SIP UserAgent (B2BUA client)——pjsip

1. sip stacks

pjsip/bell-sip/sofia-sip/libeXosip/libre

https://github.com/staskobzar/sip_stacks_examples

2. sip user agent and server network architecture

3. Installing pjsip on Ubuntu

https://www.pjsip.org

sudo apt-get install libasound2-dev

Linux system has two audio drivers: alsa and oss，oss is old，pjsip supports both，deflaut is oss.

如果目标系统内核使用的是alsa驱动，运行例子程序的时候会出现以下问题。

alsa_dev.c ! ca_thread_func: error reading data!   解决方法 修改config_site.h文件 在pjproject\pjlib\include\pj目录下

add #define PJMEDIA_AUDIO_DEV_HAS_ALSA 1 

4. cross-compile 

_install.sh

#compiler setup
if [[ $1 = 'arm' ]]; then
compiler=`CC=arm-linux-gnueabihf-gcc CXX=arm-linux-gnueabihf-g++`
host=arm-linux-gnueabihf
elif [[ $1 = 'himix100' ]]; then
compiler=`CC=arm-himix100-linux-gcc CXX=arm-himix100-linux-g++`
host=arm-himix100-linux
elif [[ $1 = 'himix200' ]]; then
compiler=`CC=arm-himix200-linux-gcc CXX=arm-himix200-linux-g++`
host=arm-himix200-linux
else
:
fi

#Preinstalled directory
install=${HOME}/_install

#Delete installed directory
rm -rf ${install}

#library path
pkg_config=${install}/lib/pkgconfig
lib_config=`CPPFLAGS=-I${install}/include CFLAGS=-I${install}/include LDFLAGS=-L${install}/lib`

#Delete compiled directory
for i in `ls .`
    do
    if [ -d $i ];then
    rm -rf $i
    fi
done

tar xvf alsa-lib-*
cd alsa-lib-*
./configure ${compiler} --prefix=${install} --host=${host}
make && make install
cd ..

tar xvf alsa-utils-*
cd alsa-utils-*
./configure ${compiler} --prefix=${install} --host=${host} PKG_CONFIG_PATH=${pkg_config} ${lib_config} --enable-static --enable-shared --disable-alsamixer --disable-xmlto
touch ./alsaconf/po/t-ja.gmo
touch ./alsaconf/po/t-ru.gmo
make && make install
cd ..

tar xvf pjproject-*
cd pjproject-*
./configure ${compiler} --prefix=${install} --host=${host} PKG_CONFIG_PATH=${pkg_config} ${lib_config} --enable-static --enable-shared --disable-libwebrtc
make dep
make && make install
cd ..

export PATH=$PATH:${install}/bin
export LD_LIBRARY_PATH=${install}/lib:$LD_LIBRARY_PATH
export ALSA_CONFIG_PATH=${install}/share/alsa/alsa.conf

dong@ubuntu:~/pjsip$ su
Password:
root@ubuntu:/home/dong/pjsip# ./_install.sh arm
root@ubuntu:~# tar cvf _install.tar.gz _install/
root@ubuntu:~# mv _install.tar.gz /home/dong/
dong@ubuntu:~$ sudo chmod 777 _install.tar.gz

4）如果用的海思arm-himix200-linux-gcc编译器，如果报错的话，可能需要将

alsa-utils-1.1.9/alsamixer/alsa-utils-1.1.9/alsamixer/volume_mapping.c里的三行exp10改成exp

5）声卡设备验证

#查看音频设备

dong@dong-ubuntu:~$ arecord -l
**** CAPTURE 硬體裝置清單 ****
card 0: PCH [HDA Intel PCH], device 0: ALC255 Analog [ALC255 Analog]
  子设备: 1/1
  子设备 #0: subdevice #0

#列举出机器所有的声卡
dong@dong-ubuntu:~$ cat /proc/asound/cards
 0 [PCH            ]: HDA-Intel - HDA Intel PCH
                      HDA Intel PCH at 0xa4220000 irq 132

#列举每个声卡的card number和device number
dong@dong-ubuntu:~$ aplay -l
**** PLAYBACK 硬體裝置清單 ****
card 0: PCH [HDA Intel PCH], device 0: ALC255 Analog [ALC255 Analog]
  子设备: 1/1
  子设备 #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0]
  子设备: 1/1
  子设备 #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1]
  子设备: 1/1
  子设备 #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2]
  子设备: 1/1
  子设备 #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3]
  子设备: 1/1
  子设备 #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 10: HDMI 4 [HDMI 4]
  子设备: 1/1
  子设备 #0: subdevice #0

#录音
dong@dong-ubuntu:~$ arecord -D "plughw:0,0" -f S16_LE -r 16000 -d 5 -t wav file.wav
正在录音 WAVE 'file.wav' : Signed 16 bit Little Endian, 频率16000Hz， Mono

#播放录音文件
dong@dong-ubuntu:~$ aplay file.wav
正在播放 WAVE 'file.wav' : Signed 16 bit Little Endian, 频率16000Hz， Mono

一般alsa设置了一个defaults设备，音频播放软件默认使用defaults设备输出声音。defaults设备定义在alsa.conf中，内容如下：

[plain]  
#  
# defaults  
#  
  
# show all name hints also for definitions without hint {} section  
defaults.namehint.showall off  
# show just basic name hints  
defaults.namehint.basic on  
# show extended name hints  
defaults.namehint.extended off  
#  
defaults.ctl.card 0  
defaults.pcm.card 0  
defaults.pcm.device 0  
defaults.pcm.subdevice -1  
……   

defaults会默认匹配card number和device number比较小的声卡。
如果要修改，则修改/etc/asound.conf或~/.asoundrc。比如我要把defaults匹配到card 1,device 3上，则添加一下几行：

[plain]  
$ sudo vim /etc/asound.conf   
defaults.pcm.card 1  
defaults.pcm.device 3  
defaults.ctl.card 1   

5) 添加声卡节点

if [ ! -d /dev/snd/ ]; then
    echo "make snd node."
    mkdir /dev/snd
    ln -s /dev/pcmC0D0c  /dev/snd/pcmC0D0c
    ln -s /dev/pcmC0D0p /dev/snd/pcmC0D0p
    ln -s /dev/timer /dev/snd/timer
    ln -s /dev/controlC0 /dev/snd/controlC0
    ln -s /dev/mixer /dev/snd/mixer
fi

5) 录音播音测试
arecord -D "plughw:0,0" -f S16_LE -r 16000 -d 5 -t wav file.wav
aplay file.wav

arecord -d 10 -f cd -r 8000 -c 2 -t wav test.wav
arecord -D "plughw:0,0" -f U16_LE -r 16000 -d 5 -t wav file.wav
arecord -d 10 -t raw -f S16_LE test.pcm
aplay -t raw -c 1 -f S16_LE -r 8000 test.pcm
arecord -d 10 -t raw -f U16_LE test.pcm
aplay -t raw -c 1 -f U16_LE -r 8000 test.pcm
arecord -d 10 -t raw -f U8 test.pcm
aplay -t raw -c 1 -f U8 -r 8000 test.pcm
aplay -t raw -r 8000 test.g711
arecord -D "plughw:0,0" -f U16_LE -r 16000 -d 5 -t wav file.wav
arecord -f S16_LE | aplay -f S16_LE

3、pjsua应用实例

pjsua Manual Page

https://www.pjsip.org/pjsua.htm

1) register --- make call
./pjsua-x86_64-unknown-linux-gnu --config-file paul.cfg
m
sip:alice@120.78.180.208:5060

2) peer to peer
./pjsua-x86_64-unknown-linux-gnu --null-audio
./pjsua-x86_64-unknown-linux-gnu sip:120.78.180.208 --null-audio

3) register cli
>>> +a
Your SIP URL: (empty to cancel):sip:100@120.78.180.208:5060
URL of the registrar: (empty to cancel):sip:120.78.180.208:5060
Auth Realm: (empty to cancel): *
Auth Username: (empty to cancel): 100
Auth Password: (empty to cancel): 100
 

sip:104@120.78.180.208

4) alice.cfg

# This is a comment in the config file.
--id sip:alice@xx.xxx.xx.236:5060
--registrar sip:xx.xxx.xx.236:5060
--realm *
--username alice
--password secret

启动程序，自动应答

./pjsua-arm-unknown-linux-gnueabihf --config-file alice.cfg --auto-answer 200

指定“紧凑模式”（去掉SIP协议的一些非必须的字段），采样率和编码优先级，我在cortex-a5上以默认参数启动，会出现pb和ca线程不同步underrun！

./pjsua-arm-unknown-linux-gnueabihf --config-file alice.cfg --use-compact-form --clock-rate=8000 --add-codec=pcmu --auto-answer 200

其他配置参考官网应用手册https://www.pjsip.org/pjsua.htm

5) 临时建群，多方通话

查看pjsua应用手册https://www.pjsip.org/pjsua.htm

的Call Commands和Conference Commands

cl查看已拨号的通道， 0是拨号的主叫号码本身，1和2分别是ringback和ring，3和4是被叫号码

M

2

sip:101@xx.xxx.xx.236:5060

sip:101@xx.xxx.xx.236:5060

cc 3 4

cc 4 3

6) freepbx语音会议

配置好freepbx语音会议后，直接拨号会议号，然后输入密码进入会议室

./pjsua-x86_64-unknown-linux-gnu --config-file 106.cfg --use-compact-form --clock-rate=8000 --add-codec=pcmu --auto-answer 200

也可以自动登陆会议室，auto-conf是自动登陆会议室

./pjsua-x86_64-unknown-linux-gnu --config-file 106.cfg --use-compact-form --clock-rate=8000 --add-codec=pcmu --auto-answer 200 --auto-conf sip:2018@xx.xxx.xx.236:5060

PC/android/ios也是一样的操作

4、 py_pjsua应用实例

https://trac.pjsip.org/repos/wiki/Py_PJSUA

python实例在pjsip-apps/src/python/samples/

5、Python_SIP编译安装
1) ./configure CFLAGS='-O2 -fPIC' --enable-shared
https://trac.pjsip.org/repos/wiki/Python_SIP/Build_Install

 2) pjsua2安装

sudo apt-get install -y software-properties-common # To make add-apt-repository work
sudo add-apt-repository ppa:dennis.guse/sip-tools 
sudo apt-get update 
sudo apt-get install python-pjsua
sudo apt-get install python-pjsua2

To check if everything went well:

python
import pjsua 
https://stackoverflow.com/questions/20195542/python-how-to-get-the-import-pjsua-giving-no-module-named-pjsua

3）Building PJSUA2 and SWIG

The PJSUA2 C++ library will be built by default by PJSIP build system. Standard C++ library is required. If you intend to use Python SWIG module (see below), you need to configure PJSIP with --enable-shared option, i.e.:

./configure --enable-shared
make dep & make
sudo make install
https://www.pjsip.org/docs/book-latest/html/intro_pjsua2.html#building-python-and-java-swig-modules

装好pjsua2就可以启动pygui测试了拨号了，这个例子带图形界面，ui用的Tkinter，另外一个带gui的例子是vidgui，ui用的qt

6、Using OpenCORE AMR NB and WB Codecs

https://trac.pjsip.org/repos/wiki/Using-OpenCORE-AMR-NB-WB-Codec

#opencore-amr 和 vo-amrwbenc
./configure --prefix=$(pwd)/_install --host=arm-linux-gnueabihf CC=/usr/local/arm/gcc-4.9.4-arm-buildroot-linux-gnueabihf/bin/arm-linux-gnueabihf-gcc
make
make install

装好之后将两个库的_install目录合并，合并到opencore-amr-0.1.3/_install

#pjproject
./configure --prefix=/home/dong/pjproject-2.8/_install --host=arm-linux-gnueabihf CC=/usr/local/arm/gcc-4.9.4-arm-buildroot-linux-gnueabihf/bin/arm-linux-gnueabihf-gcc --disable-libwebrtc CFLAGS=-I/home/dong/alsa/include LDFLAGS=-L/home/dong/alsa/lib --with-opencore-amr=$(pwd)/third_party/opencore-amr-0.1.3/_install

make dep
make
make install

 ln -s libopencore-amrnb.so.0.0.3 libopencore-amrnb.so.0
 ln -s libopencore-amrwb.so.0.0.3 libopencore-amrwb.so.0
 ln -s libvo-amrwbenc.so.0.0.4 libvo-amrwbenc.so.0

7、Ubuntu 16.04 下编译pjsip，支持视频通话

https://blog.csdn.net/wywf4/article/details/82260582

https://blog.csdn.net/SUKHOI27SMK/article/details/18667465

https://github.com/sxcong/pjsipvideo_demo

2. js sip user agent

http://www.jssip.net/

https://sipjs.com/

基于pjsip实现p2p语音对讲

https://blog.csdn.net/voidreturn/article/details/65634390

Pjsip中一些有用的基本函数实现

https://blog.csdn.net/croop520/article/details/84634160

pjsua播放rtp音频包问题

https://blog.csdn.net/rankun1/article/details/52034323

可以推荐一些基于Sip协议的开源IP电话客户端吗？

https://www.zhihu.com/question/20093472?sort=created

freepbx对接gms网关

https://blog.csdn.net/weixin_30656145/article/details/97707013

 freeswitch对接asterisk案例

https://blog.csdn.net/educast/article/details/25310819

end

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