No idea about Zabbix api, at the moment, maybe is the same as the Nagios one.

If not, it will be easy to implement, just by changing exit status and the way perf data are printed (if Zabbix supports perf data).

I'll add this to the TODO.

Thanks Daniel!

-giovanni

answered from mobile, please pardon terseness and typos, -giovanni

On Wed, Apr 21, 2021, 14:51 Daniel-Constantin Mierla miconda@gmail.com wrote:

Hello,

On 21.04.21 11:42, Giovanni Maruzzelli wrote:

...

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!)

thanks for letting everyone here know about this tool!

Do you know if other monitoring platforms (e.g., Zabbix) have the same plugin API and can be used with? Or it is for Nagios only?

Cheers, Daniel

-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone)

• https://www.asipto.com/sw/kamailio-advanced-training-online/

Yep, I believe monitoring systems are in Nagios compatible family (most of them), or in Zabbix clan.

From a quick check seems in Zabbix sipnagios would be an entity called

"external check", and it seems Zabbix does not gather multiple performance data, just one value.

But I will check it better.

Btw, sipnagios can be used also as a standalone tool to originate a call and gather it's results, like sipsak for OPTIONS et al.

-giovanni

answered from mobile, please pardon terseness and typos, -giovanni

On Wed, Apr 21, 2021, 15:04 Daniel-Constantin Mierla miconda@gmail.com wrote:

Personally I do not need such feature (e.g., Zabbix), I was more curious if there is sort of common plugin API between some of the monitoring systems (like seems to be now for web browsers) -- if I am not wrong, some of these monitoring systems have common roots.

Cheers, Daniel On 21.04.21 14:56, Giovanni Maruzzelli wrote:

No idea about Zabbix api, at the moment, maybe is the same as the Nagios one.

If not, it will be easy to implement, just by changing exit status and the way perf data are printed (if Zabbix supports perf data).

I'll add this to the TODO.

Thanks Daniel!

-giovanni

answered from mobile, please pardon terseness and typos, -giovanni

On Wed, Apr 21, 2021, 14:51 Daniel-Constantin Mierla miconda@gmail.com wrote:

...

Hello,

On 21.04.21 11:42, Giovanni Maruzzelli wrote:

...

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!)

thanks for letting everyone here know about this tool!

Do you know if other monitoring platforms (e.g., Zabbix) have the same plugin API and can be used with? Or it is for Nagios only?

Cheers, Daniel

-- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone)

• https://www.asipto.com/sw/kamailio-advanced-training-online/

--

Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone)

• https://www.asipto.com/sw/kamailio-advanced-training-online/

On Wednesday 21 April 2021 at 15:04:04, Daniel-Constantin Mierla wrote:

Personally I do not need such feature (e.g., Zabbix), I was more curious if there is sort of common plugin API between some of the monitoring systems (like seems to be now for web browsers) -- if I am not wrong, some of these monitoring systems have common roots.

This is correct - the Nagios plugin protocol has more or less become a standard format, and is shared by Icinga, Icinga2 and Shinken at least.

I'm not sure about Zabbix - it's a long time I used that.

Antony.

On 21.04.21 14:56, Giovanni Maruzzelli wrote:

...

No idea about Zabbix api, at the moment, maybe is the same as the Nagios one.

If not, it will be easy to implement, just by changing exit status and the way perf data are printed (if Zabbix supports perf data).

I'll add this to the TODO.

Thanks Daniel!

-giovanni

...

On Wed, Apr 21, 2021, 14:51 Daniel-Constantin Mierla wrote: Hello,

On 21.04.21 11:42, Giovanni Maruzzelli wrote:
> 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>

> <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!)

thanks for letting everyone here know about this tool!

Do you know if other monitoring platforms (e.g., Zabbix) have the
same plugin API and can be used with? Or it is for Nagios only?

Cheers,
Daniel

On Thu, Apr 22, 2021 at 12:36 AM Giovanni Maruzzelli gmaruzz@gmail.com wrote:

On Wed, Apr 21, 2021 at 5:19 PM Julien Chavanton jchavanton@gmail.com wrote:

...

Few notes on the mos-lq (listening quality), it consider both losses from jitter (discarded) and never received. I tried to keep the equation and variables as defined in the ITU, but it is relatively simple in the end. One thing missing is the delay impairment to have mos-cq this would be RTT + jitter buffer size of both endpoints. This way we will correctly account for jitter impairment in terms of loss and delay.

Nice!

What I would liker to do, in an undefined future :), is to use the pjsip machinery for streaming audio from and to a file, so it will have their tried and true rtcp, rtcp-xr, etc implementation. It will be easier and more precise to combine these statistics to obtain better accuracy.

Makes sense to me, RTCP-XR is proposing many new metrics, some related to transmission other to signal and filters. Burst density seems interesting to extrapolate the impact of transmission problems since PLC can only help for up to 60ms large bursts are more likely to impact intelligibility than spreaded losses. Even if all of this will always be a rough estimate since as an example the impact on intelligibility will always be quite random, I still think the ITU did more R&D than anyone else in this field.

...

More context on jitter, as I recently went back looking at some MOS score computation. Since we compute MOS in the endpoint it can be more precise when it comes to jitter. In most cases, when done in a relay, I found that jitter is hard (or not accounted) for properly, since we extrapolate adaptive / static buffers that will receive the packets. What I found in most cases was jitter x 2 like in rtp-engine seems like the best option but should endup underestimating the impairment as this would mean an adaptive buffer and assume not too much jitter of jitter meaning the size of the buffer based on estimate is always fine with the given jitter and not dropping late packets and it must drop packets when it shrinks.

How do you judge sipjs implementation of jitter measurements?

sipjs, I haven't looked, but I usually expect to find the interarrival jitter defined in the RTP 3550, which is an estimation / EWMA

...

Let's remember to keep each other posted if we improve this further.

Definitely!!! And thanks again for VoIP Patrol!

-giovanni

-- Sincerely,

Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18