160 lines
8.3 KiB
XML
Executable File
160 lines
8.3 KiB
XML
Executable File
<?xml version="1.0" encoding="UTF-8"?>
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<alimrcp-server xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:noNamespaceSchemaLocation="alimrcp-server.xsd" version="1.0">
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<properties>
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<!--<ip type="auto"/>-->
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<!-- <ip type="iface">eth0</ip>-->
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<ip>0.0.0.0</ip>
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<!-- <ext-ip>a.b.c.d</ext-ip> -->
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</properties>
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<components>
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<resource-factory>
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<resource id="speechsynth" enable="true"/>
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<resource id="speechrecog" enable="true"/>
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<resource id="recorder" enable="false"/>
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<resource id="speakverify" enable="false"/>
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</resource-factory>
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<!-- SofiaSIP MRCPv2 signaling agent -->
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<sip-uas id="SIP-Agent-1" type="SofiaSIP">
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<!-- By default, "ip" and "ext-ip" addresses, set in the properties, are used. These parameters can
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explicitly be specified per "sip-uas" by means of "sip-ip" and "sip-ext-ip" correspondingly. -->
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<!-- <sip-ip>10.10.0.1</sip-ip> -->
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<!-- <sip-ext-ip>a.b.c.d</sip-ext-ip> -->
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<sip-port>7010</sip-port>
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<sip-transport>udp,tcp</sip-transport>
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<!-- <force-destination>true</force-destination> -->
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<ua-name>UniMRCP SofiaSIP</ua-name>
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<sdp-origin>AliMrcpServer</sdp-origin>
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<!-- <sip-t1>500</sip-t1> -->
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<!-- <sip-t2>4000</sip-t2> -->
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<!-- <sip-t4>4000</sip-t4> -->
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<!-- <sip-t1x64>32000</sip-t1x64> -->
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<!-- default 600 = 5min, now 3 years-->>
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<sip-session-expires>189216000</sip-session-expires>
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<sip-min-session-expires>120</sip-min-session-expires>
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<!-- <sip-message-output>true</sip-message-output> -->
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<!-- <sip-message-dump>sofia-sip-uas.log</sip-message-dump> -->
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<!-- <mrcp-draft11-version>true</mrcp-draft11-version> -->
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</sip-uas>
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<!-- UniRTSP MRCPv1 signaling agent -->
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<rtsp-uas id="RTSP-Agent-1" type="UniRTSP">
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<rtsp-port>1554</rtsp-port>
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<resource-map>
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<param name="speechsynth" value="speechsynthesizer"/>
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<param name="speechrecog" value="speechrecognizer"/>
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</resource-map>
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<max-connection-count>100</max-connection-count>
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<inactivity-timeout>600</inactivity-timeout>
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<sdp-origin>AliMrcpServer</sdp-origin>
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</rtsp-uas>
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<!-- MRCPv2 connection agent -->
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<mrcpv2-uas id="MRCPv2-Agent-1">
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<!-- <mrcp-ip>10.10.0.1</mrcp-ip> -->
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<mrcp-port>1544</mrcp-port>
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<max-connection-count>300</max-connection-count>
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<max-shared-use-count>100</max-shared-use-count>
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<force-new-connection>true</force-new-connection> <!-- 是否禁止连接复用，true表示禁止 -->
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<rx-buffer-size>1024</rx-buffer-size> <!-- 表示接收到的MRCP消息的最大长度，超过则多次接收 -->
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<tx-buffer-size>1024</tx-buffer-size> <!-- 表示发送时的MRCP消息的最大长度，超过则多次发送 -->
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<inactivity-timeout>600</inactivity-timeout>
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<termination-timeout>3</termination-timeout>
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</mrcpv2-uas>
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<media-engine id="Media-Engine-1">
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<realtime-rate>1</realtime-rate>
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</media-engine>
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<!-- Factory of RTP terminations -->
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<rtp-factory id="RTP-Factory-1">
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<rtp-port-min>10000</rtp-port-min>
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<rtp-port-max>20000</rtp-port-max>
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</rtp-factory>
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<!-- Factory of plugins (MRCP engines) -->
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<plugin-factory>
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<engine id="alimrcp-tts" name="alimrcp_tts" enable="true">
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<max-channel-count>100</max-channel-count>
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<param name="sdk-log-level" value="5"/>
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<param name="sdm-metrics-host" value="0.0.0.0:7009"/> <!-- 默认监听到本机所有ip上的7009端口, 可修改或置空(不监控) -->
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<param name="tts-save-audio" value="0"/> <!-- 默认不保存tts合成的录音, 如需保存则改为1 -->
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<param name="text-encoding-gb2312" value="0"/> <!-- 表示接收到ivr的合成文本编码方式, 默认为0, 即utf8, 如果是gb2312需修改该参数为1 -->
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</engine>
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<engine id="alimrcp-asr" name="alimrcp_asr" enable="true">
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<max-channel-count>100</max-channel-count>
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<param name="sdk-log-level" value="5"/>
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<param name="sdm-metrics-host" value="0.0.0.0:7009"/> <!-- 默认监听到本机所有ip上的7009端口, 可修改或置空(不监控) -->
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<param name="text-encoding-gb2312" value="0"/>
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<param name="wav-uri-prefix" value=""/>
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<param name="recognize-mode-continuous" value="0"/> <!-- 无需修改, 如要修改，建议同时修改sip-session-expires为一个无限大的值 -->
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<param name="no-input-timeout" value="5000"/> <!-- 无话超时时间, 单位毫秒, 不可过低, 如果ivr传递了, 则以ivr传递的为准 -->
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<param name="speech-complete-timeout" value="800"/> <!-- vad尾点断句间隔, 单位毫秒, 范围[200,2000], 如果ivr传递了, 则以ivr传递的为准 -->
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<!--<param name="sensitivity-level" value="0.2"/>--> <!-- 噪音阈值参数、灵敏度参数，可以参考标准RFC协议，取值范围[-1,1]，等同asr的speech_noise_threshold， -->
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<param name="speech-cache-count" value="10"/> <!-- should be greater than 0, suggest [1,20] -->
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<param name="save-waveform" value="1"/> <!-- 是否保存用户录音，默认保存，但还受IVR传参影响 -->
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<param name="ignore-ivr-save-waveform" value="0"/> <!-- 是否忽略IVR传递的save-waveform参数，默认不忽略，若要忽略请置为1 -->
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<param name="return-more-detail" value="1"/> <!-- 是否返回更多ASR相关信息, 默认返回 -->
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<param name="support-message-body" value="0"/> <!-- 当no-input-timeout或者no-match等状态时，是否返回消息体，默认没有-->
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<param name="force-stop-recognize" value="1"/>
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<param name="xml-type" value="application/nlsml+xml"/>
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<param name="asr-result-template" value="../conf/asr_result_template.xml"/>
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<!--<param name="grammar-transformer" value="../script/RequestTransformer.lua"/>-->
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<param name="script-when-hangup" value="HangupNotifierDemo"/> <!-- python script -->
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<param name="script-function-when-hangup" value=""/> <!-- function of python script -->
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<param name="script-of-nlu" value="NluClientDemo"/> <!-- nlu python script -->
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<param name="script-function-of-nlu" value=""/> <!-- nlu function of python script -->
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</engine>
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</plugin-factory>
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</components>
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<settings>
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<!-- RTP/RTCP settings -->
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<rtp-settings id="RTP-Settings-1">
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<jitter-buffer>
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<adaptive>1</adaptive>
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<playout-delay>50</playout-delay>
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<max-playout-delay>600</max-playout-delay>
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<time-skew-detection>1</time-skew-detection>
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</jitter-buffer>
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<ptime>20</ptime>
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<codecs own-preference="false">PCMU PCMA L16/96/8000 telephone-event/101/8000</codecs>
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<!-- Enable/disable RTCP support -->
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<rtcp enable="false">
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<!--
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RTCP BYE policies (RTCP must be enabled first)
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0 - disable RTCP BYE
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1 - send RTCP BYE at the end of session
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2 - send RTCP BYE also at the end of each talkspurt (input)
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-->
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<rtcp-bye>1</rtcp-bye>
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<!-- RTCP transmission interval in msec (set 0 to disable) -->
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<tx-interval>5000</tx-interval>
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<!-- Period (timeout) to check for new RTCP messages in msec (set 0 to disable) -->
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<rx-resolution>1000</rx-resolution>
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</rtcp>
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</rtp-settings>
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</settings>
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<profiles>
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<!-- MRCPv2 default profile -->
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<mrcpv2-profile id="uni2">
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<sip-uas>SIP-Agent-1</sip-uas>
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<mrcpv2-uas>MRCPv2-Agent-1</mrcpv2-uas>
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<media-engine>Media-Engine-1</media-engine>
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<rtp-factory>RTP-Factory-1</rtp-factory>
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<rtp-settings>RTP-Settings-1</rtp-settings>
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</mrcpv2-profile>
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<!-- MRCPv1 default profile -->
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<mrcpv1-profile id="uni1">
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<rtsp-uas>RTSP-Agent-1</rtsp-uas>
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<media-engine>Media-Engine-1</media-engine>
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<rtp-factory>RTP-Factory-1</rtp-factory>
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<rtp-settings>RTP-Settings-1</rtp-settings>
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</mrcpv1-profile>
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</profiles>
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</alimrcp-server>
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