Hi all,

I would like to use SIPp to do stress testing against our Asterisk server.
We would like to test the scenario of logging into Asterisk queue by dialing
5000 with authentication, then after a few seconds, we will pick it up
automatically using eyebeam.  Therefore I tried to develop my own test case
with the demo XML files in SIPp.

I've attached my XML file and my CSV.  However, I keep getting unexpected
message on 407.  Can someone give me a hand on this?
Below is the SIPp output.


  Call-rate(length)     Port   Total-time  Total-calls  Remote-host
   1.0(0 ms)/5.000s   5062      20.98 s            4  10.31.0.51:5060(UDP)

  0 new calls during 0.970 s period      1 ms scheduler resolution
  0 calls (limit 5)                      Peak was 1 calls, after 5 s
  0 Running, 0 Paused, 0 Woken up
  20 out-of-call msg (discarded)
  1 open sockets
  0 Total RTP pckts sent                 0.000 last period RTP rate (kB/s)

                                 Messages  Retrans   Timeout
Unexpected-Msg
      INVITE ---------->         4         0         0
         407 <----------         0         0                   4
         ACK ---------->         0         0
      INVITE ---------->         0         0         0
         100 <----------         0         0                   0
         180 <----------         0         0                   0
         200 <----------  E-RTD1 0         0                   0

Regards,
Johnny

Attachment: test.csv
Description: Binary data

<?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 'uac' scenario with pcap (rtp) play           -->
<!--                                                                    -->

<scenario name="UAC with media">
  <!-- 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]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: [field1] <sip:[EMAIL PROTECTED]:[local_port]>;tag=[pid]SIPpTag09[call_number]
      To: [field0] <sip:[EMAIL PROTECTED]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:[EMAIL PROTECTED]:[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[local_ip_type] [local_ip]
      t=0 0
      m=audio [auto_media_port] RTP/AVP 8 101
      a=rtpmap:8 G729/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-11,16

    ]]>
  </send>

  <recv response="407" auth="true">
  </recv>

  <send>
    <![CDATA[

      ACK sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: [field1] <sip:[EMAIL PROTECTED]:[local_port]>;tag=[pid]SIPpTag09[call_number]
      To: [field0] <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>

  <send retrans="500">
    <![CDATA[

      INVITE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[pid]SIPpTag09[call_number]
      To: sut <sip:[EMAIL PROTECTED]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 2 INVITE
      Contact: sip:[EMAIL PROTECTED]:[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[local_ip_type] [local_ip]
      t=0 0
      m=audio [auto_media_port] RTP/AVP 8 101
      a=rtpmap:8 G729/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-11,16

    ]]>
  </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" crlf="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:[EMAIL PROTECTED]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[pid]SIPpTag09[call_number]
      To: sut <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 2 ACK
      Contact: sip:[EMAIL PROTECTED]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

  <!-- Play a pre-recorded PCAP file (RTP stream)                       -->
  <nop>
    <action>
      <exec play_pcap_audio="pcap/g729.pcap"/>
    </action>
  </nop>

  <!-- Pause 8 seconds, which is approximately the duration of the      -->
  <!-- PCAP file                                                        -->
  <pause milliseconds="8000"/>

  <!-- Play an out of band DTMF '1'                                     -->
  <nop>
    <action>
      <exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
    </action>
  </nop>

  <pause milliseconds="1000"/>

  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
  <send retrans="500">
    <![CDATA[

      BYE sip:[EMAIL PROTECTED]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[pid]SIPpTag09[call_number]
      To: sut <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 2 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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