SIPp之认证注册
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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:
SEQUENTIAL50000;[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
!
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