Difference between revisions of "Uac msg.xml"

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  published by the Free Software Foundation; either version 2 of the
 
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  Free Software Foundation, Inc.,                                   
 
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  59 Temple Place, Suite 330, Boston, MA  02111-1307 USA           
 
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<!--
 
                                                                   
 
-->
 
 
<!--
 
                  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" start_rtd="true"> 
 
  
 +
<scenario name="Basic Sipstone UAC">
 +
 +
        <send retrans="500" start_rtd="true">
 +
              MESSAGE sip:[call_number]@[remote_ip]:[remote_port] SIP/2.0
 +
              Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 +
              From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
 +
              To: sut <sip:[call_number]@[remote_ip]:[remote_port]>
 +
              Call-ID: [call_id]
 +
              CSeq: 1 MESSAGE
 +
              Contact: sip:sipp@[local_ip]:[local_port]
 +
              Max-Forwards: 70
 +
              Subject: Performance Test
 +
              Content-Type: application/sdp
 +
              Content-Length: [len]
 +
 +
              hello!
 +
        </send>
 +
 +
        <recv response="404" optional="true" rtd="true">
 +
                <action>
 +
                        <exec int_cmd="stop_call"/>
 +
                </action>
 +
        </recv>
 +
 +
        <recv response="200" crlf="true" rtd="true">
 +
        </recv>
 +
 +
        <ResponseTimeRepartition value="10, 50, 100, 150, 200, 500, 1000"/>
 +
</scenario>
  
      MESSAGE sip:[call_number]@[remote_ip]:[remote_port] SIP/2.0
 
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
 
      To: sut <sip:[call_number]@[remote_ip]:[remote_port]>
 
      Call-ID: [call_id]
 
      CSeq: 1 MESSAGE
 
      Contact: sip:sipp@[local_ip]:[local_port]
 
      Max-Forwards: 70
 
      Subject: Performance Test
 
      Content-Type: application/sdp
 
      Content-Length: [len]
 
  
      hello!
 
  
   
+
==Pranala Menarik==
</send>
+
 
+
* [[VoIP: Transaction Oriented Test]]
<recv response="404" optional="true" rtd="true">
+
* [[Test Performance OpenSER mengunakan SIPp]]
+
* [[OpenSER Softswitch]]
<action>
+
* [[OpenSIPS Softswitch]]
<exec int_cmd="stop_call"/>
+
* [[VoIP]]
</action>
 
</recv>
 
<recv response="200" crlf="true" rtd="true">
 
  </recv>
 
 
<!--
 
  definition of the response time repartition table (unit is ms) 
 
-->
 
<ResponseTimeRepartition value="10, 50, 100, 150, 200, 500, 1000"/>
 
</scenario>
 

Latest revision as of 09:27, 14 February 2010

<scenario name="Basic Sipstone UAC">

        <send retrans="500" start_rtd="true">
              MESSAGE sip:[call_number]@[remote_ip]:[remote_port] SIP/2.0
              Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
              From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
              To: sut <sip:[call_number]@[remote_ip]:[remote_port]>
              Call-ID: [call_id]
              CSeq: 1 MESSAGE
              Contact: sip:sipp@[local_ip]:[local_port]
              Max-Forwards: 70
              Subject: Performance Test
              Content-Type: application/sdp
              Content-Length: [len]

              hello!
        </send>

        <recv response="404" optional="true" rtd="true">
                <action>
                        <exec int_cmd="stop_call"/>
                </action>
        </recv>

        <recv response="200" crlf="true" rtd="true">
        </recv>

        <ResponseTimeRepartition value="10, 50, 100, 150, 200, 500, 1000"/>
</scenario>


Pranala Menarik