要在网页端打电话需要做的事情是
- 安装库libsrtp-dev apt-get install libsrtp-dev
- 安装pjproject 在asterisk13.8以上 pjproject跟asterisk绑定编译了,命令在asterisk源码目录下 执行 ./configure --with-pjproject-bundled 然后在执行 make menuselect 进入模块选项Resource Modules 确保res_srtp和res_http_websocket 模块被选中,若无法选择,则是srtp库没有安装成功。选择后保存退出,执行make,make install
- 接下来就需要进入/etc/asterisk 目录下,更改http.conf配置文件,设置enabled=yes bindaddr=0.0.0.0 bindport=8088,这个是http的若为https 则设置tlsenable=yes tlsbindaddr=0.0.0.0:8089,设置tlscertfile=/etc/asterisk/keys/asterisk.pem tlsprivatekey=/etc/asterisk/keys/asterisk.pem
- 进入sip.conf 配置文件中 设置
- transport=ws,wss
- avpf=yes
- nat=yes,force_rport
- nat=yes,force_rport
- directmedia=no
- encryption=yes
- icesupport
- force_avp=yes
- dtlsverify=no
- dtlsenable=yes
- dtlscertfile=/etc/asterisk/keys/asterisk.pem
- dtlsprivatekey=/etc/asterisk/keys/asterisk.pem
- dtlssetup=actpass
附上sipjs源码
http://download.csdn.net/download/u010939285/9799712