Have you looked  at the link characteristics? Does your log indicate packet drops? Out of order packets? To rule out scom's wimax, can you put up a call endpoint that is local? (Say in scom's datacenter). If you still get a crappy experience, then it could be that your wimax/scom's internal routing is at fault. If not rinse repeat with a further node
My two cents says that you'll probably have a much better experience using a FTH w/out VPN. Can you setup that scenario and use that as the gold standard?
- http://en.wikipedia.org/wiki/VoIP_VPN
- http://www.voip-info.org/wiki/view/VOIP+and+VPN
- http://www.voiptroubleshooter.com/problems/loadshare.html
- http://www.networkworld.com/community/node/39577

On Fri, Jul 13, 2012 at 4:06 PM, Odhiambo Washington <odhiambo@gmail.com> wrote:
Okay, just to clarify further, I have IP PBXs across the VPN tunnel. Between them, they do H323, not SIP. With H323 I expect each channel to be 16Kbps, meaning that in a 512Kbps I should be able to carry 32 channels. Even if I assumed that is simplex, then I should be able to carry 16 duplex channels, right?
My problem is that carry a single duplex channel is still a problem, hence my beef with Safaricom.



On Fri, Jul 13, 2012 at 1:50 PM, Jangita Nyagudi <jangita.nyagudi@gmail.com> wrote:
Ondiek,

It all depends on quality vs quantity. Usually SIP works fine BUT you
have to choose a good RTP codec that is suitable for what you want.
The lower the bit rate the more the caller sounds like he is talking
from a hole, but the more channels you can pack using a specific
bandwidth. I've found the gsm codec (which fits about 35 channels on
1mbps satisfactory)

Have a look at this

http://www.en.voipforo.com/codec/codecs.php

For example G.729 codec streams aat 8kbps, which means that in a
512kpbs link would hold 64 channels.
Also note that most codecs cannot carry DTMF unless it is specifically
provided for since they are all lossy. have a look at this

http://anetworkerblog.com/2011/02/06/dtmf-on-voip/



On 13 July 2012 13:41, Edwin Ngige <edwinngige@gmail.com> wrote:
> It all depend on your location, what`s available and such i.e lastmile
>
> Regards,
>
>
>
> On Fri, Jul 13, 2012 at 11:52 AM, Odhiambo Washington <odhiambo@gmail.com>
> wrote:
>>
>> Hello Skunkies,
>>
>> I am wondering if any of you knows someone who runs the technical aspects
>> of a call centre in KE.
>>
>> I need to know who provides them with connectivity that they use for VoIP.
>>
>> I have a WiMax connection with Safaricom of 512Kbps (symmetric) which I
>> use for VoIP only - via a VPN Tunnel to a co-lo facility in the UK.
>>
>> My problem is that this link can barely cary even a single voice channel.
>> The quality is very poor, to an extent that conversation even goes off.
>> If you call an interactive service, the DTMF will never work - always ends
>> up with errors of input not being detected.
>>
>> I am considering ditching Safaricom for a much better service.
>>
>>
>> --
>> Best regards,
>> Odhiambo WASHINGTON,
>> Nairobi,KE
>> +254733744121/+254722743223
>> _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _
>> I can't hear you -- I'm using the scrambler.
>>
>>
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>
>
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--
Best regards,
Odhiambo WASHINGTON,
Nairobi,KE
+254733744121/+254722743223
_ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _
I can't hear you -- I'm using the scrambler.


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