[asterisk-users] RTP over TCP
Zeeshan Zakaria
zishanov at gmail.com
Sat Apr 24 13:01:05 CDT 2010
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.
--
_____________________________________________________________________
-- Bandwidth and Colocati...
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100424/c2365b2c/attachment.htm
More information about the asterisk-users
mailing list