WireShark Filters with IPaddress & Port

Filter for  ip address:

Ip.addr ==  148.147.XX.XX

Ip.src  == 148.147.XX.XX   (Source IP address filter)

Ip.dst == 148.147.XX.XX      (Destination IP address filter)

Ip.src == 148.147.XX.XX &&  Ip.dst == 148.147.XX.XX  (Source and destination IP address filter)

Filter for Port:

Tcp.port == XXX  (TCP port filter)

Udp.port == XXX  (UDP Port filter)

Tcp.srcport==XXX (TCP port filter for source)

Udp.srcport==XXX  (TCP port filter for source)

Tcp.dstport==XXX(TCP port filter for destination)

Udp.dstport==XXX(UDP port filter for destination)

SIPp Commands

SIPp Commands:

Register  UAC

sudo ./sipp 1XX.1XX.2XX.2XX(Server IP) -t u1 -sf  Register.xml -inf  Register.csv -i 1XX.1XX.2XX.1XX (Local IP) -m 1 -trace_err -trace_msg -trace_logs 

Invite UAC

sudo ./sipp 1XX.1XX.2XX.2XX(Server IP) -t u1 -sf  Invite.xml -inf  Invite.csv -i 1XX.1XX.2XX.1XX (Local IP) -m 1 -trace_err -trace_msg -trace_logs

Register UAS

./sipp -sn Register.xml

Invite UAS

./sipp -sn Invite.xml

Use Command for 1000 Calls:-

sudo ./sipp 1XX.1XX.2XX.2XX(Server IP) -t u1 -sf  Invite.xml -inf  Invite.csv -i 1XX.1XX.2XX.1XX (Local IP) -m 1000 -l 10 -r 10 -rp 10000 -trace_err -trace_msg -trace_logs

-m   = Max number of calls you want generate (100,1000...etc)

-l    = Max Concurrent calls you want to generate , If Max number of call is high it may take more time to generate 

Example:-

-r   = Set the Call rate per second

-rp  =  rate period in milliseconds for call rate.

Example:-   -r 10  -rp 10000   =>   This means 10 calls for every 10 secs.

Register.XML UAS as Server Mode

Register.XML UAS as Server Mode


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

<scenario name="Register UAS">
<recv request="REGISTER" />
<send><![CDATA[
      SIP/2.0 401 Authorization Required
      [last_Via:]
      [last_From:]
      [last_To:];tag=[pid]SIPpTag01[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      WWW-Authenticate: Digest realm="test.example.com", nonce="47ebe028cda119c35d4877b383027d28da013815"
      Content-Length: [len]

    ]]>
  </send>
  <recv request="REGISTER" />
  </recv>
  <send retrans="500">
    <![CDATA[

       SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Type: application/sdp
      Content-Length: 136
    ]]>
  </send>

SIPP(Invite.xml) as UAS Server mode

UAS  -  INVITE.xml :



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

<!-- You will need to compile SIPp with OpenSSL support 'make ossl' to use this call scenario -->

<!-- Execute this script with SIPp using the following command assuming your UAS is 10.0.0.10 -->
<!-- Replace 10.0.0.10 with your SIP proxy’s address. The command will generate 10 calls (-r) per 10000 -->
<!-- milliseconds (-rp), max 100 concurrent calls (-l) and make a max of 100000 calls (-m) -->

<!-- ./sipp 10.0.0.10 -sf invite-auth-sdp-nomedia.xml -inf user-accounts.csv -m 100000 -l 100 -r 10 -rp 10000 -->

<scenario name="UAS INVITE">
 <recv request="INVITE" crlf="true">
      </recv>
 
   <!-- send 180 then trying if variable 3 is set -->
    <send>
    <![CDATA[

       SIP/2.0 180 Ringing
       [last_Via:]
       [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0

    ]]>
    </send>

  <send retrans="500">
    <![CDATA[

       SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Type: application/sdp
      Content-Length: 136

      v=0
     o=user1 53655765 2353687637 IN IP4 127.0.0.1
      s=-
      t=0 0
      c=IN IP4 [media_ip]
      m=audio [media_port] RTP/AVP 0
      a=rtpmap:0 PCMU/8000

    ]]>
  </send>
<recv request="ACK"
       optional="true"
        rtd="true"
       crlf="true">
  </recv>

  <recv request="BYE">
  </recv>

  <send>
    <![CDATA[

     SIP/2.0 200 OK
     [last_Via:]
      [last_From:]
     [last_To:];tag=[call_number]
     [last_Call-ID:]
     [last_CSeq:]
     Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      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>