使用SIPP测试重复DTMF数字

Question

I am wanting to create a Load Testing with SIPP to my SIP Server(Restcomm),

This is my sipp script,Works well... The Call is successfully, The DTFM is playing

I can receive the digit 1 one time, but cannot receive 2 times... If I change in the second 'play' a distinct digit, works well but if both are the same digit I only can receive the first.

Is it a SIPP Bug? or Have I have some wrong with my script?

I believe that the problem is here

<exec play_pcap_audio="pcap/dtmf_2833_1.pcap" />

That is my complete script

<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<scenario name="UAC with media">
    <send retrans="500">
<![CDATA[

  INVITE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
  Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  From: sipp <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
  To: sut <sip:[field0]@[remote_ip]:[remote_port]>
  Call-ID: [call_id]
  CSeq: 1 INVITE
  Contact: sip:[field1]@[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[local_ip_type] [local_port]
  t=0 0
  m=audio [auto_media_port] RTP/AVP 96 0 9 8 101 13
  a=rtpmap:8 PCMA/8000
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-1
]]>
    </send>

    <recv response="100" 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" crlf="true">
    </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 sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
  Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
  From: sipp <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
  To: sut <sip:[field0]@[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
]]>
    </send>


    <pause milliseconds="1000" />

    <nop>
            <action>
                    <exec play_pcap_audio="pcap/dtmf_2833_1.pcap" />
            </action>
    </nop>

    <pause milliseconds="100" />

   <nop>
            <action>
                     <!-- <exec play_pcap_audio="pcap/dtmf_2833_5.pcap"/> -->
                    <exec rtp_stream="resume" />
                    <exec play_pcap_audio="pcap/dtmf_2833_1.pcap" />
            </action>
    </nop>

    <pause milliseconds="100" />

    <!-- definition of the response time repartition table (unit is ms) -->
    <ResponseTimeRepartition value="100, 500,1000,3000,4000,5000,6000" />

    <!-- definition of the call length repartition table (unit is ms) -->
    <CallLengthRepartition value="500,1000,2500,5000,6000,7000,9000,10000" />

Update:

Adding Restcomm Side

I can access to this Page with the url: http://192.168.148.1:18080/test/index.jsp

I can hear the the audio (infinite loop to test)

1) Testing with SIP Client (JITSI): I can press 1,1 and hear 2 times 2) Testing with SIPP Script: I only hear 1 time

<%@ page language="java" contentType="text/xml; charset=ISO-8859-1"
    import="java.util.*" pageEncoding="ISO-8859-1"%>
<%
        if (request.getParameter("reply")!=null){
        System.out.println("===========  REPLY                    ==============="+request.getParameter("Digits"));
    }else{
             System.out.println("=======================================================");
        System.out.println("===========  NEW CALL  =================");
              System.out.println("=======================================================");
    }
  %>
 <Response>
        <Gather action="index.jsp?reply=1" method="GET" numDigits="1"   timeout="20">
 <Play>http://192.168.148.1:8080/restcomm/audio/demo-prompt.wav</Play>
 </Gather>
</Response>

Answer 1

I suggest increasing the pause time of SIPp when playing the DTMF to 139ms (instead of current 100ms), so that the Media Server can receive the packets with end-of-event flag set to true.

Also, set numDigits="2" in the RCML Gather verb.

Hope this helps.