#34 Change to allow SIPP to support binary ISUP message body

[Andy Aicken]

Code modifications to allow binary input including \x00 to be specified in a scenario. This was needed to support SIP-I and SIP-T to allow an encapsulated binary ISUP message in the message body.

The current implementation of SIPP was treating the data it processed from the scenario file as a string, so any attempt to input a binary 0 as \x00 was being interpreted as a null terminator and truncating the message at this point.

I haven’t had to make any changes on the receive message processing to get this working. I guess things are just working ok because any received ISUP data is in the body at the end of the message so doesn't impact on any string processing on the headers.

I’ve done limited testing on this feature, just running the following scripts back to back, but these seem to be working ok:

myuac.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 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/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
--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>

myuas.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 default 'uas' scenario. -->
<!-- -->

<scenario name="Basic UAS responder">
<!-- 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 request="INVITE" crlf="true">
</recv>

<!-- The '[last_*]' keyword is replaced automatically by the -->
<!-- specified header if it was present in the last message received -->
<!-- (except if it was a retransmission). If the header was not -->
<!-- present or if no message has been received, the '[last_*]' -->
<!-- keyword is discarded, and all bytes until the end of the line -->
<!-- are also discarded. -->
<!-- -->
<!-- If the specified header was present several times in the -->
<!-- message, all occurences are concatenated (CRLF seperated) -->
<!-- to be used in place of the '[last_*]' keyword. -->

<send>
<![CDATA[

SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: multipart/mixed;boundary=testing
Content-Length: [len]
MIME-Version: 1.0

--testing
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
--testing
Content-Type: application/isup;base=itu-t92+;version=itu
Content-Disposition: session;handling=optional

\x06\x14\x01\x00
--testing--

]]>
</send>

<send retrans="500">
<![CDATA[

SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: multipart/mixed;boundary=testing
Content-Length: [len]
MIME-Version: 1.0

--testing
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
--testing
Content-Type: application/isup;base=itu-t92+;version=itu
Content-Disposition: session;handling=optional

\x09\x00
--testing--

]]>
</send>

<recv request="ACK"
optional="true"
rtd="true"
crlf="true">
</recv>

<recv request="BYE">
</recv>

<send>
<![CDATA[

SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/isup;base=itu-t92+;version=itu
Content-Disposition: session;handling=optional
Content-Length: [len]

\x10\x00

]]>
</send>

<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<timewait milliseconds="4000"/>

<!-- 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>

[Andy Aicken]

diff against SIPp v3.1, version svn450M

[Andy Aicken]

diff against SIPp v3.1, version svn522M, also includes a segmentation violation fix in the SIP-I code (could cause core dump on traffic run with retransmissions)

[Andy Aicken]

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user_id=2089235
Originator: YES

File Added: diff.txt