[Freeswitch-users] Load test - performance not even matching Asterisk

[Jon Bruel]

The protocol is SIP. The profiles relevant for the incoming call test
with echo are here:

Dialplan:
*********

<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->

<include>

  <context name="public">

    <extension name="From_Asterisk">

      <condition field="destination_number"
expression="^(K00003333100004444\d{3})$">

            <action application="answer"/>

            <action application="echo"/>

      </condition>

    </extension>

  </context>

</include>

 

Sip Profiles:
*************

<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files --> 

<profile name="external">

  <!-- This profile is only for outbound registrations to providers -->

  <gateways>

    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>

  </gateways>

 

  <aliases>

    <alias name="outbound"/>

  </aliases>

 

  <domains>

    <domain name="$${domain}" parse="true"/>

  </domains>

 

  <settings>

    <param name="debug" value="1"/>

<param name="comfort-noise" value="false"/>

    <param name="sip-trace" value="no"/>

    <param name="rfc2833-pt" value="101"/>

    <param name="sip-port" value="5080"/>

    <param name="dialplan" value="XML"/>

    <param name="context" value="public"/>

    <param name="dtmf-duration" value="100"/>

    <param name="codec-prefs" value="$${outbound_codec_prefs}"/>
(comment: using PCMA)

    <param name="hold-music" value="$${hold_music}"/>

    <param name="use-rtp-timer" value="true"/>

    <param name="rtp-timer-name" value="soft"/>

    <param name="manage-presence" value="false"/>

    <param name="aggressive-nat-detection" value="false"/>

    <param name="inbound-codec-negotiation" value="generous"/>

    <param name="nonce-ttl" value="60"/>

    <param name="auth-calls" value="false"/>

    <param name="rtp-timeout-sec" value="1800"/>

    <param name="rtp-ip" value="$${local_ip_v4}"/>

    <param name="sip-ip" value="$${local_ip_v4}"/>

    <param name="ext-rtp-ip" value="$${local_ip_v4}"/>

    <param name="ext-sip-ip" value="$${local_ip_v4}"/>

    <param name="rtp-timeout-sec" value="300"/>

    <param name="rtp-hold-timeout-sec" value="1800"/>

    <!--<param name="enable-3pcc" value="true"/>-->

  </settings>

</profile>

 

Gateway (one of them):
**********************

<include>

      </gateway>

      <gateway name="100004444300">

            <param name="username" value="100004444300"/>

            <param name="realm" value="10.3.1.21"/>

            <param name="from-user" value="100004444300"/>

            <param name="from-domain" value="10.3.1.21"/>

            <param name="password" value="XXXXXX"/> 

            <param name="extension" value="K00003333100004444300"/>

            <param name="proxy" value="10.3.1.21"/>

            <param name="expire-seconds" value="600"/>

            <param name="caller-id-in-from" value="false"/>

      </gateway>        

</include>

 

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