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Communications Networking

Pro-Active VoIP Management Solutions? 30

Posted by Cliff from the simulate-the-problems-before-your-users-find-them dept.
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)."
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Pro-Active VoIP Management Solutions? More

Pro-Active VoIP Management Solutions?

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  • Simulating voice calls (Score:3, Interesting)

    by hummassa ( 157160 ) on Tuesday August 09, 2005 @03:13PM (#13281026) Homepage Journal
    Computer #1 in one building, #2 in another.
    Cron job:
    Computer #1 voice-calls computer #2 and plays a complex and long sound.
    Computer #2 records the sound it received.
    Computer #2 compares the sound it received with the original file.
    Log errors; if error-rate > x, page you, sleep short time, repeat cron job.
    Simple, ain't it?
    • Simple, ain't it?

      And wrong. Telephony systems aren't like file systems. You can't just receive a file, compare the output with the original, and assume that there's a problem if it doesn't pass. In telephony systems, there may be format translations or echo cancellation done on the voice channel, which may change the output, but are not indicative of a problem. Depending on how his network is set up, the voice channel may even be routed through analog equipment. If this is the case, you won't ever ge

      • Sorry, he isn't wrong.

        You don't do exact byte-for-byte comparisons. You do regular signal analysis and see how close the audio is to each other. If it exceeds a given deviation, you alert them.

        And yes, it's not dead as in doornail simple, but it is a fairly straightforward problem.

        • Sorry, he isn't wrong. You don't do exact byte-for-byte comparisons. You do regular signal analysis and see how close the audio is to each other. If it exceeds a given deviation, you alert them.

          First of all, the standard for speech quality is PESQ or Perceptual Evaluation of Speech Quality (ITU-T P.862). In PESQ, you don't just, "see how close the audio is to each other". There's a lot of things that need to be done to both the input and reference signals before you can begin analysis.

          First, the inpu

    • Is measuring the "error rate" between a known sound sample and a re-sampled version trivial? You sure make it sound like it is!

      -Peter
      • Talk to the ogg-vorbis people, and check their mailing list archives. I believe they have some tools that do the moral equivalent of:

        $ compress foo.wav > foo.ogg

        $ compare foo.wav foo.ogg
        18% different

        Some interesting quick googling turned up the following: http://www.abde.net/projects/ogg_mp3/ [abde.net]

        Original google search:
        http://www.google.com/search?hl=en&lr=&q=ogg+vorbi s+quantitative&btnG=Search [google.com]

        Term I seem to recall is "quantitative" comparision of audio quality (vs. "qualitative" ... ie: "it sou
      • Assuming it's all digital- and Voice over IP is- then yes, it is relatively trivial. Especially if you get to pick the sound- what you want is a siren. That way you get a big blast of sound running through all the frequencies you're interested in, and you can use the (relatively) quiet to loud transition as a good zero indicator. From there, it's just a byte comparison.
        • Except that, if this guy is smart, he's compressing the voice over the line. I'm guessing a siren run through G.729a won't sound exactly like you say.

          Remember these codecs are specifically designed to compress speech with minimal losses of intelligibility. Good luck measuring how well a human can understand speech without a mark I human ear.

          As for the OPs actual question, Cisco Callmanager can be configured to collect call statistics on every call (jitter, delay, etc.). It can also be set to simulate calls

  • :gag: (Score:4, Funny)

    by Anonymous Coward on Tuesday August 09, 2005 @03:14PM (#13281036)
    Pro-Active VoIP Management Solutions?

    You're going to hell.

  • SNMP (Score:4, Interesting)

    by QuantumRiff ( 120817 ) on Tuesday August 09, 2005 @03:28PM (#13281157)
    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)

  • Network General InfiniStream (Score:3, Informative)

    by arnie_apesacrappin ( 200185 ) on Tuesday August 09, 2005 @03:33PM (#13281196)
    The box is a sniffer with a huge array of disks. It records all traffic that you send it. I have used the product before, but not for VoIP. Here is what the Network General site says about the VoIP option for the InfiniStream:

    The Voice Option is a value-added package that integrates with InfiniStream Network Management to provide additional insights into voice- and video-over-IP converged traffic. Voice-over-IP (VoIP) Experts automatically detect and help resolve key problems seen on VoIP networks--jitter, packet loss, packet-sequencing errors, and latency. These VoIP Experts and call-tracking capabilities, along with the traditional Expert system, help ensure successful VoIP network rollouts while maintaining "toll-quality" voice and high-quality data for all users.

    The product URL is here [networkgeneral.com]

    They make a couple of versions. The last time I looked, the 1 TB version was around 25K and the 4 TB version was around 95K. I didn't buy one, but it was a fun toy to play with.

  • It sounds like you're talking about monitoring your network for application performance and watching for telltales that precede degredation. You might check out a product from this company:

    NEXVU Technologies [nexvu.com]

    Because that's exactly what they do :)

    Unlike a general packet sniffer or network monitor, they aim exactly at your kind of problem.

    Disclaimer: until I entered the glorious realm [uiuc.edu] of academic programming, I was employed by NEXVU, and I still have stock and stock options. Even though they no longer

  • You shoulda stayed with the olde way of doing things. This new fangled VOIP cr*p is nothing but a load of POS foisted upon yee by sales and marketing droids to give them a reason to exist. And sell yee more cr*p.
  • If you have to prove to them that their is a problem, they aren't likely to do anything about it when you do come up with evidence.

    Sorry to burst your bubble, but unless you call your sales rep and threaten to leave, your not going to get anywhere.

  • Similar problem (Score:3, Interesting)

    by Omega1045 ( 584264 ) on Tuesday August 09, 2005 @05:52PM (#13282120)
    A friend had a similar problem. They were sure that the only available telco in the area was not providing the level of service to which they had agreed. They could not get the telco to help at all.

    His solution? He got his board of directors to approve the purchase of some wifi radio equipment, which they mounted on nearby towers. I am not a hardware or radio guy, but this was not Linksys crap that I run in my home. He got some professional stuff. Each office had LOS to a local tower, and the towers to each other. Last I heard, they are running all of their voice and all of their data over their new links. Routers at both ends are configured for QoS, and thing are running very well. The cost of the equipment has already been paid for with the savings since what they pay for the towers is a fraction of the cost of the circuits they were running between offices. They maintain a few landlines that the phone systems on each end can use in the case of emergency to route voice traffic, and I believe he also has a couple of redundant DSL lines for data.

    • It was probably microwave. I don't know much about it, but I have seen it in plenty of places and understand it is a good way to handle digital video signals, so I have to assume it would do well with VoIP also.
  • I don't know of an existing tool for this, but you want to measure "known traffic" across the route and report when it degrades.

    Setup an "application ping server" on the far end. This will be a C program that posts a UDP listen on a port in the RTP range. When it gets an inbound packet, it returns exactly that same packet to whom sent it to them. This needs to be in C (or similar) because the latency needs to be very low. It should also run on a very low utilization server.

    On the near end, write a simil

  • Cheap options and Expensive options (Score:4, Informative)

    by kasparov ( 105041 ) * on Tuesday August 09, 2005 @06:57PM (#13282572)
    Cheap option: Linux box hooked up to an ethernet tap at interconnects with the telco's lines. Run ethereal's [ethereal.com] 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 [die.net] (which usually ships in distros with tcpdump) to grab specific chunks that you would like to look at.

    Expensive option:Empirx Hammer XMS [empirix.com]. 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.

  • http://www.prognosis.com/ [prognosis.com] Also, consider IP SLA (Cisco) GrpA
  • Fibre = Fibre Channel protocol as used in a SAN

    Fiber = Fiber optic (as in the physical cabling)

    Sorry. It's an annoying habit of mine.
  • Give Smokeping a try. Smokeping:ping::MRTG:bandwidth.

    http://people.ee.ethz.ch/~oetiker/webtools/smokepi ng/ [ee.ethz.ch]

    It shows latency, loss, and jitter in a combined easy to read graph. After using it for a while, you can spot many normally invisible network anomolies on these graphs long before they become visible to users. They're also great for post mortem analysis.

    They don't have anything to do specifically with VoIP, but I think they're invaluable tools for any network admin.
  • http://edgewaternetworks.com/ [edgewaternetworks.com]

    It's specifically designed for VOIP quality monitoring.

    And as a disclaimer, I do some work for the company.
  • Chariot does both monitoring and call load simulation... exactly what you want.

    I'm in charge of network assessments for a very large voip hardware manufacturer... we've used this tool to do what you're describing.

    Give tech support a call and get them to walk you through it. It's a great tool.

    Make sure you've got QoS properly set up on all your devices too, regardless if it's across the internet or not. You still need QoS!

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