Hi there, Sounds like codec ptime mismatch...what codec are you using? If you are using g729 make sure that you and your provider is giving the same ptime.
On 10/29/2013 11:55 AM, Stelios Koroneos wrote: > On Mon, 2013-10-28 at 14:29 -0400, Eddie Mikell wrote: >> All, >> >> >> The users in our organization are well, quite frankly, sick of phone >> service that is being provided. The choppy phone calls, and drop outs >> are detrimental to our sales force. >> >> >> I've tried about everything I can think of. >> >> >> Moved the asterisk server from VM machine to dedicated machine >> More than enough bandwidth >> Setting 802.1p = 7 >> Set Dedicated voice traffic 35% of bandwidth. >> >> >> Not sure what option would be the best >> >> >> Put analog lines in the conference room to avoid the dropouts >> - leave the sip lines in place for day to day use >> Hire a consultant >> Ditch the system and buy a pre-packaged system - RingCentral >> or some such. >> >> >> There are no local asterisk professionals who can help, and we are a >> little leery of opening up our system to outside consultants. >> >> >> Anyone else face the above, and finally abandoned Asterisk for a >> commercial system? >> >> >> We have 167 users. >> I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the >> conference rooms. >> >> >> Suggestions welcome. >> >> > A general rule of thump after several years with voip > > Voip turns out to be the "canary in the coal-mine" of a network. The > smallest change or problem will manifest itself as a voip issue no > matter what. > > > Now to some practical advice > > Voip was designed for LAN's, The moment voip packets leave your lan and > go into a WAN of any sort, it could be the source of frustration for > many reasons. > > 1) Lots of routers/modems are not build to handle intense voip traffic. > voip generates lots of small in size UPD packages. In most of the cases > the routers/modems bridging your lan with the wan have no problem > handling them BUT what i have found is that once you get over a > threshold of traffic its possible the routers/modem can not cope with > it, mainly because the large number of packets they have to process. > In most enterprise grade routers the specs give you 2 numbers for the > size of data the router can handle. > total throughput and pps (packets per second). > Usually total throughput is calculated using a packet size of around > 1500bytes and it takes the router the same resources to process a 1500 > bytes package as it does a 90bytes packet of a g729 call, as it just > looks at the headers and not the payload.So yes your router can handle > 60Mbits (of 1500byte frames) which is about 5000 packers per second but > for voip that translates to less than 4Mbits of data (5000 packets of 90 > bytes) > I think you can get the picture > > > 2) Because of 1) its possible that your ISP has issues, especially if > its handling lots of voip traffic while its equipment is not optimized > for that. > > > 3) QOS and queing in general > Whatever you do with QOS to get a better priority/quality, the dirty > secret is, you can only control what YOU send, not what you receive. > And even that is true till your modem/router. Once the packet is gone > you have no control of how it will be handle by all intermediates till > it reaches its destination. > You have no idea if qos is honored by ALL hops and what kind of queuing > they apply (if they do) to that port/service/qos mark > That beeing said, its possible that you *might* have much better luck > with sip and sip rtp than with iax rtp if your isp and all its > interconnects bother to offer qos for rtp. > Now for receiving it can be even harder if your isp does not provide > correct priority queuing for the rtp stream, as latencies can build fast > especially on "busy hours" (which happen to be the same hours people use > their phones the most...) where people download stuff,emails etc. > > ping.icmp and all the other networking monitoring tools/protocols could > be an indicator BUT its most probable that they will be handled by the > isp and its interconnects at the higher qos priority > The only way to see how rtp traffic is handled is to run rtp traffic. > > The only way around this is a "dedicated circut" MPLS or similar between > the points of interest (i.e offices), with specific SLA which usually > means much much higher costs. >> > Finally my 2 cents for troubleshouting. > Check the network first ! > Find what triggers the problem. > Is it something that happens all time regardless of traffic ? > is it periodic ? (when bw goes over X percent, or at a specific time of > day ?) > Try different qos settings/priority queuing on the router > >
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