[ANNOUNCE]: sipnagios, a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc)
Hello fellow VoIPers and RTCers, on GitHub there is an early release of sipnagios, opensource. check it out: https://github.com/gmaruzz/sipnagios sipnagios is a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc) sipnagios implements the Nagios plugin API for monitoring and performance data.sipnagios.c is a modification of the original siprtp.c sample in pjproject distribution. Supposedly, it works on Linux, Windows, and anywhere you can compile pjproject on.It makes a call, checks all the various resulting values (mos, rtt, pdd, tta, jitter, packet loss, bytes and packets transferred, and so on). It verifies these values are included into acceptable, warning, or critical ranges.If the call has gone well, sipnagios print performance data for Nagios graphs, and returns 0.If the call fails, or if its measured values are not inside acceptable ranges, it exits with Nagios conventional WARNING or CRITICAL values. mos calculation is scraped from Julien Chavanton work (VoIP Patrol, on GitHub too) I can't even understand :) (merci Julien!) Enjoy! -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18
I'm testing the plugin on Nagios but no output... /usr/src/pjproject-2.11/pjsip-apps/bin/samples/x86_64-unknown-linux-gnu/sipnagios -d 60 --local-port=5060 --ip-addr=PBXIP --local-siprealm=asterisk --local-user=100 --local-password=password sip:0749941093@acme.cloudpbx.opentelecom.it:5030 Any hint? --- I'm SoCIaL, MayBe El 21/04/2021 a las 4:40 a. m., Giovanni Maruzzelli escribió:
Hello fellow VoIPers and RTCers,
on GitHub there is an early release of sipnagios, opensource.
check it out: https://github.com/gmaruzz/sipnagios <https://github.com/gmaruzz/sipnagios>
sipnagios is a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc)
sipnagios implements the Nagios plugin API for monitoring and performance data.sipnagios.c is a modification of the original siprtp.c sample in pjproject distribution. Supposedly, it works on Linux, Windows, and anywhere you can compile pjproject on.It makes a call, checks all the various resulting values (mos, rtt, pdd, tta, jitter, packet loss, bytes and packets transferred, and so on). It verifies these values are included into acceptable, warning, or critical ranges.If the call has gone well, sipnagios print performance data for Nagios graphs, and returns 0.If the call fails, or if its measured values are not inside acceptable ranges, it exits with Nagios conventional WARNING or CRITICAL values.
mos calculation is scraped from Julien Chavanton work (VoIP Patrol, on GitHub too) I can't even understand :) (merci Julien!)
Enjoy!
-giovanni -- Sincerely,
Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Services sales@freeswitch.com https://freeswitch.com
Official FreeSWITCH Sites https://freeswitch.com/oss https://freeswitch.org/confluence https://cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com
On Wed, Apr 21, 2021 at 8:38 PM Social Boh <social@bohboh.info> wrote:
I'm testing the plugin on Nagios but no output...
/usr/src/pjproject-2.11/pjsip-apps/bin/samples/x86_64-unknown-linux-gnu/sipnagios -d 60 --local-port=5060 --ip-addr=PBXIP --local-siprealm=asterisk --local-user=100 --local-password=password sip:0749941093@acme.cloudpbx.opentelecom.it:5030
Execute it from the command line, and see what it prints. If needed, set logging options to higher values. local-port is the port you want to use to go out from sipnagios (the udp socket sipnagios creates) ip-addr is the local ip address sipnagios uses to create the udp socket, MUST be the ip address of the machine sipnagios is running on (eg: 192.168.1.23) local-siprealm is the sip domain sipnagios will use to authenticate the invite on the remote sip server local user is the auth and login user on the remote sip server password is its password the last argument is the full sip uri sipnagios will send the invite to, and is not for sure the one you used. MUST be a sip uri that is reachable by the remote sip server For any other problem, please open a github issue, so we don't squat this mailing list :) Have a nice day, -giovanni
Any hint?
--- I'm SoCIaL, MayBe
El 21/04/2021 a las 4:40 a. m., Giovanni Maruzzelli escribió:
Hello fellow VoIPers and RTCers,
on GitHub there is an early release of sipnagios, opensource.
check it out: https://github.com/gmaruzz/sipnagios
sipnagios is a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc)
sipnagios implements the Nagios plugin API for monitoring and performance data.sipnagios.c is a modification of the original siprtp.c sample in pjproject distribution. Supposedly, it works on Linux, Windows, and anywhere you can compile pjproject on.It makes a call, checks all the various resulting values (mos, rtt, pdd, tta, jitter, packet loss, bytes and packets transferred, and so on). It verifies these values are included into acceptable, warning, or critical ranges.If the call has gone well, sipnagios print performance data for Nagios graphs, and returns 0.If the call fails, or if its measured values are not inside acceptable ranges, it exits with Nagios conventional WARNING or CRITICAL values.
mos calculation is scraped from Julien Chavanton work (VoIP Patrol, on GitHub too) I can't even understand :) (merci Julien!)
Enjoy!
-giovanni -- Sincerely,
Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18
_________________________________________________________________________
The FreeSWITCH project is sponsored by SignalWire https://signalwire.com Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. Build your next product on our scalable cloud platform.
Join our online community to chat in real time https://signalwire.community
Professional FreeSWITCH Servicessales@freeswitch.comhttps://freeswitch.com
Official FreeSWITCH Siteshttps://freeswitch.com/osshttps://freeswitch.org/confluencehttps://cluecon.c...
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttps://freeswit...
-- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18
participants (2)
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Giovanni Maruzzelli
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Social Boh