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使用FreeSWITCH接受用户手机呼叫,通过UniMRCP Server集成讯飞开放平台(xfyun)插件将用户语音进行语音识别(ASR),并根据自定义业务逻辑调用语音合成(TTS),构建简单的端到端语音呼叫中心。

Batchfile 0.08% Makefile 1.27% M4 1.26% Shell 1.85% C 75.65% Perl 0.43% ActionScript 0.58% HTML 1.65% JavaScript 3.86% CMake 0.06% Smarty 0.27% QMake 0.01% C++ 12.30% CSS 0.13% Assembly 0.01% PHP 0.04% Python 0.38% Awk 0.05% XSLT 0.01% Java 0.13%

freeswitch_unimrcp_xfyun's Introduction

安装环境 ubuntu 16.04 1.下载 

freeswitch-1.6.5

2.安装依赖

apt-get install -y autoconf libtool g++ zlib1g-dev libjpeg-dev libsqlite3-dev libcurl4-openssl-dev libpcre3-dev libspeex-dev libspeexdsp-dev libldns-dev libedit-dev libssl-dev pkg-config yasm liblua50-dev libopus-dev libsndfile1-dev libpq-dev libreadline-dev lua5.2 lua5.2-doc liblua5.2-dev libtiff5 libtiff5-dev

(1)下载libyuv源码并编译

        cd freeswitch/libs         git clone https://freeswitch.org/stash/scm/sd/libyuv.git         cd libyuv         make -f linux.mk CXXFLAGS="-fPIC -O2 -fomit-frame-pointer -Iinclude/"         make install         cp /usr/lib/pkgconfig/libyuv.pc /usr/lib64/pkgconfig/

(2)下载libvpx源码并编译

        cd ..         git clone https://freeswitch.org/stash/scm/sd/libvpx.git         cd libvpx         ./configure --enable-pic --disable-static --enable-shared         make         make install         cp /usr/local/lib/pkgconfig/vpx.pc /usr/lib64/pkgconfig/

3.编译安装

cd freeswitch-1.6.5/ ./configure --prefix=/usr/local/freeswitch make make install make cd-sounds-install make cd-moh-install

4.运行

cd /usr/local/freeswitch/bin ./freeswitch

FreeSWITCH默认配置1000-1019(20个)用户,默认密码1234,您可以提前跳转到“第四步 测试与验证” 的验证步骤,登录并拨打5000,可以听到默认IVR的示例语音菜单指引。

第二步 配置编译UniMRCP Server 本次示例的UniMRCP Server在CentOS 7中进行源码编译安装,感谢由Github用户cotinyang提供的已经写好的集成讯飞SDK的UniMRCP Server源码。

1.下载 UniMRCP Server Plugin Demo 源码:

cd /opt git clone https://github.com/cotinyang/MRCP-Plugin-Demo.git MRCP-Plugin-Demo 2.编译准备环境

cd MRCP-Plugin-Demo/unimrcp-deps-1.5.0

编译可能出现错误, 注释掉:107 ~ 109, getopt的set,其中存在不识别的option

编译生成apr, apr-util, target path: ./libs

./build-dep-libs.sh 注:1.过程中需要输入两次y,并确认;2.另外,我们为该Demo工程Fork了一个自己维护的工程,地址为https://github.com/wangkaisine/MRCP-Plugin-Demo 您也可以使用这个地址的源码。

3.编译安装unimrcp

cd unimrcp-1.5.0 ./bootstrap

如果不能自动检测apr,apr-util,请在configure中增加 option:--with-apr=/path/apr --with-apr-util=/path/apr-util/

apr, apr-util由./build-dep-libs.sh 生成

./configure make make install 即可在/usr/local/中看到安装好的unimrcp。

4.测试运行

cd /usr/local/unimrcp/bin ./unimrcpserver -o 3 可以使用client进行验证

cd /usr/local/unimrcp/bin ./unimrcpclient

help usage:

  • run [app_name] [profile_name] (run demo application) app_name is one of 'synth', 'recog', 'bypass', 'discover' profile_name is one of 'uni2', 'uni1', ...

     examples: 
         run synth
         run recog
         run synth uni1
         run recog uni1
    
  • loglevel [level] (set loglevel, one of 0,1...7)

  • quit, exit 输入help回车,给出了使用方法,输入run recog运行语音识别测试,run synth进行语音合成测试。

第三步 集成讯飞开放平台SDK1.讯飞开发平台SDK下载 由于从讯飞开放平台下载的SDK包和用户以及用户创建的应用相关联,因此需要将third-party/xfyun中的文件和文件夹全部删除,重新下载解压属于自己的SDK,目录与源代码基本一致。 您需要注册并登录 讯飞开放平台 ,进入控制台页面,并创建应用; 在“我的应用”界面获得你的APPID,并为该应用“添加新服务”,选择需要的“语音听写”和”在线语音合成“服务(本示例需要); 点击右侧“SDK下载”,在跳转页面中确认“选择应用”已经选中了您创建的应用,“选择您需要的AI能力”选中上述两项服务,并点击“SDK下载”等待SDK生成与完成下载。 将下载的zip包,解压并替换MRCP-Plugin-Demo/unimrcp-1.5.0/plugins/third-party/xfyun/下的所有文件及文件夹。 注:创建应用页面中的应用平台选择“Linux”。 2.plugin编写与编译 本步骤将告诉您如何编写unimrcp的插件代码,即现在MRCP-Plugin-Demo/unimrcp-1.5.0/plugins文件夹下xfyun_recog、xfyun_xynth文件夹下的文件及其相关配置是如何得到的,如果您当前还不关注此细节,可以跳过本步骤至第四步。 实际上,上述MRCP-Plugin-Demo代码是在 Unimrcp官网 下载 Unimrcp 1.5.0 和 Unimrcp Deps 1.5.0 并在此基础上添加的plugin代码。 首先编辑MRCP-Plugin-Demo/unimrcp-1.5.0/configure.ac文件,会在后面的Makefile中使用到的宏定义。XFyun recognizer plugin的添加如下:

dnl XFyun recognizer plugin. UNI_PLUGIN_ENABLED(xfyunrecog)

AM_CONDITIONAL([XFYUNRECOG_PLUGIN],[test "${enable_xfyunrecog_plugin}" = "yes"])

...

plugins/xfyun-recog/Makefile

...

echo XFyun recognizer plugin....... : $enable_xfyunrecog_plugin 注:其中 ··· 是该文件中的其它默认配置,请找到对应位置填写。

新增源码与目录 在 MRCP-Plugin-Demo/unimrcp-1.5.0/plugins 目录下,新建 xfyun-recog 目录,并在该目录下新建 src 目录,可以将 MRCP-Plugin-Demo/unimrcp-1.5.0/plugins/demo-recog/src/demo_recog_engine.cdemo_recog_engine.c 拷贝到MRCP-Plugin-Demo/unimrcp-1.5.0/plugins/xfyun-recog/src目录下改名为 xfyun_recog_engine.c,将xfyun_recog_engine.c文件进行修改(已知一个修改的部分:将appid修改成你自己下载sdk的appid,不然会报错:QISRAudioWrite failed! error code:10407),xfyun-synth目录下对应创建并修改。 在xfyun-recog文件夹下新建Makefile.am文件,内容如下:

AM_CPPFLAGS = $(UNIMRCP_PLUGIN_INCLUDES)

plugin_LTLIBRARIES = xfyunrecog.la

xfyunrecog_la_SOURCES = src/xfyun_recog_engine.c xfyunrecog_la_LDFLAGS = $(UNIMRCP_PLUGIN_OPTS)
-L$(top_srcdir)/plugins/third-party/xfyun/libs/x64
-lmsc -ldl -lpthread -lrt -lstdc++ xfyunrecog_ladir = $(libdir) xfyunrecog_la_DATA = $(top_srcdir)/plugins/third-party/xfyun/libs/x64/libmsc.so

include $(top_srcdir)/build/rules/uniplugin.am

UNIMRCP_PLUGIN_INCLUDES += -I$(top_srcdir)/plugins/third-party/xfyun/include 对应地,在xfyun-synth文件夹下新建Makefile.am文件夹,内容如下:

AM_CPPFLAGS = $(UNIMRCP_PLUGIN_INCLUDES)

plugin_LTLIBRARIES = xfyunsynth.la

xfyunsynth_la_SOURCES = src/xfyun_synth_engine.c xfyunsynth_la_LDFLAGS = $(UNIMRCP_PLUGIN_OPTS)
-L$(top_srcdir)/plugins/third-party/xfyun/libs/x64
-lmsc -ldl -lpthread -lrt xfyunsynth_ladir = $(libdir)

include $(top_srcdir)/build/rules/uniplugin.am

UNIMRCP_PLUGIN_INCLUDES += -I$(top_srcdir)/plugins/third-party/xfyun/include 修改plugins文件夹下Makefile.am文件,xfyun-recog添加内容如下:

if XFYUNRECOG_PLUGIN SUBDIRS += xfyun-recog endif 对应地,xfyun-synth添加内容如下:

if XFYUNRECOG_PLUGIN SUBDIRS += xfyun-synth endif 修改/usr/local/unimrcp/conf/unimrcpserver.xml文件,从默认启用demo engine改为启用我们的两个engine。 xfyun-recog修改如下:

对应地,xfyun-synth修改如下: 同时,如果您已经准备好将UniMRCP Server和FreeSWITCH对接,您应该在conf/unimrcpserver.xml中配置好server的ip地址,即当前unimrcp安装的子网访问地址。 重新编译安装unimrcp(第二步 3)。 当你启动时出现如下问题时: 1. Failed to Load DSO: /usr/local/unimrcp/lib/libmsc.so: undefined symbol: _ZTVN10__cxxabiv117__class_type_infoE

fix:-lstdc++ 1. ./unimrcpserver: error while loading shared libraries: libsofia-sip-ua.so.0: cannot open shared object file: No such file or directory

fix:

在etc/ld.so.conf 内容增加: /usr/local/lib ldconfig 将ld.so.conf读入cache

第四步 配置与验证配置配置FreeSWITCH 我们需要将处理用户语音呼入的FreeSWITCH与向xfyun engine发请求的unimrcp server两者连接起来。 1.配置unimrcp模块并自动加载;

编辑freeswitch-1.6.5/modules.conf文件,找到要安装的模块,去掉前面的注释符号#

vim modules.conf #asr_tts/mod_unimrcp asr_tts/mod_unimrcp

执行make mod_xxx-install命令,这样就编译相应模块,并把编译后的动态库安装的/usr/local/freeswitch/mod目录下

make mod_unimrcp-install

编辑/usr/local/freeswitch/etc/freeswitch/autoload_configs/modules.conf.xml,去掉注释符号,如果没有发现对应模块,则添加

2.设置profile文件与conf文件; 在/usr/local/freeswitch/etc/freeswitch/mrcp_profiles目录新建unimrcpserver-mrcp-v2.xml配置文件:

配置/usr/local/freeswitch/etc/freeswitch/autoload_configs/unimrcp.conf.xml文件: 注:1.unimrcpserver-mrcp-v2.xml中server-ip为unimrcpserver启动的主机ip;2.client-ip和rtp-ip为FreeSWITCH启动的主机,client-port仕FreeSWITCH作为客户端访问unimrcpserver的端口,手机作为客户端访问的FreeSWITCH端口默认为5060,两者不同;3.unimrcpserver-mrcp-v2.xml中的profile name应和unimrcp.conf.xml中的default-tts-profile与default-asr-profile的value一致(有些文档的分析中称mrcp_profiles中的xml文件名也必须和这两者一致,实际上是非必须的)。 Attenion: unimrcpserver 和 freeswitch 部署在同一个网段很重要,最好部署测试的时候在同一台物理机器上进行 3.配置IVR与脚本。 在/usr/local/freeswitch/etc/freeswitch/dialplan里新增如下配置: 在/usr/local/freeswitch/share/freeswitch/scripts目录下新增names.lua脚本:

session:answer()

--freeswitch.consoleLog("INFO", "Called extension is '".. argv[1]"'\n") welcome = "ivr/ivr-welcome_to_freeswitch.wav" menu = "ivr/ivr-this_ivr_will_let_you_test_features.wav"-- grammar = "hello" no_input_timeout = 80000 recognition_timeout = 80000 confidence_threshold = 0.2-- session:streamFile(welcome) --freeswitch.consoleLog("INFO", "Prompt file is \n")

tryagain = 1 while (tryagain == 1) do

session:execute("play_and_detect_speech",menu .. "detect:unimrcp {start-input-timers=false,no-input-timeout=" .. no_input_timeout .. ",recognition-timeout=" .. recognition_timeout .. "}" .. grammar) xml = session:getVariable('detect_speech_result')

if (xml == nil) then freeswitch.consoleLog("CRIT","Result is 'nil'\n") tryagain = 0 else freeswitch.consoleLog("CRIT","Result is '" .. xml .. "'\n") tryagain = 0 endend

-- put logic to forward call here

session:sleep(250) session:set_tts_params("unimrcp", "xiaofang"); session:speak("今天天气不错啊"); session:hangup() 我们需要在/usr/local/freeswitch/share/freeswitch/grammar目录新增hello.gram语法文件,可以为空语法文件须满足语音识别语法规范1.0标准(简称 SRGS1.0),该语法文件 ASR 引擎在进行识别时可以使用。

<grammar version="1.0" xml:lang="zh-cn" root="Menu" tag-format="semantics/1.0"     xmlns=http://www.w3.org/2001/06/grammar     xmlns:sapi="http://schemas.microsoft.com/Speech/2002/06/SRGSExtensions"><!- 这些都是必不可少的--> 北京 上海 到 至 飞往 out.date = reles.latest(); 从 out.city = rulels.latest(); out.cross = rulels.latest(); out.city = rulels.latest(); 注:lua脚本中,”play_and_detect_speech” 调用了 ASR 服务,”speak” 调用了 TTS 服务。 通过lua脚本实现话术功能

freeswitch 呼叫用户 进入 /usr/local/freeswitch/bin 目录, ./fs_cli 启动 fs_cli originate user/1004 &echo 呼叫1004用户 originate user/1000 &lua(names.lua) 呼叫1004用户,接通后转到lua脚本,lua脚本默认存放在 /usr/local/freeswitch/share/freeswitch/scripts 目录下

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