Hi Chandran,
I have used some scripts for Invite with call establishment. Try using this.
And change the extension number in the script file, I have used 2001. This
worked for me with my ip phone and I established a call.
On Wed, Mar 30, 2011 at 5:37 AM, mayamatakeshi <mayamatake...@gmail.com>wrote:
>
>
> On Wed, Mar 30, 2011 at 3:50 AM, chandan kumar <chandan_...@yahoo.co.in>wrote:
>
>> Hi,
>>
>> Iam pretty new to Sipp tool.Could some one send me the xml script for
>> Outgoing INVITE and also response for Incomming Request
>>
>>
>>
>
> For simple tests, you can just use the embedded uac and uas scenarios:
> sipp -i local_ip -p local_port -sn uac server_address
> sipp -i local_ip -p local_port -sn uas
>
> If you need to test more complex cases, you can always dump a scenario to a
> file
> sipp -sd uac > uac.xml
> and edit it according to your needs. Just read the documentation:
> http://sipp.sourceforge.net/doc/reference.html
> It if very clear and will save you a lot of time.
>
> Also, you can search the mailing lists for scenarios:
> http://www.mail-archive.com/sipp-users@lists.sourceforge.net/msg05136.html
>
>
>
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--
Thank you with regards,
Gopalakrishnan A.N.
VoIP call - sip:sai...@gtalk2voip.com
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!--./sipp -i 192.168.0.83 -p 6060 -sf client1.xml -d 20000 192.168.0.58 -trace_err -->
<scenario name="UAClient">
<send retrans="500">
<![CDATA[
INVITE sip:2001@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:2001@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:2001@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:2001@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 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
a=rtpmap:0 PCMU/8000
]]>
</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 sip:2001@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:2001@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:2001@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:2001@[local_ip]:[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 sip:2001@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:2001@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:2001@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:2001@[local_ip]:[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"/>
<pause milliseconds="5000"/>
</scenario>------------------------------------------------------------------------------
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