<?xml version="1.0" encoding="UTF-8"?> <alimrcp-server xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:noNamespaceSchemaLocation="alimrcp-server.xsd" version="1.0"> <properties> <!--<ip type="auto"/>--> <!-- <ip type="iface">eth0</ip>--> <ip>0.0.0.0</ip> <!-- <ext-ip>a.b.c.d</ext-ip> --> </properties> <components> <resource-factory> <resource id="speechsynth" enable="true"/> <resource id="speechrecog" enable="true"/> <resource id="recorder" enable="false"/> <resource id="speakverify" enable="false"/> </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>7010</sip-port> <sip-transport>udp,tcp</sip-transport> <!-- <force-destination>true</force-destination> --> <ua-name>UniMRCP SofiaSIP</ua-name> <sdp-origin>AliMrcpServer</sdp-origin> <!-- <sip-t1>500</sip-t1> --> <!-- <sip-t2>4000</sip-t2> --> <!-- <sip-t4>4000</sip-t4> --> <!-- <sip-t1x64>32000</sip-t1x64> --> <!-- default 600 = 5min, now 3 years-->> <sip-session-expires>189216000</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> --> <!-- <mrcp-draft11-version>true</mrcp-draft11-version> --> </sip-uas> <!-- UniRTSP MRCPv1 signaling agent --> <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> <inactivity-timeout>600</inactivity-timeout> <sdp-origin>AliMrcpServer</sdp-origin> </rtsp-uas> <!-- MRCPv2 connection agent --> <mrcpv2-uas id="MRCPv2-Agent-1"> <!-- <mrcp-ip>10.10.0.1</mrcp-ip> --> <mrcp-port>1544</mrcp-port> <max-connection-count>300</max-connection-count> <max-shared-use-count>100</max-shared-use-count> <force-new-connection>true</force-new-connection> <!-- 是否禁止连接复用,true表示禁止 --> <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 --> <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 --> <inactivity-timeout>600</inactivity-timeout> <termination-timeout>3</termination-timeout> </mrcpv2-uas> <media-engine id="Media-Engine-1"> <realtime-rate>1</realtime-rate> </media-engine> <!-- Factory of RTP terminations --> <rtp-factory id="RTP-Factory-1"> <rtp-port-min>10000</rtp-port-min> <rtp-port-max>20000</rtp-port-max> </rtp-factory> <!-- Factory of plugins (MRCP engines) --> <plugin-factory> <engine id="alimrcp-tts" name="alimrcp_tts" enable="true"> <max-channel-count>100</max-channel-count> <param name="sdk-log-level" value="5"/> <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) --> <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 --> <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 --> </engine> <engine id="alimrcp-asr" name="alimrcp_asr" enable="true"> <max-channel-count>100</max-channel-count> <param name="sdk-log-level" value="5"/> <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) --> <param name="text-encoding-gb2312" value="0"/> <param name="wav-uri-prefix" value=""/> <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 --> <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 --> <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 --> <!--<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, --> <param name="speech-cache-count" value="10"/> <!-- should be greater than 0, suggest [1,20] --> <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 --> <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 --> <param name="return-more-detail" value="1"/> <!-- 是否返回更多ASR相关信息, 默认返回 --> <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--> <param name="force-stop-recognize" value="1"/> <param name="xml-type" value="application/nlsml+xml"/> <param name="asr-result-template" value="../conf/asr_result_template.xml"/> <!--<param name="grammar-transformer" value="../script/RequestTransformer.lua"/>--> <param name="script-when-hangup" value="HangupNotifierDemo"/> <!-- python script --> <param name="script-function-when-hangup" value=""/> <!-- function of python script --> <param name="script-of-nlu" value="NluClientDemo"/> <!-- nlu python script --> <param name="script-function-of-nlu" value=""/> <!-- nlu function of python script --> </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> <!-- 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> </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> </profiles> </alimrcp-server>