<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
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<!-- 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>
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<recv response="100" ptional="true">
</recv>
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<!-- 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>
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<!-- 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。-->
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<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>
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<recv response="100" ptional="true">
</recv>
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<!-- 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 -->
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<recv response="200" next="2">
</recv>
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<label id="2"/>
<pause milliseconds="1000"/>
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<!-- 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) -->
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<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
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<!-- definition of the call length repartition table (unit is ms) -->
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<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>
$ 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)
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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
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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 --------------------------------
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----------------------------- 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 --------------------------------
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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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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！
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，正在调整语音流！能播放语音流就又成功了一大步，昨天晚上不能上网，晚上把脚本贴出来。
