[Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message

[Tihomir Culjaga]

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 at gmail.com>wrote:

> On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi at gmail.com>
> wrote:
> > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga at gmail.com>
> wrote:
> >>
> >> sipp_cmd:         sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
> >> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1
> -l
> >> 4000
> >> scenario file:      uac_redirect.xml
> >> FS dialplan:       public.xml
> >> SIP trace:          trace.log
> >
> > The Via definition in your SIPp scenario differs between the INVITE and
> the ACK:
> >
> > INVITE:
> > Via: SIP/2.0/[transport] [local_ip];branch=[branch]
> >
> > ACK:
> > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
> >
> >
> > In the INVITE, you are not adding the [local_port] as you do in the ACK.
> > Just adding the [local_port] in the INVITE makes FreeSWITCH accept the
> ACK.
> > So it seems FS is not checking just the ACK's branch against the
> > INVITE's; it seems it is checking the whole Via header.
> > I don't know if this is in accordance to SIP specs.
> > Another thing, about the way you are calling SIPp: do no use "-sn uac"
> > and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
> > means "use the internal (embedded) scenario named xxx". So this
> > conflicts with the other parameter "-sf" which specifies an external
> > profile.
>
> I mean, an external scenario (file).
>
>  It seems this doesn't cause any problem (probably because in
> > the sipp startup, -sf overrides -sn), but it is misleading.
> >
> > regards,
> > takeshi
> >
>
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