Hi All,
I am trying to make TCP calls using Sipp as my 2 end users within my IMS cloud.
As TCP ports are differing for Registration & Invite, I thought of putting both
my Registration & INVITE in one xml file.
To my strange I found it in this execution procedure my UDP calls are also not
working.
*UDP calls are working fine when each scenarios are executed separately.
UAS sip is not receiving the INVITE message in this case & generating
"Discarding message which can't be mapped to a known SIPp call"
The issue remains same for both UDP & TCP calls.
In the stack "/usr/local/src/sipp-3.3.990/src/socket.cpp:
WARNING("Discarding message which can't be mapped to a known SIPp call:\n%s",
msg); is getting called.
I found inside the listen.cpp file, the call-id check is failing.
I have attached the used XML files.
Thanks
Samarpita
============================================================================================================================
Disclaimer: This message and the information contained herein is proprietary
and confidential and subject to the Tech Mahindra policy statement, you may
review the policy at http://www.techmahindra.com/Disclaimer.html externally
http://tim.techmahindra.com/tim/disclaimer.html internally within TechMahindra.
============================================================================================================================
<?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=" Register with challange 401 & followed by a call">
<!-- 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[
REGISTER sip:[field0];transport=[transport] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]-[pid]
From: "[field1]" <sip:[field1]@techma.com>;tag=396696254-[pid]
To: <sip:[field1]@techma.com>
Max-Forwards: 70
Expires: 3600
Call-ID: [call_id]
CSeq: [cseq] REGISTER
P-Preferred-Identity: "[field1]" <sip:[field1]@techma.com>
P-Access-Network-Info: IEEE-802.11
User-Agent: IMS-Communicator 081209
Supported: path
Contact: <sip:[field1]@[local_ip]:[local_port];Transport=[transport]>
Authorization: Digest username="[field1]@techma.com",realm="techma.com",uri="techma.com",nonce="",response=""
Content-Length: 0
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="401" auth="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[field0];transport=[transport] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]-[pid]
From: "[field1]" <sip:[field1]@techma.com>;tag=396696254-[pid]
To: <sip:[field1]@techma.com>
Call-ID: [call_id]
CSeq: [cseq+1] REGISTER
Max-Forwards: 70
Contact: <sip:[field1]@[local_ip]:[local_port];Transport=[transport]>
Expires: 3600
P-Preferred-Identity: "[field1]" <sip:[field1]@techma.com>
P-Access-Network-Info: IEEE-802.11
User-Agent: IMS-Communicator 081209
Supported: path
[field2]
Content-Length: 0
]]>
</send>
<recv response="200" rtd="true">
<action>
<ereg regexp=".*" search_in="hdr" header="Call-ID:" assign_to="cid" />
</action>
</recv>
<pause milliseconds="5000"/>
<recv request="INVITE">
</recv>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_Record-Route:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
Call-ID:[$cid]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<pause milliseconds="100"/>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_Record-Route:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[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>
<!-- 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>
<?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=" Register with challange 401 & followed by a call">
<!-- 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[
REGISTER sip:[field0];transport=[transport] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]-[pid]
From: "[field1]" <sip:[field1]@techma.com>;tag=396696254-[pid]
To: <sip:[field1]@techma.com>
Max-Forwards: 70
Expires: 3600
CSeq: [cseq] REGISTER
Call-ID: [call_id]
P-Preferred-Identity: "[field1]" <sip:[field1]@techma.com>
P-Access-Network-Info: IEEE-802.11
User-Agent: IMS-Communicator 081209
Supported: path
Contact: <sip:[field1]@[local_ip]:[local_port];Transport=[transport]>
Authorization: Digest username="[field1]@techma.com",realm="techma.com",uri="techma.com",nonce="",response=""
Content-Length: 0
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="401" auth="true">
</recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[field0];transport=[transport] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]-[pid]
From: "[field1]" <sip:[field1]@techma.com>;tag=396696254-[pid]
To: <sip:[field1]@techma.com>
Call-ID: [call_id]
CSeq: [cseq+1] REGISTER
Max-Forwards: 70
Contact: <sip:[field1]@[local_ip]:[local_port];Transport=[transport]>
Expires: 3600
P-Preferred-Identity: "[field1]" <sip:[field1]@techma.com>
P-Access-Network-Info: IEEE-802.11
User-Agent: IMS-Communicator 081209
Supported: path
[field2]
Content-Length: 0
]]>
</send>
<recv response="200" rtd="true">
<action>
<ereg regexp=".*" search_in="hdr" header="Call-ID:" assign_to="cid" />
</action>
</recv>
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[field0] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Route: <sip:10.193.30.22;lr=on>
Route: <sip:.iIiIiI.10.193.14.40@10.193.14.40:5060;transport=UDP;lr;orig>
From: <sip:[field1]@[field0]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[service]@[field0]>
Call-ID: [$cid]
CSeq: 21 INVITE
Contact: <sip:[field1]@[local_ip]:[local_port]>
Max-Forwards: 70
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="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" rrs="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]@[field0] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
[routes]
From: <sip:[field1]@[field0]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[service]@[field0]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 21 ACK
Contact: <sip:[field1]@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- 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>
------------------------------------------------------------------------------
_______________________________________________
Sipp-users mailing list
Sipp-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/sipp-users
