Re: [Asterisk-Users] PSGW 2.2 Skype gateway?
2005/11/17, Michael Graves [EMAIL PROTECTED]: Call quality is ok, but it seems to add considerable latency. I suspect that the call is fully decoded back to analogue (or maybe not quite that far) on one of the audio devices in the OS, then encoded into SIP for the outbound leg. That would imply additional delay in all cases. It uses the skype api, then is the api (the skype propietary client) who decodes the sound (adding some latency). Then the sip part may be including some extra latency. It sould use a low latency codec, compression is not needed in a local machine... But there is a big problem with this program: it needs windows to run, adding failure point to the circuit... and then needing of an extra machine only for acting as gateway: bad solution. I think we will need to wait until someone hacked the skype protocol... If there is someone interested on doing it. -- Alejandro Vargas ___ --Bandwidth and Colocation sponsored by Easynews.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] PSGW 2.2 Skype gateway?
On Fri, November 18, 2005 11:14, Alejandro Vargas said: 2005/11/17, Michael Graves [EMAIL PROTECTED]: Call quality is ok, but it seems to add considerable latency. I suspect that the call is fully decoded back to analogue (or maybe not quite that far) on one of the audio devices in the OS, then encoded into SIP for the outbound leg. That would imply additional delay in all cases. It uses the skype api, then is the api (the skype propietary client) who decodes the sound (adding some latency). Then the sip part may be including some extra latency. It sould use a low latency codec, compression is not needed in a local machine... But there is a big problem with this program: it needs windows to run, adding failure point to the circuit... and then needing of an extra machine only for acting as gateway: bad solution. I think we will need to wait until someone hacked the skype protocol... If there is someone interested on doing it. Which is going to be a pain, as it is encrypted... :-( Reverse-engineering may be the best option, and that is: 1) Not trivial 2) Not always legal /me sighs... I'll try getting my friends on to VoipBuster instead! ;-) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.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] PSGW 2.2 Skype gateway?
2005/11/18, Francesco Peeters [EMAIL PROTECTED]: I think we will need to wait until someone hacked the skype protocol... If there is someone interested on doing it. Which is going to be a pain, as it is encrypted... :-( Reverse-engineering may be the best option, and that is: 1) Not trivial 2) Not always legal /me sighs... I'll try getting my friends on to VoipBuster instead! ;-) ¡Right!. Thiis is the explanation of why there is not an easy way to link skype with asterisk. But there is people still thinking on doing it, because switchng from skype to voipbuster is as sifficult as switching from micro$oft-messenger to jabber: there is no reason for not doing it, but people doesn't. On other way, I must accept that skype codec has a very good compression. -- Alejandro Vargas ___ --Bandwidth and Colocation sponsored by Easynews.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] PSGW 2.2 Skype gateway?
On Fri, 2005-11-18 at 12:56 +0100, Alejandro Vargas wrote: [snip] On other way, I must accept that skype codec has a very good compression. Iirc they use iLBC Wideband which is 16KHz and does not work with Asterisk which uses 8KHz. I'm not an expert though so I might have misunderstood. Regards, Patrick ___ --Bandwidth and Colocation sponsored by Easynews.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] PSGW 2.2 Skype gateway?
On Fri, 18 Nov 2005 13:10:23 +0100, Patrick wrote: On Fri, 2005-11-18 at 12:56 +0100, Alejandro Vargas wrote: [snip] On other way, I must accept that skype codec has a very good compression. Iirc they use iLBC Wideband which is 16KHz and does not work with Asterisk which uses 8KHz. I'm not an expert though so I might have misunderstood. Regards, Patrick Yesterday was very interesting with respect to PSGW. Several co-workers who are Skype users in the UK called me. I'm in Texas. Those who called from our corp offices are behind a MS Proxy Server (ISA) and using MS proxy clients. These calls suffered latency issues that were bad. Not quote useless, but generally unacceptable. Later on one of them called me via Skype from his home, with only a firewall and no proxy server. That call was MUCH better. This leads me to beleive that perhaps that PSGW software is not the entire problem.but it's surely part of it. It's clearly a less than ideal solution. I suspect that it's better than the VTA-1000, which would break the Skype call out to a FXS, requiring me to bridge into * via an FXO. I hate FXOs. I've gone to considerable lengths to test FXO devices (TDM400, X101, SPA-3000, etc) and found none viable long term solutions. I now call forward my remaining POTS lines to DID provided by an ITSP. That's been much better than fighting with FXO interfaces for one or two lines. 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 fwd 54245 ___ --Bandwidth and Colocation sponsored by Easynews.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] PSGW 2.2 Skype gateway?
Has anyone else on list tried this yet? I bought the personal edition and would like to compare experience with others. The vendor is www.rsdevs.com. In general the application does what it promises. It receives Skype calls through an actual Skype client, then uses a virtual audio patch cord to forward that call to a SIP device. Call quality is ok, but it seems to add considerable latency. I suspect that the call is fully decoded back to analogue (or maybe not quite that far) on one of the audio devices in the OS, then encoded into SIP for the outbound leg. That would imply additional delay in all cases. Michael Graves -- 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 sponsored by Easynews.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
