Hi all,
 
Im relatively new to sipp, but I kinda like how it works. I played with scapy a 
lot to build scenarios and traffic, but sipp is the tool for VOIP!
 
Let me come to my question.
 
< VM Ubuntu RECEIVER *.66 > - - - - - - - - - - - - -  < VM Ubuntu SENDER *.77>
 
What I want to do is, I want to set a invite first, and then have a call for 
like 20 secs, and then terminate the call with a BYE. I did accomplish to 
REGISTER to a 3CX server. But I cant get it the 2 VM’s with sip talk to each 
other. Im busy like in days to get it work, but I cant see what im doing wrong. 
Here is what im running on the 2 VM’s.
The receiver starts and waits for INVITE, the SENDER sends INVITE, but it looks 
like the RECEIVER in server mode, ignores the packets.
 
 
RECEIVER 192.168.25.66
 
Command :
 
sipp -nd -sf case-receiver-uas.xml -inf case-receiver.csv -mp 5060 -rtp_echo 
-trace_stat -trace_err -trace_msg
 
 
csv:
 
SEQUENTIAL
2566;2566;
 
 
XML:
 
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="RECEIVER-">
  <recv request="INVITE">
  </recv>
  <send retrans="500">
    <![CDATA[
      SIP/2.0 100 Trying
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
    ]]>
  </send>
</scenario>
 
 
 
 
SENDER 192.168.25.77
 
 
Command:
sipp 192.168.25.66:5060 -nd -sf case-uac.xml -inf case.csv -i 192.168.25.77 -m 
1 -l 1 -r 1 -trace_stat -trace_err -trace_msg
 
 
csv:
 
SEQUENTIAL
2566;2566;
 
 
XML:
 
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="case2">
  <pause milliseconds="2000"/>  <send retrans="500">
    <![CDATA[
      INVITE sip:[field0]@[remote_ip] SIP/2.0
      Content-Type: application/sdp
      Contact: <sip:[field1]@[local_ip]>
      Allow: ACK,INFO,BYE,CANCEL,INVITE,OPTIONS,NOTIFY,PRACK,UPDATE
      Supported: replaces, timer, norefersub
      Session-Expires:14400
      MIN-SE:181
      User-Agent: SIPp/Win32
      Content-Length:[len]
      Max-Forwards:70
      From: <sip:[field1]@[local_ip]>;tag=[call_number]
      To: <sip:[field0]@[remote_ip]>
      CSeq: [cseq] INVITE
      Call-ID: [call_id]
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
     
      v=0
      o=[field1] 1785200710 1 IN IP4 [local_ip]
      s=-
      c=IN IP4 [media_ip]
      t=0 0
      m=audio [auto_media_port] RTP/AVP 8 18 0 4
      a=rtpmap:8 PCMA/8000
      a=rtpmap:18 G729/8000
      a=rtpmap:0 PCMU/8000
      a=rtpmap:4 G723/8000
    ]]>
  </send>
 
 
 
 
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