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[tculjaga at gmail.com]

Hello Takeshi,

Thanks for your hint... it worked out... so to be precise:

VIA header of both INVITE and ACK messages MUST be identical (IP:PORT + branch)... and you are right... it might not be according to SIP specification. Anyhow, i get FS understand my ACK message.


Finally, here is what i used and I'm getting some poor results .. but this is another topic


Thanks for your help.
Tihomir.


sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000


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


<scenario name="Basic Sipstone UAC">
  <send retrans="500" start_rtd="1" start_rtd="2">

    <![CDATA[

      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      Max-Forwards: 70
      Contact: <sip:[field1]@[local_ip]>
      From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
      To: [service] <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      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
      a=rtpmap:0 PCMU/8000

    ]]>
  </send>

  <recv response="100"
        optional="true" rtd="1">
  </recv>


  <recv response="302" rtd="2">
  </recv>

  <send>
    <![CDATA[

      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
      From: [field1] <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number]
      To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Max-Forwards: 70
      Content-Length: 0

    ]]>
  </send>

  <!-- 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>



On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi@gmail.com (mayamatakeshi@gmail.com)> wrote:

[mike at jerris.com]

I beleive this is following the right rfc rules for dialog matching. If it is not, please open up a bug on jira.freeswitch.org with references of what exactly is not right.


Mike

On Aug 25, 2009, at 2:51 AM, Tihomir Culjaga <tculjaga@gmail.com (tculjaga@gmail.com)> wrote:

[anthony.minessale at g...]

I wish I had a nickel for every guy struggling with sipp load testing vs real world traffic.


On Tue, Aug 25, 2009 at 1:51 AM, Tihomir Culjaga <tculjaga@gmail.com (tculjaga@gmail.com)> wrote:

[bjbrashier at gmail.com]

Well, you'd have another nickel from over here, then. 

If I can get this working before I'm tasked with something else I'll write up something more on the wiki about "Freeswitch and SIPp", but I'm not sure I'll get that chance.


BB

On Tue, Aug 25, 2009 at 11:05 AM, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:

[anthony.minessale at g...]

here's the one i use for making a call waiting x seconds and hanging up
http://www.freeswitch.org/eg/load_test/dft_cap.xml

This requires that the sipp terminate all the calls.

careful with sipp, it's like a roach motel, you can get stuck trying to make it work and never get it to produce real-life situations.



On Tue, Aug 25, 2009 at 3:23 PM, Bradley Brashier <bjbrashier@gmail.com (bjbrashier@gmail.com)> wrote: