Re: [Sipp-users] Fwd: Reg:Sipp Register and 200(ok) should be optional

[Volkan K. <kumbasar@NETAS.com.tr>]

The “to tag” in your 200 OK response is missing:

<send>
    <![CDATA[

      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:];tag=[pid]         <====
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0

    ]]>
</send>

From: Logeshwaran G [mailto:log...@gm...]
Sent: Thursday, February 09, 2017 11:07 AM
To: Volkan KUMBASAR
Subject: Re: [Sipp-users] Fwd: Reg:Sipp Register and 200(ok) should be optional

Hi VK,
Thanks a lot for your quick response.
I do have one more doubt, as per your advice I have removed the optional from Register.
Then the Register is succeeded but I am not able to make call to other users. Then I have noticed that invite is not reaching properly.
Please find below my xml file. correct me if I am wrong. Waiting for your solution.
issue message is:
                                 Messages  Retrans   Timeout   Unexpected-Msg
  ----------> REGISTER           2         0         0         1

  <---------- 200                2         0
  ----------> INVITE             0         0         0         0

  <---------- 100                0         0
  <---------- 180                0         0
  <---------- 200                0         0         0
  ----------> ACK         E-RTD1 0         0         0         0

              [ NOP ]
  ----------> BYE                0         0         0         0
  <---------- 200                0         0
  [   4000ms] Pause              0                             0
------------------------------ Test Terminated --------------------------------

xml file is:
<?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 'uas' scenario.                       -->
<!--                                                                    -->

<scenario name="Basic UAS responder">

<recv request="REGISTER" crlf="true" rrs="true">
</recv>

<send>
    <![CDATA[

      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0

    ]]>
</send

  <!-- 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 request="INVITE" crlf="true" optional="true" rrs="true">
  </recv>

  <!-- The '[last_*]' keyword is replaced automatically by the          -->
  <!-- specified header if it was present in the last message received  -->
  <!-- (except if it was a retransmission). If the header was not       -->
  <!-- present or if no message has been received, the '[last_*]'       -->
  <!-- keyword is discarded, and all bytes until the end of the line    -->
  <!-- are also discarded.                                              -->
  <!--                                                                  -->
  <!-- If the specified header was present several times in the         -->
  <!-- message, all occurences are concatenated (CRLF seperated)        -->
  <!-- to be used in place of the '[last_*]' keyword.                   -->



 <send>
    <![CDATA[

      SIP/2.0 100 Trying
      [last_Via:]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0

    ]]>
  </send>



  <send>
    <![CDATA[

      SIP/2.0 180 Ringing
      [last_Via:]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>

    ]]>
  </send>


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

      SIP/2.0 200 OK
      Record-Route: <sip:198.8.93.48:5060;lr>,<sip:31.24.230.108:5060;lr>
      [last_Via:]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Type: application/sdp
      Content-Length: [len]

      v=18
      o=mani@198.8.93.48<mailto:mani@198.8.93.48> 0 0 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio 5060  RTP/AVP 18
      a=rtpmap:18 G729/8000

    ]]>
  </send>

  <recv request="ACK"
    rtd="true"
    crlf="true">
  </recv>

  <!-- Play a pre-recorded PCAP file (RTP stream)                       -->
   <nop>
    <action>
      <exec play_pcap_audio="pcap/issue1.pcap"/>
    </action>
   </nop>

 <recv request="BYE">
  </recv>

  <send>
    <![CDATA[

      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      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.               -->
  <timewait 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>

Thanks & Kind Regards,
LogEshwaran G

On Thu, Feb 9, 2017 at 1:24 PM, Volkan KUMBASAR <kum...@ne...<mailto:kum...@ne...>> wrote:
Hi Logeshwaran,

Actually the error message says everything: "<recv> before <send> sequence without a mandatory message. Please remove one 'optional=true' (element 2)..

You have to add a mandatory <recv> before an optional <recv>.  So, your first REGISTER request must be mandatory.

Thanks, v.








From: Logeshwaran G [mailto:log...@gm...<mailto:log...@gm...>]
Sent: Thursday, February 09, 2017 10:48 AM
To: sip...@li...<mailto:sip...@li...>
Subject: [Sipp-users] Fwd: Reg:Sipp Register and 200(ok) should be optional


Hi Team,
I am facing an issue of setting the receiving response of register message as optional one. Please find below my xml file and kindly correct me if I am wrong.


<scenario name="Basic UAS responder">

<recv request="REGISTER" crlf="true" optional="true" rrs="true">
</recv>

<send>
    <![CDATA[

      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0

    ]]>
</send

  <!-- 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 request="INVITE" crlf="true" optional="true" rrs="true">
  </recv>

  <!-- The '[last_*]' keyword is replaced automatically by the          -->
  <!-- specified header if it was present in the last message received  -->
  <!-- (except if it was a retransmission). If the header was not       -->
  <!-- present or if no message has been received, the '[last_*]'       -->
  <!-- keyword is discarded, and all bytes until the end of the line    -->
  <!-- are also discarded.                                              -->
  <!--                                                                  -->
  <!-- If the specified header was present several times in the         -->
  <!-- message, all occurences are concatenated (CRLF seperated)        -->
  <!-- to be used in place of the '[last_*]' keyword.                   -->



 <send>
    <![CDATA[

      SIP/2.0 100 Trying
      [last_Via:]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0

    ]]>
  </send>
Issue is : while I am executing the sipp command
./sipp -sf sipp_uas_pcap_g729.xml -p 5060 -i 198.8.93.48
i am getting  "<recv> before <send> sequence without a mandatory message. Please remove one 'optional=true' (element 2).."

Thanks & Kind Regards,
LogEshwaran G

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Bu e-posta mesajı ve ekleri gönderildiği kişi ya da kuruma özeldir ve gizlidir. Ayrıca hukuken de gizli olabilir. Hiçbir şekilde üçüncü kişilere açıklanamaz ve yayınlanamaz. Eğer mesajın gönderildiği alıcı değilseniz bu elektronik postanın içeriğini açıklamanız, kopyalamanız, yönlendirmeniz ve kullanmanız kesinlikle yasaktır ve bu elektronik postayı ve eklerini derhal silmeniz gerekmektedir. NETAŞ TELEKOMÜNİKASYON A.Ş. bu mesajın içerdiği bilgilerin doğruluğu veya eksiksiz olduğu konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne şekilde olursa olsun içeriğinden, iletilmesinden, alınmasından, saklanmasından ve kullanılmasından sorumlu değildir. Bu mesajdaki görüşler gönderen kişiye ait olup, NETAŞ TELEKOMÜNİKASYON A.Ş.’nin görüşlerini yansıtmayabilir.
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This e-mail and its attachments are private and confidential and intended for the exclusive use of the individual or entity to whom it is addressed. It may also be legally confidential. Any disclosure, distribution or other dissemination of this message to any third party is strictly prohibited. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited, and the e-mail should immediately be deleted. NETAŞ TELEKOMÜNİKASYON A.Ş. makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the transmission, reception, storage or use of such information in any way whatsoever. The opinions expressed in this message are those of the sender and may not necessarily reflect the opinions of NETAŞ TELEKOMÜNİKASYON A.Ş.