This might have been talked about before, but I'm posting anyhow. I've got down to testing Asterisk yesterday, and I couldn't help but compare it with Skype (a Windoze only product, yet, but extremely efficient for some reason). Skype has almost unperceptible delay (LAN), while there is almost half a second of delay regardless of the codec on Asterisk. An even if we were to eliminate the latency on asterisk, Skype has at least 2x the quality of PSTN, let alone any codec on asterisk. So I ask: are there no royalty-free codecs that come close to those from GIPS? (I understand these people are supplying the codecs for Skype). Or is there something I'm doing wrong? What configuration yields the highest quality and the lowest bitrate? I'm talking LAN communication here: plenty of bandwidth, no latency. (I'm aware that this could be a flamebait. Bring em on.) -- Mathieu Nantel TecLinux.com mnantel@teclinux.com
A) Skype uses iLBC as a codec which Asterisk already has B) Asterisk won't be adding latency on LAN->LAN, it's your end clients - try Firefly - www.virbiage.com (insert bias comment here) and see if that's addresses your issues. Mathieu Nantel wrote:> This might have been talked about before, but I'm posting anyhow. > > I've got down to testing Asterisk yesterday, and I couldn't help but > compare it with Skype (a Windoze only product, yet, but extremely > efficient for some reason). > > Skype has almost unperceptible delay (LAN), while there is almost half a > second of delay regardless of the codec on Asterisk. > > An even if we were to eliminate the latency on asterisk, Skype has at > least 2x the quality of PSTN, let alone any codec on asterisk. > > So I ask: are there no royalty-free codecs that come close to those from > GIPS? (I understand these people are supplying the codecs for Skype). > > Or is there something I'm doing wrong? What configuration yields the > highest quality and the lowest bitrate? I'm talking LAN communication > here: plenty of bandwidth, no latency. > > (I'm aware that this could be a flamebait. Bring em on.) > > -- > Mathieu Nantel > TecLinux.com > mnantel@teclinux.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 >
I have to throw my 2 cents in here, I'm always disappointed when I use Skype how good it sounds as a peer to peer service in comparison to using my asterisk and IAX. I'd be willing to pay for that codec but then unless who I'm talking to has it as well there not much point. Cheers, Dean -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Adam Hart Sent: Friday, 30 April 2004 9:23 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk VS. Skype A) Skype uses iLBC as a codec which Asterisk already has B) Asterisk won't be adding latency on LAN->LAN, it's your end clients - try Firefly - www.virbiage.com (insert bias comment here) and see if that's addresses your issues. Mathieu Nantel wrote:> This might have been talked about before, but I'm posting anyhow. > > I've got down to testing Asterisk yesterday, and I couldn't help but > compare it with Skype (a Windoze only product, yet, but extremely > efficient for some reason). > > Skype has almost unperceptible delay (LAN), while there is almost halfa> second of delay regardless of the codec on Asterisk. > > An even if we were to eliminate the latency on asterisk, Skype has at > least 2x the quality of PSTN, let alone any codec on asterisk. > > So I ask: are there no royalty-free codecs that come close to thosefrom> GIPS? (I understand these people are supplying the codecs for Skype). > > Or is there something I'm doing wrong? What configuration yields the > highest quality and the lowest bitrate? I'm talking LAN communication > here: plenty of bandwidth, no latency. > > (I'm aware that this could be a flamebait. Bring em on.) > > -- > Mathieu Nantel > TecLinux.com > mnantel@teclinux.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
OK, I'll add my 2 cents. Firefly is the best soft client that I have tried. I bought X-Lite,a dn tried a handful of OSS alternatives. Firefly is the best in my experience. I will buy at least one of their hard phones when it ships on the basis of my Firefly experience. Michael Graves Sr Product Specialist Pixel Power Inc mgraves@pixelpower.com> -------- Original Message -------- > Subject: RE: [Asterisk-Users] Asterisk VS. Skype > From: "Matt" <matt@powderdays.com> > Date: Thu, April 29, 2004 4:46 pm > To: asterisk-users@lists.digium.com > > >>B) Asterisk won't be adding latency on LAN->LAN, it's your end clients > - > try Firefly - www.virbiage.com (insert bias comment here) and see if > >>that's addresses your issues. > In my (limited) experience I've found softclients always produced poor > results (choppy lagged voice); CISCO-ATA and CISCO7960 however > performed > perfectly and usually better than PSTN call quality. I spent two > hours > chatting to my friend in the US (I live in the UK) and he was stunned > at the > end of the call when I told him it was a voip->pstn call via > voicepulse. > > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Adam Hart > Sent: 30 April 2004 00:23 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Asterisk VS. Skype > > A) Skype uses iLBC as a codec which Asterisk already has > B) Asterisk won't be adding latency on LAN->LAN, it's your end clients > - try > Firefly - www.virbiage.com (insert bias comment here) and see if > that's > addresses your issues. > > > Mathieu Nantel wrote: > > This might have been talked about before, but I'm posting anyhow. > > > > I've got down to testing Asterisk yesterday, and I couldn't help but > > > compare it with Skype (a Windoze only product, yet, but extremely > > efficient for some reason). > > > > Skype has almost unperceptible delay (LAN), while there is almost > half > > a second of delay regardless of the codec on Asterisk. > > > > An even if we were to eliminate the latency on asterisk, Skype has at > > > least 2x the quality of PSTN, let alone any codec on asterisk. > > > > So I ask: are there no royalty-free codecs that come close to those > > from GIPS? (I understand these people are supplying the codecs for > Skype). > > > > Or is there something I'm doing wrong? What configuration yields the > > > highest quality and the lowest bitrate? I'm talking LAN > communication > > here: plenty of bandwidth, no latency. > > > > (I'm aware that this could be a flamebait. Bring em on.) > > > > -- > > Mathieu Nantel > > TecLinux.com > > mnantel@teclinux.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 > > _______________________________________________ > 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 is a PBX solution, Skype is just a softphone. How can you compare the two? Shouldn't you be comparing XLite or something similar? I'm perplexed. Mathieu Nantel> This might have been talked about before, but I'm posting anyhow. > > I've got down to testing Asterisk yesterday, and I couldn't help but > compare it with Skype (a Windoze only product, yet, but extremely > efficient for some reason). > > Skype has almost unperceptible delay (LAN), while there is almost half a > second of delay regardless of the codec on Asterisk. > > An even if we were to eliminate the latency on asterisk, Skype has at > least 2x the quality of PSTN, let alone any codec on asterisk. > > So I ask: are there no royalty-free codecs that come close to those from > GIPS? (I understand these people are supplying the codecs for Skype). > > Or is there something I'm doing wrong? What configuration yields the > highest quality and the lowest bitrate? I'm talking LAN communication > here: plenty of bandwidth, no latency. > > (I'm aware that this could be a flamebait. Bring em on.) > > -- > Mathieu Nantel > TecLinux.com > mnantel@teclinux.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 >Ronald R. McDaniel Southern Computer Services, Inc. rmcdaniel@southerncomp.net (251) 444-3136 office (251) 446-3137 fax (251) 294-1202 cell
You know what: I just figured out my problem. I was using Firefly on one end and IAX Phone at the other end. And guess what, IAX Phone doesn't support G711, yielding to mediocre quality and latency for what I suppose was translating G711 <-> GSM> >>B) Asterisk won't be adding latency on LAN->LAN, it's your end clients - > try Firefly - www.virbiage.com (insert bias comment here) and see if > >>that's addresses your issues. > In my (limited) experience I've found softclients always produced poor > results (choppy lagged voice); CISCO-ATA and CISCO7960 however performed > perfectly and usually better than PSTN call quality. I spent two hours > chatting to my friend in the US (I live in the UK) and he was stunnedat the> end of the call when I told him it was a voip->pstn call via voicepulse. > > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Adam Hart > Sent: 30 April 2004 00:23 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Asterisk VS. Skype > > A) Skype uses iLBC as a codec which Asterisk already has > B) Asterisk won't be adding latency on LAN->LAN, it's your end clients- try> Firefly - www.virbiage.com (insert bias comment here) and see if that's > addresses your issues. > > > Mathieu Nantel wrote: > > This might have been talked about before, but I'm posting anyhow. > > > > I've got down to testing Asterisk yesterday, and I couldn't help but > > compare it with Skype (a Windoze only product, yet, but extremely > > efficient for some reason). > > > > Skype has almost unperceptible delay (LAN), while there is almost half > > a second of delay regardless of the codec on Asterisk. > > > > An even if we were to eliminate the latency on asterisk, Skype has at > > least 2x the quality of PSTN, let alone any codec on asterisk. > > > > So I ask: are there no royalty-free codecs that come close to those > > from GIPS? (I understand these people are supplying the codecs forSkype).> > > > Or is there something I'm doing wrong? What configuration yields the > > highest quality and the lowest bitrate? I'm talking LAN communication > > here: plenty of bandwidth, no latency. > > > > (I'm aware that this could be a flamebait. Bring em on.) > > > > -- > > Mathieu Nantel > > TecLinux.com > > mnantel@teclinux.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 > > _______________________________________________ > 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 > >-- Mathieu Nantel, RHCE TecLinux.com mnantel@teclinux.com
http://gigaom.com/thevoipdaily/archives/2004/03/why_skype_is_no_different.html NEXT!!! bkw ----- Original Message ----- From: "Mathieu Nantel" <mnantel@teclinux.com> To: <asterisk-users@lists.digium.com> Sent: Thursday, April 29, 2004 5:09 PM Subject: [Asterisk-Users] Asterisk VS. Skype> This might have been talked about before, but I'm posting anyhow. > > I've got down to testing Asterisk yesterday, and I couldn't help but > compare it with Skype (a Windoze only product, yet, but extremely > efficient for some reason). > > Skype has almost unperceptible delay (LAN), while there is almost half a > second of delay regardless of the codec on Asterisk. > > An even if we were to eliminate the latency on asterisk, Skype has at > least 2x the quality of PSTN, let alone any codec on asterisk. > > So I ask: are there no royalty-free codecs that come close to those from > GIPS? (I understand these people are supplying the codecs for Skype). > > Or is there something I'm doing wrong? What configuration yields the > highest quality and the lowest bitrate? I'm talking LAN communication > here: plenty of bandwidth, no latency. > > (I'm aware that this could be a flamebait. Bring em on.) > > -- > Mathieu Nantel > TecLinux.com > mnantel@teclinux.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 >
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Thursday 29 April 2004 07:09 pm, Mathieu Nantel wrote:> This might have been talked about before, but I'm posting anyhow. > > I've got down to testing Asterisk yesterday, and I couldn't help but > compare it with Skype (a Windoze only product, yet, but extremely > efficient for some reason). > > Skype has almost unperceptible delay (LAN), while there is almost half a > second of delay regardless of the codec on Asterisk.Hmm, when I talk coast to coast I've never noticed any lag over *. Now if you are really talking about 500 ms it's a very short lag, less than a sattellite link. I have * in each end and it's so dead quiet I'm even thinking of adding white noise as a now-you-are-connected background noise.> An even if we were to eliminate the latency on asterisk, Skype has at > least 2x the quality of PSTN, let alone any codec on asterisk. > > So I ask: are there no royalty-free codecs that come close to those from > GIPS? (I understand these people are supplying the codecs for Skype). > > Or is there something I'm doing wrong? What configuration yields the > highest quality and the lowest bitrate? I'm talking LAN communication > here: plenty of bandwidth, no latency. > > (I'm aware that this could be a flamebait. Bring em on.) > > -- > Mathieu Nantel > TecLinux.com > mnantel@teclinux.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- -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin Franklin -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFAkcUsljK16xgETzkRAv49AJ9B9ziccjP22tSqICvHgpBn6UFUswCgsWW7 bINAV/Y5Y8U7C44W+zSj2B8=2kw0 -----END PGP SIGNATURE-----
Mathieu Nantel <mnantel@teclinux.com> wrote: This might have been talked about before, but I'm posting anyhow. I've got down to testing Asterisk yesterday, and I couldn't help but compare it with Skype (a Windoze only product, yet, but extremely efficient for some reason). ---Maybe you need more testing :) Do you know what most people use Asterisk for or what Asterisk is? I'm surprised that you can even come up with the comparisons. Skype has almost unperceptible delay (LAN), while there is almost half a second of delay regardless of the codec on Asterisk. ---Are you sure you don't work for Skype? You might be the only one who experience this :) An even if we were to eliminate the latency on asterisk, Skype has at least 2x the quality of PSTN, let alone any codec on asterisk. So I ask: are there no royalty-free codecs that come close to those from GIPS? (I understand these people are supplying the codecs for Skype). ---You need to do more research and answer this easy question yourself. Or is there something I'm doing wrong? What configuration yields the highest quality and the lowest bitrate? I'm talking LAN communication here: plenty of bandwidth, no latency. (I'm aware that this could be a flamebait. Bring em on.) ---You're ridiculous. If you wanna compare something with Skype, compare it with SIP in general. -- Mathieu Nantel TecLinux.com mnantel@teclinux.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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040430/b8eb9357/attachment.htm
Sorry development version crashes on startup for me too. Dean ________________________________ From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Ron McMillin Sent: Saturday, 1 May 2004 1:55 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Asterisk VS. Skype I think the released version doesn't support SIP and that's why it crashes. The development version works just fine on XP and certainly Firefly isn't annoying like xlite. gARetH baBB <hick.asterisk@gink.org> wrote: On Thu, 29 Apr 2004 mgraves@mstvp.com wrote:> OK, I'll add my 2 cents. Firefly is the best soft client that I have > tried. I bought X-Lite,a dn tried a handful of OSS alternatives.Firefly> is the best in my experience. I will buy at least one of their hardIt's the worst I've tried - it just crashes and burns everytime when I try to use SIP with it, and the UI is very poor. It doesn't even come up as a proper application which you can Alt-TAB to. Though Xlite can be annoying, it does a better job than Firefly - if only IAX2 was added to Xlite ... _______________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040430/261d39c6/attachment.htm