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Home > Archive > Voice Over IP in UK > December 2006 > Forcing inbound Codec to G711
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Forcing inbound Codec to G711
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| Pet @ www.gymratz.co.uk ;¬) 2006-12-08, 1:11 pm |
| At a guess I presume inbound numbers from public PSTN routed from ones
VOIP supplier would be routed on the most economical codec eg G712A/B
If that's the case, then if I were to force the voip ports to a single
codec e.g. G711MU, presumably the inbound call would be forced to
upscale to this single codec rather than the lowest it could get away with?
Would there be any forseeable disadvantages doing this?
I see it as a way of getting max. call quality in inbound calls esp.
where downstream bandwidth is not such an issue as upstream.
Any thoughts?
Pete
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| "Pet @ www.gymratz.co.uk ;¬)" wrote:
> At a guess I presume inbound numbers from public PSTN routed from ones
> VOIP supplier would be routed on the most economical codec eg G712A/B
What do you mean by "economical"? Some codecs cost less bandwidth but more
CPU time. Some codecs are patented [whether legally or not] and require
licensing.
> If that's the case, then if I were to force the voip ports to a single
> codec e.g. G711MU, presumably the inbound call would be forced to
> upscale to this single codec rather than the lowest it could get away
> with?
If it's SIP we're talking about, both ends have to agree on a codec. If they
cannot, the call will fail.
> Would there be any forseeable disadvantages doing this?
Higher bandwidth use.
> I see it as a way of getting max. call quality in inbound calls esp.
> where downstream bandwidth is not such an issue as upstream.
When budgeting bandwidth, it makes sense to assume that it will be
symmetrical. When I watch the bandwidth graph on my Asterisk server [SIP,
G711u], it's always level in both directions, ie it doesn't have peaks and
troughs that correlate with speech and silence. This of course may depend
on silence suppression and voice activity detection settings etc on the
endpoints.
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| Philippe Deleye 2006-12-09, 7:11 am |
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"Pet @ www.gymratz.co.uk ;¬)" <PeTe33@gymratz.co.uk> wrote in message
news:fmgeh.13632$k74.7182@text.news.blueyonder.co.uk...
> If that's the case, then if I were to force the voip ports to a single
> codec e.g. G711MU, presumably the inbound call would be forced to
> upscale to this single codec rather than the lowest it could get away
with?
If the other end of the call (your Providers PSTN Gateway) does not support
the codec you try to enforce, then the call will fail.
> I see it as a way of getting max. call quality in inbound calls esp.
> where downstream bandwidth is not such an issue as upstream.
There is no connection between inbound calls and downstream bandwith. Both
Inbound calls and outbound are using the same (symmetric) bandwith, if using
the same codec.
With good downstraem bandwith, you will experience good sound quality, but
with a bad upstraeam bandwith the other party will experience bad quality
(what you hear is downstream, what the other party hears is upstream - not
related to incoming our outgoing calls ...
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| alexd brought next idea :
>
> When budgeting bandwidth, it makes sense to assume that it will be
> symmetrical. When I watch the bandwidth graph on my Asterisk server [SIP,
> G711u],
How'd you do that?
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| Pet @ www.gymratz.co.uk ;¬) wrote:
> At a guess I presume inbound numbers from public PSTN routed from ones
> VOIP supplier would be routed on the most economical codec eg G712A/B
Not in my experience, most stick to G.711a/u. It's still only needs
about 80kbps per call which is fine for the vast majority of Internet
connections these days. G729 would save on bandwidth but it's CPU
intensive, this might not matter for you but it'll make a big difference
for a service provider. It is also not a free codec and the license
holder must be paid for it. I dare say that most importantly, it's
lower quality than G711 (and therefore the PSTN) so the customer would
perceive the service provider's service as poorer quality.
>
> If that's the case, then if I were to force the voip ports to a single
> codec e.g. G711MU, presumably the inbound call would be forced to
> upscale to this single codec rather than the lowest it could get away with?
Yes if you set your VoIP hardware to only accept g711a/u then either the
service providers server will agree to this or the call will fail.
>
> Would there be any forseeable disadvantages doing this?
> I see it as a way of getting max. call quality in inbound calls esp.
> where downstream bandwidth is not such an issue as upstream.
Less calls through your Internet connection. The calls require the same
amount of bandwidth in either direction regardless of whether they are
incoming calls or outgoing calls.
>
> Any thoughts?
> Pete
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| ale.cx 2006-12-11, 7:11 am |
| On Dec 10, 11:59 am, Jono <notha...@blueyonder.invalid> wrote:
> How'd you do that?
gkrellmd on the Asterisk box, gkrellm on my desktop :-D
alexd
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| PeterW 2006-12-18, 1:11 pm |
| Paul <nomailforme@polog40.org.uk> wrote in
news:457d2a89$0$624$5a6aecb4@news.aaisp.net.uk:
> Pet @ www.gymratz.co.uk ;¬) wrote:
>
> Not in my experience, most stick to G.711a/u. It's still only needs
> about 80kbps per call which is fine for the vast majority of Internet
> connections these days. G729 would save on bandwidth but it's CPU
> intensive, this might not matter for you but it'll make a big
> difference for a service provider. It is also not a free codec and
> the license holder must be paid for it. I dare say that most
> importantly, it's lower quality than G711 (and therefore the PSTN) so
> the customer would perceive the service provider's service as poorer
> quality.
>
>
> Yes if you set your VoIP hardware to only accept g711a/u then either
> the service providers server will agree to this or the call will fail.
>
>
> Less calls through your Internet connection. The calls require the
> same amount of bandwidth in either direction regardless of whether
> they are incoming calls or outgoing calls.
>
>
Most offer G.711a (this is the European/worldwide standard). It provides
the best 8khz sampled audio available.
G.711u is a US standard. G.726/729 or other compression standards (GSM,
Speex etc.) is only useful where bandwidth is a big issue.
peter
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