Jun,
I have used sipX a lot in testing our SIP-based products here. Some of our
tests use SIPp as a UA.
Here are two scenario files that, together, establish a basic call between two
instances of SIPp (of course, this assumes you have already registered SIPp on
both sides). These work with sipX as a proxy (I am using sipX 3.8.1-011585).
I run them side by side in tow different terminal windows, and execute them one
right after the other, running the B side first, then the A side (this way the
B side script is ready to accept the INVITE sent from the A side script). You
will also need to have the CSV file (also included in this email) in order to
populate the information in the [field] parameters in the scripts.
Here are the actual scenario files:
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="successful_call_single_proxy_sideA">
<send>
<![CDATA[
INVITE sip:[fiel...@[field4] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 70
From: <sip:[fiel...@[field1]>;tag=from-[call_number]
To: <sip:[fiel...@[field4]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact: sip:[fiel...@[local_ip]:[local_port]
User-Agent: SIPp/Linux
Subject: Test 3.2.1.1.7
Content-Type: application/sdp
Content-Length: [len]
v=0
o=sipp1 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>
<recv response="200">
</recv>
<send>
<![CDATA[
ACK sip:[fiel...@[field4] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
[last_From:]
[last_To:]
[last_Call-ID:]
CSeq: [cseq] ACK
Contact: sip:[fiel...@[local_ip]:[local_port]
User-Agent: SIPp/Linux
Content-Length: 0
]]>
</send>
<pause milliseconds="1000"/>
<send>
<![CDATA[
BYE sip:[fiel...@[field4] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
[last_From:]
[last_To:]
[last_Call-ID:]
CSeq: [cseq] BYE
Contact: sip:[fiel...@[local_ip]:[local_port]
User-Agent: SIPp/Linux
Max-Forwards: 70
Content-Length: 0
]]>
</send>
<recv response="200">
</recv>
</scenario>
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="successful_call_single_proxy_sideB">
<recv request="INVITE">
</recv>
<send>
<![CDATA[
SIP/2.0 180 UA2 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=to-[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: sip:[fiel...@[local_ip]:[local_port]
User-Agent: SIPp/Linux
Content-Length: 0
]]>
</send>
<pause milliseconds="500"/>
<send>
<![CDATA[
SIP/2.0 200 OK UA2 Answered
[last_Via:]
[last_From:]
[last_To:];tag=to-[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: sip:[fiel...@[local_ip]:[local_port]
User-Agent: SIPp/Linux
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 request="ACK">
</recv>
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK UA2 Goodbye
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: sip:[fiel...@[local_ip]:[local_port]
User-Agent: SIPp/Linux
Content-Length: 0
]]>
</send>
</scenario>
Here is the CSV file. You will have to modify the individual fields to match
your domain, and the SIPp usernames and passwords as set up in sipX. The
domain (e.g. test.azteknetworks.net) has to be the same as what you have it set
up for in sipX.
SEQUENTIAL
sipp1;test.azteknetworks.net;[authentication username=sipp1
password=sipp1];sipp2;test.azteknetworks.net;[authentication username=sipp2
password=sipp2]
Best of luck!
Jeffrey Wright
System Test Engineering Manager
Aztek Networks, Inc.
________________________________
From: Wen Jun [mailto:jun.wen.s...@gmail.com]
Sent: Fri 12/4/2009 2:18 AM
To: sipp-users@lists.sourceforge.net
Subject: [Sipp-users] Sipx with SIPp
Hi, it might be a frequently asked question about how to test SIPX with SIPp.
Does any have successful experience about that ? Appreciated if you can share
some hints to me .
I've well done the registration from SIPp UAC to SIPX but the call from SIPp
UAC to SIPX was struck yet.
Regards,
Jun
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