Re: [Asterisk-Users] Skype vs. an Xlite registered to Asterisk
2006/2/26, hugolivude <[EMAIL PROTECTED]>: > I have a bunch of road warriors who I've set up with Xlite clients. > Unfortunately the sound quality has been intermittent at best. What codec dis you use?? I think xlite support speex, that is the better codec I've tested when connections are under hevy traffic (p2p applications). G729 is good too, but Speex really worked great in my tests. > Sometimes > I was thinking about trying an Xlite client that can support G729. Anyone > had experience with that? Does it significantly improve voice quality? What you need to improve (or decress) is the bandwidth usage. Check this: http://www.voip-info.org/wiki/view/Bandwidth+consumption but try speex if it is supported by your sip phone. It is free, variable bitrate and adapts to the available bandwidth, (it is based on ogg codec). -- Alejandro Vargas ___ --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] Skype vs. an Xlite registered to Asterisk
On Feb 26, 2006, at 5:43 PM, hugolivude wrote: Say, thanks to all you for your time in responding. I hope I don't sound unappreciative (I have no time for flamers) but I don't understand how changing from SIP to IAX would make any difference. I don't have any problems with the signalling (i.e. phones ring when I make and receive calls), the problem is with the media. Aren't the signalling (IAX/SIP) and media (presumably a function of the codec) two seperate and distinct domains? Maybe I have it all wrong, so I'd be keen to learn what I'm missing. I think you have it all wrong ;~) SIP uses multiple ports to communicate from client to host. The negotiation usually occurs on port 5060, but the audio is elsewhere. IAX2 uses a single port for everything, hence all NAT/firewall routing issues are simplified. I also prefer IAX2, and just to chime in, there is an excellent new Softphone for Mac OSX called JackenIAX (dumb name) that seems to be superb although it's still in beta. Marty ___ --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] Skype vs. an Xlite registered to Asterisk
Say, thanks to all you for your time in responding. I hope I don't sound unappreciative (I have no time for flamers) but I don't understand how changing from SIP to IAX would make any difference. I don't have any problems with the signalling (i.e. phones ring when I make and receive calls), the problem is with the media. Aren't the signalling (IAX/SIP) and media (presumably a function of the codec) two seperate and distinct domains? Maybe I have it all wrong, so I'd be keen to learn what I'm missing. Thanks again, Hugh On 2/26/06, Guillermo Salas M <[EMAIL PROTECTED]> wrote: On Sun, 2006-02-26 at 15:50 -0600, Michael Graves wrote:> I second this...as a road warrior myself I find too many places where> SIP clients just won't work. So I rely on Firely over IAX2 which has> been 100% reliable. >I'm using idefisk softphone with iLBC on my Debian roadwarrior laptopand works very nice.> Also, John Todd has been using the PSGW Skype<>SIP gateway software in> new and different ways. Perhaps that's an option. >> On Sun, 26 Feb 2006 18:30:07 +0100, asterisk wrote:>> >Hi> >I have good results in using, the old very (free) of firefly (IAX2),> >with g729!> >> >rgds > >Jesper Langpap> >> >hugolivude wrote:> >>> >> I have a bunch of road warriors who I've set up with Xlite clients.> >> Unfortunately the sound quality has been intermittent at best. > >> Sometimes it's great other times completely unusable. When it's bad> >> one usually hears harsh static when the other party speaks or their> >> voice gets "clipped" to static if they speak too loudly. > >>> >> Many of these users have migrated to Skype – much to my chagrin! I'd> >> like to get them back using a SIP client so they can take advantage of> >> all Asterisk can offer. > >>> >> Anyone else had trouble with voice quality with Xlite? Any work arounds?> >>> >> I was thinking about trying an Xlite client that can support G729.> >> Anyone had experience with that? Does it significantly improve voice > >> quality?> >>> >> I also read that SJ Phone is better than XLite, but is it really the> >> client application that makes the biggest difference or the codec?> >> Perhaps it's a combination or something entirely different? Anyone > >> with experience with an SJ Phone and G729 codec?> >>> >> Any suggestions welcome!> >>> >> Yours,> >> Hugh> >>> >> P.S> Asterisk 1.2 on Redhat 9.0> >>> >> > >>> >> ___ > >> --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> >>> >> >___ > >--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> >> >>> --> Michael Graves [EMAIL PROTECTED]> Sr. Product Specialist www.pixelpower.com> Pixel Power Inc. [EMAIL PROTECTED]>> o713-861-4005> o800-905-6412> c713-201-1262> fwd 54245 ___ > --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--Guillermo Salas M.Telconet S.A. MantaCalle 15 y Av. 24 Esq.Phone : 593 5 262 8071Mobile: 593 9 985 5138SIP : [EMAIL PROTECTED]e-mail: [EMAIL PROTECTED]www : http://www.telconet.net http://www.telcocarrier.netLinux User: 255902Soporte en Linea en http://www.manta.telconet.netPlease avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Skype vs. an Xlite registered to Asterisk
On Sun, 2006-02-26 at 15:50 -0600, Michael Graves wrote: > I second this...as a road warrior myself I find too many places where > SIP clients just won't work. So I rely on Firely over IAX2 which has > been 100% reliable. > I'm using idefisk softphone with iLBC on my Debian roadwarrior laptop and works very nice. > Also, John Todd has been using the PSGW Skype<>SIP gateway software in > new and different ways. Perhaps that's an option. > > On Sun, 26 Feb 2006 18:30:07 +0100, asterisk wrote: > > >Hi > >I have good results in using, the old very (free) of firefly (IAX2), > >with g729! > > > >rgds > >Jesper Langpap > > > >hugolivude wrote: > >> > >> I have a bunch of road warriors who I've set up with Xlite clients. > >> Unfortunately the sound quality has been intermittent at best. > >> Sometimes it's great other times completely unusable. When it's bad > >> one usually hears harsh static when the other party speaks or their > >> voice gets "clipped" to static if they speak too loudly. > >> > >> Many of these users have migrated to Skype – much to my chagrin! I'd > >> like to get them back using a SIP client so they can take advantage of > >> all Asterisk can offer. > >> > >> Anyone else had trouble with voice quality with Xlite? Any work arounds? > >> > >> I was thinking about trying an Xlite client that can support G729. > >> Anyone had experience with that? Does it significantly improve voice > >> quality? > >> > >> I also read that SJ Phone is better than XLite, but is it really the > >> client application that makes the biggest difference or the codec? > >> Perhaps it's a combination or something entirely different? Anyone > >> with experience with an SJ Phone and G729 codec? > >> > >> Any suggestions welcome! > >> > >> Yours, > >> Hugh > >> > >> P.S> Asterisk 1.2 on Redhat 9.0 > >> > >> > >> > >> ___ > >> --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 > >> > > > >___ > >--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 > > > > > > -- > Michael Graves [EMAIL PROTECTED] > Sr. Product Specialist www.pixelpower.com > Pixel Power Inc. [EMAIL PROTECTED] > > o713-861-4005 > o800-905-6412 > c713-201-1262 > fwd 54245 > > > > ___ > --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 -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] Skype vs. an Xlite registered to Asterisk
I second this...as a road warrior myself I find too many places where SIP clients just won't work. So I rely on Firely over IAX2 which has been 100% reliable. Also, John Todd has been using the PSGW Skype<>SIP gateway software in new and different ways. Perhaps that's an option. On Sun, 26 Feb 2006 18:30:07 +0100, asterisk wrote: >Hi >I have good results in using, the old very (free) of firefly (IAX2), >with g729! > >rgds >Jesper Langpap > >hugolivude wrote: >> >> I have a bunch of road warriors who I've set up with Xlite clients. >> Unfortunately the sound quality has been intermittent at best. >> Sometimes it's great other times completely unusable. When it's bad >> one usually hears harsh static when the other party speaks or their >> voice gets "clipped" to static if they speak too loudly. >> >> Many of these users have migrated to Skype much to my chagrin! I'd >> like to get them back using a SIP client so they can take advantage of >> all Asterisk can offer. >> >> Anyone else had trouble with voice quality with Xlite? Any work arounds? >> >> I was thinking about trying an Xlite client that can support G729. >> Anyone had experience with that? Does it significantly improve voice >> quality? >> >> I also read that SJ Phone is better than XLite, but is it really the >> client application that makes the biggest difference or the codec? >> Perhaps it's a combination or something entirely different? Anyone >> with experience with an SJ Phone and G729 codec? >> >> Any suggestions welcome! >> >> Yours, >> Hugh >> >> P.S> Asterisk 1.2 on Redhat 9.0 >> >> >> >> ___ >> --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 >> > >___ >--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 > > -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ --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] Skype vs. an Xlite registered to Asterisk
Hi I have good results in using, the old very (free) of firefly (IAX2), with g729! rgds Jesper Langpap hugolivude wrote: I have a bunch of road warriors who I've set up with Xlite clients. Unfortunately the sound quality has been intermittent at best. Sometimes it's great other times completely unusable. When it's bad one usually hears harsh static when the other party speaks or their voice gets "clipped" to static if they speak too loudly. Many of these users have migrated to Skype – much to my chagrin! I'd like to get them back using a SIP client so they can take advantage of all Asterisk can offer. Anyone else had trouble with voice quality with Xlite? Any work arounds? I was thinking about trying an Xlite client that can support G729. Anyone had experience with that? Does it significantly improve voice quality? I also read that SJ Phone is better than XLite, but is it really the client application that makes the biggest difference or the codec? Perhaps it's a combination or something entirely different? Anyone with experience with an SJ Phone and G729 codec? Any suggestions welcome! Yours, Hugh P.S> Asterisk 1.2 on Redhat 9.0 ___ --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 ___ --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
