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>

posted @ 2023-01-03 20:31 garvenc 阅读(1593) 评论(0) 编辑收藏举报

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使用SIPp进行压力测试

安装

使用

示例

注册测试

呼叫测试（使用默认场景）

呼叫测试（使用场景文件）

公告

常用链接

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	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

	<?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>

	SEQUENTIAL
	1001;[authentication username=1001 password=1234]
	1002;[authentication username=1002 password=1234]
	1003;[authentication username=1003 password=1234]

	<?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>