Re: [asterisk-users] Codec Conversion

[Miguel Molina]

El 09/08/10 05:30, michel freiha escribió:

Hello Miguel molina,

I did what you asked, but still the voice is too bad

Regards

On Thu, Aug 5, 2010 at 11:38 PM, Miguel Molina 
mailto:mmol...@millenium.com.co>> wrote:


El 05/08/10 14:50, Tim Nelson escribió:

- "michel freiha" 
 wrote:
>
> Dear Sir,
>
> I tried to convert ilbc to ulaw and get the same result...Bad
Voice Quality
>
> Regards
>

Again, iLBC is poor quality to begin with. You can't take a poor
audio sample and make it better by converting it to a codec with
better 'resolution'. An audio sample full of robot voice is going
to sound like the same robot voice even if you transcode it to a
better quality codec, whether that is G.729, G.711u, or the
latest 'HD Voice' codecs.

--Tim

This just made me remember some comment on the iax.conf sample file...

disallow=lpc10; Icky sound quality...  Mr. Roboto.

Cheers,

-- 
Ing. Miguel Molina

Grupo de Tecnología
Millenium Phone Center
 



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users



Hi,

I didn't ask nothing... but as Tim said you are encouraged to change the 
iLBC codec to other (could be GSM) and do some tests. Play with several 
codecs and see which one fits your needs or whether this is not a codec 
or transcoding issue.


Regards,

--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Codec Conversion

[michel freiha]

Hello Miguel molina,

I did what you asked, but still the voice is too bad

Regards

On Thu, Aug 5, 2010 at 11:38 PM, Miguel Molina wrote:

>  El 05/08/10 14:50, Tim Nelson escribió:
>
> - "michel freiha"   wrote:
> >
> > Dear Sir,
> >
> > I tried to convert ilbc to ulaw and get the same result...Bad Voice
> Quality
> >
> > Regards
> >
>
>  Again, iLBC is poor quality to begin with. You can't take a poor audio
> sample and make it better by converting it to a codec with better
> 'resolution'. An audio sample full of robot voice is going to sound like the
> same robot voice even if you transcode it to a better quality codec, whether
> that is G.729, G.711u, or the latest 'HD Voice' codecs.
>
>  --Tim
>
> This just made me remember some comment on the iax.conf sample file...
>
> disallow=lpc10; Icky sound quality...  Mr. Roboto.
>
> Cheers,
>
> --
> Ing. Miguel Molina
> Grupo de Tecnología
> Millenium Phone Center
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Codec Conversion

[Jeff Brower]

Steve-

>   On 08/07/2010 03:15 AM, Jeff Brower wrote:
>> Steve-
>>
>>> El 05/08/10 14:50, Tim Nelson escribió:
 - "michel freiha"wrote:
> Dear Sir,
>
> I tried to convert ilbc to ulaw and get the same result...Bad Voice
 Quality
> Regards
>
 Again, iLBC is poor quality to begin with. You can't take a poor audio
 sample and make it better by converting it to a codec with better
 'resolution'. An audio sample full of robot voice is going to sound
 like the same robot voice even if you transcode it to a better quality
 codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs.

 --Tim
>>> This just made me remember some comment on the iax.conf sample file...
>>>
>>> disallow=lpc10; Icky sound quality...  Mr. Roboto.
>> LPC10 is a very old codec, from early 1980s.  LPC10 doesn't do a good 
>> job with pitch detection so it tends to
>> have
>> a
>> 'robotic' sound.  With advent of MELPe, anyone needing bitrates 2400 or 
>> less should not be using LPC10.
>>
>> -Jeff
> MELPe is patent encumbered,
 Not if used for govt/defense purposes.  For commercial-only purposes, TI 
 will waive royalty fees if their chip is
 used
 in the product.  It would have been nice if Digium had considered the many 
 advantages of using a DSP pioneer such
 as
 TI before putting a Mindspeed chip on their TC400B card.
>>> I think all the IP for MELP is now in the hands of Compandent, and TI no
>>> longer has the ability to waive royalties.
>> That is not correct.  Compandent has filed copyrights on certain files 
>> associated with a C549 chip assembly language
>> implementation they did under contract to NSA around 2001.  TI has patent 
>> rights on 2400 bps, TI + Microsoft on 1200
>> bps, and TI + Microsoft + Thales Group on 600 bps.  Microsoft's IP came 
>> about as a result of acquiring a company
>> called SignalCom around 2001.  If the noise pre-processor is used, then 
>> there is some AT&T IP.  To verify this, you
>> can search dsprelated.com (specifically, look for posts discussing this 
>> issue on comp.dsp), and you can also read
>> the
>> "Compandent IPR" section of the MELPe Wikipedia page
>> (http://en.wikipedia.org/wiki/Mixed_Excitation_Linear_Prediction).  That 
>> section was authored by the Compandent's
>> founder, Oded Gottesman.  Oded is a super sharp, very hard working guy.
>>
>> Compandent also claims a copyright on some C code in the file melp_syn.c 
>> (synthesis filter).  I have read
>> discussions
>> by DSP experts indicating the copyrighted section of code can be implemented 
>> in alternative ways, but Oded may say
>> that's not accurate.
> That guy is PITA. He must have driven a lot of people away from MELP by
> the way he acts. He really annoys the regulars in the comp.dsp group by
> posting astroturf questions about MELP, and giving astroturf replies
> about how fantastic it is. That probably shapes a lot of my attitude to
> MELP. :-)
>>> Either way, government use
>>> and use with TI silicon are two niches that might work out well, and
>>> everything else is a problem for several more years. If you are going to
>>> pay royalties for a low bit rate codec, IMBE is probably a better option.
>> I would disagree because IMBE source is not available.  MELPe source is 
>> available and can be downloaded online.
> Depends what you mean by available. IMBE is patented, just like MELP is
> patented. Licence either, and implementations are available.

I meant that MELPe C source code is available for non-commercial purposes 
(academic, R&D, bug fixes and other source
level improvements) without payment and without signing a license agreement 
with a corporation (such as Digital Voice
with IMBE).

> IMBE has
> the great benefit of being widely used for commercial and amateur low
> bit rate channels. For example, amateur radio uses IMBE - an anomaly
> which is one of the drivers for David Rowe's work on an open low bit
> rate codec. Transcoding at low bit rates is a disaster, so using a codec
> you won't need to transcode is a big plus.

Yes all good points.  IMBE and AMBE have surely been successful, testaments to 
the Digital Voice guys and their
pioneering work in the LBR codec area.

>>> TI is a good option, but what do you have against Mindspeed? Choosing a
>>> good option for this kind of card is mostly about managing the patent
>>> licence fees. I assume Mindspeed gave Digium the best option for doing
>>> that, within Digium's volume constraints.
>> My understanding in talking to Digium engineers at Globalcom and other trade 
>> shows back in 2006 is they were worried
>> about interfacing the TI TNET series devices over the PCI bus.  They would 
>> have needed an FPGA with some non-trivial
>> logic programming, so I understand their decision.  But if they had got past 
>> their FPGA "writer's

Re: [asterisk-users] Codec Conversion

[Steve Underwood]

  On 08/07/2010 03:15 AM, Jeff Brower wrote:
> Steve-
>
>> El 05/08/10 14:50, Tim Nelson escribió:
>>> - "michel freiha"wrote:
 Dear Sir,

 I tried to convert ilbc to ulaw and get the same result...Bad Voice
>>> Quality
 Regards

>>> Again, iLBC is poor quality to begin with. You can't take a poor audio
>>> sample and make it better by converting it to a codec with better
>>> 'resolution'. An audio sample full of robot voice is going to sound
>>> like the same robot voice even if you transcode it to a better quality
>>> codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs.
>>>
>>> --Tim
>> This just made me remember some comment on the iax.conf sample file...
>>
>> disallow=lpc10; Icky sound quality...  Mr. Roboto.
> LPC10 is a very old codec, from early 1980s.  LPC10 doesn't do a good job 
> with pitch detection so it tends to have
> a
> 'robotic' sound.  With advent of MELPe, anyone needing bitrates 2400 or 
> less should not be using LPC10.
>
> -Jeff
 MELPe is patent encumbered,
>>> Not if used for govt/defense purposes.  For commercial-only purposes, TI 
>>> will waive royalty fees if their chip is
>>> used
>>> in the product.  It would have been nice if Digium had considered the many 
>>> advantages of using a DSP pioneer such as
>>> TI before putting a Mindspeed chip on their TC400B card.
>> I think all the IP for MELP is now in the hands of Compandent, and TI no
>> longer has the ability to waive royalties.
> That is not correct.  Compandent has filed copyrights on certain files 
> associated with a C549 chip assembly language
> implementation they did under contract to NSA around 2001.  TI has patent 
> rights on 2400 bps, TI + Microsoft on 1200
> bps, and TI + Microsoft + Thales Group on 600 bps.  Microsoft's IP came about 
> as a result of acquiring a company
> called SignalCom around 2001.  If the noise pre-processor is used, then there 
> is some AT&T IP.  To verify this, you
> can search dsprelated.com (specifically, look for posts discussing this issue 
> on comp.dsp), and you can also read the
> "Compandent IPR" section of the MELPe Wikipedia page
> (http://en.wikipedia.org/wiki/Mixed_Excitation_Linear_Prediction).  That 
> section was authored by the Compandent's
> founder, Oded Gottesman.  Oded is a super sharp, very hard working guy.
>
> Compandent also claims a copyright on some C code in the file melp_syn.c 
> (synthesis filter).  I have read discussions
> by DSP experts indicating the copyrighted section of code can be implemented 
> in alternative ways, but Oded may say
> that's not accurate.
That guy is PITA. He must have driven a lot of people away from MELP by 
the way he acts. He really annoys the regulars in the comp.dsp group by 
posting astroturf questions about MELP, and giving astroturf replies 
about how fantastic it is. That probably shapes a lot of my attitude to 
MELP. :-)
>> Either way, government use
>> and use with TI silicon are two niches that might work out well, and
>> everything else is a problem for several more years. If you are going to
>> pay royalties for a low bit rate codec, IMBE is probably a better option.
> I would disagree because IMBE source is not available.  MELPe source is 
> available and can be downloaded online.
Depends what you mean by available. IMBE is patented, just like MELP is 
patented. Licence either, and implementations are available. IMBE has 
the great benefit of being widely used for commercial and amateur low 
bit rate channels. For example, amateur radio uses IMBE - an anomaly 
which is one of the drivers for David Rowe's work on an open low bit 
rate codec. Transcoding at low bit rates is a disaster, so using a codec 
you won't need to transcode is a big plus.


>> TI is a good option, but what do you have against Mindspeed? Choosing a
>> good option for this kind of card is mostly about managing the patent
>> licence fees. I assume Mindspeed gave Digium the best option for doing
>> that, within Digium's volume constraints.
> My understanding in talking to Digium engineers at Globalcom and other trade 
> shows back in 2006 is they were worried
> about interfacing the TI TNET series devices over the PCI bus.  They would 
> have needed an FPGA with some non-trivial
> logic programming, so I understand their decision.  But if they had got past 
> their FPGA "writer's block", they could
> have put one TNETV3010 chip on there, even smaller than the Mindspeed and 
> without the heat sink, and had twice the
> channel capacity as they do now.
TI have had DSP chips with a PCI interface for years, so that 
explanation doesn't make a lot of sense. Of course, these days you need 
a PCI-E interface. I'm not so sure about the status of those in DSP chips.
>> so there is still a place for LPC10 [...]

> e>>  I haven't seen an LPC10 implementation with MOS higher than 2.5.  Due to 
> its 

Re: [asterisk-users] Codec Conversion

[Jeff Brower]

Steve-

> El 05/08/10 14:50, Tim Nelson escribió:
>> - "michel freiha"   wrote:
>>> Dear Sir,
>>>
>>> I tried to convert ilbc to ulaw and get the same result...Bad Voice
>> Quality
>>> Regards
>>>
>> Again, iLBC is poor quality to begin with. You can't take a poor audio
>> sample and make it better by converting it to a codec with better
>> 'resolution'. An audio sample full of robot voice is going to sound
>> like the same robot voice even if you transcode it to a better quality
>> codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs.
>>
>> --Tim
> This just made me remember some comment on the iax.conf sample file...
>
> disallow=lpc10; Icky sound quality...  Mr. Roboto.
 LPC10 is a very old codec, from early 1980s.  LPC10 doesn't do a good job 
 with pitch detection so it tends to have
 a
 'robotic' sound.  With advent of MELPe, anyone needing bitrates 2400 or 
 less should not be using LPC10.

 -Jeff
>>> MELPe is patent encumbered,
>> Not if used for govt/defense purposes.  For commercial-only purposes, TI 
>> will waive royalty fees if their chip is
>> used
>> in the product.  It would have been nice if Digium had considered the many 
>> advantages of using a DSP pioneer such as
>> TI before putting a Mindspeed chip on their TC400B card.
>
> I think all the IP for MELP is now in the hands of Compandent, and TI no
> longer has the ability to waive royalties.

That is not correct.  Compandent has filed copyrights on certain files 
associated with a C549 chip assembly language
implementation they did under contract to NSA around 2001.  TI has patent 
rights on 2400 bps, TI + Microsoft on 1200
bps, and TI + Microsoft + Thales Group on 600 bps.  Microsoft's IP came about 
as a result of acquiring a company
called SignalCom around 2001.  If the noise pre-processor is used, then there 
is some AT&T IP.  To verify this, you
can search dsprelated.com (specifically, look for posts discussing this issue 
on comp.dsp), and you can also read the
"Compandent IPR" section of the MELPe Wikipedia page
(http://en.wikipedia.org/wiki/Mixed_Excitation_Linear_Prediction).  That 
section was authored by the Compandent's
founder, Oded Gottesman.  Oded is a super sharp, very hard working guy.

Compandent also claims a copyright on some C code in the file melp_syn.c 
(synthesis filter).  I have read discussions
by DSP experts indicating the copyrighted section of code can be implemented in 
alternative ways, but Oded may say
that's not accurate.

> Either way, government use
> and use with TI silicon are two niches that might work out well, and
> everything else is a problem for several more years. If you are going to
> pay royalties for a low bit rate codec, IMBE is probably a better option.

I would disagree because IMBE source is not available.  MELPe source is 
available and can be downloaded online.

> TI is a good option, but what do you have against Mindspeed? Choosing a
> good option for this kind of card is mostly about managing the patent
> licence fees. I assume Mindspeed gave Digium the best option for doing
> that, within Digium's volume constraints.

My understanding in talking to Digium engineers at Globalcom and other trade 
shows back in 2006 is they were worried
about interfacing the TI TNET series devices over the PCI bus.  They would have 
needed an FPGA with some non-trivial
logic programming, so I understand their decision.  But if they had got past 
their FPGA "writer's block", they could
have put one TNETV3010 chip on there, even smaller than the Mindspeed and 
without the heat sink, and had twice the
channel capacity as they do now.

>>> so there is still a place for LPC10 [...]
e>> I haven't seen an LPC10 implementation with MOS higher than 2.5.  Due to 
its age and expiration of patents, LPC10
>> might be a basis for a 2400 bps open source codec.  But enormous improvement 
>> would be needed to come close to MELPe
>> performance.
>>
>>
> MELPe is definitely a compandent thing, and TI cannot waive fees for
> that. MELP and MELPe are derived from LPC10. Any attempt to improve
> LPC10 would take you down a similar road, though you would need to skirt
> around the patents.

Again, not correct.  Suggest to research the many online independent sources, 
or contact NSA (who initiated the
overall MELPe effort in the 1990s, in response to a need to significantly 
improve over LPC10) and who can give you a
complete IP list.

> Do you really consider MELPe to be an enormous improvement over LPC10?
> Its still pretty lousy compared to a number of options at about 5kbps,
> and RTP overheads mean the gain from going lower than 5k isn't that big.
> The main reason LPC10 and MELPe offer a low bit rate in RTP is the
> minimum packet you can pack 22.5ms frames into sanely is a 90ms one.

In MOS terms, yes.  In VoIP terms, I agree it's not clear cut.  At 2400 bps, a 
90 msec packet would

Re: [asterisk-users] Codec Conversion

[Steve Underwood]

  On 08/06/2010 04:43 PM, Jeff Brower wrote:
> Steve-
>
>>On 08/06/2010 05:40 AM, Jeff Brower wrote:
>>> Miguel-
>>>
 El 05/08/10 14:50, Tim Nelson escribió:
> - "michel freiha"   wrote:
>> Dear Sir,
>>
>> I tried to convert ilbc to ulaw and get the same result...Bad Voice
> Quality
>> Regards
>>
> Again, iLBC is poor quality to begin with. You can't take a poor audio
> sample and make it better by converting it to a codec with better
> 'resolution'. An audio sample full of robot voice is going to sound
> like the same robot voice even if you transcode it to a better quality
> codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs.
>
> --Tim
 This just made me remember some comment on the iax.conf sample file...

 disallow=lpc10; Icky sound quality...  Mr. Roboto.
>>> LPC10 is a very old codec, from early 1980s.  LPC10 doesn't do a good job 
>>> with pitch detection so it tends to have a
>>> 'robotic' sound.  With advent of MELPe, anyone needing bitrates 2400 or 
>>> less should not be using LPC10.
>>>
>>> -Jeff
>> MELPe is patent encumbered,
> Not if used for govt/defense purposes.  For commercial-only purposes, TI will 
> waive royalty fees if their chip is used
> in the product.  It would have been nice if Digium had considered the many 
> advantages of using a DSP pioneer such as
> TI before putting a Mindspeed chip on their TC400B card.

I think all the IP for MELP is now in the hands of Compandent, and TI no 
longer has the ability to waive royalties. Either way, government use 
and use with TI silicon are two niches that might work out well, and 
everything else is a problem for several more years. If you are going to 
pay royalties for a low bit rate codec, IMBE is probably a better option.

TI is a good option, but what do you have against Mindspeed? Choosing a 
good option for this kind of card is mostly about managing the patent 
licence fees. I assume Mindspeed gave Digium the best option for doing 
that, within Digium's volume constraints.
>> so there is still a place for LPC10 [...]
> I haven't seen an LPC10 implementation with MOS higher than 2.5.  Due to its 
> age and expiration of patents, LPC10
> might be a basis for a 2400 bps open source codec.  But enormous improvement 
> would be needed to come close to MELPe
> performance.
>
>
MELPe is definitely a compandent thing, and TI cannot waive fees for 
that. MELP and MELPe are derived from LPC10. Any attempt to improve 
LPC10 would take you down a similar road, though you would need to skirt 
around the patents.

Do you really consider MELPe to be an enormous improvement over LPC10? 
Its still pretty lousy compared to a number of options at about 5kbps, 
and RTP overheads mean the gain from going lower than 5k isn't that big. 
The main reason LPC10 and MELPe offer a low bit rate in RTP is the 
minimum packet you can pack 22.5ms frames into sanely is a 90ms one. 
90ms RTP *really* cuts the overheads, compared to the more typical 20ms 
or 30ms packets used for G.729.

As others have mentioned, David Rowe is working on a modern 2400bps 
codec. He did a burst of work some time ago, and then put it aside while 
busy with other things. He recently told me he is restarting the work, 
and he wants to get that codec into good shape before the end of this year.

Steve

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Codec Conversion

[Michael Graves]

On Fri, 06 Aug 2010 07:40:44 -0500, Michael Graves wrote:

>On Fri, 6 Aug 2010 03:43:33 -0500 (CDT), Jeff Brower wrote:
>
>
>
>>> MELPe is patent encumbered,
>>
>>Not if used for govt/defense purposes.  For commercial-only purposes, TI will 
>>waive royalty fees if their chip is used
>>in the product.  It would have been nice if Digium had considered the many 
>>advantages of using a DSP pioneer such as
>>TI before putting a Mindspeed chip on their TC400B card.
>>
>>> so there is still a place for LPC10 [...]
>>
>>I haven't seen an LPC10 implementation with MOS higher than 2.5.  Due to its 
>>age and expiration of patents, LPC10
>>might be a basis for a 2400 bps open source codec.  But enormous improvement 
>>would be needed to come close to MELPe
>>performance.
>>
>>-Jeff
>
>I wonder where David Rowe's newer CODEC2 fits into this discussion?
>(http://codec2.org/)
>
>Clearly it's not implemented anywhere yet, but it may prove yet useful
>in very bandwidth constrained applications. Oh yes. It's completely
>open source and should not be subject to patent issues.
>
>Michael

The more appropriate link should have been
http://www.rowetel.com/blog/?page_id=452

Michael
--
Michael Graves
mgravesmstvp.com
http://www.mgraves.org
o713-861-4005
c713-201-1262
sip:mgra...@mstvp.onsip.com
skype mjgraves
Twitter mjgraves




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Codec Conversion

[Michael Graves]

On Fri, 6 Aug 2010 03:43:33 -0500 (CDT), Jeff Brower wrote:



>> MELPe is patent encumbered,
>
>Not if used for govt/defense purposes.  For commercial-only purposes, TI will 
>waive royalty fees if their chip is used
>in the product.  It would have been nice if Digium had considered the many 
>advantages of using a DSP pioneer such as
>TI before putting a Mindspeed chip on their TC400B card.
>
>> so there is still a place for LPC10 [...]
>
>I haven't seen an LPC10 implementation with MOS higher than 2.5.  Due to its 
>age and expiration of patents, LPC10
>might be a basis for a 2400 bps open source codec.  But enormous improvement 
>would be needed to come close to MELPe
>performance.
>
>-Jeff

I wonder where David Rowe's newer CODEC2 fits into this discussion?
(http://codec2.org/)

Clearly it's not implemented anywhere yet, but it may prove yet useful
in very bandwidth constrained applications. Oh yes. It's completely
open source and should not be subject to patent issues.

Michael
--
Michael Graves
mgravesmstvp.com
http://www.mgraves.org
o713-861-4005
c713-201-1262
sip:mgra...@mstvp.onsip.com
skype mjgraves
Twitter mjgraves




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Codec Conversion

[Jeff Brower]

Steve-

>   On 08/06/2010 05:40 AM, Jeff Brower wrote:
>> Miguel-
>>
>>> El 05/08/10 14:50, Tim Nelson escribió:
 - "michel freiha"  wrote:
> Dear Sir,
>
> I tried to convert ilbc to ulaw and get the same result...Bad Voice
 Quality
> Regards
>
 Again, iLBC is poor quality to begin with. You can't take a poor audio
 sample and make it better by converting it to a codec with better
 'resolution'. An audio sample full of robot voice is going to sound
 like the same robot voice even if you transcode it to a better quality
 codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs.

 --Tim
>>> This just made me remember some comment on the iax.conf sample file...
>>>
>>> disallow=lpc10; Icky sound quality...  Mr. Roboto.
>> LPC10 is a very old codec, from early 1980s.  LPC10 doesn't do a good job 
>> with pitch detection so it tends to have a
>> 'robotic' sound.  With advent of MELPe, anyone needing bitrates 2400 or less 
>> should not be using LPC10.
>>
>> -Jeff
> MELPe is patent encumbered,

Not if used for govt/defense purposes.  For commercial-only purposes, TI will 
waive royalty fees if their chip is used
in the product.  It would have been nice if Digium had considered the many 
advantages of using a DSP pioneer such as
TI before putting a Mindspeed chip on their TC400B card.

> so there is still a place for LPC10 [...]

I haven't seen an LPC10 implementation with MOS higher than 2.5.  Due to its 
age and expiration of patents, LPC10
might be a basis for a 2400 bps open source codec.  But enormous improvement 
would be needed to come close to MELPe
performance.

-Jeff


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Codec Conversion

[Steve Underwood]

  On 08/06/2010 05:40 AM, Jeff Brower wrote:
> Miguel-
>
>> El 05/08/10 14:50, Tim Nelson escribió:
>>> - "michel freiha"  wrote:
 Dear Sir,

 I tried to convert ilbc to ulaw and get the same result...Bad Voice
>>> Quality
 Regards

>>> Again, iLBC is poor quality to begin with. You can't take a poor audio
>>> sample and make it better by converting it to a codec with better
>>> 'resolution'. An audio sample full of robot voice is going to sound
>>> like the same robot voice even if you transcode it to a better quality
>>> codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs.
>>>
>>> --Tim
>> This just made me remember some comment on the iax.conf sample file...
>>
>> disallow=lpc10; Icky sound quality...  Mr. Roboto.
> LPC10 is a very old codec, from early 1980s.  LPC10 doesn't do a good job 
> with pitch detection so it tends to have a
> 'robotic' sound.  With advent of MELPe, anyone needing bitrates 2400 or less 
> should not be using LPC10.
>
> -Jeff
MELPe is patent encumbered, so there is still a place for LPC10. LPC10 
should sound a lot better than the one in Asterisk. The Asterisk codec 
is broken.

Steve


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Codec Conversion

[Miguel Molina]

>> This just made me remember some comment on the iax.conf sample file...
>>
>> disallow=lpc10; Icky sound quality...  Mr. Roboto.
>>  
> LPC10 is a very old codec, from early 1980s.  LPC10 doesn't do a good job 
> with pitch detection so it tends to have a
> 'robotic' sound.  With advent of MELPe, anyone needing bitrates 2400 or less 
> should not be using LPC10.
>
> -Jeff
>
>
OK, on years I have working with asterisk I never have used, tested or 
even heard that old codec. I was just quoting the funny comment...

Cheers,

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Codec Conversion

[Jeff Brower]

Miguel-

> El 05/08/10 14:50, Tim Nelson escribió:
>> - "michel freiha"  wrote:
>> >
>> > Dear Sir,
>> >
>> > I tried to convert ilbc to ulaw and get the same result...Bad Voice
>> Quality
>> >
>> > Regards
>> >
>>
>> Again, iLBC is poor quality to begin with. You can't take a poor audio
>> sample and make it better by converting it to a codec with better
>> 'resolution'. An audio sample full of robot voice is going to sound
>> like the same robot voice even if you transcode it to a better quality
>> codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs.
>>
>> --Tim
> This just made me remember some comment on the iax.conf sample file...
>
> disallow=lpc10; Icky sound quality...  Mr. Roboto.

LPC10 is a very old codec, from early 1980s.  LPC10 doesn't do a good job with 
pitch detection so it tends to have a
'robotic' sound.  With advent of MELPe, anyone needing bitrates 2400 or less 
should not be using LPC10.

-Jeff


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Codec Conversion

[Miguel Molina]

El 05/08/10 14:50, Tim Nelson escribió:

- "michel freiha"  wrote:
>
> Dear Sir,
>
> I tried to convert ilbc to ulaw and get the same result...Bad Voice 
Quality

>
> Regards
>

Again, iLBC is poor quality to begin with. You can't take a poor audio 
sample and make it better by converting it to a codec with better 
'resolution'. An audio sample full of robot voice is going to sound 
like the same robot voice even if you transcode it to a better quality 
codec, whether that is G.729, G.711u, or the latest 'HD Voice' codecs.


--Tim

This just made me remember some comment on the iax.conf sample file...

disallow=lpc10; Icky sound quality...  Mr. Roboto.

Cheers,

--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Codec Conversion

[Jeff Brower]

Michel-

> I tried to convert ilbc to ulaw and get the same
> result...Bad Voice Quality

I think you have to be more specific when you say "bad voice quality".  Like 
what?  Worse than a cellphone call?  Gaps
of audio missing?  Robotic or "cyborg" sound?  Static?  A background tone or 
buzzing?

iLBC isn't any worse voice quality than other LBR codecs (GSM-AMR, EVRC, etc).  
If you want land-line quality and what
you're hearing is cellphone quality, then you're asking too much.  Otherwise, 
suggest to be specific and detailed in
describing your problem.

-Jeff

> On Thu, Aug 5, 2010 at 4:13 PM, Tim Nelson  wrote:
>
>> - "michel freiha"  wrote:
>> >
>> > Dear All,
>> >
>> > i would like to ask please if someone tried to make a codec conversion
>> from ilbc to g729, because i did that but the voice quality was too bad and
>> a lot of disconnection..
>> >
>> > Can i get your feedback regarding this issue please?
>> >
>> > regards
>>
>> I can't comment on your 'disconnection' as you don't say if that means the
>> call is disconnected or you're getting stuttered audio. Regardless, iLBC has
>> one of the lowest bitrates of the available codecs and as such the voice
>> quality is not spectacular to begin with. Take 'not so good' audio and try
>> to convert it to another audio format, and the deficiencies can be
>> exacerbated.
>>
>> --Tim


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Codec Conversion

[Tim Nelson]

- "michel freiha"  wrote: 
> 
> Dear Sir, 
> 
> I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality 
> 
> Regards 
> 


Again, iLBC is poor quality to begin with. You can't take a poor audio sample 
and make it better by converting it to a codec with better 'resolution'. An 
audio sample full of robot voice is going to sound like the same robot voice 
even if you transcode it to a better quality codec, whether that is G.729, 
G.711u, or the latest 'HD Voice' codecs. 


--Tim -- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Codec Conversion

[michel freiha]

Dear Sir,

I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality

Regards

On Thu, Aug 5, 2010 at 4:13 PM, Tim Nelson  wrote:

> - "michel freiha"  wrote:
> >
> > Dear All,
> >
> > i would like to ask please if someone tried to make a codec conversion
> from ilbc to g729, because i did that but the voice quality was too bad and
> a lot of disconnection..
> >
> > Can i get your feedback regarding this issue please?
> >
> > regards
>
> I can't comment on your 'disconnection' as you don't say if that means the
> call is disconnected or you're getting stuttered audio. Regardless, iLBC has
> one of the lowest bitrates of the available codecs and as such the voice
> quality is not spectacular to begin with. Take 'not so good' audio and try
> to convert it to another audio format, and the deficiencies can be
> exacerbated.
>
> --Tim
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Codec Conversion

[Tim Nelson]

- "michel freiha"  wrote: 
> 
> Dear All, 
> 
> i would like to ask please if someone tried to make a codec conversion from 
> ilbc to g729, because i did that but the voice quality was too bad and a lot 
> of disconnection.. 
> 
> Can i get your feedback regarding this issue please? 
> 
> regards 


I can't comment on your 'disconnection' as you don't say if that means the call 
is disconnected or you're getting stuttered audio. Regardless, iLBC has one of 
the lowest bitrates of the available codecs and as such the voice quality is 
not spectacular to begin with. Take 'not so good' audio and try to convert it 
to another audio format, and the deficiencies can be exacerbated. 


--Tim -- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Codec Conversion

[michel freiha]

Dear All,

i would like to ask please if someone tried to make a codec conversion from
ilbc to g729, because i did that but the voice quality was too bad and a lot
of disconnection..

Can i get your feedback regarding this issue please?

regards
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] codec conversion

[Jeff LaCoursiere]

On Tue, 2 Feb 2010, Steve Edwards wrote:

> On Tue, 2 Feb 2010, wassim darwich wrote:
>
>> Thanks for?your reply,ill give?you my situation, iam using my asterisk box 
>> as a switch ,so my client is sending me ulaw and my voip provider?only 
>> accept g723 ,So what i have to do is to receive?g711?codec and convert them 
>> to g723 at?asterisk ,i tried this before but i saw the cpu?usage is 
>> overloaded when doing conversion ,Iam using asterisk 1.4.22 ,So what do you 
>> advice me.
>
> Get your client to switch to g723 or your provider to switch to ulaw. If that 
> is not possible, get more CPU resources:
>
> 1) Eliminate any unrelated processes (X, Apache, MySQL, Tetris). Make sure 
> Asterisk is running with elevated priority.
>
> 2) If your other processes (AGIs?) are written in scripting languages (Perl, 
> PHP), re-code them in compiled languages (C).
>
> 3) Use more powerful processors (faster clock, more cores, more processors).
>
> 4) Split the load across multiple hosts. This has the added advantage of not 
> putting all your eggs in one basket -- you can take a host out of service for 
> maintenance or upgrades.
>
> 5) If you are swapping, more RAM may help.
>

Don't forget the fancy Digium codec translator card thingy!

j

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] codec conversion

[Steve Edwards]

On Tue, 2 Feb 2010, wassim darwich wrote:

Thanks for?your reply,ill give?you my situation, iam using my asterisk 
box as a switch ,so my client is sending me ulaw and my voip 
provider?only accept g723 ,So what i have to do is to receive?g711?codec 
and convert them to g723 at?asterisk ,i tried this before but i saw the 
cpu?usage is overloaded when doing conversion ,Iam using asterisk 1.4.22 
,So what do you advice me.


Get your client to switch to g723 or your provider to switch to ulaw. If 
that is not possible, get more CPU resources:


1) Eliminate any unrelated processes (X, Apache, MySQL, Tetris). Make sure 
Asterisk is running with elevated priority.


2) If your other processes (AGIs?) are written in scripting languages 
(Perl, PHP), re-code them in compiled languages (C).


3) Use more powerful processors (faster clock, more cores, more 
processors).


4) Split the load across multiple hosts. This has the added advantage of 
not putting all your eggs in one basket -- you can take a host out of 
service for maintenance or upgrades.


5) If you are swapping, more RAM may help.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] codec conversion

[wassim darwich]

Hi:
Thanks for your reply,ill give you my situation, iam using my asterisk box as a 
switch ,so my client is sending me ulaw and my voip provider only accept g723 
,So what i have to do is to receive g711 codec and convert them to g723 
at asterisk ,i tried this before but i saw the cpu usage is overloaded when 
doing conversion ,Iam using asterisk 1.4.22 ,So what do you advice me.


  -- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] codec conversion

[Russell Bryant]

On Tue, 2006-08-01 at 18:23 -0400, Wasif wrote:
> What is the best utility to convert GSM files into G729 files for batch
> processing.

I don't think sox supports G729.  However, you can actually use Asterisk
to do this for you if you use the trunk, or upcoming 1.4 release.  In
the trunk, there is a "convert" CLI command.

First, you will need to download codec_g729a.so from Digium.  You will
also need some licenses to use it.

Then, to convert a directory a bunch of gsm files, you could do
something like this ...

   # for n in `ls *.gsm`; do asterisk -rx "convert $n `basename
$n .gsm`.g729"; done

-- 
Russell Bryant
Software Developer
Digium, Inc.

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] codec conversion

[Wasif]

Hello,

What is the best utility to convert GSM files into G729 files for batch
processing.


Thanks

WAzb

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Codec conversion

[Asterisk guy]

for sip calls,   asterisk is able to convert a incoming g729 cal to a outgoing G.711 call. 
 
For oh323,  I am unable to get asterisk to convert a incoming g729 call to a outgoing G711 call .
 
 
my question is : For h323,   how to configure asterisk to convert a incoming h323/g729 calls to a outgoing h323 g.711 call  ?
 
any suggest are welcome.
 
 
On 1/17/05, Helder Rogério [MICROREDE] <[EMAIL PROTECTED]> wrote:

Hi!
 
Is there any way to receive in * server a call from a Terminal adapter in G.723/
G.729 and then convert it to G.711?
 
I'm wondering this because I can only place all thru Broadvoice in G.711 but most of customers have ADSL connection with 128k upstream, so the result is that they can hear in excellent conditions but can't be heard very well the sound is all choppy. even directly to broadvoice thru Xten sip client.

 
So the idea was to act as "proxy" and "codec converter" so that the communication coming out their router is the smaller it can get. I've mentioned G729 or 
G.723 becuase their routers have it, (Draytek 2600V).
 
Thanks in advance for your suggestions
Helder Rogerio___Asterisk-Users mailing list
Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Codec conversion sip peer <> Asterisk

[Helder Rogério [MICROREDE]]

Hi!
 
There's any way to set up a call using 
G726 (sip peer) receive it on Asterisk convert it to G711Mu to send it 
to PSTN broadband termination?
 
I've put  the following in 
sip.conf:
 
 
disalow=all 
allow=gsm
allow=g726 (my TAs use G726 
32K)
 
 
best regards,
Helder
 
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Codec conversion

[Eric Wieling aka ManxPower]

Helder Rogério [MICROREDE] wrote:
Hi!
Is there any way to receive in * server a call from a Terminal adapter in 
G.723/G.729 and then convert it to G.711?
I'm wondering this because I can only place all thru Broadvoice in G.711 but 
most of customers have ADSL connection with 128k upstream, so the result is 
that they can hear in excellent conditions but can't be heard very well the 
sound is all choppy. even directly to broadvoice thru Xten sip client.
So the idea was to act as "proxy" and "codec converter" so that the communication coming out their router is the smaller it can get. I've mentioned G729 or G.723 becuase their routers have it, (Draytek 2600V).
Asterisk does not support G723.1.  You can purchase G729 licenses from 
Digium for $10/channel.  Both codecs are patented.  The G723.1 patent 
holders don't want to license their codec to smaller companies like 
Digium.  The G729 patent holders were more interested in working with 
Digium, so they can sell the G729 codec.

Yes, Asterisk will "transcode" between codecs it supports.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Codec conversion

[Brian Wilkins]

Mr. Colp, 
Try turning-off HTML email before flaming someone, that way people will 
take your comments more seriously instead of seeming like they are coming 
from a newbie. 

On Monday 17 January 2005 05:24 pm, Joshua Colp wrote:
> Hello There,
>
>
>
> Asterisk does not support G723, and it only supports G729 with the usage of
> licenses. These licenses cost $10 per concurrent channel. How did I know
> this? It’s called reading! Did you try Google? Did you try
> http://www.asterisk.org/? Did you try http://www.voip-info.org/? Probably
> not, but all of this was outlined there… so please before asking questions
> check the sites.
>
>
>
> In regards to you using Broadvoice for service with customers… I wouldn’t
> expect to be in business for a long time. Their terms of service
> specifically prevents stuff like this. Have fun.
>
>
>
> - Joshua Colp.
>
>
>
>   _
>
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Helder
> Rogério [MICROREDE]
> Sent: Monday, January 17, 2005 12:34 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Codec conversion
>
>
>
> Hi!
>
>
>
> Is there any way to receive in * server a call from a Terminal adapter in
> G.723/G.729 and then convert it to G.711?
>
>
>
> I'm wondering this because I can only place all thru Broadvoice in G.711
> but most of customers have ADSL connection with 128k upstream, so the
> result is that they can hear in excellent conditions but can't be heard
> very well the sound is all choppy. even directly to broadvoice thru Xten
> sip client.
>
>
>
> So the idea was to act as "proxy" and "codec converter" so that the
> communication coming out their router is the smaller it can get. I've
> mentioned G729 or G.723 becuase their routers have it, (Draytek 2600V).
>
>
>
> Thanks in advance for your suggestions
>
> Helder Rogerio

-- 
Brian Wilkins
Software Engineer
[EMAIL PROTECTED]

Heritage Communications Corporation
  Melbourne, FL USA 32935
321.308.4000 x33
http://www.hcc.net

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Codec conversion

[Brian C. Fertig]

In your SIP.CONF you need to tell * what
codecs to use.   

 

sip.conf

[broadvoice]

disallow=all

allow=ulaw

 

[phone]

disallow=all

allow=g729

 

Then in your extensions.conf you just have
it dial as usual.

 



 

 

.o---o.

Brian Fertig

Network Engineer

Planet Telecom, Inc.

Tampa, FL Office





 







From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Helder Rogério
[MICROREDE]
Sent: Monday, January 17, 2005
11:34 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Codec
conversion



 



Hi!





 





Is there any way to receive in * server a call from a
Terminal adapter in G.723/G.729 and then convert it to G.711?





 





I'm wondering this because I can only place all thru
Broadvoice in G.711 but most of customers have ADSL connection with 128k upstream,
so the result is that they can hear in excellent conditions but can't be heard
very well the sound is all choppy. even directly to broadvoice thru Xten sip
client.





 





So the idea was to act as "proxy" and "codec
converter" so that the communication coming out their router is the
smaller it can get. I've mentioned G729 or G.723 becuase their routers have it,
(Draytek 2600V).





 





Thanks in advance for your suggestions





Helder Rogerio








___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Codec conversion

[Joshua Colp]

Hello There,

 

Asterisk does not support G723, and it
only supports G729 with the usage of licenses. These licenses cost $10 per
concurrent channel. How did I know this? It’s called reading! Did you try
Google? Did you try http://www.asterisk.org/?
Did you try http://www.voip-info.org/? Probably
not, but all of this was outlined there… so please before asking
questions check the sites.

 

In regards to you using Broadvoice for
service with customers… I wouldn’t expect to be in business for a
long time. Their terms of service specifically prevents stuff like this. Have
fun.

 

- Joshua Colp.

 









From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Helder Rogério
[MICROREDE]
Sent: Monday, January 17, 2005
12:34 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Codec
conversion



 



Hi!





 





Is there any way to receive in * server a call from a
Terminal adapter in G.723/G.729 and then convert it to G.711?





 





I'm wondering this because I can only place all thru
Broadvoice in G.711 but most of customers have ADSL connection with 128k upstream,
so the result is that they can hear in excellent conditions but can't be heard
very well the sound is all choppy. even directly to broadvoice thru Xten sip
client.





 





So the idea was to act as "proxy" and "codec
converter" so that the communication coming out their router is the
smaller it can get. I've mentioned G729 or G.723 becuase their routers have it,
(Draytek 2600V).





 





Thanks in advance for your suggestions





Helder Rogerio








___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Codec conversion

[Helder Rogério [MICROREDE]]

Hi!
 
Is there any way to receive in * server a call from 
a Terminal adapter in G.723/G.729 and then convert it to G.711?
 
I'm wondering this because I can only place all 
thru Broadvoice in G.711 but most of customers have ADSL connection with 128k 
upstream, so the result is that they can hear in excellent conditions but can't 
be heard very well the sound is all choppy. even directly to broadvoice thru 
Xten sip client.
 
So the idea was to act as "proxy" and "codec 
converter" so that the communication coming out their router is the smaller it 
can get. I've mentioned G729 or G.723 becuase their routers have it, (Draytek 
2600V).
 
Thanks in advance for your suggestions
Helder Rogerio
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Codec Conversion

[Lyle Giese]

Doesn't g729 require a license?

Lyle
- Original Message - 
From: "Sean Cook" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Thursday, December 02, 2004 8:12 PM
Subject: Re: [Asterisk-Users] Codec Conversion


> I think that all you have to do is where you define the codecs for the
> extention/protocol and asterisk will take care of the rest...
>
> [sip2101]   [sip2102]
> allow=g711allow=g729
>
>
> Asterisk will make the conversion on its own...  I could be wrong 
> but I think that is the way it works
>
>
> Sean
>
> kido noagbodji wrote:
>
> > Hello,
> >
> > Is there an utility for asterisk for codec conversion? I tried google
> > but i haven' got anything.
> > I am trying to initiate a call with G711 codec to asterisk and i would
> > like asterisk to call a gateway with an g729 codec, therefore making a
> > codec conversion from g711 to g729. I know chan_oh323 does it by
> > specifying the OUT_CODEC variable, but chan_h323 does not. And i was
> > wondering is there is a general way of doing that.
> >
> > Thanks
> >
> > K.
> >
> >
> >
> >___
> >Asterisk-Users mailing list
> >[EMAIL PROTECTED]
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Codec Conversion

[Sean Cook]

I think that all you have to do is where you define the codecs for the 
extention/protocol and asterisk will take care of the rest...

[sip2101]   [sip2102]
allow=g711allow=g729
Asterisk will make the conversion on its own...  I could be wrong  
but I think that is the way it works

Sean
kido noagbodji wrote:
Hello,
 
Is there an utility for asterisk for codec conversion? I tried google 
but i haven' got anything.
I am trying to initiate a call with G711 codec to asterisk and i would 
like asterisk to call a gateway with an g729 codec, therefore making a 
codec conversion from g711 to g729. I know chan_oh323 does it by 
specifying the OUT_CODEC variable, but chan_h323 does not. And i was 
wondering is there is a general way of doing that.
 
Thanks
 
K.


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Codec Conversion

[kido noagbodji]

Hello,
 
Is there an utility for asterisk for codec 
conversion? I tried google but i haven' got anything.
I am trying to initiate a call with G711 codec to 
asterisk and i would like asterisk to call a gateway with an g729 codec, 
therefore making a codec conversion from g711 to g729. I know chan_oh323 does it 
by specifying the OUT_CODEC variable, but chan_h323 does not. And i was 
wondering is there is a general way of doing that.
 
Thanks
 
K.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users