[asterisk-users] RTP over TCP

[bruce bruce]

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 <zishanov at 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" <dbackeberg at gmail.com> wrote:
>
> On Fri, Apr 23, 2010 at 3:21 PM, <adamk at 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.
>
>
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