Hi All,

I'm contacting you because I've a strange issue with PCAP and RTP packets.
I'm sending a PCAP file to be reproduced when I run sipp.

Unfortunately, sometime I can heard noise and tones but sometimes not.

I don't understand very well this problem and I don't know if my script is
wrong or I am missing something.
Because it works sometimes as I said.

I copied my script below and the command line I execute to run it.

If anybody can help me I will appreciate that.

SCRIPT:

<?xml version="1.0" encoding="ISO-8859-1" ?>

<scenario name="register_client">
  <send retrans="500">
    <![CDATA[
      REGISTER sip:[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[EMAIL PROTECTED]>;tag=[call_number]
      To: <sip:[EMAIL PROTECTED]>
      Call-ID: [call_id]
      CSeq: 1 REGISTER
      Contact: sip:[EMAIL PROTECTED]:[local_port]
      Max-Forwards: 5
      Expires: 3600
      User-Agent: SIPp/Linux
      Content-Length: 0
    ]]>
  </send>

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

  <recv response="401" auth="true">
  </recv>

  <send retrans="500">
    <![CDATA[
      REGISTER sip:[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[EMAIL PROTECTED]>;tag=[call_number]
      To: <sip:[EMAIL PROTECTED]>
      Call-ID: [call_id]
      CSeq: 2 REGISTER
      Contact: sip:[EMAIL PROTECTED]:[local_port]
      [field2]
      Supported: path
      Max-Forwards: 5
      Expires: 3600
      User-Agent: SIPp/Linux
      Content-Length: 0

    ]]>
  </send>

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

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

 <send retrans="500">
    <![CDATA[
      INVITE sip:[EMAIL PROTECTED]<[EMAIL PROTECTED]>SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      [routes]
      Max-Forwards: 70
      Contact: <sip:[EMAIL PROTECTED]:[local_port]>
      To: <sip:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
>
      From: <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
      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] [local_ip]
    t=0 0
       m=audio [auto_media_port] RTP/AVP 107 119 100 106 6 0 97 105 98 8 102
3 5 101
    a=rtpmap:0 PCMU/8000
       a=rtpmap:107 BV32/16000
       a=rtpmap:119 BV32-FEC/16000
       a=rtpmap:100 SPEEX/16000
       a=rtpmap:106 SPEEX-FEC/16000
       a=rtpmap:97 SPEEX/8000
       a=rtpmap:105 SPEEX-FEC/8000
       a=rtpmap:98 iLBC/8000
       a=rtpmap:102 L16/16000
       a=rtpmap:101 telephone-event/8000
    ]]>
  </send>

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

<recv response="180" optional="true">
  </recv>

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

 <send>
   <![CDATA[

     ACK sip:[EMAIL PROTECTED]<[EMAIL PROTECTED]>SIP/2.0
     [routes]
     Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
     To: sut <sip:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
>;[peer_tag_param]
     From: sipp <sip:[EMAIL PROTECTED]:
[local_port]>;tag=[call_number]

     Call-ID: [call_id]
     CSeq: 1 ACK
     Content-Length: 0
   ]]>
 </send>

<nop>
  <action>
    <exec play_pcap_audio="/tmp/sipp.svn/pcap/g711a.pcap"/>
  </action>
</nop>
<pause milliseconds="8000"/>

</scenario>


COMMAND LINE
./sipp -sn uac 172.27.82.43 -i 172.27.82.45 -sf
/tmp/goosip/register_client222.xml -inf /tmp/goosip/register_client222.csv
-m 1 -r 1 -rp 1000 -d 1000 -auth_uri 172.27.82.43 -trace_err -trace_msg

THANKS!!!
martin
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