概述
SIP压测工具sipp,免费,开源,功能足够强大,配置灵活,优点多。
本文档介绍sipp工具的常用参数和测试脚本。
环境
centos7.9
sipp v3.6.2_rc1
常用参数
-sf 加载自定义的脚本XML格式
-i 设置本地ip 地址,用于指定 'Contact:','Via:', and ' 的地址
-p 指定本地端口
-bg 在后台运行sipp
-aa 使能SIPp 对 INFO, UPDAT E and NOTIFY 自动回 200ok
-inf 在呼叫过程中,从一个外部CSV 文件引入值到脚本中去。文件的第一行表明数据的读取顺序。
-rtp_echo 启用rtp 回送功能。将由 mp 参数指定的端口收到的 rtp 流回送给原发送者,默认6000。
-r 设置呼叫速率(设置多少个呼叫每秒)
-rp 设置呼叫速率的周期,默认是1000 毫秒。例如 r 7 rp 2000表示 2 秒中 7 个呼叫
-l 设置最大的并发呼叫量
-m 设置最本最大的呼叫个数,当sipp 达到该指定值会自动退出
号码csv
新增号码csv文件,在呼叫中自动设置到脚本变量中。
vi call-test.csv
SEQUENTIAL
12345678;13712345678;
常用脚本
在sipp.3.6.2/docs目录下有常用的默认脚本,uas.xml和uac.xml分别是server端和client端。
为了把号码设置到信令中,需要修改uac.xml文件中的from头域和to头域,将对应的值修改为field0和field1。
vi uac-test.xml
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic Sipstone UAC">
<send retrans="500">
<![CDATA[
INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [field1] <sip:[field1]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 8 18
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=ptime:20
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="183" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="200" rtd="true">
</recv>
<send>
<![CDATA[
ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [field1] <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<pause milliseconds="10000"/>
<send retrans="500">
<![CDATA[
BYE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [field1] <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
为了测试被叫侧的不同场景,需要修改uas.xml文件,通过分支功能实现。
在下面的脚本例子中,被叫侧响应有2种编码方式,第1种是PCMA,G729,第2种是PCMA,两种各占50%的比例。
分支功能使用next指定跳转的label ID号,使用chance指定跳转的比例。
vi uas-test729.xml
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="Basic UAS responder">
<recv request="INVITE" crlf="true" >
</recv>
<label id="1"/>
<send next="3" chance="0.5">
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag08b[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<label id="2"/>
<pause milliseconds="1000"/>
<send>
<![CDATA[
SIP/2.0 183 Session Progress
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 8 18
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=ptime:20
]]>
</send>
<pause milliseconds="1000"/>
<send next="5">
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<label id="3"/>
<pause milliseconds="1000"/>
<send>
<![CDATA[
SIP/2.0 183 Session Progress
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
]]>
</send>
<pause milliseconds="1000"/>
<send next="5">
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<label id="5"/>
<pause milliseconds="1000"/>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 8 18
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
]]>
</send>
<recv request="ACK"
optional="true"
rtd="true"
crlf="true">
</recv>
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<timewait milliseconds="4000"/>
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
测试
服务端uas。
sudo sipp -i 10.55.55.138 -p 5555 -sf uas-test729.xml -rtp_echo
客户端uac。
sudo sipp -i 10.55.55.138 -p 6666 -inf call-test.csv -sf uac-test.xml 10.55.55.137:3060 -t un -rp 1000 -r 20 -l 8192 -m 99999
总结
sipp的github地址:https://github.com/SIPp/sipp
后续会有更多的压测方案和脚本慢慢写。
空空如常
求真得真
标签:SIP,last,压测,ip,Content,call,sipp3.6,local,分支 From: https://www.cnblogs.com/qiuzhendezhen/p/17252150.html