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#150 sipp v3.4.1 memory leak issue

v3.4
open
nobody
None
5
2015-05-27
2015-05-20
Brad Hong
No

When we started ipv6 load test with sipp v3.4.1, there was critical memory leak issue at UAS side. sipp process only is using up more than gigabyte of memory. I don't know what is the problem. it might be the problem of sipp v3.4.1 itself or UAS scenario problemn but i am not sure.

see below UAS scenario


<scenario name="Basic Sipstone UAC">


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

  INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
  Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch];rport
  From: <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
  To: <sip:[field1]@[remote_ip]:[remote_port]>
  Call-ID: [call_id]
  CSeq: 1 INVITE
  Contact: sip:1023;tgrp=1111@[local_ip]:[local_port]
  Max-Forwards: 70
  Subject: Performance Test
  Supported: 100rel
  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 8 101
  a=rtpmap:0 pcmu/8000
  a=sendrecv
  a=rtcp:5377
  a=maxptime:20
  a=ptime:20
  a=rtpmap:8 PCMA/8000
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-15,36

]]>

</send>

<recv response="100" optional="true">
</recv>

<recv response="200">
</recv>



<send crlf="true">
<![CDATA[

  ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
  Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  From: <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
  To: <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
  Call-ID: [call_id]
  CSeq: 1 ACK
  Contact: sip:sipp@[local_ip]:[local_port]
  Max-Forwards: 70
  Subject: Performance Test
  Content-Length: 0

]]>

</send>




<pause milliseconds="15000"/>


<send retrans="500">
<![CDATA
BYE sip:[field1@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sip:[field0]@[local_ip]:[local_port];tag=[call_number]
To: sip:[field1]@[remote_ip]:[remote_port][peer_tag_param]
Call-ID: [call_id]
CSeq: 3 BYE
Contact: sip:sipp;tgrp=1111@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

]]>

</send>

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


<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>


<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

</scenario>

Discussion

  • Rob Day

    Rob Day - 2015-05-20

    Can you raise this at https://github.com/SIPp/sipp/issues, including the command line and any CSV files you were using to run SIPp?

    It also looks like you have a UAC scenario, not a UAS one - have you posted the right one?

     

    • Brad Hong

      Brad Hong - 2015-05-20

      Hi Rob
      I posted uac scenario as you mentioned. it's my mistake. :)
      Here is my answer for your question.

      1. uac command line: ../sipp -r 300 -rp 1000 -inf usr.csv -sf uac.xml
        -i [fd00:0:0:7ea1:10:1:1:80] -trace_err [fd00:0:0:7ea1:10:1:1:91]:5060

      2. uas command line: ../sipp -sn uas -trace_err -sf mrf.xml -p 5010 -i
        [fd00:0:0:7ea1:10:1:1:80]

      3. usr.csv file
        RANDOM,PRINTF=300
        01088880%03d;01099990%03d

      4.uas scenario file
      <?xml version="1.0" encoding="ISO-8859-1" ?>
      <!DOCTYPE scenario SYSTEM "sipp.dtd">

      <scenario name="uas_ct">



      <recv request="INVITE">


      </recv>










      <send>
      <![CDATA[

        SIP/2.0 100 Trying
  [last_Via:]
  [last_From:]
  [last_To:];tag=1234567890_[call_number]
  [last_Call-ID:]
  [last_CSeq:]
  Content-Length: 0

]]>

      </send>


      <send>
      <![CDATA[

        SIP/2.0 200 OK
  [last_Via:]
  [last_From:]
  [last_To:];tag=1234567890_[call_number]
  [last_Call-ID:]
  [last_CSeq:]
  Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, PRACK, UPDATE
  Accept: application/sdp, text/*, application/msml+xml,

      application/moml+xml
      Contact: sip:msml@[local_ip]:[local_port];transport=[transport]
      Content-Length: [len]
      Content-Type: application/sdp

        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 96
  a=rtpmap:0 PCMU/8000
  a=rtpmap:96 telephone-event/8000
  a=fmtp:96 0-15,32,36
  a=sendrecv
]]>

      </send>

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

      <recv request="INFO">
      </recv>

      <send crlf="true">
      <![CDATA[

        SIP/2.0 200 OK
  [last_Via:]
  [last_From:]
  [last_To:]
  [last_Call-ID:]
  [last_CSeq:]
  Accept: application/sdp, text/*, application/msml+xml,

      application/moml+xml
      Contact: sip:msml@[local_ip]:[local_port];transport=[transport]
      Content-Length: [len]
      Content-Type: application/msml+xml

        <?xml version="1.0" encoding="US-ASCII"?>
  <msml version="1.1">
          <result response="200">
          </result>
  </msml>
]]>

      </send>

      <!--send retrans="500">
      <![CDATA
      BYE sip:mrfc@10.1.1.92:5071 SIP/2.0
      Via: [$1
      From: [$3];tag=1234567890_[call_number]
      To: [$2]
      [last_Call-ID:]
      CSeq: 10 BYE
      Contact: sip:sipp;tgrp=1111@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

      ]]>
      </send>

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

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

      <send crlf="true">
      <![CDATA[

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

]]>

      </send>




      <timewait milliseconds="4000"/>


      <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>


      <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

      </scenario>

      2015-05-20 16:47 GMT+09:00, Rob Day arcady-91@users.sf.net:

      Can you raise this at https://github.com/SIPp/sipp/issues, including the
      command line and any CSV files you were using to run SIPp?

      It also looks like you have a UAC scenario, not a UAS one - have you posted
      the right one?

      [bugs:#150] sipp v3.4.1 memory leak issue

      Status: open
      Group: v3.4
      Created: Wed May 20, 2015 04:59 AM UTC by Brad Hong
      Last Updated: Wed May 20, 2015 04:59 AM UTC
      Owner: nobody

      When we started ipv6 load test with sipp v3.4.1, there was critical memory
      leak issue at UAS side. sipp process only is using up more than gigabyte of
      memory. I don't know what is the problem. it might be the problem of sipp
      v3.4.1 itself or UAS scenario problemn but i am not sure.

      see below UAS scenario

      <?xml version="1.0" encoding="ISO-8859-1" ?>
      <!DOCTYPE scenario SYSTEM "sipp.dtd">


















      <scenario name="Basic Sipstone UAC">


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

        INVITE sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
  Via: SIP/2.0/[transport]

      [local_ip]:[local_port];branch=[branch];rport
      From: sip:[field0]@[local_ip]:[local_port];tag=[call_number]
      To: sip:[field1]@[remote_ip]:[remote_port]
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:1023;tgrp=1111@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Supported: 100rel
      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 8 101
  a=rtpmap:0 pcmu/8000
  a=sendrecv
  a=rtcp:5377
  a=maxptime:20
  a=ptime:20
  a=rtpmap:8 PCMA/8000
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-15,36

]]>

      </send>

      <recv response="100" optional="true">
      </recv>

      <recv response="200">
      </recv>



      <send crlf="true">
      <![CDATA[

        ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
  Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  From: <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
  To: <sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]
  Call-ID: [call_id]
  CSeq: 1 ACK
  Contact: sip:sipp@[local_ip]:[local_port]
  Max-Forwards: 70
  Subject: Performance Test
  Content-Length: 0

]]>

      </send>




      <pause milliseconds="15000"/>


      <send retrans="500">
      <![CDATA
      BYE sip:[field1      @[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sip:[field0]@[local_ip]:[local_port];tag=[call_number]
      To: sip:[field1]@[remote_ip]:[remote_port][peer_tag_param]
      Call-ID: [call_id]
      CSeq: 3 BYE
      Contact: sip:sipp;tgrp=1111@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

      ]]>

      </send>

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


      <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>


      <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

      </scenario>

      Sent from sourceforge.net because you indicated interest in
      https://sourceforge.net/p/sipp/bugs/150/

      To unsubscribe from further messages, please visit
      https://sourceforge.net/auth/subscriptions/

       

  • Brad Hong

    Brad Hong - 2015-05-20

    I posted uac scenario as you mentioned. it's my mistake. :)
    Here is my answer for your question.

    1. uac command line: ../sipp -r 300 -rp 1000 -inf usr.csv -sf uac.xml
      -i [fd00:0:0:7ea1:10:1:1:80] -trace_err [fd00:0:0:7ea1:10:1:1:91]:5060

    2. uas command line: ../sipp -sn uas -trace_err -sf mrf.xml -p 5010 -i
      [fd00:0:0:7ea1:10:1:1:80]

    3. usr.csv file
      RANDOM,PRINTF=300
      01088880%03d;01099990%03d

    4.uas scenario file
    <?xml version="1.0" encoding="ISO-8859-1" ?>
    <!DOCTYPE scenario SYSTEM "sipp.dtd">

    <scenario name="uas_ct">



    <recv request="INVITE">


    </recv>










    <send>
    <![CDATA[

      SIP/2.0 100 Trying
  [last_Via:]
  [last_From:]
  [last_To:];tag=1234567890_[call_number]
  [last_Call-ID:]
  [last_CSeq:]
  Content-Length: 0

]]>

    </send>


    <send>
    <![CDATA[

      SIP/2.0 200 OK
  [last_Via:]
  [last_From:]
  [last_To:];tag=1234567890_[call_number]
  [last_Call-ID:]
  [last_CSeq:]
  Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, PRACK, UPDATE
  Accept: application/sdp, text/*, application/msml+xml,

    application/moml+xml
    Contact: sip:msml@[local_ip]:[local_port];transport=[transport]
    Content-Length: [len]
    Content-Type: application/sdp

      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 96
  a=rtpmap:0 PCMU/8000
  a=rtpmap:96 telephone-event/8000
  a=fmtp:96 0-15,32,36
  a=sendrecv
]]>

    </send>

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

    <recv request="INFO">
    </recv>

    <send crlf="true">
    <![CDATA[

      SIP/2.0 200 OK
  [last_Via:]
  [last_From:]
  [last_To:]
  [last_Call-ID:]
  [last_CSeq:]
  Accept: application/sdp, text/*, application/msml+xml,

    application/moml+xml
    Contact: sip:msml@[local_ip]:[local_port];transport=[transport]
    Content-Length: [len]
    Content-Type: application/msml+xml

      <?xml version="1.0" encoding="US-ASCII"?>
  <msml version="1.1">
          <result response="200">
          </result>
  </msml>
]]>

    </send>

    <!--send retrans="500">
    <![CDATA
    BYE sip:mrfc@10.1.1.92:5071 SIP/2.0
    Via: [$1
    From: [$3];tag=1234567890_[call_number]
    To: [$2]
    [last_Call-ID:]
    CSeq: 10 BYE
    Contact: sip:sipp;tgrp=1111@[local_ip]:[local_port]
    Max-Forwards: 70
    Subject: Performance Test
    Content-Length: 0

    ]]>
    </send>

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

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

    <send crlf="true">
    <![CDATA[

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

]]>

    </send>




    <timewait milliseconds="4000"/>


    <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>


    <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

    </scenario>

     

  • Brad Hong

    Brad Hong - 2015-05-27

    please give me the clue or updates. is this sipp 3.4.1 bug, right?

     

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