Hello,
i'm trying to use freeswitch as a redirecting server meaning FS has to
receive an INVITE and according to some rules it will redirect calls to
other destinations.
CALLING_USER FREESWITCH SOMEWHERE
INVITE ------------------------------->
<------------------------------ 100 Trying
<------------------------------ 302 Moved Temporary
ACK ------------------------------->
INVITE--------------------------------------------------------------------------------->
Well, wverything works well except i have perfromance issues .... on my HW
FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary). When
i increase the rate, FS starts delaying 302 response. Right at 50 CPS i see
"calls" being build up in FS and the delay begining to grow.
When i observe the machine, load average is almost nothing (load average:
1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread taking
most load... all others are just sitting there with 1-5 % CPU time.
This looks to me as FS handles 302 messages in a single thread?!?!
tculj...@fs:/usr/local/freeswitch/conf/dialplan$ top -H
top - 10:41:37 up 167 days, 20:42, 3 users, load average: 1.41, 0.61, 0.60
Tasks: 83 total, 2 running, 81 sleeping, 0 stopped, 0 zombie
Cpu(s): 25.3%us, 1.5%sy, 0.0%ni, 30.3%id, 42.7%wa, 0.0%hi, 0.2%si,
0.0%st
Mem: 2074520k total, 571244k used, 1503276k free, 259604k buffers
Swap: 2650684k total, 3020k used, 2647664k free, 153868k cached
PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+
COMMAND
4814 root 20 0 34188 20m 3780 S 38 1.0 3:10.29
freeswitch
4800 root 20 0 34188 20m 3780 S 6 1.0 0:08.26
freeswitch
4798 root 20 0 34188 20m 3780 R 5 1.0 0:24.46
freeswitch
4787 root 20 0 34188 20m 3780 S 2 1.0 0:11.24
freeswitch
4794 root 20 0 34188 20m 3780 S 1 1.0 0:11.42
freeswitch
4803 root 20 0 34188 20m 3780 S 1 1.0 0:11.74
freeswitch
4788 root 20 0 34188 20m 3780 S 1 1.0 0:02.96
freeswitch
4804 root 20 0 34188 20m 3780 S 1 1.0 0:01.64
freeswitch
4807 root 20 0 34188 20m 3780 S 1 1.0 0:01.68
freeswitch
4811 root 20 0 34188 20m 3780 S 1 1.0 0:02.50 freeswitch
cat /proc/cpuinfo
processor : 0
vendor_id : GenuineIntel
cpu family : 6
model : 15
model name : Intel(R) Xeon(R) CPU 5140 @ 2.33GHz
stepping : 6
cpu MHz : 2333.560
cache size : 4096 KB
physical id : 0
siblings : 2
core id : 0
cpu cores : 2
apicid : 0
initial apicid : 0
fdiv_bug : no
hlt_bug : no
f00f_bug : no
coma_bug : no
fpu : yes
fpu_exception : yes
cpuid level : 10
wp : yes
flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3 cx16
xtpr dca lahf_lm
bogomips : 4670.78
clflush size : 64
power management:
processor : 1
vendor_id : GenuineIntel
cpu family : 6
model : 15
model name : Intel(R) Xeon(R) CPU 5140 @ 2.33GHz
stepping : 6
cpu MHz : 2333.560
cache size : 4096 KB
physical id : 0
siblings : 2
core id : 1
cpu cores : 2
apicid : 1
initial apicid : 1
fdiv_bug : no
hlt_bug : no
f00f_bug : no
coma_bug : no
fpu : yes
fpu_exception : yes
cpuid level : 10
wp : yes
flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3 cx16
xtpr dca lahf_lm
bogomips : 4666.82
clflush size : 64
power management:
uname -a
Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009 i686
GNU/Linux
Of course, i've tuned the machine up
ulimit -c unlimited
ulimit -d unlimited
ulimit -f unlimited
ulimit -i unlimited
ulimit -n 999999
ulimit -q unlimited
ulimit -u unlimited
ulimit -v unlimited
ulimit -x unlimited
ulimit -s 240
ulimit -l unlimited
ulimit -a
Started FS with minimum modules but still 40 CPS seems to be the limit.
So, is there any way to improve performance?
Tihomir.
<!-- 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="$${domain}" parse="true"/>
</domains>
<settings>
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="rfc2833-pt" value="101"/>
<param name="sip-port" value="5080"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${outbound_codec_prefs}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<param name="manage-presence" value="false"/>
<param name="aggressive-nat-detection" value="true"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="nonce-ttl" value="60"/>
<param name="auth-calls" value="false"/>
<param name="rtp-timeout-sec" value="1800"/>
<param name="rtp-ip" value="$${local_ip_v4}"/>
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
<!--<param name="enable-3pcc" value="true"/>-->
</settings>
</profile>
<!-- http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files -->
<profile name="internal" domain="$${domain}">
<!--aliases are other names that will work as a valid profile name for this profile-->
<aliases>
<alias name="$${domain}"/>
<alias name="default"/>
</aliases>
<!-- Outbound Registrations -->
<gateways>
<X-PRE-PROCESS cmd="include" data="internal/*.xml"/>
</gateways>
<domains>
<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
<!--<domain name="$${domain}" parse="true"/>-->
<!-- indicator to parse the directory for domains with parse="true" to get gateways and alias every domain to this profile -->
<!--<domain name="all" alias="true" parse="true"/>-->
</domains>
<settings>
<!-- <param name="user-agent-string" value="FreeSWITCH Rocks!"/> -->
<param name="debug" value="0"/>
<param name="sip-trace" value="no"/>
<param name="context" value="public"/>
<param name="rfc2833-pt" value="101"/>
<!-- port to bind to for sip traffic -->
<param name="sip-port" value="5060"/>
<param name="dialplan" value="XML"/>
<param name="dtmf-duration" value="100"/>
<param name="codec-prefs" value="$${global_codec_prefs}"/>
<param name="use-rtp-timer" value="true"/>
<param name="rtp-timer-name" value="soft"/>
<!-- ip address to use for rtp -->
<param name="rtp-ip" value="$${local_ip_v4}"/>
<!-- ip address to bind to -->
<param name="sip-ip" value="$${local_ip_v4}"/>
<param name="hold-music" value="$${hold_music}"/>
<param name="apply-nat-acl" value="rfc1918"/>
<param name="aggressive-nat-detection" value="true"/>
<!--<param name="enable-timer" value="false"/>-->
<!--<param name="enable-100rel" value="false"/>-->
<!--<param name="apply-inbound-acl" value="domains"/>-->
<!--<param name="apply-register-acl" value="domains"/>-->
<!--<param name="dtmf-type" value="info"/>-->
<param name="record-template" value="$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<!--enable to use presense and mwi -->
<param name="manage-presence" value="true"/>
<!-- This setting is for AAL2 bitpacking on G726 -->
<!-- <param name="bitpacking" value="aal2"/> -->
<!--max number of open dialogs in proceeding -->
<!--<param name="max-proceeding" value="1000"/>-->
<!--session timers for all call to expire after the specified seconds -->
<!--<param name="session-timeout" value="120"/>-->
<!--<param name="multiple-registrations" value="true"/>-->
<!--set to 'greedy' if you want your codec list to take precedence -->
<param name="inbound-codec-negotiation" value="generous"/>
<!-- if you want to send any special bind params of your own -->
<!--<param name="bind-params" value="transport=udp"/>-->
<!--<param name="unregister-on-options-fail" value="true"/>-->
<!-- TLS: disabled by default, set to "true" to enable -->
<param name="tls" value="false"/>
<!-- additional bind parameters for TLS -->
<param name="tls-bind-params" value="transport=tls"/>
<!-- Port to listen on for TLS requests. (5061 will be used if unspecified) -->
<param name="tls-sip-port" value="5061"/>
<!-- Location of the agent.pem and cafile.pem ssl certificates (needed for TLS server) -->
<param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/>
<!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may not work with TLSv1 -->
<param name="tls-version" value="tlsv1"/>
<!--If you don't want to pass through timestampes from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
<!--<param name="rtp-rewrite-timestamps" value="true"/>-->
<!--<param name="pass-rfc2833" value="true"/>-->
<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
<!--Uncomment to set all inbound calls to no media mode-->
<!--<param name="inbound-bypass-media" value="true"/>-->
<!--Uncomment to set all inbound calls to proxy media mode-->
<!--<param name="inbound-proxy-media" value="true"/>-->
<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
<!--<param name="inbound-late-negotiation" value="true"/>-->
<!-- this lets anything register -->
<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
<param name="accept-blind-reg" value="true"/>
<!-- accept any authentication without actually checking (not a good feature for most people) -->
<param name="accept-blind-auth" value="true"/>
<!-- supress CNG on this profile or per call with the 'supress_cng' variable -->
<!-- <param name="supress-cng" value="true"/> -->
<!--TTL for nonce in sip auth-->
<param name="nonce-ttl" value="60"/>
<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
that the originator is using-->
<!--<param name="disable-transcoding" value="true"/>-->
<!-- Used for when phones respond to a challenged ACK with method INVITE in the hash -->
<!--<param name="NDLB-broken-auth-hash" value="true"/>-->
<!-- add a ;received="<ip>:<port>" to the contact when replying to register for nat handling -->
<!--<param name="NDLB-received-in-nat-reg-contact" value="true"/>-->
<param name="auth-calls" value="false"/>
<!-- on authed calls, authenticate *all* the packets not just invite -->
<param name="auth-all-packets" value="false"/>
<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> -->
<!-- <param name="ext-sip-ip" value="$${external_sip_ip}"/> -->
<!-- rtp inactivity timeout -->
<param name="rtp-timeout-sec" value="300"/>
<param name="rtp-hold-timeout-sec" value="1800"/>
<!-- VAD choose one (out is a good choice); -->
<!-- <param name="vad" value="in"/> -->
<!-- <param name="vad" value="out"/> -->
<!-- <param name="vad" value="both"/> -->
<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
<!--all inbound reg will look in this domain for the users -->
<!--<param name="force-register-domain" value="cluecon.com"/>-->
<!-- disable register and transfer which may be undesirable in a public switch -->
<!--<param name="disable-transfer" value="true"/>-->
<!--<param name="disable-register" value="true"/>-->
<!--<param name="enable-3pcc" value="true"/>-->
<!-- use stun when specified (default is true) -->
<!--<param name="stun-enabled" value="true"/>-->
<!-- use stun when specified (default is true) -->
<!-- set to true to have the profile determine stun is not useful and turn it off globally-->
<!--<param name="stun-auto-disable" value="true"/>-->
</settings>
</profile>
<?xml version="1.0" encoding="utf-8"?>
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
<context name="default">
<extension name="show_info">
<condition field="destination_number" expression="12...@public">
<action application="answer"/>
<action application="info"/>
<action application="sleep" data="250"/>
<action application="hangup"/>
</condition>
</extension>
<!-- <extension name="my test">
<action application="bridge" data="sofia/sip/[email protected]:5060"/>
</extension> -->
</context>
</include>
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
<context name="public">
<extension name="LNP">
<condition field="destination_number" expression="(^300000)(.*)">
<action application="lnp_getprefix" data="in $2, out reroutingalias"/>
<action application="redirect" data="sip:${[email protected]:5060"/>
</condition>
</extension>
<extension name="LBS">
<condition field="destination_number" expression="(^300010)(.*)">
<action application="lbs_getpublicphone" data="in ${caller_id_number}, in $2, out reroutingalias"/>
<action application="redirect" data="sip:${[email protected]:5060"/>
</condition>
</extension>
<extension name="CPS">
<condition field="destination_number" expression="(^300020)(.*)">
<action application="cps_verifyphone" data="in ${caller_id_number}, in $2, out radiusacc"/>
</condition>
<condition field="radiusacc" expression="1">
<action application="redirect" data="sip:${[email protected]:5060"/>
<anti-action application="respond" data="403 Forbidden"/>
</condition>
</extension>
<extension name="ServiceLookup">
<condition field="destination_number" expression="(^300030)(.*)">
<!--action application="lookup_service_destination" data="in ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, in $1, in pgw01.ot.hr:5060, out red_contact, out authResult"/-->
<action application="log" data="INFO ######################## ServiceLookup ########################\n"/>
<action application="log" data="INFO ######################## contact = '${red_contact}' ##############\n"/>
<action application="log" data="INFO ######################## CallerNum = '${caller_id_number:6:16}' ##########\n"/>
<action application="log" data="INFO ######################## RADIUS auth = '${authResult}' ##########\n"/>
<!--action application="ring_ready"/-->
<!--action application="pre_answer"/-->
<action application="execute_extension" data="doRedirect XML public"/>
</condition>
</extension>
<extension name="doRedirect">
<condition field="destination_number" expression="^doRedirect$"/>
<condition field="${authResult}" expression="^0$|^60$">
<action application="log" data="INFO ######################## RADIUS auth OK!!!' ##########\n"/>
<!--action application="redirect" data="sip:${red_contact}"/-->
<!--action application="answer"/-->
<action application="redirect" data="sip:[email protected]:5060"/>
<!--anti-action application="answer"/-->
<!--anti-action application="sleep" data="2000"/-->
<action application="hangup" data="USER_BUSY"/>
<anti-action application="redirect" data="sip:[email protected]:5060"/>
<anti-action application="log" data="INFO ######################## RADIUS auth NOK!! ##########\n"/>
<!--anti-action application="respond" data="403 Forbidden"/-->
<anti-action application="hangup" data="USER_BUSY"/>
</condition>
</extension>
</context>
</include>
<configuration name="switch.conf" description="Core Configuration">
<cli-keybindings>
<key name="1" value="help"/>
<key name="2" value="status"/>
<key name="3" value="show channels"/>
<key name="4" value="show calls"/>
<key name="5" value="sofia status"/>
<key name="6" value="reloadxml"/>
<key name="7" value="console loglevel 0"/>
<key name="8" value="console loglevel 7"/>
<key name="9" value="sofia status profile internal"/>
<key name="10" value="fsctl pause"/>
<key name="11" value="fsctl resume"/>
<key name="12" value="version"/>
</cli-keybindings>
<settings>
<!--Colorize the Console -->
<param name="colorize-console" value="true"/>
<!--Most channels to allow at once -->
<param name="max-sessions" value="4000"/>
<!--Most channels to create per second -->
<param name="sessions-per-second" value="500"/>
<!-- Default Global Log Level - value is one of debug,info,notice,warning,err,crit,alert -->
<param name="loglevel" value="0"/>
<!--Try to catch any crashes that can be recoverable (in the context of a call)-->
<param name="crash-protection" value="false"/>
<!--<param name="max_dtmf_duration" value="192000"/>-->
<!--<param name="default_dtmf_duration" value="8000"/>-->
<param name="auth-calls" value="false"/>
<param name="manage-presence" value="false"/>
<!--
If you want to send out voicemail notifications via Windows you'll need to change the mailer-app
variable to the setting below:
<param name="mailer-app" value="msmtp"/>
Donot change mailer-app-args.
You will also need to download a sendmail clone for Windows (msmtp). This version works without issue:
http://msmtp.sourceforge.net/index.html. Download and copy the .exe to %winddir%\system32.
You'll need to create a small config file for smtp credentials (host name, authentication, tls, etc.) in
%USERPROFILE%\Application Data\ called "msmtprc.txt". Below is a sample copy of this file:
###################################
# The SMTP server of the provider.
account provider
host smtp.myisp.com
from [email protected]
auth login
user johndoe
password mypassword
# Set a default account
account default : provider
###################################
-->
<param name="mailer-app" value="sendmail"/>
<param name="mailer-app-args" value="-t"/>
<param name="dump-cores" value="yes"/>
<!--RTP port range -->
<!--<param name="rtp-start-port" value="16384"/>-->
<!--<param name="rtp-end-port" value="32768"/>-->
</settings>
</configuration>
<configuration name="modules.conf" description="Modules">
<modules>
<!-- Loggers (I'd load these first) -->
<load module="mod_console"/>
<!--load module="mod_logfile"/-->
<!-- <load module="mod_syslog"/> -->
<!--<load module="mod_yaml"/>-->
<!-- Multi-Faceted -->
<!-- mod_enum is a dialplan interface, an application interface and an api command interface -->
<load module="mod_enum"/>
<!-- XML Interfaces -->
<!-- <load module="mod_xml_rpc"/> -->
<!-- <load module="mod_xml_curl"/> -->
<!-- <load module="mod_xml_cdr"/> -->
<!-- Event Handlers -->
<!--load module="mod_cdr_csv"/-->
<!-- <load module="mod_event_multicast"/> -->
<load module="mod_event_socket"/>
<!-- <load module="mod_zeroconf"/> -->
<!-- Directory Interfaces -->
<load module="mod_ldap"/>
<!-- Endpoints -->
<!-- <load module="mod_dingaling"/> -->
<!-- <load module="mod_iax"/> -->
<!-- <load module="mod_portaudio"/> -->
<!-- <load module="mod_alsa"/> -->
<load module="mod_sofia"/>
<!-- <load module="mod_wanpipe"/> -->
<!-- <load module="mod_woomera"/> -->
<!-- <load module="mod_openzap"/> -->
<!-- Applications -->
<load module="mod_commands"/>
<!--load module="mod_conference"/-->
<load module="mod_dptools"/>
<load module="mod_expr"/>
<!--load module="mod_fifo"/-->
<!--load module="mod_voicemail"/-->
<!--load module="mod_limit"/-->
<!--load module="mod_esf"/-->
<!--load module="mod_fsv"/-->
<!-- SNOM Module -->
<!--<load module="mod_snom"/>-->
<!-- Dialplan Interfaces -->
<!-- <load module="mod_dialplan_directory"/> -->
<load module="mod_dialplan_xml"/>
<!--load module="mod_dialplan_asterisk"/-->
<!-- Codec Interfaces -->
<load module="mod_voipcodecs"/>
<load module="mod_g723_1"/>
<load module="mod_g729"/>
<load module="mod_amr"/>
<!--load module="mod_ilbc"/-->
<!--load module="mod_speex"/-->
<!--load module="mod_h26x"/-->
<!-- File Format Interfaces -->
<!--load module="mod_sndfile"/-->
<load module="mod_native_file"/>
<!--For icecast/mp3 streams/files-->
<!--<load module="mod_shout"/>-->
<!--For local streams (play all the files in a directory)-->
<load module="mod_local_stream"/>
<load module="mod_tone_stream"/>
<!-- Timers -->
<!-- Languages -->
<!--load module="mod_spidermonkey"/-->
<!-- <load module="mod_perl"/> -->
<!-- <load module="mod_python"/> -->
<!-- <load module="mod_java"/> -->
<!--load module="mod_lua"/-->
<!-- ASR /TTS -->
<!-- <load module="mod_flite"/> -->
<!-- <load module="mod_pocketsphinx"/> -->
<!-- <load module="mod_cepstral"/> -->
<!-- <load module="mod_openmrcp"/> -->
<!-- <load module="mod_rss"/> -->
<!-- Say -->
<load module="mod_say_en"/>
<!-- <load module="mod_say_zh"/> -->
</modules>
</configuration>
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