Hi Vanessa and all sipp expert:
I meet some issues for sipp 3pcc extension mode, could you give me the help and
the support? I want to know if the extracted vlaue stored to one varible for
sip head by regular expression in twin sipp master instance, then by SendCmd
sending to sipp slave instance, in sipp slave instance, can I load the value
stored in the varible in some sip message, I had try it, extracted value for
sip message body(SDP) can be used in sipp slave instance.
I extracted the the caller and callee number from "From" ($5) and "To"($4)
head by regular expression then send to another 3pcc extension slave SIPp
instance, then checked the log that has sent to slave, but it cannot replace
the varible in "INVITE" sip head with extracted value, but the vlaue extracted
for $1 (SDP) can be used in the "INVITE" message.
the following is sipp slave instance log
----------------------------------------------- 2009-03-23 11:11:11:53.455
TCP control message received [210] bytes :
Call-ID: [email protected] <BLOCKED::mailto:[email protected]>
From: m
55557772
55557774
m=audio 50000 RTP/AVP 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:31C397D875C040A5A805A758906BDF5C
----------------------------------------------- 2009-03-23 11:11:11:53.455
UDP message sent (1009 bytes):
INVITE sip:[email protected]:42232 SIP/2.0
Via: SIP/2.0/UDP 135.251.25.238:50000;branch=z9hG4bK-3084-1-1
From: <sip:@135.251.205.87;user=phone>;tag=3084-sipp3-1
To: <sip:[email protected]>
Call-ID: [email protected] <mailto:[email protected]>
CSeq: 1 INVITE
Contact: <sip:135.251.25.238:50000>
User-Agent: ALCATE ISAM
Max-Forwards: 68
Accept: application/broadsoft,application/sdp,application/simple-message-summary
Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE
Content-Length: 238
P-Asserted-Identity: <sip:+8621
@fs5k8.shanghai.com;user=phone>
Request-Disposition: no-fork
Record-Route: <sip:135.251.25.238:50000;lr;lsstag=pt-15674-15674>
P-Called-Party-ID: <sip:[email protected];user=phone>
Content-Type: application/sdp
v=0
o=sipp 53655765 2353687637 IN IP4 135.251.205.87
s=-
c=IN IP4 135.251.205.87
t=0 0
m=audio 50000 RTP/AVP 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:31C397D875C040A5A805A758906BDF5C
the original scenario xml (master)
<scenario name="FS5000-call-flow">
<recv request="INVITE">
<action>
<ereg regexp="m=audio.*" search_in="msg" assign_to="1"/>
<ereg regexp="[0-9]{8}" search_in="hdr" header="To:" check_it="true"
assign_to="4"/>
<ereg regexp="5555[0-9]{4}" search_in="hdr" header="From:"
check_it="true" assign_to="5"/>
</action>
</recv>
<sendCmd dest="s1">
<![CDATA[
Call-ID: [call_id]
From: m
[$4]
[$5]
[$1]
]]>
</sendCmd>
.........
the original scenario xml (slave)
<recvCmd src="m" start_rtd="1">
<action>
<ereg regexp="m=audio.*" search_in="msg" assign_to="1"/>
<ereg regexp="[0-9]{8}" search_in="hdr" header="To:" check_it="true"
assign_to="4"/>
<ereg regexp="5555[0-9]{4}" search_in="hdr" header="From:"
check_it="true" assign_to="5"/>
</action>
</recvCmd>
<send retrans="1000" rtd="1">
<![CDATA[
INVITE sip:[field5 line=1][...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[...@[remote_ip];user=phone>;tag=[pid]-sipp3-[call_number]
To: <sip:[field5 line=1][...@[remote_ip]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: <sip:[local_ip]:[local_port]>
User-Agent: ALCATE ISAM
Max-Forwards: 68
Accept:
application/broadsoft,application/sdp,application/simple-message-summary
Allow:
INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE
Content-Length: [len]
P-Asserted-Identity: <sip:[field5
line=0][[email protected];user=phone>
Request-Disposition: no-fork
Record-Route: <sip:[local_ip]:[local_port];lr;lsstag=pt-15674-15674>
P-Called-Party-ID: <sip:[field5 line=1][[email protected];user=phone>
Content-Type: application/sdp
v=0
o=sipp 53655765 2353687637 IN IP4 [remote_ip]
s=-
c=IN IP4 [remote_ip]
t=0 0
[$1]
]]>
</send>
________________________________
From: Vanessa Tejada Muñoz [mailto:[email protected]]
Sent: 2008年10月16日 20:52
To: ZHOU Gaofeng A
Cc: [email protected]
Subject: Re: [Sipp-users] sipp start_rtd and rtp issue!
Only an idea ok?
Why don not you change rtd value? I mean, for example 3 instead of true.
On Thu, Oct 16, 2008 at 11:13 AM, ZHOU Gaofeng A
<[email protected]> wrote:
Hi :
who can help me?, Now I try to write a xml scenario simulated
registrar server, but I cannot run it ok with the following error. How can I
modify it?
C:\Program Files\SIPp>sipp -d 10 -i 135.251.25.238 -p 5060 -sf
D:\call-flow\pack\
registrar.xml
2008-10-16 17:04:52:334 1224147892.334594: You have started
Response Time
Duration 1, but have never stopped it!.
C:\Program Files\SIPp>
registrar.xml:
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or
-->
<!-- modify it under the terms of the GNU General Public License as
-->
<!-- published by the Free Software Foundation; either version 2 of the
-->
<!-- License, or (at your option) any later version.
-->
<!--
-->
<!-- This program is distributed in the hope that it will be useful,
-->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of
-->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
-->
<!-- GNU General Public License for more details.
-->
<!--
-->
<!-- You should have received a copy of the GNU General Public License
-->
<!-- along with this program; if not, write to the
-->
<!-- Free Software Foundation, Inc.,
-->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
-->
<!--
-->
<!-- Sipp default 'branchs' scenario.
-->
<!--
-->
<scenario name="Registrar Server">
<!--<recv request="REGISTER" start_rtd="true">
-->
<recv request="REGISTER" start_rtd="true">
<action>
<ereg regexp="^5[0-9]*[0-9]$" 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
-------------------------------------------------------------------------
This SF.Net email is sponsored by the Moblin Your Move Developer's
challenge
Build the coolest Linux based applications with Moblin SDK & win great
prizes
Grand prize is a trip for two to an Open Source event anywhere in the
world
http://moblin-contest.org/redirect.php?banner_id=100&url=/
_______________________________________________
Sipp-users mailing list
[email protected]
https://lists.sourceforge.net/lists/listinfo/sipp-users
--
Vanessa Tejada
------------------------------------------------------------------------------
Apps built with the Adobe(R) Flex(R) framework and Flex Builder(TM) are
powering Web 2.0 with engaging, cross-platform capabilities. Quickly and
easily build your RIAs with Flex Builder, the Eclipse(TM)based development
software that enables intelligent coding and step-through debugging.
Download the free 60 day trial. http://p.sf.net/sfu/www-adobe-com
_______________________________________________
Sipp-users mailing list
[email protected]
https://lists.sourceforge.net/lists/listinfo/sipp-users
