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

Mon Aug 24 15:13:47 PDT 2009

Hello Brian, Dave

Still nothing... i've changed ip_addresses (remote_ip, local_ip) and changed
branch within ACK message to meet INVITE's one....but it is still not
enough...
Also i checked RFC and this is how should it be ... (ACK without contact
taking care to have correct TAGs and branch)... what can it be?

   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
>
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060>
   Call-ID: 1-7079 at 10.4.4.252
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length:   131

   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 325 bytes to udp/[10.4.4.252]:5060 at 21:56:08.152812:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060>
   Call-ID: 1-7079 at 10.4.4.252
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0

   ------------------------------------------------------------------------
send 718 bytes to udp/[10.4.4.252]:5060 at 21:56:08.159929:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
>;tag=cFS6jHj9DgjjF
   Call-ID: 1-7079 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
recv 342 bytes from udp/[10.4.4.252]:5060 at 21:56:08.160166:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7079-1-0
   From: 22222238515000403 <sip:22222238515000403 at 110.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
>;tag=cFS6jHj9DgjjF
   Call-ID: 1-7079 at 10.4.4.252
   CSeq: 1 ACK
   Max-Forwards: 70
   Content-Length: 0

   ------------------------------------------------------------------------
send 718 bytes to udp/[10.4.4.252]:5060 at 21:56:08.661299:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
>;tag=cFS6jHj9DgjjF
   Call-ID: 1-7079 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

here is a scenario i use:

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

<scenario name="Basic Sipstone UAC">
  <send retrans="500">
    <![CDATA[

      INVITE sip:[service]@[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip];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">
  </recv>

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

---------- Forwarded message ----------
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Date: Mon, 24 Aug 2009 15:42:31 -0500
Subject: Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand
ACK message
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[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

Use that.. your scenario has some hard coded IP's in the fields that
shouldn't be there.

/b

On Aug 24, 2009, at 3:37 PM, Tihomir Culjaga wrote:

Hello Brian,

it doesn't work .. tried this today as well:
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