使用SIPp进行压力测试
本文更新于2022-05-14，使用SIPp v3.5.3。
官网：http://sipp.sourceforge.net/。
中文文档：http://sipp.sourceforge.net/doc/cn-reference.pdf。
安装
其它版本请于官网下载。
wget https://github.com/SIPp/sipp/releases/download/v3.5.3/sipp-3.5.3.tar.gz
tar -xzf sipp-3.5.3.tar.gz
cd sipp-3.5.3/
./build.sh
sudo cp sipp /usr/local/bin/sipp
使用
sipp HOST[:PORT] [OPTIONS]
OPTIONS可为：
- -aa：对INFO、NOTIFY、OPTIONS、UPDATE自动回复200 OK。
- -d：每个呼叫的持续时间，单位为毫秒。
- -inf CSVFILENAME：CSV数据文件。
- -p PORT：sipp监听的端口。默认为5060（与freeswitch默认的internal Profile端口相同）。
- -r N：每秒发起的请求数。
- -rtp_echo：将收到的RTP流原样返回。
- -s USERNAME：被呼方用户名。默认为service。
- -sf XMLFILENAME：XML场景文件。
- -sn SCENARIO：使用默认的场景文件。uac作为UAC（即SIP客户端）。
运行时可使用1、2、3、4按键切换界面的显示信息。
示例
注册测试
sipp 192.168.1.1:5060 -aa -sf reg.xml -inf users.csv -r 10 -p 6060
reg.xml场景文件的内容如下（官网有示例文件http://sipp.sourceforge.net/doc/branchc.xml，不过需修改才能使用）：
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="register">
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 70
Contact: sip:[field0]@[local_ip]:[local_port]
To: [field0] <sip:[field0]@[remote_ip]:[remote_port]>
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
Call-ID: [call_id]
CSeq: 1 REGISTER
Expires: 3600
User-Agent: SIPp
Content-Length: 0
]]>
</send>
<recv response="401" auth="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 70
Contact: sip:[field0]@[local_ip]:[local_port]
[field1]
To: [field0] <sip:[field0]@[remote_ip]:[remote_port]>
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
Call-ID: [call_id]
CSeq: 2 REGISTER
Expires: 3600
User-Agent: SIPp
Content-Length: [len]
]]>
</send>
<recv response="200">
</recv>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the 200 again. -->
<timewait milliseconds="500"/>
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
users.csv数据文件的内容如下（该文件提供注册用户的用户名和密码，SEQUENTIAL表示顺序执行）：
SEQUENTIAL
1001;[authentication username=1001 password=1234]
1002;[authentication username=1002 password=1234]
1003;[authentication username=1003 password=1234]
呼叫测试（使用默认场景）
sipp 192.168.1.1:5080 -sn uac -r 1 -d 10000 -rtp_echo -s 9664
需先在public Dialplan中为9664添加相应的路由。
呼叫测试（使用场景文件）
sipp 192.168.1.1:5080 -sf uac.xml -r 1 -d 10000 -rtp_echo -s 9664
需先在public Dialplan中为9664添加相应的路由。
uac.xml场景文件可以在官网查看：http://sipp.sourceforge.net/doc/uac.xml.html。文件内容如下：
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[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 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[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>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<pause/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[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>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
