SIPp uac.xml 之我见
https://sipp.sourceforge.net/doc/uac.xml.html这个 uac.xml 有没有问题呢？
有！
问题之一是：
<recv response="200" rtd="true" rrs="true">
要加 rrs, 仔细看解释就能看到
问题之二是：
ACK sip:@: SIP/2.0
这是不对的，应该是 ACK SIP/2.0
问题之三还是 ACK 发的不对，要加 头
还有，encoding 配置为 UTF-8 更好，如允许以增加中文解释
至于为什么，多看看自然就知道了
附完备的 uac.xml：
<?xml version="1.0" encoding="UTF-8" ?>
<!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, MA02111-1307 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 keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:@: SIP/2.0
Via: SIP/2.0/ :;branch=
From: sipp <sip:sipp@:>;tag=
To: sut <sip:@:>
Call-ID:
CSeq: 1 INVITE
Contact: sip:sipp@:
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length:
v=0
o=user1 53655765 2353687637 IN IP
s=-
c=IN IP
t=0 0
m=audio 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" rrs="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be percent. -->
<send>
<![CDATA[
ACK sip: SIP/2.0
Via: SIP/2.0/ :;branch=
From: sipp <sip:sipp@:>;tag=
To: sut <sip:@:>
Call-ID:
CSeq: 1 ACK
Contact: sip:sipp@:
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:@: SIP/2.0
Via: SIP/2.0/ :;branch=
From: sipp <sip:sipp@:>;tag=
To: sut <sip:@:>
Call-ID:
CSeq: 2 BYE
Contact: sip:sipp@:
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>
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