SIP UserAgent (B2BUA client)——libosip2 libeXosip2

1. libosip2 libeXosip2

http://www.antisip.com/download/exosip2/

Features:
--------
eXosip2 has support for:
 * registrations. (REGISTER)
 * call initiation and modification. (INVITE, re-INVITE)
 * other methods within calls (INFO, OPTIONS, UPDATE)
 * call transfer. (REFER)
 * reliability for provisionnal response. (PRACK)
 * sip event package. (SUBSCRIBE/NOTIFY)
 * event state publication. (PUBLISH)
 * instant messaging. (MESSAGE)
 * ...

eXosip2 does not contain:
 * RTP.
 * audio interface
 * sdp negotiation.
 * ...

2. Build libexosip2-5.1.0.tar.gz + libosip2-5.1.0.tar.gz

ln -s /usr/lib/x86_64-linux-gnu/libssl.so.1.0.0 /usr/lib/libssl.so
ln -s /usr/lib/x86_64-linux-gnu/libcrypto.so.1.0.0 /usr/lib/libcrypto.so

_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

#http://www.mirrorservice.org/pub/gnu/osip/
tar xvf libosip2-*
cd libosip2-*
./configure --prefix=${install} ${compiler} PKG_CONFIG_PATH=${pkg_config} ${lib_config} --host=${host} --enable-static --enable-shared --disable-openssl
make && make install
cd ..

#http://mirror.ossplanet.net/nongnu/exosip/
tar xvf libexosip2-*
cd libexosip2-*
./configure --prefix=${install} ${compiler} PKG_CONFIG_PATH=${pkg_config} ${lib_config} --host=${host} --enable-static --enable-shared --disable-openssl
make && make install
cd ..

媒体库

#JRTPLIB-3.11.1.tar.gz
cmake -DCMAKE_INSTALL_PREFIX=./_install -DCMAKE_C_COMPILER=/usr/local/gcc-linaro-arm-linux-gnueabihf/bin/arm-linux-gnueabihf-gcc -DCMAKE_CXX_COMPILER=/usr/local/gcc-linaro-arm-linux-gnueabihf/bin/arm-linux-gnueabihf-g++
make
make install

#faac-1.29.9.2.tar.gz
./configure CXX=arm-linux-gnueabihf-g++ CC=arm-linux-gnueabihf-gcc --prefix=${PWD}/_install --host=arm-linux-gnueabihf --with-mp4v2=no --enable-shared --enable-static
make
make install

#h264bitstream-0.2.0.tar.gz
sudo apt-get install build-essential libtool
autoreconf -i
./configure CXX=arm-linux-gnueabihf-g++ CC=arm-linux-gnueabihf-gcc --prefix=${PWD}/_install --host=arm-linux-gnueabihf --enable-shared --enable-static
make
make install

dong@ubuntu:~/libexosip2-5.1.0$ su
Password:
root@ubuntu:/home/dong/libexosip2-5.1.0# ./_install.sh
root@ubuntu:~# tar cvf _install.tar.gz _install/
root@ubuntu:~# mv _install.tar.gz /home/dong/
dong@ubuntu:~$ sudo chmod 777 _install.tar.gz

3. example

Freepbx or freeswitch can be used as SIP server

libexosip2-5.1.0/tools/sip_reg.c

add eXosip event

    /*zhoudd add*/
    case EXOSIP_CALL_CANCELLED:
        syslog_wrapper (LOG_INFO, "call cancelled");
        break;
    case EXOSIP_CALL_ACK:
        syslog_wrapper (LOG_INFO, "call ack");
        break;
    case EXOSIP_CALL_CLOSED:
        syslog_wrapper (LOG_INFO, "call closed");
        break;
    case EXOSIP_CALL_PROCEEDING:
        syslog_wrapper (LOG_INFO, "call proceeding");
        break;
    case EXOSIP_CALL_RINGING:
        syslog_wrapper (LOG_INFO, "call ringing");
        break;
    case EXOSIP_CALL_ANSWERED:
        syslog_wrapper (LOG_INFO, "call answered");
        break;
    case EXOSIP_CALL_SERVERFAILURE:
    case EXOSIP_CALL_RELEASED:
        syslog_wrapper (LOG_INFO, "call release");
        break;

compile example

export LD_LIBRARY_PATH=$(pwd)/_install/lib:$LD_LIBRARY_PATH

gcc -O2 -I./_install/include -L./_install/lib sip_reg.c \
-o sip_reg \
-leXosip2 -losip2 -losipparser2 -lpthread

test.sh

./sip_reg \
-r sip:182.61.147.213:5060 \
-u sip:1002@182.61.147.213 \
-c sip:1002@182.61.147.213:5060 \
-U 1002 \
-P 1234  \
-p 5060 \
-e 1800 \
-d

caller mode

dong@ubuntu:~/doing/demo$ ./test.sh
SipReg v1.0 up and running
proxy: sip:182.61.147.213:5060
fromuser: sip:1002@182.61.147.213
contact: sip:1002@182.61.147.213:5060
expiry: 3600
local port: 5060
transport: UDP
username: 1002
password: [removed]
r   register
u   unregister
i   call
h   hangup
a   answer
q   quit
f   info
m   message
Please input the command:

registrered failure
registrered successfully
registrered successfully
registrered successfully
i
Please input the command:

call proceeding
call requestfailure
call proceeding
call ringing
call answered
call message new
BYE
recieved message eXosip event (type, did, cid) = (14, 2, 1)
call closed
call release

Please input the command:

q
dong@ubuntu:~/doing/demo$

callee mode

dong@ubuntu:~/doing/demo$ ./test.sh
SipReg v1.0 up and running
proxy: sip:182.61.147.213:5060
fromuser: sip:1002@182.61.147.213
contact: sip:1002@182.61.147.213:5060
expiry: 3600
local port: 5060
transport: UDP
username: 1002
password: [removed]
r   register
u   unregister
i   call
h   hangup
a   answer
q   quit
f   info
m   message
Please input the command:

registrered failure
registrered successfully
registrered successfully
registrered successfully
call invite
call cancelled
call closed

Please input the command:

call release
q
dong@ubuntu:~/doing/demo$

4. rport && contact

    case EXOSIP_CALL_INVITE:
      {
        eXosip_masquerade_contact (context_eXosip, event->request->req_uri->host, atoi(event->request->req_uri->port));

        osip_message_t *answer = NULL;
        int i;
        /*180 ring*/
        ...

        /*200 ok*/
        ...

        break;
      }

    case EXOSIP_CALL_ACK:
      {
        /*ack*/
        ...

        conference_bridge_creat_run(remote_audio_ip, atoi(remote_audio_port), local_audio_ip, atoi(local_audio_port));
        break;
      }

    case EXOSIP_CALL_ANSWERED:
      {
        /*ack*/
        ...

        /*200 ok*/
        ...      

        conference_bridge_creat_run(remote_audio_ip, atoi(remote_audio_port), local_audio_ip, atoi(local_audio_port));
        break;
      }

#register into gb28181 server

34020000001320000001.sh

./sip_reg  \
-r sip:182.61..xxx: \
-u sip:@192.168.127.128 \
-c sip:@192.168.127.128: \
-U  \
-P   \
-p  \
-e  \
-d

5. User Agent Application
1) register
libexosip2-5.1.0/tools/sip_reg.c

2) invite
libexosip2-5.1.0/help/doxygen/ht0-initialize.dox
libexosip2-5.1.0/help/doxygen/ht1-callcontrol.dox
libexosip2-5.1.0/help/doxygen/ht2-registration.dox

3) rtp mediastream
rtp_over_udp: jrtplib-3.11.1/examples/example1.cpp
rtp_over_tcp:  jrtplib-3.11.1/tests/tcptest.cpp

4) audio encode/decode && conference bridge
https://trac.pjsip.org/repos/wiki/media-flow?from=groupmessage