asterisk users - Aug 2003 - What is the highest quality codec I can use for recording voice messages?

I have looked at the codec's available but I don't know how get the
highest
quality recorded message.

If a user calls in over the normal telephone network is this limited to the
carriers codec or the codec at the asterisk side?

Would I get a higher quality result using VoIP rather than the normal
network?

Any help would be appreciated.

Thanks
Fats.

The highest quality codec is ulaw or alaw (otherwise know as G.711).. These are
the same as what comes in on your PSTN line..

If you want high Quality voice prompts your best bet os to record them on a PC
with a good quality mic and then copy then to your server.. You juast have to
make sure thay are recorded at the correct sample rate ( IIRC its 8khz 16bit
mono but I could be wrong on that)..

Later..
> 
> I have looked at the codec's available but I don't know how get the
highest
> quality recorded message.
> 
> If a user calls in over the normal telephone network is this limited to the
> carriers codec or the codec at the asterisk side?
> 
> Would I get a higher quality result using VoIP rather than the normal
> network?
> 
> Any help would be appreciated.
> 
> Thanks
> Fats.
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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Hi list,

Can anyone give me a hint on how i can introduce some delay in dialing an
extension and thus preventing timeout, when i press # to transfer a call.
 And also in which context should i write that timeoutfunction.

Rgds
Manoj K Gupta

try using DigitTimeout application

regards
Martin

On Thu, 14 Aug 2003, Manoj K Gupta wrote:
> Hi list,
>
> Can anyone give me a hint on how i can introduce some delay in dialing an
> extension and thus preventing timeout, when i press # to transfer a call.
>  And also in which context should i write that timeoutfunction.
>
> Rgds
> Manoj K Gupta
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

> I was interested in getting the highest quality over a normal phone line
> because I want users to be able to record their messages at the highest
> quality. They do it not me hence my question about the highest quality
> codec.
> 
> If for example they used VoIP software on a computer could they get higher
> quality than over a normal phone line?
> 
> If they did use a normal phone line can I increase the quality of is this a
> limitation at the carriers end?
> 
> Any help would be appreciated.
> 
> Thanks.
> Fats.
> 
Using alaw or ulaw over VoIP will give you best quality you are going to get
over VoIP.. If your PSTN carrier is giving a bad quality line then you should
probably change carrier... :)

Most prompts will be converted to GSM files anyway so that will be the quality
of the end result..

I wouldn't worry about it too much, just record some and take a listen, they
should be fine..

Later..
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> Thanks for the fast reply.
> 
> I assume you mean that alaw or ulaw is what the carrier delivers. Because
my
> customers use the phones they have I do not have control over the carrier
> they use, I just meant it to mean all carriers. As in what is the standard
> format they deliver in.
> 
> I assume that I have no control over how they deliver but at the Asterisk
> end I wanted to ensure the highest quality recording.
> 
> If I used a direct connection from one computer to asterisk could I
increase
> the quality by using a different codec, assuming they have broadband access
> at their end. I have read that the g711 is the highest quality in the
sample
> rate but was not sure if this would still work with Asterisk or if I should
> use G729 as it seems to be double the bandwidth sampling at between 8 and
12
> kps whereas g711 seems to sample at 64kbs but that may flood the
connection.
> 
> Any ideas?
> 
> Thanks
> Fats.
> 
Basically G.711 applies the least amout of compression and therefore the least
amout of quality loss.. similarly G729 applies a very high level of compression
so in theory would have a high quality loss..

Unfortuantely its not always that simple as not all codecs are created equal..
:)

And as you said the G.711 having the highest quality also has the highest
bandwidth requirement of 64Kbps (this increases to just over 80Kbps after you
add IP overhead) so if you can sustain that leavel of throughput then the use of
G.711 is irrelevant..

Like I said if you are really concerned then the best way is to test it and then
listen to the results and see if you are happy with it..

Later..
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On 15/8/03 20:15, "WipeOut ." <wipeout@linuxmail.org> wrote:
>> Thanks for the fast reply.
>> 
>> I assume you mean that alaw or ulaw is what the carrier delivers.
Because my
>> customers use the phones they have I do not have control over the
carrier
>> they use, I just meant it to mean all carriers. As in what is the
standard
>> format they deliver in.
>> 
>> I assume that I have no control over how they deliver but at the
Asterisk
>> end I wanted to ensure the highest quality recording.
>> 
>> If I used a direct connection from one computer to asterisk could I
increase
>> the quality by using a different codec, assuming they have broadband
access
>> at their end. I have read that the g711 is the highest quality in the
sample
>> rate but was not sure if this would still work with Asterisk or if I
should
>> use G729 as it seems to be double the bandwidth sampling at between 8
and 12
>> kps whereas g711 seems to sample at 64kbs but that may flood the
connection.
>> 
>> Any ideas?
>> 
>> Thanks
>> Fats.
>> 
> 
> Basically G.711 applies the least amout of compression and therefore the
least
> amout of quality loss.. similarly G729 applies a very high level of
> compression so in theory would have a high quality loss..
> 
> Unfortuantely its not always that simple as not all codecs are created
equal..
> :)
> 
> And as you said the G.711 having the highest quality also has the highest
> bandwidth requirement of 64Kbps (this increases to just over 80Kbps after
you
> add IP overhead) so if you can sustain that leavel of throughput then the
use
> of G.711 is irrelevant..
> 
> Like I said if you are really concerned then the best way is to test it and
> then listen to the results and see if you are happy with it..
> 
> Later..
Yes you probably right.
Test and see.

So how do I get and use each codec to test them. I understand some are under
tight copyright control. Do they have testing variations rather than buying
first, find it it does not work, and your stuck with the cost?


Thanks for you help.

As always a fast reply.
Thanks

Fats.

> 
> Yes you probably right.
> Test and see.
> 
> So how do I get and use each codec to test them. I understand some are
under
> tight copyright control. Do they have testing variations rather than buying
> first, find it it does not work, and your stuck with the cost?
> 
> 
> Thanks for you help.
> 
> As always a fast reply.
> Thanks
> 
> Fats.
> 
If you want to test under Asterisk using a SIP client I would suggest you
download X-Lite.. This will give you G.711(64Kbps) and GSM(16Kbps) codecs that
are both usable with Asterisk.. That should give you a basis for your
comparison..

If you want to try out G.729 you will have to buy a licence for it from Digium
and get a SIP phone that supports it..

Have fun..

Doubt you will get another fast reply cos its 2200 so bed will be calling soon..
:)

Later..

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