[Freeswitch-users] DTMF Digits Lost when Under Load

Michael Toop michaelt at voxcore.voxtelecom.co.za
Mon Nov 30 04:12:18 PST 2009


Hi All,

 Thought I would share my solution to this DTMF problem:  it turns out my
ISP was capping my bandwidth & dropping packets to keep the connection &
1Mbps, so the experienced DTMF loss was actually packets being discarded.

 On my way to this discovery I tested Freeswitch & DTMF quite thoroughly &
never actually found any problems even at hundreds of concurrent calls.
Here is how I tested, who knows this might be useful to someone:

   - I used SIPp to generate calls & a Python script to log the received
   DTMF digits
   - SIPp command line:
      - sipp -sf dtmfSenario.xml -d 10000 -s 451  -l 96  -mp 5606 -i
      xxx.xxx.xxx.xxx
   - dtmfSenario.xml below
   - Dialplan:
      -     <extension name="test_dtmf_capture_test">
            <!--Grab calls for dialing -->
            <condition field="destination_number" expression="(^100100$)">
              <action application="answer"/>
              <action application="python" data="writeDtmfStats"/>
            </condition>
          </extension>
   - Python:
      - import sys
      from freeswitch import *


      def get_number(session,invalid,num=20):
              digits = session.getDigits(num, "", 15000)

              consoleLog("info","Got '%s' digits from user.\n" % digits)
              if digits == '':
                      # Invalid call
                      if invalid == 3:
                              consoleLog("info","Three invalid
      attempts!!\n")

      session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/misc/8000/invalid_extension.wav")
                              session.hangup()
                              sys.exit(0)
                      else:

      session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/misc/8000/invalid_extension.wav")
                              get_number(session,invalid + 1)
              else:
                      consoleLog("info","Got a valid number: %s,
      proceeding...\n" % digits)
                      return digits



      def handler(session, args):

      session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav")
          numberToDial = get_number(session,2,num=10)
          consoleLog('info','Got 10 DTMF digits. Writing "1" to file...\n')
          fo = open('/tmp/dtmfData.csv','a')
          fo.write('"1"\n')
          fo.close()
          # Do some stuff & wait for SIPP to hangup

      session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav")

      session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please_enter_extension_followed_by_pound.wav")
          return
   - DTMF senario file:
      - # cat dtmfSenario.xml
      <?xml version="1.0" encoding="ISO-8859-1"?>
      <!DOCTYPE scenario SYSTEM "sipp.dtd">
      <!-- This program is free software; you can redistribute it
      and/or      -->
      <!-- modify it under the terms of the GNU General Public License
      as     -->
      <!-- published by the Free Software Foundation; either version 2 of
      the -->
      <!-- License, or (at your option) any later
      version.                    -->
      <!--
      -->
      <!-- This program is distributed in the hope that it will be
      useful,    -->
      <!-- but WITHOUT ANY WARRANTY; without even the implied warranty
      of     -->
      <!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See
      the      -->
      <!-- GNU General Public License for more
      details.                       -->
      <!--
      -->
      <!-- You should have received a copy of the GNU General Public
      License  -->
      <!-- along with this program; if not, write to
      the                      -->
      <!-- Free Software Foundation,
      Inc.,                                    -->
      <!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307
      USA             -->
      <!--
      -->
      <!--                 Sipp 'uac' scenario with pcap (rtp)
      play           -->
      <!--
      -->
      <scenario name="UAC with media">
        <!-- 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=[call_number]
            To: sut <sip:[service]@[remote_ip]:[remote_port]>
            Call-ID: [call_id]
            CSeq: 1 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[local_ip_type] [local_ip]
            t=0 0
            m=audio [auto_media_port] RTP/AVP 18 100
            a=rtpmap:18 G729/8000
            a=fmtp:18 annexb=no
            a=rtpmap:100 telephone-event/8000
            a=fmtp:100 0-16
            a=silenceSupp:off - - - -
            a=ptime:20
            a=sendrecv
          ]]>
        </send>

      <!--      a=rtpmap:8 PCMA/8000 -->
        <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:[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=[call_number]
            To: sut
      <sip:[service]@[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>

        <!-- Play a pre-recorded PCAP file (RTP
      stream)                       -->
      <!--
        <nop>
          <action>
            <exec play_pcap_audio="g711a.pcap"/>
          </action>
        </nop>
      -->

        <!-- Pause 8 seconds, which is approximately the duration of
      the      -->
        <!-- PCAP
      file                                                        -->
        <pause milliseconds="5000"/>

        <!-- Play an out of band DTMF
      '1'                                     -->
        <nop>
          <action>
            <exec play_pcap_audio="dtmf_2833_1.pcap"/>
          </action>
        </nop>

        <pause milliseconds="1000"/>

        <!-- Play an out of band DTMF
      '1'                                     -->
        <nop>
          <action>
            <exec play_pcap_audio="dtmf_2833_1.pcap"/>
          </action>
        </nop>
        <pause milliseconds="1000"/>

        <!-- Play an out of band DTMF
      '1'                                     -->
        <nop>
          <action>
            <exec play_pcap_audio="dtmf_2833_1.pcap"/>
          </action>
        </nop>
        <pause milliseconds="500"/>

        <!-- Play an out of band DTMF
      '1'                                     -->
        <nop>
          <action>
            <exec play_pcap_audio="dtmf_2833_1.pcap"/>
          </action>
        </nop>
        <pause milliseconds="500"/>

        <!-- Play an out of band DTMF
      '1'                                     -->
        <nop>
          <action>
            <exec play_pcap_audio="dtmf_2833_1.pcap"/>
          </action>
        </nop>
        <pause milliseconds="500"/>

        <!-- Play an out of band DTMF
      '1'                                     -->
        <nop>
          <action>
            <exec play_pcap_audio="dtmf_2833_1.pcap"/>
          </action>
        </nop>
        <pause milliseconds="500"/>

        <!-- Play an out of band DTMF
      '1'                                     -->
        <nop>
          <action>
            <exec play_pcap_audio="dtmf_2833_1.pcap"/>
          </action>
        </nop>
        <pause milliseconds="500"/>

        <!-- Play an out of band DTMF
      '1'                                     -->
        <nop>
          <action>
            <exec play_pcap_audio="dtmf_2833_1.pcap"/>
          </action>
        </nop>
        <pause milliseconds="500"/>

        <!-- Play an out of band DTMF
      '1'                                     -->
        <nop>
          <action>
            <exec play_pcap_audio="dtmf_2833_1.pcap"/>
          </action>
        </nop>
        <pause milliseconds="500"/>

        <!-- Play an out of band DTMF
      '1'                                     -->
        <nop>
          <action>
            <exec play_pcap_audio="dtmf_2833_1.pcap"/>
          </action>
        </nop>

        <pause milliseconds="2000"/>

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

            BYE 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=[call_number]
            To: sut
      <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
            Call-ID: [call_id]
            CSeq: 2 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>

Cheers,

Michael

On Mon, Nov 16, 2009 at 5:25 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> That's a pretty small problem description to be so sure about something.
> It would probably be better to capture some evidence of the exact problem
> you are having since we are using computers and we need to see the computers
> in action doing something specifically incorrect to diagnose any sort of
> problem.  Take the time to describe the origin and destination of your
> calls, the call flow, the hardware in use on both ends of the call, detailed
> console logs on debug level, (maybe even uncomment the 2833 debug ifded in
> switch_rtp.c) and gather something to go on besides "I seem to be losing
> dtmf) maybe a packect capture of the networking interface on both ends of
> these calls.
>
> Also problems should be reported to http://jira.freeswitch.org not this
> mailing list.
> Save us a step if you report a jira and provide all the info above or we
> will just have to ask for it again.
>
>
> On Mon, Nov 16, 2009 at 1:07 AM, Michael Toop <
> michaelt at voxcore.voxtelecom.co.za> wrote:
>
>> Hi All,
>>
>>  I have an issue that when my call volumes on my FS IVR box > 30 calls
>> DTMF digits are lost (using RFC2833).  It is definitely load related as it
>> all works perfectly under 30 calls.
>>
>>  Any pointers or a solution to the problem?
>>
>> Thanks,
>>
>> Michael
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091130/227f9d35/attachment-0002.html 


More information about the FreeSWITCH-users mailing list