Hello SIPp team, I'm using SIPp 3.1.
I would like to set a binary part into the body. I'm using the following syntax
into my script:
<send retrans="500">
<![CDATA[
INVITE sip:respons...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:s...@[local_ip]:[local_port];cpc=333>;tag=[call_number]
To: sut <sip:[servi...@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:s...@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: [len]
Content-Type: multipart/mixed;boundary=ssboundary
--ssboundary
Content-Length: 139
Content-Type: application/sdp
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
--ssboundary
Content-Length: 28
Content-Type: application/isup;version=itu-t92+
\x01\x00\x48\x00\x0a\x03\x02\x08\x06\x83\x90\x81\x99\x99\x03\x08\x01\x00\x0a\x07\x03\x13\x12\x52\x00\x27\x47\x00
--ssboundary--
]]>
</send>
The problem is that it stops after the first \x00. So the remaining data are
missing (including the \x00).
Attached are the full script and the logs.
Could you help on that ?
Thank you in advance.
Best regards.
David.
David PEYRTON<mailto:david.peyr...@hp.com>
CMS EMEA Delivery<http://h20208.www2.hp.com/opencall/contact/index.jsp>
Communication and Media Solutions Business
Unit<http://h20208.www2.hp.com/opencall/index.jsp?jumpid=reg_R1002_USEN>
HewlettPackard<http://www.hp.com/>
[cid:984564014@23062009-2052]
Telnet: 614 60 80
Tel.: +33 4 76 14 60 80
Mob: +33 6 72 99 17 43
Loc.: B3,N2,H1
------------------------------------------
When the brothers work together, the mountains change into gold.
Chinese proverb.
-------------------------------------------
It is the essential role of the professor to waken the joy of work and to know.
Albert Einstein.
-------------------------------------------
Qui rit le dernier pense le moins vite !
<<inline: image001.jpg>>
<?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:respons...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:s...@[local_ip]:[local_port];cpc=333>;tag=[call_number]
To: sut <sip:[servi...@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:s...@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: [len]
Content-Type: multipart/mixed;boundary=ssboundary
--ssboundary
Content-Length: 139
Content-Type: application/sdp
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
--ssboundary
Content-Length: 28
Content-Type: application/isup;version=itu-t92+
\x01\x00\x48\x00\x0a\x03\x02\x08\x06\x83\x90\x81\x99\x99\x03\x08\x01\x00\x0a\x07\x03\x13\x12\x52\x00\x27\x47\x00
--ssboundary--
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" 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>
<pause milliseconds="200"/>
<!-- 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:respons...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:s...@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[servi...@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:s...@[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 milliseconds="5000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:respons...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:s...@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[servi...@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:s...@[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>
uac_2584_messages.log
Description: uac_2584_messages.log
------------------------------------------------------------------------------
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