i want know the what is the command to excecute sipp with PCAP
Regards
darshan
On 08/08/2008, [EMAIL PROTECTED] <
[EMAIL PROTECTED]> wrote:
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> Today's Topics:
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> 1. Re: SUBSCRIBE scenario (??????? ?.?.)
> 2. SIPp problem with pcap ( Jes?s Camacho Rodr?guez )
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 07 Aug 2008 18:35:49 +0400
> From: ??????? ?.?. <[EMAIL PROTECTED]>
> Subject: Re: [Sipp-users] SUBSCRIBE scenario
> To: [email protected], ims.asuser ims.asuser"
> <[EMAIL PROTECTED]>
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=windows-1251
>
>
> Hi Khaldon,
>
> i did perfomance tests of OpenIMSCore using SIPp 2.1 and SIPp IMS Bench.
> Normal Registration signaling flow looks like:
>
> > 1) register ->
> > 2) <-401
> > 3) register->
> > 4) <-200
>
> it's enough to provide the registration for user. Pay attention to
> P-Headers ;)
>
> In addition you'll need to fill the FoHSS with a number of subscriber
> entities, to handle this issue in automatic mode analyse the SQL scripts
> used for creation of Alice and Bob.
>
> Regards,
> Ivan Kuzmin.
> State University of Telecomunications, St.Petersburg, Russia.
>
>
> -----Original Message-----
> From: "ims.asuser ims.asuser" <[EMAIL PROTECTED]>
> To: [email protected]
> Date: Thu, 7 Aug 2008 16:16:40 +0200
> Subject: [Sipp-users] SUBSCRIBE scenario
>
> >
> > Hi all,
> >
> > I would like to test the OpenIMSCore platform (I will perform as much as
> I
> > can). These tests will be then given to the IMS community.
> > First of all, did anyone perform OpenIMSCOre tests using SIPp?
> >
> > I would like to perform first in different stage:
> > 1) Registration
> > 2) Calls
> > 3) Instant Messaging
> >
> > So far, I'm only on the first step (registering a user). I only managed
> to
> > do a simple registration (register ->, <-401, register->, <-200). But I
> have
> > problems when I want to go further.
> > In ims, the scenario looks like that:
> > 1) register ->
> > 2) <-401
> > 3) register->
> > 4) <-200
> > 5) subscribe->
> > 6) <-notify
> > 7) 200->
> >
> > The IMS Core is behaving really eerily when I send a subscribe message
> then
> > the simulation stops
> > Is anyone know why is it working like that?
> >
> > Attached are my script and wireshark trace.
> >
> > Many thanks,
> > Khaldon
> >
> > ATTACHMENT: text/xml (Registerbob1.xml)
> > ATTACHMENT: application/octet-stream (SIPp_SUBSCRIBE)
> >
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> >
>
>
>
>
> ------------------------------
>
> Message: 2
> Date: Thu, 7 Aug 2008 20:33:24 +0200
> From: " Jes?s Camacho Rodr?guez " <[EMAIL PROTECTED]>
> Subject: [Sipp-users] SIPp problem with pcap
> To: [email protected]
> Message-ID:
> <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi all,
>
> I am doing Asterisks performance tests on a pico-itx box. I compiled SIPp
> with pcap play support in a Debian Etch machine so I could send RTP
> packages
> through a recorded pcap file for simulating traffic. I run SIPp in client
> mode in one machine and I call to a VoIP phone (actually I wanted to call
> to
> a SIPp in server mode but I am calling to a phone for debugging purposes).
> The problem I have is that SIPp in client mode sends the RTP packages to
> Asterisk (at least that is what SIPp shows in the "Total RTP pckts sent"),
> but if I activate the RTP debug mode in Asterisk, I see that the
> communication is only in one direction (communication is established but
> RTP
> packages are only sent to the machine with SIPp client). SIP messages
> between the boxes seems to arrive fine, and the communication after a while
> closes without problem, telling the SIPp client that it was a Succesful
> call. First I thought it could be Asterisk, but when I call from another
> source (one more VoIP phone I have here) the communication is perfect and
> RTP packages are running in both directions. So I think it can be SIPp. I
> have used my own xml scenario based on uac_pcap.xml (it has little
> changes),
> but even if I run the original uac_pcap, it doesn't work still. Anyway I
> add
> my uac_pcap.xml:
>
> <?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 'uac' scenario with pcap (rtp) play -->
> <!-- -->
>
> <scenario name="UAC with media">
> <!-- In client mode (sipp placing calls), the Call-ID MUST be -->
> <!-- generated by sipp. To do so, use [call_id] keyword.
> -->
> <send retrans="500" start_rtd="1">
> <![CDATA[
>
> INVITE sip:[EMAIL PROTECTED]:[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:[EMAIL PROTECTED]:[remote_port]>
> Call-ID: [call_id]
> CSeq: 1 INVITE
> Contact: sip:[EMAIL PROTECTED]:[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[local_ip_type] [local_ip]
> t=0 0
> m=audio [auto_media_port] RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>
> ]]>
> </send>
>
> <recv response="100" optional="true" rtd="1" start_rtd="2">
> </recv>
>
> <recv response="180" optional="true" rtd="2">
> </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" crlf="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:[EMAIL PROTECTED]:[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:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
> Call-ID: [call_id]
> CSeq: 1 ACK
> Contact: sip:[EMAIL PROTECTED]:[local_port]
> Max-Forwards: 70
> Subject: Performance Test
> Content-Length: 0
>
> ]]>
> </send>
>
> <!-- Play a pre-recorded PCAP file (RTP stream) -->
> <nop>
> <action>
> <exec play_pcap_audio="pcap/call_g711a.pcap"/>
> </action>
> </nop>
>
> <!-- Pause 3 minutes, which is approximately the duration of the -->
> <!-- PCAP file -->
> <pause milliseconds="180000"/>
>
> <!-- Play an out of band DTMF '1' -->
> <nop>
> <action>
> <exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
> </action>
> </nop>
>
> <pause milliseconds="1000"/>
>
> <!-- The 'crlf' option inserts a blank line in the statistics report. -->
> <send retrans="500">
> <![CDATA[
>
> BYE sip:[EMAIL PROTECTED]:[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:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
> Call-ID: [call_id]
> CSeq: 2 BYE
> Contact: sip:[EMAIL PROTECTED]:[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="50, 100, 200, 500, 1100, 2100, 3100, 4100,
> 5100, 6100, 10000"/>
>
> <!-- definition of the call length repartition table (unit is ms) -->
> <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
>
> </scenario>
>
>
> I am quite crazy about the whole problem, it makes me mad that it works
> using two phones and that it doesn't when I generate the call with SIPp.
> Maybe I have to add something to the SIP messages in the script that I am
> not realizing or I don't know...
>
> Thanks, greetings,
> Jes?s
>
> PS. I am not using NAT, all the machines and phones are in the same private
> network.
> -------------- next part --------------
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> End of Sipp-users Digest, Vol 27, Issue 3
> *****************************************
>
--
Darshan B N
Thanks & Regards
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