server-ip = 192.168.1.187 ;mrcp server ip address
server-port = 6060 ;mrcp server sip port
; server-username = test
force-destination = 1
client-ip = 192.168.1.188 ;mrcp client ip address
; client-ext-ip = auto
client-port = 6060 ;mrcp client sip port
sip-transport = udp ;either udp or tcp
; ua-name = Asterisk
; sdp-origin = Asterisk
rtp-ip = 192.168.1.188 ;mrcp client ip address
; rtp-ext-ip = auto
rtp-port-min = 9000 ;mrcp client lower port range
rtp-port-max = 9010 ;mrcp client upper port range
playout-delay = 50
; min-playout-delay = 20
max-playout-delay = 200
ptime = 20
codecs = PCMU PCMA L16/96/8000 telephone-event/101/8000
rtcp = 1
rtcp-bye = 2
rtcp-tx-interval = 5000
rtcp-rx-resolution = 1000

[tts-flite]
version = 2 ;mrcp server
server-ip = 192.168.1.187 ;mrcp server ip address
server-port = 6060 ;mrcp server sip port
; server-username = test
force-destination = 1
client-ip = 192.168.1.188 ;mrcp client ip address
; client-ext-ip = auto
client-port = 6060 ;mrcp client sip port
sip-transport = udp ;either udp or tcp
; ua-name = Asterisk
; sdp-origin = Asterisk
rtp-ip = 192.168.1.188 ;mrcp client ip address
; rtp-ext-ip = auto
rtp-port-min = 9000 ;mrcp client lower port range
rtp-port-max = 9010 ;mrcp client upper port range
playout-delay = 50
; min-playout-delay = 20
max-playout-delay = 200
ptime = 20
codecs = PCMU PCMA L16/96/8000 telephone-event/101/8000
rtcp = 1
rtcp-bye = 2
rtcp-tx-interval = 5000
rtcp-rx-resolution = 1000

THIS IS UNIMRCP.XML

<?xml version="1.0" encoding="UTF-8"?>
<unimrcpclient xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
xsi:noNamespaceSchemaLocation="../unimrcpclient.xsd"
version="1.0">
<settings>
<sip-settings id="UniMRCP-SIP-Settings">
<server-ip>192.168.1.187</server-ip>
<server-port>6060</server-port>
</sip-settings>
<rtsp-settings id="UniMRCP-RTSP-Settings">
<server-port>1554</server-port>
<resource-location>media</resource-location>
<resource-map>
<param name="speechsynth" value="speechsynthesizer"/>
<param name="speechrecog" value="speechrecognizer"/>
</resource-map>
</rtsp-settings>
</settings>
<profiles>
<mrcpv2-profile id="uni2">
<sip-uac>SIP-Agent-1</sip-uac>
<mrcpv2-uac>MRCPv2-Agent-1</mrcpv2-uac>
<media-engine>Media-Engine-1</media-engine>
<rtp-factory>RTP-Factory-1</rtp-factory>
<sip-settings>UniMRCP-SIP-Settings</sip-settings>
<rtp-settings>RTP-Settings-1</rtp-settings>
</mrcpv2-profile>
<mrcpv1-profile id="uni1">
<rtsp-uac>RTSP-Agent-1</rtsp-uac>
<media-engine>Media-Engine-1</media-engine>
<rtp-factory>RTP-Factory-1</rtp-factory>
<rtsp-settings>UniMRCP-RTSP-Settings</rtsp-settings>
<rtp-settings>RTP-Settings-1</rtp-settings>
</mrcpv1-profile>
</profiles>
</unimrcpclient>

THIS IS UNIMRCPCLIENT.XML

<?xml version="1.0" encoding="UTF-8"?>
<unimrcpclient xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
xsi:noNamespaceSchemaLocation="unimrcpclient.xsd"
version="1.0"
subfolder="client-profiles">
<properties>
<ip type="auto"/>
<ip>192.168.1.188</ip>
<ext-ip>192.168.1.188</ext-ip>
<server-ip>192.168.1.187</server-ip>
</properties>
<components>
<resource-factory>
<resource id="speechsynth" enable="true"/>
<resource id="speechrecog" enable="true"/>
<resource id="recorder" enable="true"/>
<resource id="speakverify" enable="true"/>
</resource-factory>
<sip-uac id="SIP-Agent-1" type="SofiaSIP">
<sip-port>6060</sip-port>
<sip-transport>udp</sip-transport>
<ua-name>UniMRCP SofiaSIP</ua-name>
<sdp-origin>UniMRCPClient</sdp-origin>
</sip-uac>
<rtsp-uac id="RTSP-Agent-1" type="UniRTSP">
<max-connection-count>100</max-connection-count>
<sdp-origin>UniMRCPClient</sdp-origin>
</rtsp-uac>
<mrcpv2-uac id="MRCPv2-Agent-1">
<max-connection-count>100</max-connection-count>
<offer-new-connection>false</offer-new-connection>
<rx-buffer-size>1024</rx-buffer-size>
<tx-buffer-size>1024</tx-buffer-size>
</mrcpv2-uac>
<media-engine id="Media-Engine-1">
<realtime-rate>1</realtime-rate>
</media-engine>
<rtp-factory id="RTP-Factory-1">
<rtp-port-min>5000</rtp-port-min>
<rtp-port-max>6000</rtp-port-max>
</rtp-factory>
</components>
<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>PCMU PCMA L16/96/8000 telephone-event/101/8000</codecs>
<rtcp enable="false">
<rtcp-bye>1</rtcp-bye>
<tx-interval>5000</tx-interval>
<rx-resolution>1000</rx-resolution>
</rtcp>
</rtp-settings>
</settings>
</unimrcpclient>

THIS IS UNIMRCPSERVER.XML

<?xml version="1.0" encoding="UTF-8"?>
<unimrcpserver xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:noNamespaceSchemaLocation="unimrcpserver.xsd" version="1.0">
<properties>
<ip type="auto"/>
<ip>192.168.1.187</ip>
<ext-ip>192.168.1.187</ext-ip>
</properties>
<components>
<resource-factory>
<resource id="speechsynth" enable="true"/>
<resource id="speechrecog" enable="true"/>
<resource id="recorder" enable="true"/>
<resource id="speakverify" enable="true"/>
</resource-factory>
<sip-uas id="SIP-Agent-1" type="SofiaSIP">
<sip-port>6060</sip-port>
<sip-transport>udp,tcp</sip-transport>
<ua-name>UniMRCP SofiaSIP</ua-name>
<sdp-origin>UniMRCPServer</sdp-origin>
</sip-uas>
<rtsp-uas id="RTSP-Agent-1" type="UniRTSP">
<rtsp-port>1554</rtsp-port>
<resource-map>
<param name="speechsynth" value="speechsynthesizer"/>
<param name="speechrecog" value="speechrecognizer"/>
</resource-map>
<max-connection-count>100</max-connection-count>
<sdp-origin>UniMRCPServer</sdp-origin>
</rtsp-uas>
<mrcpv2-uas id="MRCPv2-Agent-1">
<mrcp-port>1544</mrcp-port>
<max-connection-count>100</max-connection-count>
<force-new-connection>false</force-new-connection>
<rx-buffer-size>1024</rx-buffer-size>
<tx-buffer-size>1024</tx-buffer-size>
</mrcpv2-uas>
<media-engine id="Media-Engine-1">
<realtime-rate>1</realtime-rate>
</media-engine>
<rtp-factory id="RTP-Factory-1">
<rtp-port-min>5000</rtp-port-min>
<rtp-port-max>6000</rtp-port-max>
</rtp-factory>
<plugin-factory>
<engine id="Flite-1" name="mrcpflite" enable="true"/>
<engine id="PocketSphinx-1" name="mrcppocketsphinx" enable="true"/>
<engine id="Demo-Synth-1" name="demosynth" enable="true"/>
<engine id="Demo-Recog-1" name="demorecog" enable="true"/>
<engine id="Demo-Verifier-1" name="demoverifier" enable="true"/>
<engine id="Recorder-1" name="mrcprecorder" enable="true"/>
</plugin-factory>
</components>
<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>
<rtcp enable="false">
<rtcp-bye>1</rtcp-bye>
<tx-interval>5000</tx-interval>
<rx-resolution>1000</rx-resolution>
</rtcp>
</rtp-settings>
</settings>
<profiles>
<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>
</mrcpv2-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>
</profiles>
</unimrcpserver>

[Arsen Chaloyan]

The configuration seems correct. You may need to check the server side logs to see where 503 comes from.

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