webrtc接入freeswitch的sip音视频传输

Posted 2021-03-18 dong1

1、安装freeswitch

https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7

dong@ubuntu:~/freeswitch$ vi freeswitch_v1.6_install_centos7.6.sh

#freeswitch_v1.6_install_centos7.6.sh

yum install -y http://files.freeswitch.org/freeswitch-release-1-6.noarch.rpm epel-release

yum install -y git alsa-lib-devel autoconf automake bison broadvoice-devel bzip2 curl-devel db-devel e2fsprogs-devel flite-devel g722_1-devel gcc-c++ gdbm-devel gnutls-devel ilbc2-devel ldns-devel libcodec2-devel libcurl-devel libedit-devel libidn-devel libjpeg-devel libmemcached-devel libogg-devel libsilk-devel libsndfile-devel libtheora-devel libtiff-devel libtool libuuid-devel libvorbis-devel libxml2-devel lua-devel lzo-devel mongo-c-driver-devel ncurses-devel net-snmp-devel openssl-devel opus-devel pcre-devel perl perl-ExtUtils-Embed pkgconfig portaudio-devel postgresql-devel python26-devel python-devel soundtouch-devel speex-devel sqlite-devel unbound-devel unixODBC-devel wget which yasm zlib-devel

# 
cd /usr/local/src
#git clone -b v1.6 https://github.com/signalwire/freeswitch.git freeswitch
git clone -b v1.6 https://gitee.com/dong2/freeswitch.git freeswitch
cd /usr/local/src/freeswitch
./bootstrap.sh -j
./configure
make -j
make -j install

# 
#make -j cd-sounds-install
#make -j cd-moh-install

# 
ln -sf /usr/local/freeswitch/bin/freeswitch /usr/bin/ 
ln -sf /usr/local/freeswitch/bin/fs_cli /usr/bin/

# 
#freeswitch
#freeswitch -nc
#freeswitch -nonat -nonatmap
#freeswitch -nonat -nonatmap -nosql

# 
#netstat -anp|grep 5060

# 
#fs_cli

 

2、配置freeswitch

1) Change password
cd /usr/local/freeswitch/conf
vi vars.xml
    Change:  <X-PRE-PROCESS cmd="set" data="default_password=1"/> {!!set it to something different!!}
    Save and close (<Esc> :wq!)

2) Delete IPv6
cd /usr/local/freeswitch/conf/sip_profiles
mv internal-ipv6.xml internal-ipv6.xml.removed   {disables ipv6 support}
mv external-ipv6.xml external-ipv6.xml.removed  {disables ipv6 support}

3) Configuring ext-rtp-ip
cd /usr/local/freeswitch/conf/sip_profiles/
vi internal.xml
    <param name="ext-rtp-ip" value="182.61.xx.25"/>
    <param name="ext-sip-ip" value="182.61.xx.25"/>

vi external.xml
    <param name="ext-rtp-ip" value="182.61.xx.25"/>
    <param name="ext-sip-ip" value="182.61.xx.25"/>

 

4）stun

  <X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:182.61.xx.25:3478"/>
  <X-PRE-PROCESS cmd="stun-set" data="external_sip_ip=stun:182.61.xx.25:3478"/>

 

3、下载webrtc客户端sipml5(sipjs/jssip也类似)

https://github.com/DoubangoTelecom/sipml5

 

4、在sipml5根目录启动一个http服务

1）python -m SimpleHTTPServer 8000 &

2）访问http://192.168.18.130:8000/

 

3）进入Enjoy our live demo

 

4）填写sip账号信息和ws接入信息

 

5）拨号

 

5、stun穿透

sip终端拨不通webrtc或者webrtc拨sip终端很久才能拨通，一般是nat穿透问题，在webrtc端加个stun服务一般能解决

免费的stun，我验证过的就这三个能用，最好还是自己搭一个stun/turn/ice服务

以下url填入ICE Servers即可

[{url:\'stun:stun.schlund.de\'}]
[{url:\'stun:stun.voipbuster.com\'}]
[{url:\'stun:stun.xten.com\'}]

 

coturn和Stuntman比较好用
coturn
https://github.com/coturn/coturn

Stuntman
http://www.stunprotocol.org/

stuntman_centos7.2_install.sh

yum install boost
yum install boost-devel
yum install boost-doc
wget http://www.stunprotocol.org/stunserver-1.2.7.tgz
tar -zxvf stunserver-1.2.7.tgz
make
./stuntestcode
./stunserver
#./stunclient 182.61.xx.25 3478

stuntman_ubuntu16.04_install.sh

sudo apt-get install g++
sudo apt-get install make
sudo apt-get install libboost-dev
sudo apt-get install libssl-dev
make
./stuntestcode
./stunserver
#./stunclient 182.61.xx.25 3478

装好stunserver，即可在webrtc客户端sipml5 ICE Servers配置
[{url:\'stun:182.61.xx.25:3478\'}] 

或者装好coturn

apt-get install coturn / yum install coturn

turnserver -o -a -f -v --mobility -m 10 --max-bps=1024000 --min-port=16384 --max-port=32768 --user=test:test123 -r test --cert=/usr/local/nginx/conf/SSL_Pub.pem --pkey=/usr/local/nginx/conf/SSL_Priv.pem CA-file=/usr/local/nginx/conf/SSL_CA.pem
(开放端口与freeswitch/conf/autoload_configs/switch.conf.xml配置保持同步,允许通过的比特流速率1M=1024×1024不能太小, 可以不检测密钥)

[{url:\'stun:8.134.xx.xxx:3478\'},{url:\'turn:8.134.xx.xxx:3478\', username:\'test\', credential:\'test123\'}]

除了在webrtc端添加stun服务来解决nat穿透问题，在freeswitch端配置应该也行，我还没验证。

 

6、参考设计

基于freeSWITCH的sip协议利用WebRTC 实现实时视频聊天

https://blog.csdn.net/graceup/article/details/79485976

 

 

7、freeswitch_webrtc

新版本chrome在http下已经没有操作麦克风和摄像头的权限，而且涉及webrtc的通信都需要安全传输

新版本freeswitch安装也更方便了，涉及webrtc的内容整理到github了

https://github.com/dong2/2FreeSWITCH/tree/master/VIV.freeswitch_webrtc

 

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