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? 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050724/5c156b87/attachment.htm
It has nothing to do with bandwidth. It has everything to do with your routing gear! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050724/ec2f16df/attachment.htm
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: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of BSUMRALLL@aol.com 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! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050724/b5de21ab/attachment.htm
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Eric Wieling aka ManxPower
2005-Jul-25 02:54 UTC
[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.
On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen wrote:> I don’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 “quality” sound better. > Maybe it’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
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 <steve-asterisk@gbnet.net> wrote:> On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen wrote: > > > I don’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 “quality” sound better. > > Maybe it’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 >
Eric Wieling aka ManxPower
2005-Jul-25 05:40 UTC
[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
Steve Kennedy wrote:>On Sun, Jul 24, 2005 at 11:54:17PM -0700, Storm D. J. Petersen wrote: > > > >> I don’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 “quality” sound better. >> Maybe it’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
Eric Wieling aka ManxPower
2005-Jul-25 06:15 UTC
[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’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 “quality” sound better. >>> Maybe it’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
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’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 “quality” sound better. >>>> Maybe it’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 >
>>>> >>> 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.