Hi,
I am trying to do the following scenario but I am not able to get the
INVITE msg to go out. Am I missing something or is this something that is
not supported in SIPP
remote sipp
INVITE ---------------------------------->
<----------------------------------------200 OK
ACK-------------------------------------->
<----------------------------------------INVITE ( HOLD>
200 OK--------------------------------->
<-------------------------------------------ACK
For some reason the INVITE(HOLD) is not going out at all. Is there a run
option I am missing. I am attaching the Scenario file for your reference.
--
cheers
ramesh
<?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. -->
<label id="1" />
<recv request="INVITE" crlf="true" rrs="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]>
[route]
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:]
[route]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=- 2209592090 0 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 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
]]>
</send>
<recv request="ACK" optional="true" rtd="true" crlf="true">
</recv>
<recv request="INVITE" crlf="true" >
</recv>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
[route]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=- 2209592090 0 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 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
]]>
</send>
<recv request="ACK" optional="true" rtd="true" crlf="true">
</recv>
<send retrans="500">
<![CDATA[
INVITE sip:[EMAIL PROTECTED]:[remote_port];transport=[transport] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: 41009 <sip:[EMAIL PROTECTED]>;tag=[call_number]
To: 40012 <sip:[EMAIL PROTECTED]:[remote_port]>;[peer_tag_param]
[last_Call-ID:]
CSeq: [cseq] INVITE
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Max-Forwards: 70
Route: <sip:135.9.97.22:5060;transport=TCP;lr>
Allow: INVITE,ACK,BYE,CANCEL,NOTIFY,REFER,OPTIONS,SUBSCRIBE,PRACK,UPDATE
Supported: Histinfo
P-Asserted-Identity: sip:[EMAIL PROTECTED]
[route]
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 2209592090 2 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 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendonly
]]>
</send>
<recv response="200" rtd="true" rrs="true">
</recv>
<send>
<![CDATA[
ACK sip:[EMAIL PROTECTED]:[remote_port];transport=[transport] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: 41009 <sip:[EMAIL PROTECTED]>;tag=[call_number]
To: 40012 <sip:[EMAIL PROTECTED]:[remote_port]>;tag=[peer_tag_param]
[last_Call-ID:]
CSeq: [cseq] ACK
Contact: <sip:[EMAIL PROTECTED]>
Max-Forwards: 70
[route]
Content-Length: 0
]]>
</send>
<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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