Hello all. I am new to SIP and SIPp and I needed a little hand on a new
scenario I am trying to set.
I need 2 scenarios. The first on will generate calls just like the
simple uac but with different callerids other than just sipp. This can
be easily accomplished by a small modification on the scenario and a csv
file with all the caller ids I want to generate.
The second scenario is really complicated a few things need to be
accomplished.
1. REGISTER with MD5 Digest
-> REGISTER
<- 401
-> REGISTER with AUTH
<- 100 (TRYING)
<- 200
2. INVITE with MD5 Digest
-> INVITE
<- 407
-> ACK
-> INVITE
<- 100
<- 180
<- 183
<- 200
-> ACK
<- BYE
-> 200
3. Be ready to receive INVITEs for a certain necessarily
REGISTERED
user on part 1
Same as uas.xml with modifications I would guess
Here's what I could do so far: Part 1 and Part 2 separately but not
together (same file) because I can't use the same Call-ID on both and
when prepending something with /// just won't work for the next message.
Other than that, I have no idea on how to make a UAS that keeps
registration while receiving calls or register and "turn the UAS on" for
several users on a csv file.
I am attaching the file so you could take a look.
Please, some help would be really appreciated.
Thanks,
John Mesquita
<?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="Asterisk/OpenPBX Agent login - responder">
<!-- Since I am only able to enable authentication on a recv command, -->
<!-- let's trhow a nonsense command for getting a 401 back -->
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
To: <sip:[EMAIL PROTECTED]:[remote_port]>
From: <sip:[EMAIL PROTECTED]:[remote_port]>
Contact: "[field0]" <sip:[EMAIL PROTECTED]:[local_port]>;transport=[transport]
Expires: 300
Call-ID: ABCDE///[call_id]
CSeq: 2 REGISTER
Content-Length: 0
]]>
</send>
<!-- Received TRYING -->
<recv response="100" optional="true">
</recv>
<!-- Received the 401 as expected from the trash REGISTER message -->
<recv response="401" auth="true">
</recv>
<!-- Now, lets register for real with the proper authentication taken from the csv -->
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
To: <sip:[EMAIL PROTECTED]:[remote_port]>
From: <sip:[EMAIL PROTECTED]:[remote_port]>
Contact: "[field0]" <sip:[EMAIL PROTECTED]:[local_port]>;transport=[transport]
[field1]
Expires: 300
Call-ID: ABCDE///[call_id]
CSeq: 2 REGISTER
Content-Length: 0
]]>
</send>
<!-- Receive TRYING back -->
<recv response="100" optional="true">
</recv>
<!-- Receive the OK, we are registered! -->
<recv response="200"
crlf="true">
</recv>
<!-- Now we need to log the agent on! -->
<!-- For that we need to dial the agent route -->
<!-- So let's start with an invite -->
<send retrans="500">
<![CDATA[
INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0] <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: asterisk <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[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv response="407"
auth="true">
</recv>
<!-- We have been challenged, lets show them the money! And invite again -->
<!-- Send the acnoledgement for the 407 message -->
<send>
<![CDATA[
ACK sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
From: [field0] <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: asterisk <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
[last_Call-ID]
[last_CSeq]
Contact: sip:[EMAIL PROTECTED]:[local_port]
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [field0] <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: asterisk <sip:[EMAIL PROTECTED]:[remote_port]>
Call-ID: [call_id]
CSeq: 2 INVITE
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
[field1]
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 response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" optional="true">
</recv>
<recv response="200" rtd="true">
</recv>
<!-- Here we go with the regular UAC to receive the calls! -->
<!-- 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=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
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"
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-Length: 0
]]>
</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. -->
<pause 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>
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