Adobe Air and Adobe FMS are good examples of VoIP working flawlessly over TCP. We are actually developing a flash phone which needs only TCP to transmit both signal and audio.
-Bruce On Sat, Apr 24, 2010 at 2:01 PM, Zeeshan Zakaria <zisha...@gmail.com> wrote: > RTP stands for Real-Time Transport Protocol. TCP is not designed to deal > with real-time data transfer as it takes time to acknowledge packets and > re-send them if missing. All audio video data transfer happens in real time, > and it doesn't make any sense to retransmit missing packets. Real time > packets mixed with old missing packets which are submitted would result in > an non-understandable audio and video. So how come any system can use TCP > for real time data transfer, while assuring the quality at the same time. > Does their exist any such system? I would certainly like to try it, maybe > they are doing it right using some trick which I don't know yet. > > Zeeshan A Zakaria > > -- > Sent from my Android phone with K-9 Mail. > > On 2010-04-24 1:48 PM, "David Backeberg" <dbackeb...@gmail.com> wrote: > > On Fri, Apr 23, 2010 at 3:21 PM, <ad...@3a.hu> wrote: > > i have to put an * between two other SIP ga... > > Don't do it. > > There have been any number of posts to asterisk-users begging asterisk > to bend over backwards to accommodate Microsoft's foray into the world > of VoIP. Nobody seems to be asking Microsoft to build a stack > compatible with the rest of the world of VoIP. > > I disagree that sending SIP over TCP is superior to sending it over > UDP. Think about it for a second. UDP is 'unreliable' in that lost > packets are not rebroadcast. > > Now let's say you have an 'unreliable' connection where it's just > barely good enough that the SIP call setup goes through, but the RTP > stream immediately fails. > > Why would that be superior to having the SIP call setup getting > dropped? The end result of no reliable voice is the same, but now you > have a funkier debug condition that's going to be more complex to > track down. I personally think it would be superior to see the bad > connection as early in call setup as possible. > > And of course, SIP over UDP is the way the rest of the world works. If > anybody wants to speak up about a framework that supports BOTH SIP > over UDP AND SIP over TCP, maybe somebody already built a middleware > layer that will let Microsoft join the world of voip. > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocati... > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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