171 lines
4.6 KiB
Markdown
171 lines
4.6 KiB
Markdown
# Freeswitch通过mod_unimrcp与百度mrcp-server对接（lua版）
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---
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## 参考链接
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[https://freeswitch.org/confluence/display/FREESWITCH/mod_unimrcp]
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[https://ptorch.com/news/206.html]
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[https://ptorch.com/news/207.html]
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## 安装并加载mod_unimrcp模块
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```bash
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# 在freeswitch源码目录（不是安装目录）
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make mod_unimrcp-install
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# 在freeswitch安装目录中编译modules.conf.xml文件
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cd /usr/local/freeswitch
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vim conf/autoload_configs/modules.conf.xml
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```
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```xml
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<!-- 添加如下配置 -->
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<load module="mod_unimrcp"/>
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```
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## 设置profile文件和conf文件
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```bash
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vim /usr/local/freeswitch/conf/mrcp_profiles/unimrcpserver-mrcp-v2.xml
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```
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输入以下内容：
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```xml
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<include>
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<!-- UniMRCP Server MRCPv2 -->
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<!-- 后面我们使用该配置文件，均使用 name 作为唯一标识，而不是文件名 -->
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<profile name="unimrcpserver-mrcp2" version="2">
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<!-- MRCP 服务器地址和SIP端口号 -->
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<param name="server-ip" value="192.168.16.4"/>
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<!-- mrcp服务器的sip-port -->
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<param name="server-port" value="15060"/>
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<param name="resource-location" value=""/>
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<!-- FreeSWITCH IP、端口以及 SIP 传输方式 -->
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<param name="client-ip" value="192.168.16.4" />
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<!-- freeswitch的sip-port -->
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<param name="client-port" value="5069"/>
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<param name="sip-transport" value="udp"/>
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<param name="speechsynth" value="speechsynthesizer"/>
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<param name="speechrecog" value="speechrecognizer"/>
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<!--param name="rtp-ext-ip" value="auto"/-->
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<!-- 也是freeswitch的ip和rtp端口范围（不是mrcp里配置的rtp范围） -->
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<param name="rtp-ip" value="192.168.16.4"/>
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<param name="rtp-port-min" value="4000"/>
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<param name="rtp-port-max" value="5000"/>
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<param name="codecs" value="PCMU PCMA L16/96/8000"/>
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<!-- Add any default MRCP params for SPEAK requests here -->
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<synthparams>
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</synthparams>
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<!-- Add any default MRCP params for RECOGNIZE requests here -->
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<recogparams>
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<!--param name="start-input-timers" value="false"/-->
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</recogparams>
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</profile>
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</include>
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```
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编辑`unimrcp.conf.xml`文件改`default-tts-profile`和`default-asr-profile`
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```bash
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vim /usr/local/freeswitch/conf/autoload_configs/unimrcp.conf.xml
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```
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```xml
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<!-- UniMRCP profile to use for TTS -->
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<param name="default-tts-profile" value="unimrcpserver-mrcp2"/>
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<!-- UniMRCP profile to use for ASR -->
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<param name="default-asr-profile" value="unimrcpserver-mrcp2"/>
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```
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## 设置dialplan
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```bash
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vim /usr/local/freeswitch/conf/dialplan/default.xml
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```
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添加一个extension：
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```xml
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<extension name="unimrcp">
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<condition field="destination_number" expression="^5001$">
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<action application="answer"/>
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<!-- 对应scripts/baidu.lua文件 -->
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<action application="lua" data="baidu.lua"/>
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</condition>
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</extension>
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```
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在`/usr/local/freeswitch/scripts`目录下创建`baidu.lua`文件：
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```bash
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touch /usr/local/freeswitch/scripts/baidu.lua
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vim /usr/local/freeswitch/scripts/baidu.lua
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```
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文件内容如下：
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```lua
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session:answer()
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--freeswitch.consoleLog("INFO", "Called extension is '".. argv[1]"'\n")
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welcome = "/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav"
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--
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grammar = "baidu"
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no_input_timeout = 80000
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recognition_timeout = 80000
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--
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tryagain = 1
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while (tryagain == 1) do
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--
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session:execute("play_and_detect_speech", welcome .. " detect:unimrcp {start-input-timers=false,no-input-timeout=" .. no_input_timeout .. ",recognition-timeout=" .. recognition_timeout .. "} " .. grammar)
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xml = session:getVariable('detect_speech_result')
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--
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if (xml == nil) then
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freeswitch.consoleLog("CRIT","Result is 'nil'\n")
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tryagain = 0
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else
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freeswitch.consoleLog("CRIT","Result is '" .. xml .. "'\n")
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tryagain = 0
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end
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end
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--
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-- put logic to forward call here
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--
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session:sleep(250)
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session:hangup()
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```
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> 以上脚本实现当分机用户拨打5001时，freeswitch会自动播放一段录音，并接收用户发出的声音，同时把声音传给mrcp服务器并接收返回结果
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在`/usr/local/freeswitch/grammar`目录新增`hello.gram`语法文件，内容为百度mrcp程序句中的语法文件内容：
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```xml
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<?xml version="1.0"?>
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<grammar xmlns="http://www.w3.org/2001/06/grammar" xml:lang="en-US" version="1.0" mode="voice" root="digit">
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<rule id="digit">
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<one-of>
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<item>one</item>
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<item>two</item>
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<item>three</item>
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</one-of>
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</rule>
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</grammar>
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```
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## 让配置生效并测试
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```bash
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fs_cli
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reloadxml
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```
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## 防火墙
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在freeswitch服务器和mrcp服务器都不需要额外开放端口。
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