Hi Michael,
What I can see from the ATSC M/H Transport spec is that there are
requirements on encoder and decoder clock precisions, which for sure
give a complete different environment than assumed by RFC3550 with
respect to the clock precision. If we could assure such clocks for all
devices which may potentially encode and stream layered media via multi
session transmission, the problem would have been solved anyway. But
that is not the case, as I tried to show by pointing to some of the
relevant discussions
on the mailing list.
About your second point: RTP does not require NTP to be used as
wallclock. In RTP, there is much more freedom in selecting the source
for this clock. RTP uses
the NTP timestamp format just for indicating the wallclock in the RTCP
sender reports. Furthermore draft-perkins-avt-rapid-rtp-sync-03 does
only require to use the same wallclock as used for the SRs also for the
wallclock timestamps in the
header extensions.
Another interesting point is that the ATSC Transport spec is not
mentioning SVC while mentioning AVC and AAC. If looking at the ATSC
Video Systems spec. (S4-136r7-A153-Part-7-Video-System),
there is a solution
which requires RTP timestamp synchronization for "SVC Transport in Two
RTP Sessions". This for sure solves the clock skew problem, but as
discussed in AVT, this is not a general solution
and has different implications if applied to all use cases of RTP multi
session transmission as, e.g. conversational services. Therefore we now
have a solution which seems to be agreeable by the group, which solves
the
problem and is more in line with RFC 3550, i.e. using the wallclock for
the
data alignment between the session.
Best regards,
Thomas
Michael A Dolan wrote:
Hi Thomas-
Thanks. Yes, I am familiar with the discussion.
I'm not necessarily pushing the ATSC solution, but offering some points
to consider, which apply equally to the problem at hand here, I
think.
Also, it is important to note some more things about the ATSC
text:
1. The language in the specification very carefully recommends the
clock
precision without requiring it. It is expected to work over routers
with jitter and delays and with media components from outside the
broadcast facility. The difference is that the client is expected to be
able to perform tight sync when the servers are able to provide the
recommended timing model (encoding one hop away, etc). It
gracefully degrades to "Internet" timing model
quality.
2. The clock, although "NTP" syntax, is not time of day (e.g.
reference_ID= "GPS"),
but "PPS", thus allowing the components to be encoded and
decoded without concern for end-to-end delivery delay. This only
works when NTP is explicitly signaled as part of the encoding and both
components share a clock (same or closely coupled encoders). So it will
not work in the general case. But there are many systems where it
will and perhaps consideration should be given to permitting
non-"wall clock" timebases.
The NTP hdr ext here is an encoding optimization that indeed has some
other good properties. Although the draft ties the NTP timestamp
back to the RTCP SR records, the timebase is not explicitly defined
(which is also true of RTCP SR). Some reference_ID value must be
assumed and
there is enough delta between, for example, GPS and UTC, timebases to
cause things to break if the entire system is not implicitly assumed to
be only one timebase. For example, without an explicit reference_ID,
how
could one use other timebases defined in NTPv4? This is a broader
issue than the hdr ext syntax, but since we're discussing NTP, I
thought
I'd bring it up.
I think the above and the other points below are relevant here and
not inconsistent with the RTCP SR methods. The fact that the ATSC
design
*can* be more precise is not alone cause to discard the techniques
developed there.
Regards,
Mike
At 09:34 AM 3/31/2009 +0200, Thomas.Schierl at hhi.fraunhofer.de wrote:
Dear Michael,
Thanks for the comments and the reference to the ATSC M/H.
I guess, what we are looking for in AVT is a solution which also works
for encoders having non-perfect clocks. There was a long discussion on
the list, here are two pointers:
http://www.ietf.org/mail-archive/web/avt/current/msg09144.html
http://www.ietf.org/mail-archive/web/avt/current/msg10491.html
ATSC surely can assume to have a perfect clock in the NTP streams,
which
will not be the case on PC systems that may be used as an encoder for
multi session transmission of a layered codec. This is one major
problem
we intended to solve with the header extension while keeping the whole
approach as close as possible in line with the existing RTCP SR
method.
Best regards,
Thomas
______________________________________________
Thomas Schierl, Dipl.-Ing.
Fraunhofer HHI
Senior Research Engineer
Image Processing Dept. / Multimedia Transport
Einsteinufer 37, 10587 Berlin, Germany
PHONE: +49 30 31002 227 , FAX: +49 30 392 7200
WEB: http://ip.hhi.de
,
SKYPE: thomas.schierl
EMAIL: thomas.schierl at hhi.fraunhofer.de
Michael A Dolan wrote:
Those interested in this topic
may also find this interesting:
http://www.atsc.org/standards/cs_documents/a153-2009-03-13/S4-132r13-A153-Part-3-Transport.pdf
Section 10.3.
Some differences are:
1. There is one NTP packet stream per service, rather than the encoding
optimization in the hdr ext;
2. RTCP SR records are emitted "near" video random access
points, rather than on a prescribed time schedule; and
3. The above works in multicast-only, UDLR networks (e.g. television
broadcast), rather than requiring a 2-way link.
#2 is interesting since in order to begin a *synchronized* decode
(video
+ audio properly correlated), a client must have acquired all of:
a. NTP
b. RTCP SR for each component
c. video reference frame and decoding metadata (i.e. a random access
point)
d. audio frame
So, the relative timing of the RTCP SR records to the RTP stream
content
is perhaps more important than the emission period. Having the RTCP
SR records arrive faster than the video reference frames doesn't help
of
course. And even if they are "fast enough", having them
arrive out of sync with the reference frames delays the service
acquisition by half the period on average.
Bandwidth utilization (how many bits to devote to RTCP and NTP) can
arguably be more efficiently managed by the server in most cases, not
individual client requests for faster RTCP SR packets. And, the client
would have no knowledge of the intended (video) reference frame periods
being used by the server, so it would not, in general, know what to ask
for.
In practice, won't a server that wants to enable faster acquisition and
sync do things like the above? I'm concerned that detailed requests
from clients beyond "please use more bandwidth and enable fast
sync" won't be as valuable as they seem since in the general case
the client doesn't know what to ask for exactly.
FYI, the above ATSC document is in a "candidate" state where
public comments are solicited. Independent of this thread, if
anyone has comments on it, I could convey them back to the ATSC.
Regards,
Mike
At 07:43 PM 3/9/2009 +0000, Colin Perkins wrote:
The changes in this version are
primarily editorial; we're aware that
not all the open issues have been addressed yet. Any further
comments
are welcomed.
Chairs: we'd like this to be considered as an AVT working group
draft, as indicated privately.
Cheers,
Colin
Begin forwarded message:
From:
Internet-Drafts at ietf.org
Date: 9 March 2009 17:15:02 GMT
To: i-d-announce at ietf.org
Subject: I-D Action:draft-perkins-avt-rapid-rtp-sync-03.txt
Reply-To: internet-drafts at ietf.org
A New Internet-Draft is available from the on-line Internet-Drafts
directories.
Title : Rapid
Synchronisation of RTP Flows
Author(s) : C. Perkins, T.
Schierl
Filename :
draft-perkins-avt-rapid-rtp-sync-03.txt
Pages :
17
Date :
2009-03-09
This memo outlines how RTP multimedia sessions are synchronised, and
discusses how rapidly such synchronisation can occur. We show
that
most RTP sessions can be synchronised immediately, but that the use
of video switching multipoint conference units (MCUs) or large
source
specific multicast (SSM) groups can greatly increase the initial
synchronisation delay. This increase in delay can be unacceptable
to
some applications that use layered and/or multi-description
codecs.
This memo updates the RTP Control Protocol (RTCP) timing rules to
reduce the initial synchronisation delay for SSM sessions. A
new
feedback packet is defined for use with the Extended RTP Profile for
RTCP-based Feedback (RTP/AVPF), allowing video switching MCUs to
rapidly request resynchronisation. Two new RTP header extensions
are
defined to allow rapid synchronisation of late joiners, and
guarantee
correct timestamp based decoding order recovery for layered codecs
in
the presence of clock skew.
A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-perkins-avt-rapid-rtp-
sync-03.txt
Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/
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