this is the xml file for uac that i have configured iam unable to find the
error
please help me out .thanks
-->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[127.0.0.1]:[5060] SIP/2.0
Via: SIP/2.0/[UDP] [127.0.0.1]:[1234];branch=[branch]
From: sipp <sip:sipp@[127.0.0.1]:[1234]>;tag=[call_number]
To: sut <sip:[service]@[127.0.0.1]:[5060]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[127.0.0.1]:[1234]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [141]
v=0
o=user1 53655765 2353687637 IN IP[4] [127.0.0.1]
s=Session sdp
c=IN IP[4] [127.0.0.1]
t=0 0
m=audio [1234] 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:[service]@[127.0.0.1]:[5060] SIP/2.0
Via: SIP/2.0/[transport] [127.0.0.1]:[1234];branch=[branch]
From: sipp <sip:sipp@[127.0.0.1]:[1234]>;tag=[call_number]
To: sut <sip:[service]@[127.0.0.1]:[5060]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[127.0.0.1]:[1234]
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:[service]@[127.0.0.1]:[5060] SIP/2.0
Via: SIP/2.0/[transport] [127.0.0.1]:[1234];branch=[branch]
From: sipp <sip:sipp@[127.0.0.1]:[1234]>;tag=[call_number]
To: sut <sip:[service]@[127.0.0.1]:[5060]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[127.0.0.1]:[1234]
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>
UAS scenario
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[127.0.0.1]:[5060];transport=[UDP]>
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:]
Contact: <sip:[127.0.0.1]:[5060];transport=[UDP]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[4] [127.0.0.1]
s=-
c=IN IP[4] [127.0.0.1]
t=0 0
m=audio [3456] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK" optional="true" rtd="true" crlf="true"></recv>
<recv request="BYE"></recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[127.0.0.1]:[5060];transport=[UDP]>
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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