Re: [asterisk-users] meetme conference using g729?
Tilghman Lesher wrote: On Wednesday 03 October 2007 06:09:01 Peter Fern wrote: Of course, I could be missing something obvious, please correct me if that's the case. I invite you to try it. You could make a lot of really smart people look like fools if you're able to mix compressed audio together without decompressing, or you might make yourself look like a fool, because you get back garbage for attempting to mix compressed data. Easy there tiger, that's just the kind of something obvious I was asking about, and you're right of course. Ta. ___ --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] meetme conference using g729?
On 10/3/07, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 02 October 2007 16:55:52 Brian West wrote: On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote: anyway still if there's a hack for meetme to work with g729 codec this won't be an issue. So is there a hack or patch that i can use any codec for meetme? tnx You still do not understand. It doesn't matter if the call coming in is g729 you must transcode it to signed linear, mix the frames and then code it back into g729 you end up with quality loss doing that. Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential for signal degradation. yeah i still don't understand. this is what i want to do. I want asterisk not to compress and decompress codecs. so either i can use SLIN as my codec for my SIP or IAX. or i can remove SLIN codec in meetme and change it to g729a so there's is no compression and decompression. do you get what i want to do? Thanks! ___ --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] meetme conference using g729?
Mark, Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. yeah i still don't understand. this is what i want to do. I want asterisk not to compress and decompress codecs. so either i can use SLIN as my codec for my SIP or IAX. or i can remove SLIN codec in meetme and change it to g729a so there's is no compression and decompression. do you get what i want to do? Thanks! Tilghman wrote it out: You can not mix two compressed audio streams together. You first have to uncompress them. Even if both audio streams use the same codec, they are compressed thus have to be uncompressed for the mixing of the audio to happen. Better? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] meetme conference using g729?
In article [EMAIL PROTECTED], Mark Quitoriano [EMAIL PROTECTED] wrote: yeah i still don't understand. this is what i want to do. I want asterisk not to compress and decompress codecs. so either i can use SLIN as my codec for my SIP or IAX. or i can remove SLIN codec in meetme and change it to g729a so there's is no compression and decompression. do you get what i want to do? Thanks! Yes, but it can't be done. In order to allow each conference participant to hear all the others at once, it is necessary to mix the audio by adding the contents of each channel. It is impossible to mix G.729 compressed because there is not a simple mathematical relationship between the output data and multiple input data. The mathematical way to do it would be what you are trying to avoid: convert each incoming stream to signed linear samples, then perform the mixing by adding those samples together, and then convert the outgoing mixed stream back to G.729 or whatever. This is what Asterisk does with any kind of codec that talks to Meetme, whether it be uLaw, ALaw, GSM, G.729, ILBC, and it doesn't need all participants to be using the same codec. Why were you so set on mixing G.729 without decoding/encoding? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] meetme conference using g729?
Tilghman Lesher wrote: Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential for signal degradation. Ummm, why?? Unless you can explain some technical reason for this, looks like about 11 lines to change, +3 for correct log messages, +1 for a define, +~3 to add it as a nice config option in meetme.conf. So, in all about... 18 lines worth of code to get it running on any available codec, configurable from meetme.conf, which IMHO would make a lot of sense for single-codec systems... especially for G.729 due to better use of licenses, but for others too, due to load reduction and improved audio quality... Of course, I could be missing something obvious, please correct me if that's the case. ___ --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] meetme conference using g729?
Peter Fern wrote: Tilghman Lesher wrote: Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential for signal degradation. Ummm, why?? Unless you can explain some technical reason for this, looks like about 11 lines to change, +3 for correct log messages, +1 for a define, +~3 to add it as a nice config option in meetme.conf. So, in all about... 18 lines worth of code to get it running on any available codec, configurable from meetme.conf, which IMHO would make a lot of sense for single-codec systems... especially for G.729 due to better use of licenses, but for others too, due to load reduction and improved audio quality... lol. +2 lines of comments. Could you post the patch? ;) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 ___ --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] meetme conference using g729?
On 3 Oct 2007, at 10:16, Mark Quitoriano wrote: On 10/3/07, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 02 October 2007 16:55:52 Brian West wrote: On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote: anyway still if there's a hack for meetme to work with g729 codec this won't be an issue. So is there a hack or patch that i can use any codec for meetme? tnx You still do not understand. It doesn't matter if the call coming in is g729 you must transcode it to signed linear, mix the frames and then code it back into g729 you end up with quality loss doing that. Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential for signal degradation. yeah i still don't understand. this is what i want to do. I want asterisk not to compress and decompress codecs. so either i can use SLIN as my codec for my SIP or IAX. or i can remove SLIN codec in meetme and change it to g729a so there's is no compression and decompression. do you get what i want to do? Thanks! Not exactly. Here are the facts: meetme mixes in SLIN. Any data arriving in anything other than slin will get transcoded twice, once on the way in and again on the way out. Now some opinions: The more efficient the compression of the codec, the less well it copes with decoding and re-encoding. Ulaw and Alaw are simple and not that efficient, but you don't lose any more by re-encoding than you did by decoding in the first place. Tighter codecs like 729 and GSM you will definitely hear the difference. Theory: If you have a conference where there is only _ever_ one speaker at a time, you could (in theory) optimize meetme to do without mixing, and if all the participants were using the same codec, you could get away with not re-encoding by sending out the appropriate incomming packet to all (other) members. I'm guessing that isn't the case for you. Advice: use Ulaw - it's a decent tradeoff for this sort of thing. Tim. ___ --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] meetme conference using g729?
On Wednesday 03 October 2007 06:09:01 Peter Fern wrote: Tilghman Lesher wrote: Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential for signal degradation. Ummm, why?? Unless you can explain some technical reason for this, looks like about 11 lines to change, +3 for correct log messages, +1 for a define, +~3 to add it as a nice config option in meetme.conf. So, in all about... 18 lines worth of code to get it running on any available codec, configurable from meetme.conf, which IMHO would make a lot of sense for single-codec systems... especially for G.729 due to better use of licenses, but for others too, due to load reduction and improved audio quality... Of course, I could be missing something obvious, please correct me if that's the case. I invite you to try it. You could make a lot of really smart people look like fools if you're able to mix compressed audio together without decompressing, or you might make yourself look like a fool, because you get back garbage for attempting to mix compressed data. While I won't go so far as to say mixing compressed audio is impossible without decompressing first, it is not *simple* by any means whatsoever. In fact, I would go so far as to say that not only are you likely to degrade the audio even further, but the CPU time it would take is an order of magnitude higher than the current methodology. -- Tilghman ___ --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] meetme conference using g729?
I have been following this discussion. You do have a point. However, the way * works right now. If a channel does not require trans-coding to get into a conference, coder usage is counted. So I really do not know what difference putting the transcoding in meetme is going to make. I mean, how could this better contribute to better use of G729 licenses. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Fern Sent: Wednesday, October 03, 2007 7:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] meetme conference using g729? Tilghman Lesher wrote: Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential for signal degradation. Ummm, why?? Unless you can explain some technical reason for this, looks like about 11 lines to change, +3 for correct log messages, +1 for a define, +~3 to add it as a nice config option in meetme.conf. So, in all about... 18 lines worth of code to get it running on any available codec, configurable from meetme.conf, which IMHO would make a lot of sense for single-codec systems... especially for G.729 due to better use of licenses, but for others too, due to load reduction and improved audio quality... Of course, I could be missing something obvious, please correct me if that's the case. ___ --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 ___ --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] meetme conference using g729?
But his preference of G729 is to save bandwidth. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Wednesday, October 03, 2007 8:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] meetme conference using g729? Not exactly. Here are the facts: meetme mixes in SLIN. Any data arriving in anything other than slin will get transcoded twice, once on the way in and again on the way out. Now some opinions: The more efficient the compression of the codec, the less well it copes with decoding and re-encoding. Ulaw and Alaw are simple and not that efficient, but you don't lose any more by re-encoding than you did by decoding in the first place. Tighter codecs like 729 and GSM you will definitely hear the difference. Theory: If you have a conference where there is only _ever_ one speaker at a time, you could (in theory) optimize meetme to do without mixing, and if all the participants were using the same codec, you could get away with not re-encoding by sending out the appropriate incomming packet to all (other) members. I'm guessing that isn't the case for you. Advice: use Ulaw - it's a decent tradeoff for this sort of thing. Tim. ___ --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 ___ --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] meetme conference using g729?
If bandwidth were not an issue, I would think everyone would opt for ulaw or alaw. Why compress and use CPU cycles and G729 licenses if there were no bandwidth issues? Thanks, Steve totaro Wai Wu wrote: But his preference of G729 is to save bandwidth. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: Wednesday, October 03, 2007 8:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] meetme conference using g729? Not exactly. Here are the facts: meetme mixes in SLIN. Any data arriving in anything other than slin will get transcoded twice, once on the way in and again on the way out. Now some opinions: The more efficient the compression of the codec, the less well it copes with decoding and re-encoding. Ulaw and Alaw are simple and not that efficient, but you don't lose any more by re-encoding than you did by decoding in the first place. Tighter codecs like 729 and GSM you will definitely hear the difference. Theory: If you have a conference where there is only _ever_ one speaker at a time, you could (in theory) optimize meetme to do without mixing, and if all the participants were using the same codec, you could get away with not re-encoding by sending out the appropriate incomming packet to all (other) members. I'm guessing that isn't the case for you. Advice: use Ulaw - it's a decent tradeoff for this sort of thing. Tim. ___ --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] meetme conference using g729?
On Wed, 3 Oct 2007 08:35:06 -0500, Tilghman Lesher wrote: I invite you to try it. You could make a lot of really smart people look like fools if you're able to mix compressed audio together without decompressing, or you might make yourself look like a fool, because you get back garbage for attempting to mix compressed data. I wholly understand the problem here. You can't, at present, mix compressed audio stream, in compressed domain. You must decode them to baseband, do the manipulation, then re-encode. OK, we get that. That's today. Such things have parallels in my day job, which is television production transmission. At least in the US the signal that a TV station delivers to its DTV transmitter (ie the new digital one, not the old analog one that the feds will make us turn off in 2008) that is a compressed stream. Typically MPEG2 @ 19.2 MPBS. There was a time when that was a signal stream that could not be manipulated. It was just the transport mechanism from the last leg before the transmitter. Many companies wanted to be able to perform what seemed simple manipulations on the stream, for example to add a station logo, without taking the very significant quality hit of decompression and recompression. Such hardware systems have become available over time. Manipulation of the transmission streams in the compressed domain is possible, but its very compute intensive...and so expensive. It's done in massively parallel hardware architecture. There are a few vendors in the broadcast business who provide such systems. And that's for high bandwidth broadcast video. It would also be possible for voice streams, but the math is very complex. Hardware acceleration of encoding is already very common, witness Digium's own encode/decode board. Given the right motivation to spur the development this could be possible. In truth I suspect that there's little economic reason to do it. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 c713-201-1262 skype mjgraves fwd 54245 ___ --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] meetme conference using g729?
On 10/2/07, Brian West [EMAIL PROTECTED] wrote: Ok Let me chime in on this one. If you can use ulaw/alaw because you'll end up with tandem encoding which will make the conference sound worse to some people. All audio coming in will get transcoded to signed linear and pushed down into zaptel then back up and out to the conference participants. You'll end up with the best audio quality if you limit the transcoding. /b meetme uses signed linear(slin) right? so if you're using ulaw/alaw codec there's still transcoding from ulaw to slin right? can i use slin for my sip channels? anyway still if there's a hack for meetme to work with g729 codec this won't be an issue. So is there a hack or patch that i can use any codec for meetme? tnx ___ --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] meetme conference using g729?
You still do not understand. It doesn't matter if the call coming in is g729 you must transcode it to signed linear, mix the frames and then code it back into g729 you end up with quality loss doing that. /b On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote: anyway still if there's a hack for meetme to work with g729 codec this won't be an issue. So is there a hack or patch that i can use any codec for meetme? tnx ___ --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 ___ --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] meetme conference using g729?
On Tuesday 02 October 2007 16:55:52 Brian West wrote: On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote: anyway still if there's a hack for meetme to work with g729 codec this won't be an issue. So is there a hack or patch that i can use any codec for meetme? tnx You still do not understand. It doesn't matter if the call coming in is g729 you must transcode it to signed linear, mix the frames and then code it back into g729 you end up with quality loss doing that. Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential for signal degradation. -- Tilghman ___ --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] meetme conference using g729?
Thanks for making it clearer :) My mind is mush today! /b On Oct 2, 2007, at 5:39 PM, Tilghman Lesher wrote: Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential for signal degradation. -- Tilghman ___ --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] meetme conference using g729?
In my experience, and theoretically by design, it doesn't matter what codec you are using when you call a meetme conference. Moj Mark Quitoriano wrote: Hi, is there a way to use g729 in meetme? Thanks! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ --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] meetme conference using g729?
but is there a way to use g729 codec in meetme? On 10/2/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: In my experience, and theoretically by design, it doesn't matter what codec you are using when you call a meetme conference. Moj ___ --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] meetme conference using g729?
Ok Let me chime in on this one. If you can use ulaw/alaw because you'll end up with tandem encoding which will make the conference sound worse to some people. All audio coming in will get transcoded to signed linear and pushed down into zaptel then back up and out to the conference participants. You'll end up with the best audio quality if you limit the transcoding. /b On Oct 1, 2007, at 6:37 PM, Mark Quitoriano wrote: but is there a way to use g729 codec in meetme? On 10/2/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: In my experience, and theoretically by design, it doesn't matter what codec you are using when you call a meetme conference. Moj ___ --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 ___ --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] meetme conference using g729?
As long as you have some g729 codecs installed, Asterisk will do this fine. PaulH On Tue, 2007-10-02 at 07:37 +0800, Mark Quitoriano wrote: but is there a way to use g729 codec in meetme? On 10/2/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: In my experience, and theoretically by design, it doesn't matter what codec you are using when you call a meetme conference. Moj ___ --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 ___ --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] meetme conference using g729?
Hello Mark, On 10/2/07, Mark Quitoriano [EMAIL PROTECTED] wrote: but is there a way to use g729 codec in meetme? You have to buy a G.729 license for each channel which I believe is at USD 10.00 if I'm not mistaken. Then, make sure that your machine is fast enough for transcoding. But the best solution that I think is a codec passthrough which I think is not supported in Asterisk. Good luck! ___ --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] meetme conference using g729?
Since the channels have to be mixed together by Asterisk, passthrough can't be supported in this case. In other circumstances, passthru works fine. PaulH On Tue, 2007-10-02 at 10:02 +0800, GNUbie wrote: Hello Mark, On 10/2/07, Mark Quitoriano [EMAIL PROTECTED] wrote: but is there a way to use g729 codec in meetme? You have to buy a G.729 license for each channel which I believe is at USD 10.00 if I'm not mistaken. Then, make sure that your machine is fast enough for transcoding. But the best solution that I think is a codec passthrough which I think is not supported in Asterisk. Good luck! ___ --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 ___ --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
