Re: Streaming Video across the IT
- From: Kevin McMurtrie <mcmurtri@xxxxxxxxxxxxxx>
- Date: Fri, 23 Jun 2006 20:41:06 -0700
In article <1151106290.798951.229480@xxxxxxxxxxxxxxxxxxxxxxxxxxxx>,
erniemac@xxxxxxxxxxxxx wrote:
First off: satellite? Whachoo talkin' about, Willis? Typical DSL and
cable internet connections don't bring satellites into the picture.
Neither do the high-capacity trunks, which rely on fiber optic
connections. Satellite delay isn't really relevant in most cases (if
you have a situation where it is, please elaborate).
Second, verification? Maybe you should read on on UDP. For audio and
video transport you don't need verification, just stream the packets and
do the best you can. See <http://en.wikipedia.org/wiki/Udp> for a
little more information on UDP.
And even if you did use TCP (which most players can, to get around e.g.
firewall issues) you just buffer a few seconds to make sure that any
packets that need to be transmitted have had a chance to be before
they're actually needed.
They're teaching network protocols in high school physics classes now?
Interesting.
--
Jerry Kindall, Seattle, WA <http://www.jerrykindall.com/>
Send only plain text messages under 32K to the Reply-To address.
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BUFFERING is not REAL TIME streaming ! All Mac QT videos use BUFFERING.
Missing a few frames here and there ? Sure, not impotant for lots of
uses, but for CRITICAL uses such as MEDICAL videos ?
Don't they teach velocity of light in High School Physics anymore? I
mentioned NOTHING about network protocols ! Irrespectibe of method of
communication, it still takes TIME to transmit packets over fiber or
sattilite or microwave. ALL packets don't necessarily go via the same
route, so receiving computer needs to sort out packets to right order.
Please read my posts properly before posting !
Ernie Lee
You never mentioned latency so why are you yelling? Latency will ALWAYS
be there, it's just a matter of how much. TCP/IP latency is a few times
the round trip time to be stable. UDP has sub-second and stable latency
but will lose packets. Error correction can compensate for lost packets
at the cost of increased bandwidth and increased latency.
Modern real-time video protocols use UDP, some error correction, data
spreading, and a loss tolerant video codec. Data spreading (required
for efficient error correction and loss tolerance) increases latency but
it's not as bad as TCP/IP retransmits. Loss tolerance in the codec
(re-synch markers) increases bandwidth but not as much as extra error
correction data would.
It's all trade-offs. If you think cellphones are real-time, try calling
yourself on another phone. There's 1/4 to 1/2 second lag.
.
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