I did something once that does not require 2 scenarios.
Using labels, you can create different flow for Register and Invite
within the same scenario.
This will also support different call-id's for these methods.
The script I have attached shows Register + handling Invite as a
redirect server, but I think you can integrate the Register and labels
portion into a basic UAS scenario.
Good luck, Itzik.
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Charles P
Wright
Sent: Sunday, June 22, 2008 12:18 AM
To: Lucian Romi
Cc: [EMAIL PROTECTED];
[email protected]
Subject: Re: [Sipp-users] How to do REGISTER and UAS(INVITE server)
together.
You need to use separate scenarios.
Charles
"Lucian Romi" <[EMAIL PROTECTED]>
Sent by: [EMAIL PROTECTED]
06/20/2008 07:06 PM
To
[email protected]
cc
Subject
[Sipp-users] How to do REGISTER and UAS(INVITE server) together.
Hi,
I tried to create one scenario like this. There are REGISTER and INVITE
server.
To make the server able to locate this UAS without expire, every 3600
second will send 1 REGISTER.
INVITE traffic is continusly sending from UAC, say 1 per second.
Because REGISTER and INVITE server have different frequency and Call-ID,
anybody tell me how to do this scenario like this. Thanks!
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<?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="REGISTER" optional="true" crlf="true" next="3">
</recv>
<recv request="INVITE" crlf="true">
<action>
<ereg regexp="[+][0-9]+|[0-9]+"
search_in="hdr"
header="To:"
check_it="false"
assign_to="1"/>
</action>
<action>
<ereg regexp="pstn_1"
serach_in="hdr"
header="From:"
check_it="false"
assign_to="2"/>
</action>
</recv>
<pause milliseconds="300"/>
<!-- 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 next="2" test="2">
<![CDATA[
SIP/2.0 100 Trying
[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 next="1">
<![CDATA[
SIP/2.0 100 Trying
[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>
<label id="2"/>
<send retrans="500">
<![CDATA[
SIP/2.0 302 Moved Temporarily
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[EMAIL PROTECTED]:5060;user=phone;epName=testing>
Content-Length: 0
]]>
</send>
<label id="1"/>
<send retrans="500">
<![CDATA[
SIP/2.0 302 Moved Temporarily
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[EMAIL PROTECTED]:5060;user=phone;epName=testing>
Content-Length: 0
]]>
</send>
<!--
Contact: <sip:[EMAIL PROTECTED]:5060;user=phone;PG=dummy_pg1>
Contact: <sip:[EMAIL PROTECTED]:5060;user=phone;PG=dummy_pg2>
Contact: <sip:[EMAIL PROTECTED]:5060;user=phone;PG=mg98_pg >
-->
<recv request="ACK"
rtd="true"
crlf="true">
</recv>
<!-- 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" next="4" crlf="true"/>
<label id="3"/>
<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
Expires: 30
]]>
</send>
<label id="4"/>
<!-- 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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