Re: [Asterisk-Users] super high bandwidth codec
On Wed, Jul 27, 2005 at 08:50:54PM -0700, Andrew C. Brown wrote: Andrew C. Brown wrote: BTW, with all this talk about wideband iLBC, we should probably start using correct terms since iLBC actually stands for internet Low Bandwidth Codec. So wb iLBC is an oxymoron. Oops. Correcting myself, iLBC stands for internet Low Bitrate Codec, which is not strictly the same thing. Even so, it is used to refer to a codec which is of the normal 8 kHz sample bandwidth. The wideband (and proprietary) GIPS codecs are called iSAC and iPCM-wb and are the codecs used by Skype and Gizmo. They both have 16 kHz sample rates. So Asterisk can't talk to gizmo? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] super high bandwidth codec
On Tue, Jul 26, 2005 at 10:42:35PM -0700, Andrew C. Brown wrote: A recent blog entry indicated that GIPS was issuing licenses for its technology from a mere $50k for unlimited licenses with respect to an agreement with Microsoft. I don't have a huge concern about bandwidth limits. If I could get better quality than G.711 in the same bandwidhth that would be great. However, since I'm using IAX2 based DIDs and termination would it really matter? That is, if the ITSPs are connection to the PSTN via TDM interconnects wouldn't any calls be limited to G.711 quality anyway? IAX2 is a protocol, not a codec, so has little impact on sampling quality. But the second assumption is correct. If you are going to PSTN at any point in the chain, you are back to 8kHz sample rate and that extra spectrum you put over iSAC or whatever is tossed out the window. And also when you use MeetMe, right? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] super high bandwidth codec
Tzafrir Cohen wrote: On Tue, Jul 26, 2005 at 10:42:35PM -0700, Andrew C. Brown wrote: A recent blog entry indicated that GIPS was issuing licenses for its technology from a mere $50k for unlimited licenses with respect to an agreement with Microsoft. I don't have a huge concern about bandwidth limits. If I could get better quality than G.711 in the same bandwidhth that would be great. However, since I'm using IAX2 based DIDs and termination would it really matter? That is, if the ITSPs are connection to the PSTN via TDM interconnects wouldn't any calls be limited to G.711 quality anyway? IAX2 is a protocol, not a codec, so has little impact on sampling quality. But the second assumption is correct. If you are going to PSTN at any point in the chain, you are back to 8kHz sample rate and that extra spectrum you put over iSAC or whatever is tossed out the window. And also when you use MeetMe, right? I'm researching that. All I've been able to find so far is http://lists.digium.com/pipermail/asterisk-users/2005-May/107214.html which says that basically, no, Asterisk can't yet handle anything but 8KHz sample rates (though I suppose that doesn't necessarily preclude reinvited peer to peer VoIP calls where Asterisk removes itself from the audio path). If you find any more references on that issue, please post them. This question of high quality voice is going to keep coming up so I'd like there to be Wiki page to bring people up to date on all this we're discussing. And frankly I'd like to help build some momentum towards increased spectrum voice telephony. Right now, few people even think to ask and VoIP to them is just about saving money rather than improving the product. ___ 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] super high bandwidth codec
Andrew C. Brown wrote: BTW, with all this talk about wideband iLBC, we should probably start using correct terms since iLBC actually stands for internet Low Bandwidth Codec. So wb iLBC is an oxymoron. Oops. Correcting myself, iLBC stands for internet Low Bitrate Codec, which is not strictly the same thing. Even so, it is used to refer to a codec which is of the normal 8 kHz sample bandwidth. The wideband (and proprietary) GIPS codecs are called iSAC and iPCM-wb and are the codecs used by Skype and Gizmo. They both have 16 kHz sample rates. ___ 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] super high bandwidth codec
Skype uses wideband-ilbc. I don't think thats right. I think it just uses iLBC over it's own proprietary Voip protocol. http://www.skype.com/help/faq/technical.html How much bandwidth does Skype use while I'm in a call? Skype automatically selects the best codec depending on the connection between yourself and the person you are calling. On average, Skype uses between 3-16 kilobytes/sec depending on bandwidth available for other party, network conditions in between, callers CPU performance, etc. ___ 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] super high bandwidth codec
Geoff Manning wrote: Skype uses wideband-ilbc. I don't think thats right. I think it just uses iLBC over it's own proprietary Voip protocol. http://www.skype.com/help/faq/technical.html How much bandwidth does Skype use while I'm in a call? Skype automatically selects the best codec depending on the connection between yourself and the person you are calling. On average, Skype uses between 3-16 kilobytes/sec depending on bandwidth available for other party, network conditions in between, callers CPU performance, etc. I don't think that's correct. I don't have the link to the Columbia paper where they tried (with only mixed success) to figure out what all nefarious stuff Skype does (hijacking port 80 being the most pernicious) but I'm pretty sure they have figured out that if possible, it will use the (proprietary) wideband version of iLBC. B. ___ 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] super high bandwidth codec
Brian Capouch wrote: Geoff Manning wrote: Skype uses wideband-ilbc. I don't think thats right. I think it just uses iLBC over it's own proprietary Voip protocol. http://www.skype.com/help/faq/technical.html How much bandwidth does Skype use while I'm in a call? Skype automatically selects the best codec depending on the connection between yourself and the person you are calling. On average, Skype uses between 3-16 kilobytes/sec depending on bandwidth available for other party, network conditions in between, callers CPU performance, etc. I don't think that's correct. I don't have the link to the Columbia paper where they tried (with only mixed success) to figure out what all nefarious stuff Skype does (hijacking port 80 being the most pernicious) but I'm pretty sure they have figured out that if possible, it will use the (proprietary) wideband version of iLBC. BTW, with all this talk about wideband iLBC, we should probably start using correct terms since iLBC actually stands for internet Low Bandwidth Codec. So wb iLBC is an oxymoron. GIPS has a few proprietary codecs. Probably the one we are referring to is iPCM-wb, which according to GIPS (globalipsound.com) ...is a wide band codec with extreme robustness against packet loss [that] provides better-than-PSTN quality across ... the public Internet. They also have iSAC which supposedly ...makes VoIP communications possible using a dial-up modem, automatically adjusting transmission rates to deliver better-than-PSTN voice quality. Skype is listed as a customer of GIPS on GIPS' website, but I don't know which codec Skype uses. It must be one of the proprietary ones though. ___ 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] super high bandwidth codec
Brian Capouch wrote: Geoff Manning wrote: Skype uses wideband-ilbc. I don't think thats right. I think it just uses iLBC over it's own proprietary Voip protocol. http://www.skype.com/help/faq/technical.html How much bandwidth does Skype use while I'm in a call? Skype automatically selects the best codec depending on the connection between yourself and the person you are calling. On average, Skype uses between 3-16 kilobytes/sec depending on bandwidth available for other party, network conditions in between, callers CPU performance, etc. I don't think that's correct. I don't have the link to the Columbia paper where they tried (with only mixed success) to figure out what all nefarious stuff Skype does (hijacking port 80 being the most pernicious) but I'm pretty sure they have figured out that if possible, it will use the (proprietary) wideband version of iLBC. FYI: One can find the columbia paper link by going to the VoIP wiki's Skype page. According to GIPS datasheets, GIPS offers two proprietary wideband codecs. iPCM-wb and iSAC. Both have 16kHZ sample rate* which is double the 8kHz of PSTN, iLBC and most of the other codecs, hence the relatively wonderous sound quality which I, among others, covet for Asterisk. The channel bit rate is respectively (it varies dynamically) iLBC (free) 13-15kbps iSAC ($)10-30kbps iPCM-wb ($) 80kbps iPCM-wb doesn't seem to offer any outright fidelity advantage over iSAC since they are the same sample rate. I presume all those extra bits are redundancy to make the quality more robust. * Ref: http://www.globalipsound.com/solutions/solutions_Codecs.php ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] super high bandwidth codec
On Tue, 26 Jul 2005 18:50:11 -0700, Andrew C. Brown wrote: Brian Capouch wrote: Geoff Manning wrote: Skype uses wideband-ilbc. I don't think thats right. I think it just uses iLBC over it's own proprietary Voip protocol. http://www.skype.com/help/faq/technical.html How much bandwidth does Skype use while I'm in a call? Skype automatically selects the best codec depending on the connection between yourself and the person you are calling. On average, Skype uses between 3-16 kilobytes/sec depending on bandwidth available for other party, network conditions in between, callers CPU performance, etc. I don't think that's correct. I don't have the link to the Columbia paper where they tried (with only mixed success) to figure out what all nefarious stuff Skype does (hijacking port 80 being the most pernicious) but I'm pretty sure they have figured out that if possible, it will use the (proprietary) wideband version of iLBC. FYI: One can find the columbia paper link by going to the VoIP wiki's Skype page. According to GIPS datasheets, GIPS offers two proprietary wideband codecs. iPCM-wb and iSAC. Both have 16kHZ sample rate* which is double the 8kHz of PSTN, iLBC and most of the other codecs, hence the relatively wonderous sound quality which I, among others, covet for Asterisk. The channel bit rate is respectively (it varies dynamically) iLBC (free)13-15kbps iSAC ($) 10-30kbps iPCM-wb ($)80kbps iPCM-wb doesn't seem to offer any outright fidelity advantage over iSAC since they are the same sample rate. I presume all those extra bits are redundancy to make the quality more robust. * Ref: http://www.globalipsound.com/solutions/solutions_Codecs.php A recent blog entry indicated that GIPS was issuing licenses for its technology from a mere $50k for unlimited licenses with respect to an agreement with Microsoft. I don't have a huge concern about bandwidth limits. If I could get better quality than G.711 in the same bandwidhth that would be great. However, since I'm using IAX2 based DIDs and termination would it really matter? That is, if the ITSPs are connection to the PSTN via TDM interconnects wouldn't any calls be limited to G.711 quality anyway? Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] super high bandwidth codec
On Tue, Jul 26, 2005 at 10:18:35PM -0500, Michael Graves wrote: A recent blog entry indicated that GIPS was issuing licenses for its technology from a mere $50k for unlimited licenses with respect to an agreement with Microsoft. I don't have a huge concern about bandwidth limits. If I could get better quality than G.711 in the same bandwidhth that would be great. Try speex with a license price of 0$ per year. However, since I'm using IAX2 based DIDs and termination would it really matter? That is, if the ITSPs are connection to the PSTN via TDM interconnects wouldn't any calls be limited to G.711 quality anyway? Please, not another patented algorithm. We have enough troubles from those already. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] super high bandwidth codec
Michael Graves wrote: On Tue, 26 Jul 2005 18:50:11 -0700, Andrew C. Brown wrote: Brian Capouch wrote: Geoff Manning wrote: Skype uses wideband-ilbc. I don't think thats right. I think it just uses iLBC over it's own proprietary Voip protocol. http://www.skype.com/help/faq/technical.html How much bandwidth does Skype use while I'm in a call? Skype automatically selects the best codec depending on the connection between yourself and the person you are calling. On average, Skype uses between 3-16 kilobytes/sec depending on bandwidth available for other party, network conditions in between, callers CPU performance, etc. I don't think that's correct. I don't have the link to the Columbia paper where they tried (with only mixed success) to figure out what all nefarious stuff Skype does (hijacking port 80 being the most pernicious) but I'm pretty sure they have figured out that if possible, it will use the (proprietary) wideband version of iLBC. FYI: One can find the columbia paper link by going to the VoIP wiki's Skype page. According to GIPS datasheets, GIPS offers two proprietary wideband codecs. iPCM-wb and iSAC. Both have 16kHZ sample rate* which is double the 8kHz of PSTN, iLBC and most of the other codecs, hence the relatively wonderous sound quality which I, among others, covet for Asterisk. The channel bit rate is respectively (it varies dynamically) iLBC (free) 13-15kbps iSAC ($) 10-30kbps iPCM-wb ($) 80kbps iPCM-wb doesn't seem to offer any outright fidelity advantage over iSAC since they are the same sample rate. I presume all those extra bits are redundancy to make the quality more robust. * Ref: http://www.globalipsound.com/solutions/solutions_Codecs.php A recent blog entry indicated that GIPS was issuing licenses for its technology from a mere $50k for unlimited licenses with respect to an agreement with Microsoft. I don't have a huge concern about bandwidth limits. If I could get better quality than G.711 in the same bandwidhth that would be great. However, since I'm using IAX2 based DIDs and termination would it really matter? That is, if the ITSPs are connection to the PSTN via TDM interconnects wouldn't any calls be limited to G.711 quality anyway? IAX2 is a protocol, not a codec, so has little impact on sampling quality. But the second assumption is correct. If you are going to PSTN at any point in the chain, you are back to 8kHz sample rate and that extra spectrum you put over iSAC or whatever is tossed out the window. ___ 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] super high bandwidth codec
I dont know if I have the same experiences. Usually my Skype calls are very garbled at first. I find that my G729 Asterisk calls are better quality. You can try using ULAW if you have the bandwidth. It. might make the quality sound better. Maybe its your SIP client/hardware phone that is giving you troubles. Storm. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Sunday, July 24, 2005 8:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] super high bandwidth codec Ive just gotten off a skype conference call and it pisses me off that the quality of skype is higher than my asterisk calls. Is there such a thing as a super high bandwidth codec? In a situation that you have the bandwidth to share is there something that I can use for important calls when the situation warrants it? TIA, Dean smime.p7s Description: S/MIME cryptographic signature ___ 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] super high bandwidth codec
Dean Collins wrote: I've just gotten off a skype conference call and it pisses me off that the quality of skype is higher than my asterisk calls. Is there such a thing as a super high bandwidth codec? Asterisk does not support wideband codecs as far as I know. Most telephony gear expects most calls to be handed at some point by a PSTN channel (FXO or FXS) or by a VoIP hardphone. The highest bandwidth those devices support is ulaw or alaw. Hardphones COULD support wideband codecs, but I don't know of any that actually do. Of course, if all legs of a call uses a wideband codec, like if you are only using Softphones, then in theory you could use a wideband codec. Skype, which until recently didn't even support PSTN or hardphones might very well use a wideband codec in order to fool users into thinking it's a better product. NOTE: CVS-HEAD allows you to specify more options for codecs. SpeeX might be able to be set for wideband mode, but that won't make much difference if your hardware/software doesn't support it. ___ 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] super high bandwidth codec
On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen wrote: I don#8217;t know if I have the same experiences. Usually my Skype calls are very garbled at first. I find that my G729 Asterisk calls are better quality. You can try using ULAW if you have the bandwidth. It. might make the #8220;quality#8221; sound better. Maybe it#8217;s your SIP client/hardware phone that is giving you troubles. Skype uses ilbc, and g.729 for PSTN breakout. Steve -- NetTek Ltd Fax +44-(0)20 7483 2455 Skype / In stevekennedyuk / UK +442088167166 / US +13106518226 Vonage UK +442079932612 / US +13108577715 / UK mob 07775 755503 Personal Blog http://stevekennedy.blogspot.com Euro Tech News Blog http://eurotechnews.blogspot.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] super high bandwidth codec
Yes and ilbc is more robust against packet loss, jitter etc. with using not very much but less more bandwith. Asterisk has support for ilbc and there are many providers offering PSTN termination with ilbc codec. And voice quality is better than g723. check out http://www.ilbcfreeware.org/ Regards Deniz On 7/25/05, Steve Kennedy [EMAIL PROTECTED] wrote: On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen wrote: I don#8217;t know if I have the same experiences. Usually my Skype calls are very garbled at first. I find that my G729 Asterisk calls are better quality. You can try using ULAW if you have the bandwidth. It. might make the #8220;quality#8221; sound better. Maybe it#8217;s your SIP client/hardware phone that is giving you troubles. Skype uses ilbc, and g.729 for PSTN breakout. Steve -- NetTek Ltd Fax +44-(0)20 7483 2455 Skype / In stevekennedyuk / UK +442088167166 / US +13106518226 Vonage UK +442079932612 / US +13108577715 / UK mob 07775 755503 Personal Blog http://stevekennedy.blogspot.com Euro Tech News Blog http://eurotechnews.blogspot.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 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] super high bandwidth codec
Deniz Pecel wrote: Yes and ilbc is more robust against packet loss, jitter etc. with using not very much but less more bandwith. Asterisk has support for ilbc and there are many providers offering PSTN termination with ilbc codec. And voice quality is better than g723. check out http://www.ilbcfreeware.org/ iLBC does not seem to support any kind of wideband mode, so it will not be any clearer than plain old G711 ulaw/alaw. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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] super high bandwidth codec
Steve Kennedy wrote: On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen wrote: I don#8217;t know if I have the same experiences. Usually my Skype calls are very garbled at first. I find that my G729 Asterisk calls are better quality. You can try using ULAW if you have the bandwidth. It. might make the #8220;quality#8221; sound better. Maybe it#8217;s your SIP client/hardware phone that is giving you troubles. Skype uses ilbc, and g.729 for PSTN breakout. Skype uses wideband-ilbc. Steve ___ 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] super high bandwidth codec
Steve Underwood wrote: Steve Kennedy wrote: On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen wrote: I don#8217;t know if I have the same experiences. Usually my Skype calls are very garbled at first. I find that my G729 Asterisk calls are better quality. You can try using ULAW if you have the bandwidth. It. might make the #8220;quality#8221; sound better. Maybe it#8217;s your SIP client/hardware phone that is giving you troubles. Skype uses ilbc, and g.729 for PSTN breakout. Skype uses wideband-ilbc. Do yu have a link for wideband-ilbc info? -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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] super high bandwidth codec
Eric Wieling aka ManxPower wrote: Steve Underwood wrote: Steve Kennedy wrote: On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen wrote: I don#8217;t know if I have the same experiences. Usually my Skype calls are very garbled at first. I find that my G729 Asterisk calls are better quality. You can try using ULAW if you have the bandwidth. It. might make the #8220;quality#8221; sound better. Maybe it#8217;s your SIP client/hardware phone that is giving you troubles. Skype uses ilbc, and g.729 for PSTN breakout. Skype uses wideband-ilbc. Do yu have a link for wideband-ilbc info? It is described on the GIPS site, along with the narrow band ilbc. The wideband one is not offered to the world on a royalty free basis, as the narrow band one is. I have never looked at how it works. I don't know how similar/different the narrow and wideband codecs are. The wideband codec operates at 16k samples/second. Regards, Steve ___ 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] super high bandwidth codec
Steve Underwood wrote: Eric Wieling aka ManxPower wrote: Do yu have a link for wideband-ilbc info? It is described on the GIPS site, along with the narrow band ilbc. The wideband one is not offered to the world on a royalty free basis, as the narrow band one is. I have never looked at how it works. I don't know how similar/different the narrow and wideband codecs are. The wideband codec operates at 16k samples/second. That would explain why I didn't know about it. Other than G729, pretty much any codec that is not free is off my radar. --Eric -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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] super high bandwidth codec
On Mon, Jul 25, 2005 at 03:14:44PM +0300, Deniz Pecel wrote: Yes and ilbc is more robust against packet loss, jitter etc. with using not very much but less more bandwith. Asterisk has support for ilbc and there are many providers offering PSTN termination with ilbc codec. And voice quality is better than g723. check out http://www.ilbcfreeware.org/ What about other codecs supported by *? Speex's homepage claims speex features integration of multiple sampling rates in the same bitstream. Is this supported anywhere? What about g726? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ 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] super high bandwidth codec
http://www.globalipsound.com Try there. /b On Jul 25, 2005, at 8:15 AM, Eric Wieling aka ManxPower wrote: Steve Underwood wrote: Steve Kennedy wrote: On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen wrote: I don#8217;t know if I have the same experiences. Usually my Skype calls are very garbled at first. I find that my G729 Asterisk calls are better quality. You can try using ULAW if you have the bandwidth. It. might make the #8220;quality#8221; sound better. Maybe it#8217;s your SIP client/hardware phone that is giving you troubles. Skype uses ilbc, and g.729 for PSTN breakout. Skype uses wideband-ilbc. Do yu have a link for wideband-ilbc info? -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ 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 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] super high bandwidth codec
It has nothing to do with bandwidth. It has everything to do with your routing gear! ___ 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] super high bandwidth codec
How do you figure? How does skype sounds so damm good on the same network/machine? I think you might be wrong. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, 25 July 2005 12:11 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] super high bandwidth codec It has nothing to do with bandwidth. It has everything to do with your routing gear! ___ 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] super high bandwidth codec
Title: Message -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: Sunday, July 24, 2005 9:11 PMTo: asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] super high bandwidth codec It has nothing to do with bandwidth. It has everything to do with your routing gear! This is completely incorrect. Skype uses a codec that uses far more bandwidth than traditional telephony provides, which is why it's audio can have morerange than even the best quality phone call. In theory, there is nothing preventing an all VOIP network from using such a codec, but as a practical matter, at least part of most phone calls are via traditional phone gear and/or networks, you don't see it widely deployed. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.4/57 - Release Date: 07/22/2005 ___ 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
