I wouldn't consider netflow/sflow ideal for this. They are great at monitoring utilization of a link, but the way they do it means they are not good for monitoring jitter, latency, or loss, which are very important for VOIP and video. Sflow (and Netflow before it) work by sampling every nth packet in hardware and forwarding meta information about that packet to a collector, that makes infererences about utilization by looking at the distance between packets and other meta information about the header. RIght there you can see you have lost all information about packet loss/drops.
What you really need is a measurement of jitter for VOIP QOS. Back when I worked at Global Crossing we had special poller/collectors running at our major WAN sites that would collect WAN jitter using off the shelf tools that just sent periodic streams of packets and looking at the mean, median, and variance in the inter-packet arrival ratios. Of course, this isn't that useful in a LAN environment where everything goes through one switch. But, in that kind of environment you can still have collectors off of different switches where you would do this. The hardware stats from each swtich that you want include Drops, various types of errors, and some other hardware counters (perhaps switch dependent) that might include things like buffer full events or pause frames or similar things. You can collect these into something like Circonus, OpenTSDB, or Graphite and then have rules that trigger (e.g. Bosun) for sending notifications to your event management and ticketing systems. Sent from my android device. Sent from my android device. Sent from my android device. -----Original Message----- From: Matt Simmons <[email protected]> To: Evan Pettrey <[email protected]> Cc: "[email protected]" <[email protected]> Sent: Fri, 08 May 2015 8:24 Subject: Re: [lopsa-discuss] VoIP Monitoring Is this something you could use netflow/sFlow to monitor? Also, this looks like an interesting slide deck from someone who has also thought about this problem: http://luca.ntop.org/OpenSourceVoipMonitoring.pdf --Matt On Fri, May 8, 2015 at 8:10 AM, Evan Pettrey <[email protected]> wrote: > Greetings folks, > > I'm currently in the process of trying to put in place better proactive > monitoring of our VoIP environment and I'm hoping to tap into the wisdom of > some of you that have more experience with this than myself. > > The challenge that we're running into is that there are simply so many > different things to monitor and it's not something that is on/off or that > relies solely on hardware utilization to fire off alerts. Currently we > typically don't know there is a problem until users report that audio on > their calls is dropping in and out. > > Essentially we need to be able to monitor SIP and RTP traffic flow > end-to-end to be alerted when there is jitter or anything else that could > affect call quality. This should include (but may not be limited to): > > - *Internal Network - *Juniper hardware (I've come across some really > good VoIP monitoring tools like the one available from Solarwinds but they > require Cisco hardware) > > - *VoIP Servers - *Running Asterisk 1.8 > > - *SIP Trunks - *We have limited ability to monitor traffic once it > hits our SIP trunks so this poses a challenge but it needs to be monitored > nonetheless > > > We've set things up like Homer <http://www.sipcapture.org/> and while > that is a great tool for retroactively troubleshooting issues reported it > does not do much to alert us to problems proactively. > > > Any guidance here would be greatly appreciated. Thanks in advance! > > > Best, > Evan > > > _______________________________________________ > Discuss mailing list > [email protected] > https://lists.lopsa.org/cgi-bin/mailman/listinfo/discuss > This list provided by the League of Professional System Administrators > http://lopsa.org/ > >
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