xml 文件 . 本文草本出自狼山客人, 保留其相关注释.
<?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 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!--狼山客人：下面这一块表示SIPp发送一个INVITE数据包到SIP server(Wavesplitter的MSP-16)-->
<send>
<![CDATA[
INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch];rport
From: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: [field1] <sip:[field1]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
Cseq: 1 INVITE
Contact: sip:[field0]@[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=-
t=0 0
c=IN IP[media_ip_type] [media_ip]
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 G726/8000
]]>
</send>
<!--狼山客人：下面这一块表示SIPp在等待SIP server返回一个100的数据包-->
<recv response="100"> optional="true"
</recv>
<!--狼山客人：下面这一块表示SIPp在等待SIP server返回一个200的数据包，如果收到，说明ViVoice公司的VENUS NW800视频电话已经接听了，用户已经提起话筒-->
<recv response="200">
</recv>
<!--狼山客人：下面这一块表示SIPp开始通话-->
<!-- 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:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!--狼山客人：下面这一块表示SIPp开始发送语音RTP stream，在VENUS NW800视频电话上可以听到不知道哪国的老外的声音-->
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<!--狼山客人：暂停10秒钟(10000)，一小时， 3600000， 等待播放语音完毕-->
<pause milliseconds="300000"/>
<!--狼山客人：下面这一块发送BYE信号，这是挂断电话信号-->
<send retrans="500">
<![CDATA[
BYE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
From: sipp <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: sip <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
Cseq: 2 BYE
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!--狼山客人：下面这一块表示挂断完毕-->
<recv response="200">
</recv>
</scenario>
里面的业务逻辑需要自己确认, 最好的方法是通过抓包(比如ethereal), 分析其工作流. 不同情况下, 返回的信息有所不同, 需要相应调整.
上面是个呼入后,播放IVR 的案例, 比较简单.
下面是register 的例子(要测试通话, 就要先注册分机号)
<?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. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<send retrans="500">
<![CDATA[
REGISTER sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch];rport
From: [field0] <sip:[field0]@[remote_ip]:[remote_port]>;tag=[call_number]
To: [field0] <sip:[field0]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: [cseq] REGISTER
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-d87543-717507386-1--d87543-;rport
Contact: <sip:[field0]@[local_ip]:[local_port]>
Expires: 1200
Max-Forwards: 70
User-Agent: eyeBeam release 3004t stamp 16741
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY,MESSAGE,SUBSCRIBE,INFO
Content-Length: 0
]]>
</send>
<recv response="100"> optional="true"
</recv>
<recv response="200" crlf="true">
</recv>
</recv>
</scenario>
