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Re: [Sipp-users] SIP-I
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From: Sakharam T. <sak...@ou...> - 2014-09-23 10:27:12
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Hi i tried to run following scenario on sipp 3.4.1, saw outgoing isup msg in wireshark as part of sip nsg.
<?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"> <!-- 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:te...@co... SIP/2.0 From: <sip:223...@si...>;tag=e27b3 To: <sip:te...@co...> Call-Id: scb2f9c3f297d0197b29f4591169fcc23 Cseq: 25077 INVITE Session-Expires: 1800 Min-Expires: 90 Content-Type:multipart/mixed;boundary=level3-boundary Content-Length: 675 Expires: 180 Date: Tue, 01 Mar 2011 04:19:12 GMT Max-Forwards: 69 User-Agent: wxCommunicator Accept-Language: en Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,INFO,PRACK,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REGISTER,REFER Supported: replaces,timer,100rel,from-change,norefersub Via: SIP/2.0/UDP 173.170.11.84:5060;branch=z9hG4bK-83c6ce099e7d;rport Contact: <sip:2233437105@173.170.11.84:5060> Content-Length: [len] MIME-Version: 1.0
--level3-boundary Content-Type: application/sdp
v=0 o=sipX 5 10 IN IP4 192.168.1.138 s=call c=IN IP4 192.168.1.138 t=0 0 m=audio 9000 RTP/AVP 96 97 a=candidate:0 t UDP 1.0 192.168.1.138 9000 a=candidate:0 t UDP 1.0 192.168.1.138 9001 a=candidate:1 t UDP 0.5 5.210.195.106 9000 a=candidate:1 t UDP 0.5 5.210.195.106 9001 a=rtpmap:96 telephone-event/8000/1 a=rtpmap:97 speex/32000/1 a=fmtp:97 mode=4 a=ptime:20 --level3-boundary Content-Type: application/isup;base=itu-t92+;version=itu Content-Disposition: session;handling=optional
\x01\x00\x20\x00\x00\x03\x02\x06\x04\x01\x10\x21\x43\x0a\x08\x01\x15\x44\x21\x43\x65\x87\x09\x00 --level3-boundary--
]]> </send>
<recv response="100" optional="true"> </recv>
<recv response="180" optional="true"> </recv>
<recv response="183" 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"> </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:[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=[pid]SIPpTag00[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK 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=[pid]SIPpTag00[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>
To copile Sipp 3.4.1 , can take look at http://techvick.blogspot.in/2014/09/how-to-intsall-sipp-on-ubuntu.html
Best Regards,Sakharam Thorat.
From: s_a...@ho...
To: sak...@ou...; fbe...@ya...
Subject: RE: [Sipp-users] SIP-I
Date: Tue, 23 Sep 2014 15:09:55 +0530
Hello Sakharam/Fabio,
I am not sure if this code merge has really worked with real soft switch or PSTN GW testing.
check this :: http://sourceforge.net/p/sipp/patches/34/
this patch seem to be tested back to back..
@fabio : is it some Network deployment you want to test ?
Thanks
-Abinash
From: sak...@ou...
To: fbe...@ya...
Date: Tue, 23 Sep 2014 13:32:21 +0530
CC: sip...@li...
Subject: Re: [Sipp-users] SIP-I
Hi,
Please take a look at following scenario,
<?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">
<!-- 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=[pid]SIPpTag00[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: multipart/mixed;boundary=level3-boundary
Content-Length: [len]
MIME-Version: 1.0--level3-boundary
Content-Type: application/sdpv=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
--level3-boundary
Content-Type: application/isup;base=itu-t92+;version=itu
Content-Disposition: session;handling=optional\x01\x00\x20\x00\x00\x03\x02\x06\x04\x01\x10\x21\x43\x0a\x08\x01\x15\x44\x21\x43\x65\x87\x09\x00
--level3-boundary--]]>
</send><recv response="100"
optional="true">
</recv><recv response="180" optional="true">
</recv><recv response="183" 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">
</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:[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=[pid]SIPpTag00[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
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=[pid]SIPpTag00[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-Type: application/isup;base=itu-t92+;version=itu
Content-Disposition: session;handling=optional
Content-Length: [len]\x0c\x02\x00\x02\x87\x90]]>
</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>
Best Regards,Sakharam Thorat.
Date: Tue, 23 Sep 2014 00:55:54 -0700
From: fbe...@ya...
Subject: SIP-I
To: sak...@ou...
Hi,
do you know how to embed ISUP packets inside SIP messages ?
I read someone made a patch. Is it mainstream ?
Do you have a small example ?
thank you
Fabio
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