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:[service]@[remote_ip]:[remote_port] SIP/2.0
这是不对的,应该是 ACK [next_url] SIP/2.0
问题之三还是 ACK 发的不对,要加 [routes] 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, MA 02111-1307 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 [call_id] keyword. -->
- <send retrans="500">
- <![CDATA[
- INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
- Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
- From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
- To: sut <sip:[service]@[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 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 [1-100] percent. -->
- <send>
- <![CDATA[
- ACK sip:[next_url] SIP/2.0
- Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
- From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
- To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
- Call-ID: [call_id]
- CSeq: 1 ACK
- [routes]
- Contact: sip:sipp@[local_ip]:[local_port]
- 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:[service]@[remote_ip]:[remote_port] SIP/2.0
- Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
- From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
- To: sut <sip:[service]@[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>
- <!-- 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>
复制代码
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