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

[Duncan Turnbull]

On 29/10/2013, at 9:55 am, Mike <mike352 at microdel.org> wrote:

> On Mon, 28 Oct 2013, 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.  
You probably need to break the problem down into more specific issues.

Does every call suffer or is it just some?
Is the CPU overworked at all and are calls being transcoded or all the same codec?
Is it time of day affected?
What load is on the network at the same time?
Do you have monitoring tools such as smoke ping tell you the network looks so you can match to issues?
Call drop outs suggest network issues or upset hardware, you shouldn’t be getting call drop outs if you have enough bandwidth and low jitter

>> 
>>      Moved the asterisk server from VM machine to dedicated machine
>> 
We have found hardware issues before on hardware and the next box worked fine. Testing the network performance to the SIP provider to see jitter and packet loss especially while on the phone and getting issues.

>>      More than enough bandwidth
Need both up and down - if you are G711 then its about 128K per channel, if you have about 20 simultaneous channels you need about 2M both directions, however if its a bandwidth issue you should be fine testing calls when the network load is quiet. If you are using GSM etc you need a lot less bandwidth. 
The issue is often the upstream as many services are asymmetric but voip needs symmetry. You are sharing with email, cloud apps, vpns and all other stuff so make sure there is no congestion on the uplink. 

Buy a link just for voip is often easier than QOS and these days not very costly.

>> 
>>      Setting 802.1p = 7
Won’t be honoured on the net
>> 
>>      Set Dedicated voice traffic 35% of bandwidth.

How are you doing this? Mikrotik routers have a good queue mechanism that works reasonably well I have found, but the main thing is to test your up and down bandwidth and check it stays available under other loads.

>> Not sure what option would be the best
>> 
Get another ISP - or hassle them over performance standards 
>>      Put analog lines in the conference room to avoid the dropouts - leave the sip lines in place for day to day use
Buy better bandwidth is another option?
Where is your sip provider in relation to your network ? Try another SIP provider - the closer and better performing network the better
>> 
>>      Hire a consultant
>> 
>>      Ditch the system and buy a pre-packaged system - RingCentral or some such.
If you buy a new system and use SIP they will make sure you buy some dedicated bandwidth for the network so they don’t have to deal with issues like this

>> 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.
Well worth getting 10 people dial into conference from desks and check quality for 5-10mins - if its good then you are looking at your network

>> Suggestions welcome.

Good luck
>> Best
>> Eddie
>> --
>