Re: [asterisk-users] Codec Conversion
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 mmol...@millenium.com.cowrote: El 05/08/10 14:50, Tim Nelson escribió: - michel freiha mich...@gmail.com mich...@gmail.com 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
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 mmol...@millenium.com.co mailto:mmol...@millenium.com.co wrote: El 05/08/10 14:50, Tim Nelson escribió: - michel freiha mich...@gmail.com mailto:mich...@gmail.com 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
Steve- On 08/07/2010 03:15 AM, Jeff Brower wrote: Steve- El 05/08/10 14:50, Tim Nelson escribió: - michel freihamich...@gmail.comwrote: 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 ATT 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, RD, 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 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
Re: [asterisk-users] Codec Conversion
Steve- On 08/06/2010 05:40 AM, Jeff Brower wrote: Miguel- El 05/08/10 14:50, Tim Nelson escribió: - michel freihamich...@gmail.com 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
On Fri, 6 Aug 2010 03:43:33 -0500 (CDT), Jeff Brower wrote: snip 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 mgravesatmstvp.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
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: snip 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 mgravesatmstvp.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
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 freihamich...@gmail.com 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
Steve- El 05/08/10 14:50, Tim Nelson escribió: - michel freihamich...@gmail.com 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 ATT 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 be 27 payload bytes. For IP/UDP/RTP usage, that much delay could well be counterproductive. Places where I have seen MELPe effectively used for VoIP include applications
Re: [asterisk-users] Codec Conversion
On 08/07/2010 03:15 AM, Jeff Brower wrote: Steve- El 05/08/10 14:50, Tim Nelson escribió: - michel freihamich...@gmail.comwrote: 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 ATT 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 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
Re: [asterisk-users] Codec Conversion
- michel freiha mich...@gmail.com 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
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 tnel...@rockbochs.com wrote: - michel freiha mich...@gmail.com 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
- michel freiha mich...@gmail.com 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- 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 tnel...@rockbochs.com wrote: - michel freiha mich...@gmail.com 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
El 05/08/10 14:50, Tim Nelson escribió: - michel freiha mich...@gmail.com 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
Miguel- El 05/08/10 14:50, Tim Nelson escribió: - michel freiha mich...@gmail.com 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
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
On 08/06/2010 05:40 AM, Jeff Brower wrote: Miguel- El 05/08/10 14:50, Tim Nelson escribió: - michel freihamich...@gmail.com 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
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
Re: [asterisk-users] codec conversion
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
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
Re: [Asterisk-Users] Codec conversion
for sip calls, asterisk is able to convert a incoming g729 cal to a outgoing G.711 call. Foroh323, I am unable toget 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
RE: [Asterisk-Users] Codec conversion
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? Its 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 wouldnt 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
RE: [Asterisk-Users] Codec conversion
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
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? Its 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 wouldnt 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
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
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
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
