Re: [Asterisk-Users] http://www.skype.com/
Alastair Maw wrote: On 05/11/03 10:14, Olle E. Johansson wrote: As I understand it they must not be fully peer-to-peer even if they use your bandwidth, there has to be media servers in their network, handling calls. Or? No, the whole point is that it's completely decentralized. More interesting to end users is that the calls are encrypted and can traverse NAT. The way Skype can bounce between peers effectively enables it to provide a few different routes for the traffic, from which it picks the least latency one. Add a nice UI, and it's not surprising that it's gathering speed rapidly. So all peers exchange traffic constantly over UDP to keep NAT bindings open? And a central server to set it all up... Hmmm. Interesting. If I have a network connection with low latency and use SKype, the risk is that my network bogs down with the automatic routing of calls to my connection... Maybe we should develop IAX3 with automatic p2p routing/latency handling? :-) /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.skype.com/
On Wed, 05 Nov 2003 10:36:01 +, Alastair Maw wrote: >On 05/11/03 10:14, Olle E. Johansson wrote: > >> As I understand it they must not be fully peer-to-peer even if they >> use your bandwidth, there has to be media servers in their network, >> handling calls. Or? > >No, the whole point is that it's completely decentralized. More >interesting to end users is that the calls are encrypted and can >traverse NAT. The way Skype can bounce between peers effectively enables >it to provide a few different routes for the traffic, from which it >picks the least latency one. Add a nice UI, and it's not surprising that >it's gathering speed rapidly. what the question is how without some for of centralisation can they have BOTH ends behind NAT ?? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.skype.com/
Philipp von Klitzing wrote: http://www.skype.com/ seesm to be the latest craze... anyone have any knowledge of their technolgy use etc ?? - closed source - WinXP and 2k only - peer-2-peer, i.e. they route foreign calls through your client (and bandwidth) if that helps the calling parties In one of our Swedish daily newspapers, like the national "Financial Times", one of the owners said that they're going to sell a commercial version with PSTN connectivity early next year. As I understand it they must not be fully peer-to-peer even if they use your bandwidth, there has to be media servers in their network, handling calls. Or? /O My limited understanding: If you have a public IP address (non-NAT) then you will see more traffic going through your session than most, since NAT'ed hosts need a "relay" on the outside of their NATs. Skype uses the Global IP Sound codecs, which are tremendously efficient. Voice quality is reportedly excellent, even under the extreme examples of multi-application use dialup connections. Skype encrypts all sessions at the management and media layers, which is a feature that I _love_ and wish Asterisk would develop more robustly. Skype is indeed proprietary, and is a for-profit company, so don't expect a chan_skype to happen soon unless they decide that they want to play nice with others (doubtful.) Skype will certainly be introducing PSTN connectivity, but I am very interested in what their numbering plan will look like for inbound calls, if such a plan is contemplated at all. These guys have to make money, so look for any new features costing $$$ - don't get too hooked yet (Anyone remember the problems .mp3 and .gif formats? Hell?) Skype has the ease of use and features to which we, as the rest of the VoIP community, should aspire. Extremely easy setup, excellent call quality, robust and distributed routing, and secure transmissions. They are certainly lacking many of the features that makes something "good", such as compliance with standards, but as a private company they can ignore those issues because they're not doing this for the betterment of anyone but themselves. If we can implement Skype-like features in our software but still develop in the open source, standards-compliant world, then that is a noble goal. Skype will certainly lead the way in showing us what features the customers want, and their system will push us towards making "real" VoIP networks of a much larger and robust (P2P) scale, but ultimately I think they'll fail due to their closed source methods. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.skype.com/
On 05/11/03 10:14, Olle E. Johansson wrote: As I understand it they must not be fully peer-to-peer even if they use your bandwidth, there has to be media servers in their network, handling calls. Or? No, the whole point is that it's completely decentralized. More interesting to end users is that the calls are encrypted and can traverse NAT. The way Skype can bounce between peers effectively enables it to provide a few different routes for the traffic, from which it picks the least latency one. Add a nice UI, and it's not surprising that it's gathering speed rapidly. -- Alastair Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.skype.com/
Philipp von Klitzing wrote: http://www.skype.com/ seesm to be the latest craze... anyone have any knowledge of their technolgy use etc ?? - closed source - WinXP and 2k only - peer-2-peer, i.e. they route foreign calls through your client (and bandwidth) if that helps the calling parties In one of our Swedish daily newspapers, like the national "Financial Times", one of the owners said that they're going to sell a commercial version with PSTN connectivity early next year. As I understand it they must not be fully peer-to-peer even if they use your bandwidth, there has to be media servers in their network, handling calls. Or? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.skype.com/
Hi! > http://www.skype.com/ > seesm to be the latest craze... anyone have any knowledge of their > technolgy use etc ?? - closed source - WinXP and 2k only - peer-2-peer, i.e. they route foreign calls through your client (and bandwidth) if that helps the calling parties Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] http://www.skype.com/
http://www.skype.com/ seesm to be the latest craze... anyone have any knowledge of their technolgy use etc ?? Gary ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
