<?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>
<ip type="auto"/>
</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">
<sip-port>8060</sip-port>
<sip-transport>udp</sip-transport>
<ua-name>UniMRCP SofiaSIP</ua-name>
<sdp-origin>UniMRCPServer</sdp-origin>
</sip-uas>
<!-- UniRTSP MRCPv1 signaling agent -->
<rtsp-uas id="RTSP-Agent-1" type="UniRTSP">
<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>
<sdp-origin>UniMRCPServer</sdp-origin>
</rtsp-uas>
<!-- MRCPv2 connection agent -->
<mrcpv2-uas id="MRCPv2-Agent-1">
<mrcp-port>1554</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 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">
<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="true"/>
<engine id="Demo-Verifier-1" name="demoverifier" enable="true"/>
<engine id="Recorder-1" name="mrcprecorder" enable="true"/>
</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 8000</codecs>
<!-- enable/disable RTCP support -->
<rtcp enable="false">
<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>
<!-- more profiles might be added here -->
</profiles>
</unimrcpserver>
您好，
我正在尝试将VBVoice应用程序连接到用于TTS的Unimrcp服务器。应用程序将Invite成功发送到服务器，然后服务器回复100和200，但它们都转到错误的端口（5060而不是8060）。我在配置文件中遗漏了什么吗？
答案 0 :(得分：1)
VBVoice可以配置为更改用于MRCP连接的端口，因为它适用于VBVMRCPClient。要修改VBVoice MRCP客户端使用的端口，请打开Pronexus控制面板，然后访问VBVConfig实用程序。在VBVConfig的左侧访问MRCP部分。在这里，您将看到ASRServerPort和TTSServerPort的选项。默认端口为5060.您可以将其设置为任何可用的端口号。完成所需更改后，使用文件下拉列表选择＆＃34;保存所有密钥＆＃34;并关闭VBVConfig。下次启动VBVoice MRCP客户端时将应用配置更改。
值得注意 - 当VBVoice IVR运行VOIP电话协议时，通常需要更改MRCP端口号，因为VBVoice使用的默认VOIP端口为5060.
同样值得注意 - 验证运行ASR / TTS系统的语音服务器是否将使用TCP或UDP进行MRCP连接。默认情况下，VBVoice配置为使用TCP。这可以在MRCP部分的VBVConfig实用程序中修改，只需查找ASRServerPortIsTCP和TTSServerPortIsTCP选项。