SIPp之认证注册

欢迎大家转载，为保留作者成果，转载请注明出处， http://blog.csdn.net/netluoriver，有些文件在资源中也可以下载！如果你没有积分，可以联系我！

无论做什么事情，只要坚持就一定会有收获的，只不过是时间的问题。

测试SIP协议的同志们可以加入群： 323827101，共同探讨哦！

群共享里可能会有你需要的资料

在经历了10天的挣扎之后，注册消息终于成功了。

最新的版本3.3试验注册有问题，我相信在不久这个BUG会修复的。

环境描述如下：

192.168.0.20是一个有注册认证的SIP服务器，服务器端口为5060；

192.168.0.101是我在windows安装的cygwin软件后 测试机器；

sipp版本： SIPp v3.2-TLS-PCAP, version unknown, built Jul 17 2013, 21:50:11

注册代码如下：

<?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 'branchc' scenario.                   -->
<!--                                                                    -->
<!-- 首先发送SIP注册消息，Register。里面的From与To是注册的号码  -->
<scenario name="branch_client">
  <send retrans="500">
    <![CDATA[
 
      REGISTER sip:[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number];rport
      To: [field0] <sip:[field0]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 REGISTER
      Contact: sip:[field0]@[local_ip]:[local_port]
      Content-Length: 0
      Expires: 300
    ]]>
  </send>

  <recv response="100" ptional="true">
  </recv>

  <!--  SIPp会收到来自AST要求验证的401 消息体，Recv意思为Receive，接收到来自AST的401要求验证的消息，Next为如果收到401，那么转至Label为1的地方进行操作 -->
  <recv response="401" auth="true" next="1">
  </recv>

  <!--  send invite with authentication messages -->
  <!--  开始发送Register消息，里面将把验证的密码消息发送给对方，在消息体里面是抓不到密码消息的，而且已经被md5方式加密过。-->

  <label id="1"/>
  <send retrans="500">
    <![CDATA[
 
      REGISTER sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port]
      From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number];rport
      To: [field0] <sip:[field0]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 2 REGISTER
      Contact: sip:[field0]@[local_ip]:[local_port]
      [field1]
      Content-Length: [len]
      Expires: 3600
    ]]>
  </send>

  <recv response="100" ptional="true">
  </recv>

  <!--   收到来自AST的200 ACK消息后，系统转至等待1000ms，或者可以直接去掉该设置 -->

  <recv response="200"  next="2">
  </recv>

  <label id="2"/>
  <pause milliseconds="1000"/>


  <!-- 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.               -->
  <pause 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>

user.csv代码如下：

SEQUENTIAL
50000;[authentication username50000 password=50000]

运行状态如下：

sipp -r 1 -i 192.168.0.101 -l 1 -sf branchc.xml -inf user.csv 192.168.0.20

Warning: open file limit > FD_SETSIZE; limiting max. # of open files to FD_SETSI                                                                                                                ZE = 64
       Resolving remote host '192.168.0.20'... Done.
------------------------------ Scenario Screen -------- [1-9]: Change Screen --
  Call-rate(length)   Port   Total-time  Total-calls  Remote-host
   1.0(0 ms)/1.000s   5060     114.59 s           20  192.168.0.20:5060(UDP)

  0 new calls during 14.353 s period     0 ms scheduler resolution
  1 calls (limit 1)                      Peak was 1 calls, after 1 s
  0 Running, 0 Paused, 0 Woken up
  0 dead call msg (discarded)            1 out-of-call msg (discarded)
  3 open sockets

                                 Messages  Retrans   Timeout   Unexpected-Msg
    REGISTER ---------->         20        0         0
         100 <----------         0         0         0         0
         401 <----------         20        0         0         0
    REGISTER ---------->         20        0         0
         100 <----------         0         0         0         0
         200 <----------         20        0         0         0
       Pause [   1000ms]         20                            0
       Pause [   4000ms]         20                            0
------------------------------ Test Terminated --------------------------------


----------------------------- Statistics Screen ------- [1-9]: Change Screen --
  Start Time             | 2013-07-20   09:55:08:618    1374285308.618236
  Last Reset Time        | 2013-07-20   09:56:48:874    1374285408.874967
  Current Time           | 2013-07-20   09:57:03:229    1374285423.229290
-------------------------+---------------------------+--------------------------
  Counter Name           | Periodic value            | Cumulative value
-------------------------+---------------------------+--------------------------
  Elapsed Time           | 00:00:14:354              | 00:01:54:611
  Call Rate              |    0.000 cps              |    0.175 cps
-------------------------+---------------------------+--------------------------
  Incoming call created  |        0                  |        0
  OutGoing call created  |        0                  |       20
  Total Call created     |                           |       20
  Current Call           |        1                  |
-------------------------+---------------------------+--------------------------
  Successful call        |        0                  |       19
  Failed call            |        0                  |        0
-------------------------+---------------------------+--------------------------
  Call Length            | 00:00:04:584              | 00:00:04:989
------------------------------ Test Terminated --------------------------------

2013-07-20      09:56:49:417    1374285409.417536: Discarding message which can't be mapped to a known SIPp call:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bK-2568-20--1

From: 50000 <sip:50000@192.168.0.101:5060>;tag=20;rport

To: 50000 <sip:50000@192.168.0.20:5060>;tag=1ea81e666

Call-ID: 20-2568@192.168.0.101

CSeq: 3 BYE

User-Agent: ASG Switch

Contact: <sip:50000@192.168.0.20:5060>

Content-Length: 0

！