[asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Daniel van den Berg]

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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