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[AVT] AD questions on draft-ietf-avt-ulp
I had several questions and concerns about this draft. I cc'd
Jonathan as he was going to do an expert review. Likewise, Mark, as
an review of this, do you still have concerns that are not addressed
that you think need to be fixed for this to go to RFC?
First of all, i raised a bunch of questions here - Do feel free to
push back hard where I am confused, wrong etc. I have not been
following this work, I am just reading it now, and have a somewhat
limited amount of time to read all the background and really come up
to speed on the whole topic. You will need to help educate me and I
would be pleased to find out the document is fine and I am confused.
Major Items ------------
Is this backwards compatible with a device that does 2733. If not I
have concerns about obsoleting it?
Can we have a way of signaling baseline and extended mode support? I
would not want to have all old 2733 devices be required to support
extended mode. Can we separate baseline and extended mode into two
specifications.
I am unclear how a receiver wold know what order to apply SRTP/FEC
processing. I am unconvinced there is a need for multiple ways of
doing this. The security section needs discussion of problems with
two time pads when doing this. It also will need security review with
respect to two time pads issues from some security person.
Is specifying the mask and m(i) and L(k) in every packet the right
design choice. This uses a lot of bandwidth and seems like
information that would not need to be specified every time.
The document needs a sane and rational applicability statement. There
are definitely deployment trade offs where this approach would not be
the best way to go. Need to explain how and when this should be
used. Given we are forming a WG to deal with FEC, this document
needs to also address when to use generic approaches vs these.
I am unclear on what level of reconstruction is provided or what we
are tying to do with the multiple levels. For example, take the
example in figure 5. Here 8 payloads are sent and 7 payloads worth of
parity bits are sent so we are sending nearly as much FEC data as
original data and introducing significant decode latency. Yet if
payload 0 and 1 are lost, it does not seem like they can be
reconstructed. I hope I am wrong here - it seems like with this
amount of bits allocated to FEC we should be apply to loose any 3
packets and still recover all the data. Please help me understand.
Can both 0 and 1 be recovered?
It is unclear to me how the receiver will know if it has enough
buffer to decode the FEC stream that will be sent to it. It needs to
know this when it decided if it wants the FEC stream or not.
The basic operation section that tries to give the overview of how
the scheme works is very confusing for people not already familiar
with the scheme. Let me be very blunt - I gave it to three different
developers all familiar with RTP and none of them could figure out
how this works or what it does. Part of the issue is that the term
level is used before it ever introduced.
Minor Items ---------------
Is a m(i) of 48 bits enough for things like mpeg?
How does this work to FEC something where the next packet may not
happen for a very long time like DTMF or text.
Given the packet overhead, would this be the recommended way to
protecting say 20 ms G.729 audio packets in an interactive voice
conversation.
Nice to point out some example audio codecs that benefit from ULP.
When doing FEC on an encrypted packet, the encryption is very likely
to make ULP useless - should carefully consider and discuss.
Would there ever be a case to have both integrity protection and ULP?
Seems weird.
I think that is the media is encrypted, then the FEC MUST be
encrypted too ( not should ) and similarly with integrity.
In section 11, the discussion of changing media encryption keys and
how that relates to decoding the FEC packets, seems, ah, wrong.
Perhaps it is right but more is needed on this.
Section 12, 2nd para. This tells implementers they MUST do something.
However, I have no idea what they need to implement here. We need to
give advice on what they implementers need to do - as it is, it does
not seem implementable.
The advice on using FEC and redundant encoding or separate stream is
somewhat unsettling. First it say SHOULD do it as separate stream and
provided no exception when you would not then goes on for a few
section on telling you how to violate this SHOULD. Would be nice to
have advice on when an implementation uses each one (or just define
one way). I will note that from a bandwidth tracking for RSVP and
flow set up for ICE point of view, redundant encoding has advantages.
Trivial Stuff ------------------
Section 16 mentions Hamming codes? Are they discussed elsewhere in
the document. (Note I think Hamming codes are a good idea and
suggesting making it clear how to use them not suggesting removing them)
Header of draft need to specify the RFCs it obsoletes.
Section 5 - the sentence ending in "the channel capacity is not fully
utilized" make no sense to me. I don't think we mean the same things
by channel capacity.
Section 7.3 - I don't think the timestamp setting is correct. I think
you want the timestamp of packet the protected packet with the
largest timestamp. Often you want to delay sending FEC packets after
the final packet they protect so that burst channel errors have
better chance of not killing the last packet and the FEC packet.
The term "monotonically increasing" has contention meanings - I
would not use this term and instead just say what you mean. (ie
every FEC packet has a TS that is the same or larger than previous
packets). I also don't think this will turn out to be true for FEC
protected video.
When using 48 bit mask where is the low bit?
IANA registration - I don't think the change controller should be a
WG as they get closed.
This document has a lot of advertising in it - for example it
suggests ULP is useful place where Reed-Solomon is too
computationally complex. Every codec type that I can think of where
ULP will be useful requires complex transforms such as frequency
space for audio or DCT approximation to frequency space for video
that make the Reed-Solomon computation complexity look like a drop in
the bucket. Are there codec that can use ULP that are used on devices
where the extra complexity of Reed-Solomon would make a significant
impact on the device?
I understand the desire to keep the title the same as the document
this replaces but this is far from a generic scheme and I think a
misleading title is worse that a changed title.
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