Hi
I have made changes in the script , so u can use the script and it may run
this time.
On Tue, May 27, 2008 at 5:56 PM, <[EMAIL PROTECTED]> wrote:
> Hi All,
>
> I am trying to simulate simple call with SIPp. I had created a script which
> can send Invite to the softswitch, and i can able to receive 1xx and 200 Ok
> from the farend,
> but once i receive 200OK my script is not able to send ACK ..I am attaching
> the call secnario script. If anybody tells what's wrong(not able to send
> ACK) in my script??
> please help me.
>
> Regards
> Srinivas
>
>
>
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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 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- 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[
INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0
From: <sip:[EMAIL PROTECTED];user=phone>;tag=8dde3b2c31;epid=B8DDA65423
To: <sip:[EMAIL PROTECTED];user=phone>
Call-ID: a96c6a5a-c5d2-4d93-a56d-4454d7e7f063
CSeq: 1 INVITE
MAX-FORWARDS: 70
Supported: 100rel
USER-AGENT: RTCC/3.0.0.0 MediationServer
Via: SIP/2.0/TCP 192.168.48.199:3899;branch=z9hG4bKfed0a31a
Contact:
<sip:192.168.48.199:5060;maddr=192.168.48.199;transport=Tcp;ms-opaque=4e1752980aafe5b7>
Allow: UPDATE,ACK,CANCEL,BYE,INVITE
Content-Type: application/sdp;charset=utf-8
Content-Length: [len]
v=0
o=- 99 1 IN IP4 192.168.48.199
s=-
t=0 0
m=audio 5404 RTP/AVP 0 8 101 111
c=IN IP4 192.168.48.199
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:111 X-nt-inforeq/8000
a=ptime:20
a=maxptime:20
a=sendrecv
]]>
</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">
</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 [next_url] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: sut <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<pause/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE [next_url] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: sut <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 4 BYE
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" 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>
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