SIPp UAC Scenario to send RTP

Please install the latest sipp (>=3.7.3)

wget https://github.com/SIPp/sipp/releases/download/v3.7.3/sipp
chmor +x sipp
mv sipp /usr/bin/sipp

use the following Sipp scenario file

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

<!--                                                                    -->
<!--                 Sipp 'uac with pcap' scenario.                     -->
<!--                                                                    -->

<scenario name="UAC with PCAP">
  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
  <!-- generated by sipp. To do so, use [call_id] keyword.                -->
  <send retrans="500">
    <![CDATA[

      INVITE 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=[pid]SIPpTag00[call_number]
      To: B-Party <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: [cseq] INVITE
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      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 [auto_media_port] RTP/AVP 8
      a=rtpmap:8 PCMA/8000

    ]]>
  </send>

  <recv response="407" auth="true">
        <action>
                <ereg regexp="tag=([^ ]*)" search_in="hdr" header="To" check_it="true" assign_to="4,3" />
                <log message="tag is [$4], [$3]"/>
        </action>
  </recv>

  <send>
   <![CDATA[

    ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
    Via: SIP/2.0/[transport] [local_ip]:[local_port]
    From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
    To: B-Party <sip:[service]@[remote_ip]:[remote_port]>;tag=[$3]
    Call-ID: [call_id]
    CSeq: [cseq] ACK
    Contact: sip:sipp@[local_ip]:[local_port]
    Max-Forwards: 70
    Subject: Performance Test
    Content-Length: 0

   ]]>
 </send>

  <send retrans="500">
    <![CDATA[

      INVITE 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=[pid]SIPpTag00[call_number]
      To: B-Party <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: [cseq] INVITE
      Contact: sip:sipp@[local_ip]:[local_port]
      [authentication]
      Max-Forwards: 70
      Subject: Performance Test
      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 [auto_media_port] RTP/AVP 8
      a=rtpmap:8 PCMA/8000

    ]]>
  </send>

  <recv response="407" optional="true">
  </recv>

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

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

  <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
  <!-- are saved and used for following messages sent. Useful to test   -->
  <!-- against stateful SIP proxies/B2BUAs.                             -->
  <recv response="200" rtd="true" rrs="true">
        <action>
                <ereg regexp="tag=([^ ]*)" search_in="hdr" header="To" check_it="true" assign_to="1,2" />
                <log message="tag is [$1], [$2]"/>
        </action>
  </recv>

  <!-- Packet lost can be simulated in any send/recv message by         -->
  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
  <send>
    <![CDATA[

      ACK [next_url] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      [routes]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
      To: B-Party <sip:[service]@[remote_ip]:[remote_port]>;tag=[$2]
      Call-ID: [call_id]
      CSeq: [cseq] ACK
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>


  <nop>
    <action>
            <exec play_pcap_audio="./g711a.pcap"/>
    </action>
  </nop>
  <!-- This delay can be customized by the -d command-line option       -->
  <!-- or by adding a 'milliseconds = "value"' option here.             -->
  <pause/>

  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
  <send retrans="500">
    <![CDATA[

      BYE [next_url] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      [routes]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
      To: B-Party <sip:[service]@[remote_ip]:[remote_port]>;tag=[$2]
      Call-ID: [call_id]
      CSeq: [cseq] BYE
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

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

  <!-- 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>

download pcap file g711a.pcap and put it in the same directory

now start sipp script with the following command

sipp -sf uac_407.xml kamailio.hbvoice.local:5060 -s +123456789 -au sipp -ap 12345 -d 30000 -m 100 -r 1 -rp 3000 -m 10 -cid_str 14OCT24-TEST1-%u-%p@%s -min_rtp_port 20000 -max_rtp_port 25000 -base_cseq 12345

Enjoy 😉