Hello Narendra
Nice to hear from you after so many days. Give me your contact number.
Find the attachment for UAS scenario file.
Thanks and Regards
Sanjeeb
________________________________
From: Narendra V [mailto:[EMAIL PROTECTED]
Sent: Fri 6/22/2007 9:16 AM
To: Sanjeeb Dash (WT01 - Telecom Applications and Solutions); [EMAIL
PROTECTED]; [email protected]
Subject: Re: [Sipp-users] re-INVITE from UAS
Hello Sanjeeb,
Please send us your UAS scenario file.
Thanks and Regards,
-Narendra.
[EMAIL PROTECTED] wrote:
Hello Olivier
Thanks for your reply.
I was trying to send a re-INVITE from UAS.
UAC UAS
INVITE -------> INVITE
100 Trying<------ 100 Trying
180 Ringing <----- 180 Ringing
200 OK <------ 200 OK
ACK -------> ACK
<---------re-INVITE
So in the aove scenario when I try to send re-INVITE it dumps. I
beleive that my re-INVITE strucure is workng. So if you can send me a sample
re-INVITE from UAS side. That would be of great help.
Thanks and Regards
Sanjeeb
________________________________
From: Boulkroune, Olivier (Non-HP:Atos Origin) [mailto:[EMAIL PROTECTED]
Sent: Fri 6/22/2007 1:25 AM
To: Sanjeeb Dash (WT01 - Telecom Applications and Solutions);
[email protected]
Subject: RE: [Sipp-users] re-INVITE from UAS
Hello Sanjeeb,
What version of sipp was in use while getting the core dump ? May I see
the scenario(s) involved ? Could you try to reproduce this with the latest
unstable version (sipp.2007-06-21.tar.gz
<http://sipp.sourceforge.net/snapshots/sipp.2007-06-21.tar.gz> ) and analyse
the core using gdb ?
Thanks and regards,
Olivier Boulkroune
________________________________
De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de [EMAIL
PROTECTED]
Envoyé : vendredi 22 juin 2007 05:46
À : [EMAIL PROTECTED]; [email protected]
Objet : Re: [Sipp-users] re-INVITE from UAS
Hello All
Can some body send a UAS file with re-INVITE. I want to send a
re-INVITE from UAS after ACK. I tried but getting coredump.
Thanks and Regards
Sanjeeb
________________________________
From: [EMAIL PROTECTED] on behalf of Bea Chan
Sent: Tue 6/19/2007 5:07 PM
To: [email protected]
Cc: Bea Chan
Subject: [Sipp-users] sipp scenario re-INVITE followed immediately by
BYE
Hi I am new to SIPP and I would appreciate if someone can provide a
SIPp scenario that re-invited followed immediately by BYE such that the call
gets terminated before processing of the re-INVITE completes.
I really appreciat the help.
Regards,
Becky
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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">
<recv request="INVITE" crlf="true">
</recv>
<!-- The '[last_*]' keyword automatically is replaced 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 recieved, 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. -->
<pause milliseconds="100"/>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
[last_Record-Route:]
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_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
[last_Record-Route:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
t=0 0
c=IN IP4 [media_ip]
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK"
rtd="true"
crlf="true">
</recv>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<pause milliseconds="10000"/>
<send retrans="1000" crlf="true">
<![CDATA[
INVITE sip:[EMAIL PROTECTED]:[local_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[EMAIL PROTECTED]:[local_port];otg=8102>;tag=[call_number]
To: <sip:[EMAIL PROTECTED]:5060>[peer_tag_param]
Call-ID: [call_id]
Cseq: [cseq] INVITE
Contact: <sip:[EMAIL PROTECTED]:[local_port]>
Accept: multipart/mixed, application/sdp, application/isup,
application/dtmf, application/dtmf-relay
MIME-Version: 1.0
Unsupported: 100rel
Max-Forwards: 70
Allow: OPTIONS, INVITE, CANCEL, ACK, BYE, PRACK, INFO
Content-Type: multipart/mixed;boundary=sonus-content-delim
Content-Length: [len]
--sonus-content-delim
Content-Type: application/sdp
v=0
o=Sonus_UAC 31928 9834 IN IP4 [local_ip]
s=SIP Media Capabilities
c=IN IP4 192.168.12.249
t=0 0
m=audio 12820 RTP/AVP 0 0
a=rtpmap:0 PCMU/8000
a=rtpmap:0 PCMU/8000
a=sendrecv
--sonus-content-delim
Content-Type: application/isup;base=gr317;version=ansi
Content-Disposition: signal;handling=optional
..H....
.......h6pPv
....U75U=......
--sonus-content-delim--
]]>
</send>
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
[last_Record-Route:]
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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