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。

有时候我们需要模拟现网的生产环境来压测，就需要同时有多个sipp脚本运行，并且需要不断的调整呼叫并发。

通过python脚本的子进程功能，我们可以很方便的实现sipp的多方案压测功能。

环境

centos7.9

freeswitch1.10.7

sipp v3.6.2_rc1

python 2.7.5

压测方案

Freeswitch部署在137服务器，作为呼叫的信令代理，开放端口5080。

Sipp和python脚本部署在138服务器，sipp的测试方案需要启动4个进程。

Sipp1，uas端，开启5555端口，接收fs137转发的呼叫，并响应接通信令。

Sipp2，uac端，开启6666端口，向fs137发送呼叫，并在接通30秒后挂断电话。

Sipp3，uac端，开启7777端口，向fs137发送呼叫，并接收fs137返回的404信令但不处理。

Sipp4，uac端，开启6668端口，向fs137发送呼叫，并在收到180振铃消息后主动cancel呼叫。

其中，sipp2，sipp3，sipp4三个uac端的进程，每间隔3分钟发起新一轮呼叫，每一轮呼叫运行2分钟。

Sipp1配置脚本

Sipp1，uas端，开启5555端口，接收fs137转发的呼叫，并响应接通信令。

监听10.55.55.138的5555端口，命令和脚本如下。

sudo sipp -i 10.55.55.138 -p 5555 -sf uas-test180cps.xml

uas-test180cps.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>

  <send>

    <![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>

  <pause milliseconds="500"/>

  <send>

    <![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-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="500"/>

  <send>

    <![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>

  <pause milliseconds="500"/>

  <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

      a=rtpmap:8 PCMA/8000

    ]]>

  </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>

  <!-- Keep the call open for a while in case the 200 is lost to be     -->

  <!-- able to retransmit it if we receive the BYE again.               -->

  <timewait milliseconds="4000"/>

  <!-- 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>

Sipp2配置脚本

Sipp2，uac端，开启6666端口，向fs137发送呼叫，并在接通30秒后挂断电话。

使用10.55.55.138的6666端口向10.55.55.137的5080端口发起呼叫，每秒发起160通呼叫，总共发起19200通呼叫后停止运行，后台运行。

sudo sipp -i 10.55.55.138 -p 6666 -inf call-test.csv -sf uac-test180cps.xml 10.55.55.137:5080 -rp 1000 -r 160 -l 8192 -m 19200 -bg

uac-test180cps.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]>;tgrp=TG-19073;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="30000"/>

  <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>

Sipp3配置脚本

Sipp3，uac端，开启7777端口，向fs137发送呼叫，并接收fs137返回的404信令但不处理。

使用10.55.55.138的7777端口向10.55.55.137的5080端口发起呼叫，每秒发起30通呼叫，总共发起3600通呼叫后停止运行，后台运行。

sudo sipp -i 10.55.55.138 -p 7777 -inf call-test.csv -sf uac-test404-noack.xml 10.55.55.137:5080 -rp 1000 -r 30 -l 8192 -m 3600 -bg

uac-test404-noack.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="404" rtd="true">

  </recv>

  <timewait milliseconds="1000"/>

  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

</scenario>

Sipp4配置脚本

Sipp4，uac端，开启6668端口，向fs137发送呼叫，并在收到180振铃消息后主动cancel呼叫。

使用10.55.55.138的6668端口向10.55.55.137的5080端口发起呼叫，每秒发起10通呼叫，总共发起1200通呼叫后停止运行，后台运行。

sudo sipp -i 10.55.55.138 -p 6668 -inf call-test.csv -sf uac-test-cancel1.xml 10.55.55.137:5080 -rp 1000 -r 10 -l 8192 -m 1200 -bg

uac-test-cancel1.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" rtd="true">

  </recv>

  <recv response="180" >

  </recv>

  <recv response="180" >

  </recv>

  <pause milliseconds="200"/>

  <send retrans="500">

    <![CDATA[

      CANCEL sip:[field1]@[remote_ip]:[remote_port] SIP/2.0

      [last_Via:]

      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 CANCEL

      Contact: sip:sipp@[local_ip]:[local_port]

      Max-Forwards: 70

      Content-Length: 0

    ]]>

  </send>

  <recv response="200" crlf="true">

  </recv>

  <recv response="487" crlf="true">

  </recv>

  <send>

    <![CDATA[

      ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0

      [last_Via:]

      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>

  <timewait milliseconds="1000"/>

  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

</scenario>

Python脚本

该Python脚本可以在python2和python3的环境下运行，测试环境只跑了python2。

脚本如下，通过subprocess子进程实现压测方案的自动启停。

sipp-auto.py

# coding=utf-8

# python3 required

import subprocess

import time

import signal

import sys

def signal_handler(sig, frame):

    continueflag = 0

    for p in processes:

        p.terminate()

    sys.exit(0)

signal.signal(signal.SIGINT, signal_handler)

commands = [

    # "sudo sipp -i 10.55.55.138 -p 5555 -sf uas-test180cps.xml",

    "sudo sipp -i 10.55.55.138 -p 6666 -inf call-test.csv -sf uac-test180cps.xml 10.55.55.137:5080 -rp 1000 -r 160 -l 8192 -m 19200 -bg",

    "sudo sipp -i 10.55.55.138 -p 7777 -inf call-test.csv -sf uac-test404-noack.xml 10.55.55.137:5080 -rp 1000 -r 30 -l 8192 -m 3600 -bg",

    "sudo sipp -i 10.55.55.138 -p 6668 -inf call-test.csv -sf uac-test-cancel1.xml 10.55.55.137:5080 -rp 1000 -r 10 -l 8192 -m 1200 -bg"

]

continueflag = 1

processes = []

commandUas = "sudo sipp -i 10.55.55.138 -p 5555 -sf uas-test180cps.xml"

p = subprocess.Popen(commandUas, shell=True)

processes.append(p)

while 1==continueflag:

    for command in commands:

        p = subprocess.Popen(command, shell=True)

        processes.append(p)

   

    time.sleep(180) # 等待3分钟

for p in processes:

    p.wait()

测试

Sipp的配置脚本和python脚本放置到相同目录下。

启动python脚本。

sudo python sipp-auto.py

观察fs资源情况和sipp运行情况。

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