Hi sipp expect:
can you tell me how to use sipp to simulator the uas which act as as both
registrar receiving sip endpoint OPTIONS heartbeat message and b2bua call based
on tcp?
for the following scenarios, probably sip gateway sent out the REGISTER ,
OPTIONS, INVITE message in disorder, I try it based on udp is ok because if
firstly received the first OPTIONS(recv register timuout jump to recv OPTIONS),
udp transaction retransmit the OPTIONS , the gap is shorter, at this time,
match. but based on tcp per dialog, it's nok because if sip gateway only send
request OPTIONS in tcp stream, the gap for retransmit the OPTIONS is decided by
tcp layer and the gap is longer.
sip gateway----------REGISTER------------------>sipp
uas(registrar+b2bua)
|<------------200 ok--------------------|
|-----------OPTIONS----------------->|
|<-------------200 ok-------------------|
|-------------INVITE------------------->|--------------------INVITE----------->|
|<------------100 tring-----------------|
|<-------------180 ring-----------------|
|------------.......------------------------|
-----Original Message-----
From: Charles P Wright [mailto:[email protected]]
Sent: 2008年10月17日 3:01
To: ZHOU Gaofeng A
Cc: [email protected]
Subject: Re: [Sipp-users] sipp start_rtd and rtp issue!
Your nop is not terminated with a </nop> (or alternatively <nop rtd="true"
/>).
Charles
"ZHOU Gaofeng A" <[email protected]>
10/16/2008 05:13 AM
To
<[email protected]>
cc
Subject
[Sipp-users] sipp start_rtd and rtp issue!
Hi :
who can help me?, Now I try to write a xml scenario simulated registrar
server, but I cannot run it ok with the following error. How can I modify it?
C:\Program Files\SIPp>sipp -d 10 -i 135.251.25.238 -p 5060 -sf
D:\call-flow\pack\ registrar.xml
2008-10-16 17:04:52:334 1224147892.334594: You have started
Response Time
Duration 1, but have never stopped it!.
C:\Program Files\SIPp>
registrar.xml:
<?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 'branchs' scenario. -->
<!-- -->
<scenario name="Registrar Server">
<!--<recv request="REGISTER" start_rtd="true"> -->
<recv request="REGISTER" start_rtd="true">
<action>
<ereg regexp="^5[0-9]*[0-9]$" search_in="hdr" header="Contact:"
check_it="true" assign_to="4"/>
<ereg regexp="^\<?.*\>?$" search_in="hdr" header="From:"
check_it="true" assign_to="5"/>
</action>
</recv>
<send>
<![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]>;expires=3600
Content-Length: 0
Allow-Events: reg
P-Associated-URI: [$5]
Path:
<sip:pcsf-stdn.imsgroup0-000.fs5k8.shanghai.com:5060;lr;bidx=0>
]]>
</send>
<nop rtd="true">
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="1000, 1040, 1080, 1120, 1160, 1200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="1000, 1100, 1200, 1300, 1400"/>
</scenario>
Thanks!
Jack
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