sipp模拟freeswitch分机测试(SIP协议调试)

1、freeswitch安装

yum install -y http://files.freeswitch.org/freeswitch-release-1-6.noarch.rpm epel-release
yum install -y freeswitch-config-vanilla freeswitch-lang-* freeswitch-sounds-*
systemctl enable freeswitch
freeswitch

2、freeswitch配置

以上默认能拨通，但是没声音，30秒左右就自动断线了，明显跟freepbx一样是NAT的故障

我偷懒用yun install安装的freeswitch，配置文件在/etc/freeswitch目录，安装文件比较分散

/var/lib/yum/repos/x86_64/7/freeswitch
/var/lib/freeswitch
/var/cache/yum/x86_64/7/freeswitch
/var/log/freeswitch
/usr/lib64/freeswitch
/usr/share/freeswitch
/usr/bin/freeswitch
/run/freeswitch
/etc/sysconfig/freeswitch
/etc/freeswitch

需要打开的端口：

udp     16384:32768
udp     4569
udp     5060
tcp     5060
udp     5080
tcp     5080
tcp     8000
udp     8000

配置conf/autoload_configs/verto.conf.xml

<param name="ext-rtp-ip" data="外网ip">

配置conf/sip_profiles/internal.xml

<param name="aggressive-nat-detection" value="true"/>
<param name="multiple-registrations" value="true"/>
<param name="ext-rtp-ip" value="外网ip"/>
<param name="ext-sip-ip" value="外网ip"/>
<param name="NDLB-received-in-nat-reg-contact" value="true"/>
<param name="NDLB-force-rport" value="true"/>
<param name="NDLB-broken-auth-hash" value="true"/>
<param name="enable-timer" value="false"/>
<param name="auth-calls" value="true"/>

配置conf/sip_profiles/external.xml

<param name="aggressive-nat-detection" value="true"/>
<param name="ext-rtp-ip" value="外网ip"/>
<param name="ext-sip-ip" value="外网ip"/>
<param name="NDLB-force-rport" value="true"/>

配置conf/autoload_configs/switch.conf.xml

<param name="rtp-start-port" value="16384"/>
<param name="rtp-end-port" value="32768"/>

2) 关闭centos 7.2防火墙，先保持网络通畅，调完了在配置防火墙端口。

systemctl stop firewalld.service #停止firewall
systemctl disable firewalld.service #禁止firewall开机启动
firewall-cmd --state #查看默认防火墙状态（关闭后显示notrunning，开启后显示running）

重启freeswitch

freeswitch -stop

freeswitch

3、freeswitch测试

sample sipp scenarios for testing freeswitch

https://github.com/os11k/sipp2freeswitch

1) 更改文件invite-accounts.csv中的SIP Proxy配置

SEQUENTIAL
# Username: 1000-1009
# Password: 1234
# SIP Proxy: 182.61.56.247, Destination Number: 9196
1000;182.61.56.247;[authentication username=1000 password=1234];9196;
1001;182.61.56.247;[authentication username=1001 password=1234];9196;
1002;182.61.56.247;[authentication username=1002 password=1234];9196;
1003;182.61.56.247;[authentication username=1003 password=1234];9196;
1004;182.61.56.247;[authentication username=1004 password=1234];9196;
1005;182.61.56.247;[authentication username=1005 password=1234];9196;
1006;182.61.56.247;[authentication username=1006 password=1234];9196;
1007;182.61.56.247;[authentication username=1007 password=1234];9196;
1008;182.61.56.247;[authentication username=1008 password=1234];9196;
1009;182.61.56.247;[authentication username=1009 password=1234];9196;

2) 更改文件register-accounts.csv中的SIP Proxy配置

SEQUENTIAL
# Username: 1000-1009
# Password: 1234
# SIP Proxy: 182.61.56.247
1000;182.61.56.247;[authentication username=1000 password=1234]
1001;182.61.56.247;[authentication username=1001 password=1234]
1002;182.61.56.247;[authentication username=1002 password=1234]
1003;182.61.56.247;[authentication username=1003 password=1234]
1004;182.61.56.247;[authentication username=1004 password=1234]
1005;182.61.56.247;[authentication username=1005 password=1234]
1006;182.61.56.247;[authentication username=1006 password=1234]
1007;182.61.56.247;[authentication username=1007 password=1234]
1008;182.61.56.247;[authentication username=1008 password=1234]
1009;182.61.56.247;[authentication username=1009 password=1234]

3)注意invite-auth.xml文件中的g711a.pcap目录

<exec play_pcap_audio="pcap/g711a.pcap"/>

4) 启动测试

sipp -i 172.16.0.6 -sf register.xml -inf register-accounts.csv 182.61.56.247:5060 -r 1 -rp 1000 -aa -trace_err
sipp -i 172.16.0.6 -sf invite-auth.xml -inf invite-accounts.csv 182.61.56.247:5060 -r 1 -rp 1000 -trace_err

测试注册，拨号，rtp数据都通了

4、FreeSwitch默认只支持音频，为支持视频，修改vars.xml如下：

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,GSM,H264,H263-1998,H263"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM,H264,H263-1998,H263"/>

5、杜金房的个人网站

http://www.dujinfang.com/

https://github.com/os11k/sipp2freeswitch