Pro-Active VoIP Management Solutions?

Adeptus_Luminati asks: "I've been running a 1000 user Mitel VoIP phone (to the desk) network which encompasses 20 buildings glued together by our Telco's _private_ fibre backbone (no Internet involved here). Once in a while we have voice quality degradation issues caused by excess latency, jitter, bandwidth saturation, QoS mis-configurations, and so forth. I've been using Ixia Chariot software to simulate VoIP calls over the WAN between our various offices and collect data of the problems, but this is only useful AFTER the problem is reported by our users, and after I am lucky enough to be around and catch the problem happening in real time; otherwise, I have no way of proving to our Telco that there IS a problem. What solutions have other network admins come up with to pro-actively manage similar private VoIP networks?" "I am looking for some sort of solution to allow me to pro-actively monitor or simulate 24/7 VoIP calls between offices and then report back to me immediately when certain thresholds of voice quality degradation have been exceeded and accumulate significant info that I can forward my Telco and get them to deal with the problem, right away. FYI, bandwidth is free on my office WAN links, we're mostly 100Mbit fibre, and we have QoS from end to end (except small parts of the telco backbone)."

Comments

:gag:

[Anonymous Coward]

Pro-Active VoIP Management Solutions?

You're going to hell.

SNMP

[QuantumRiff]

Ask your telco for "SNMP read" access to their routers that they use. Setup an MRTG page that shows traffic and latency. Is this pure fiber from building to building? or are there a bunch of Cat5 (or other cabling) to fiber converters along the path? Most Telco's offer SLA (Service Level Agreements) that garuntee a certain amount of bandwith, latency, and availabiltiy. Also, I know on our metro fiber ring we are moving to, it is all ethernet over fiber, and each company gets their own VLAN. Is your connection pure ethernet all the way through? (if you live in a big city, some of the big players give you your own wavelength, instead of VLAN.. Much nicer)

there is also the option of turning down the audio quality between buildings. (ie, 128Kb stream inside the building, 64kb stream between them.) While slightly more noisy, it still works, and uses less bandwith. I know with our old Cisco VOIP at my old job, department to department calls were low bandwith, and customer calls were setup for highest bandwith. (clearest)

Cheap options and Expensive options

[kasparov]

Cheap option: Linux box hooked up to an ethernet tap at interconnects with the telco's lines. Run ethereal's tethereal in ring buffer mode (making sure that individual files are under 2GB). You are only limited by hard drive space in how much you can store. When viewing the dumps, use etheral > 0.10.10 and go to Statistics->Voip Calls. It will allow you to choose specific calls and even graph things such as latency, jitter, etc. Since you will be dealing with lots of very large files, I recommend using tcpslice (which usually ships in distros with tcpdump) to grab specific chunks that you would like to look at.

Expensive option:Empirx Hammer XMS. It does all of the above with a nice web interface plus it gives you RTP quality metrics like r-factor and MOS. It's not cheap, but I've used and it does a good job (it is basically a SuSE Linux box with some networking gear running their network monitoring software).

All of the above I have tested only with SIP/RTP traffic. If you youse MGCP or H.323, I can't personally vouch for either of the above solutions, though both support them.