Hi,
You should know what you want to test in your VoIP phone..
You should be aware of basic SIP message flow... that you can get by repeating
manual steps on your VOIP phone and capturing network sip traffic.
Once you get your SIP message flow.
Prepare XML file for message flow.
Run the command on command prompt cmd..
sipp <SIP-CDS (Server) IP address> -sf sipMessage.xml
Here is sipMessage.xml content...
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!-- 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 >
<![CDATA[
REGISTER sip:10.230.53.225 SIP/2.0
Via: SIP/2.0/UDP 10.230.52.50:[local_port];branch=[branch];rport
Max-Forwards: 70
Contact: <sip:te...@10.230.52.50:[local_port];rinstance=df769c75a8bad123>
To: "test2"<sip:te...@10.230.53.225>
From: "test2"<sip:te...@10.230.53.225>;tag=[call_number]
Call-ID: [call_id]
CSeq: [cseq] REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Length: [len]
]]>
</send>
<recv response="200" crlf="true">
</recv>
<send >
<![CDATA[
SUBSCRIBE sip:te...@10.230.53.225 SIP/2.0
Via: SIP/2.0/UDP 10.230.52.50:[local_port];branch=[branch];rport
Max-Forwards: 70
Contact: <sip:te...@10.230.52.50:[local_port]>
To: "test2"<sip:te...@10.230.53.225>
From: "test2"<sip:te...@10.230.53.225>;tag=[call_number]
Call-ID: [call_id]
CSeq: [cseq] SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Event: message-summary
Content-Length: [len]
]]>
</send>
<recv response="501" crlf="true">
</recv>
<label id="1"/>
<send >
<![CDATA[
INVITE sip:[fiel...@10.230.53.225 SIP/2.0
Via: SIP/2.0/UDP 10.230.52.50:[local_port];branch=[branch];rport
Max-Forwards: 70
Contact: <sip:te...@10.230.52.50:[local_port]>
To: "[field0]"<sip:[fiel...@10.230.53.225>
From: "test2"<sip:te...@10.230.53.225>;tag=[call_number]
Call-ID: [call_id]
CSeq: [cseq] INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
Content-Length: [len]
v=0
o=- 9 2 IN IP4 10.230.52.50
s=CounterPath X-Lite 3.0
c=IN IP4 10.230.52.50
t=0 0
m=audio 14554 RTP/AVP 107 0 8 101
a=alt:1 3 : 8M33e0IX 0MCl+l2S 10.230.52.50 14554
a=alt:2 2 : BMV0PwaS OFqCq9q3 192.168.222.1 14554
a=alt:3 1 : RshdMeaM ZFpg93ys 192.168.81.1 14554
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv
]]>
</send>
<!-- 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>
From: chandan kumar [mailto:chandan_confide...@hotmail.com]
Sent: 30 March 2010 10:19
To: sipp-users@lists.sourceforge.net
Subject: [Sipp-users] SIPp on windows
Hi all,
Could any one please let me know how to use SIPp scripts on windows.I have
installed on windows ,I dont know how to use it.
I dont find any document regarding SIPp on windows.
IF i would like to test a existing VOIP phone & how can i use this SIPp tool
.For any particular scenarios where can i change the scripts .Please provide an
example.
Best Regards,
chandan
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