►
From YouTube: IETF-AVTCORE-20221215-1600
Description
AVTCORE meeting session at IETF
2022/12/15 1600
https://datatracker.ietf.org/meeting//proceedings/
B
D
B
E
F
F
I
can
help
when
I'm
not
presenting
and
I'm
actually
planning
on
talking
a
lot
during
the
slot
after
mine
as
well
so
and
I
can
help
the
wife.
C
B
Yeah,
so
basically,
just
as
usual,
you
know,
if
you
have
any
IPR,
you
have
to,
you,
have
to
disclose
it
and
we
are
under
or
you
have
to
disclose
it
subject
to
the
policies.
It's
you
know
the
player
for
the
details
and
we
were
under
various
code
of
conduct.
So
please
behave
appropriately.
I
think
that's
more
cover
on
the
next
slide.
B
Indeed,
yes,
so
we
are
under
the
code
of
conduct,
so
please
behave
professionally
and
you
know
we,
you
know,
debate
ideas,
not
people.
If
you
feel
like
there,
you
don't
have
any
problems.
Please
do
the
part
of
your
chairs
or
the
ombuds
Team.
B
Okay,
I've
mentioned
our
note.
Takers
are
yeah
Center
and
no
yes,
indeed,
okay.
Thank
you.
Spencer,
no.
C
So
here's
the
agenda
for
today
we
have
more
to
go
on
the
chair,
slides
covering
the
state
of
CFA
stuff
like
that,
and
then
we
have
a
presentation
on
green
metadata,
some
things
relating
to
RTP
over
quick
and
then
un
is
going
to
talk
about
codec,
agnostic,
payload
format
and
then
we'll
have
wrap-ups
and
next
steps.
D
B
Yep,
okay,
yeah
we've
had
some
things
published.
We
have
two
things
in
auth
48,
so
that's
very
cool.
Those
have,
let's
have
a
drive,
they're
are
a
few
numbers
are
assigned,
so
those
should
be
Republicans
should
be
popping
out
quite
soon.
B
Skip
is
an
ietf
last
call
so
we'll
see
what
the
comments
on
that
are
marketing
mode.
We
do
need
a
revised
ID
for
that.
So.
B
Please,
let's
see
I
see
up
here
to
the
slides
at
the
very
last
minute.
Yes,
1783
best
has
Pub
wrecked
about
an
hour
ago.
B
B
Okay,
cool,
thank
you,
I
thought
I
was
gonna
have
to
say
this
is
and
then
let's
see
yeah
so
then
hard
to
be
a
provocative
adopted.
Evc
is
adopted,
but
we
need
to
revise.
We
need
an
updated
draft.
Stefan
says
he
juggling
two
calls
so.
D
B
I
think
the
idea
there
was
to
basically
do
the
same
thing
as
VVC,
so
yeah.
So
that's
and
then
we
need
to
have
v3c.
We
publish
as
a
working
group
document
I,
don't
know
if
any
of
the
authors
of
that
are
here
today.
C
Lori
you're
here,
oh.
G
C
Go
fish
the
please
receive
email
out
of
our
inboxes
yeah.
B
I
think
Laurie
published
the
link
to
the
things
you
need
to
click
on.
So
let
me
put
that.
C
D
B
There
were
some
comments,
people
being
in
favor
of
it
and
one
thing
I
can't
so
I
think
we
have
that
on
the
agenda.
So
we
can
talk
to
that
one
we
get
to
that.
F
C
C
Yeah,
it
was
on
the
the
new
draft
for
b3c
approve
it
as
a
working
group
draft.
H
C
Okay,
Magnus
isn't
here
yeah.
I
B
C
Greetings
so
we
should
we
can
we
skip
some
of
this
and
then
I
guess
cover
it
during
your
portion.
C
All
right
so
I
guess
the
I
guess
the
question
will
be:
have
the
comments
been
addressed
and
are
we
ready
for
adoption
I?
Guess
that's
what
we're
trying
to
answer
in
in
Young's
presentation:
okay,
go.
I
Ahead,
Young
all
right,
thank
you.
So
this
is
a
presentation
for
the
rtcp
messages
for
green
metadata.
Next
slides,
please.
So
this
just
quick
summary.
The
message
we
proposed
so
the
first
is
the
request.
Message
include
the
frame
rate
picture
with
picture
height
and
then
the
second
message
is
the
notification
to
response
the
framing
rate,
picture,
width
and
height
in
the
notification
message
next
slides.
Please.
I
Yeah,
we
received
a
number
of
comments
during
the
CFA
and
the
first
comment
is
from
Laurel
from
Nokia.
He
comments
that
it
is
needed
to
clarify
how
adapting
the
best
stream
of
rtcp
mechanism
is
used
in
collaborate
with
sdp
updates.
We
think
is
this
a
good
comment
and
we
plan
to
add
sdp
definition
subclass
a
similar,
the
one
as
in
the
RFC
510,
to
define
the
rtcp
feedback
attribute
and
the
relevant
parameters
for
the
proposed
message.
I
So
the
second
command
that
we
receive
from
Magnus
is
about
the
title.
He
feels
the
current
title
may
not
be
clear,
because
the
green
metadata
is
about
energy
saving,
but
the
the
message
propose.
The
message
does
not
provide
any
energy
information,
so
he
suggested
to
to
change
it.
So
we
we
think
about
like
two
titles
for
the
group
to
consider
the
first.
We
may
change
it
to
like
rtcp
messages
for
spatial
and
temporal
resolutions.
I
The
second
candidate
title
could
be
codec
resolution
control
message
in
RTP,
audio
visual
profile
with
feedback
so
yeah.
We
think
the
the
proposed
message
can
be
used
beyond
the
green
metadata,
so
we
are
open
to
to
change
the
title
but
we'd
like
to
to
hear
from
the
group
which
title
may
be
more
appropriate.
I
So
I
may
stop
here
and
see
any
comments.
B
I
feel
like
it's
probably
best
not
to
do
too
much
of
this
bashing
on
the
on
the
on.
You
know
in
real
time,
go
ahead.
Stephanie.
B
Anyway,
obviously,
I'm
not
sure
doing
the
you
know,
you
know
the
details
of
you
know:
fashion.
I
B
Yeah
I
guess
I
mean
I
guess
the
question
is,
are
you
know
I
think?
That's
that's
probably
well
known
in
the
from
the
the
itu
session
ISO
Community,
which,
where
that
work
is
being
done,
I'm
not
sure
how
well
it's
known
in
the
ITF
Community,
so
yeah
I
mean
I.
Think
I
mean
I
feel
like
trying
to
do
the
details
of
bashing.
The
I
think
this
would
be
better
for
the
mailing
list
than
the
then
live
in
the
conference.
B
C
I
mean
yeah:
please
indicate
that
in
the
notes,
Spencer
and
Mel
yeah.
I
All
right,
so,
let's,
let's
move
on
to
next
comment.
So
next
comment
is
that
currently
the
notification
message
just
repeats
the
picture
with
height
and
frame
rate
and
the
comments
that
may
be
wasteful
or
maybe
only
send
the
ssrc
and
the
secret
number
is
enough.
So
in
that
case
we
can
save
132
bits
per
FCI
entry,
but
that's
that's
the
only
only
one
can
disable
because
we
still
have
to
keep
sending
the
ssrc
and
the
sequence
number
I
checked
the
the
current
or
the
existing
notification.
I
They
just
send
all
the
the
complete
field
so
that
the
benefit
to
keep
all
three
or
other
fields
is
to
be
consistent
with
the
current
design.
But
we
are
open
if
people
think
32-bit
saving
is
quite
significant
in
this
case.
B
Yeah
I
feel
like
it's
my
as
an
individual.
My
inclination
would
be
to
say
better
to
be
descriptive
if
it's
not
that
expensive,
and
this
also
introduces
the
possibility
of
like
you
know,
you
could
send
unsoluted
unsolicited
tsdn
if
you're
just
doing
something
unilaterally,
which
I
think
happens
for
some
of
the
other
messages,
but
I'm
not
sure.
If
that's
a
good
idea
but
open.
I
J
I
Okay
thanks.
The
last
question
is
just
the
the
the
question
that
in
case
the
media
center
will
change
the
resolution
or
frame
rate
or
the
spatial
resolution
due
to
the
congestion
Zen.
It
probably
not.
We
need
to
send
any
notification.
I
B
Yeah,
this
is
analogous
to
how
Timber
works.
Like
you
know,
Timber
is
a
Max
bit
rate,
but
this
is
that's.
There
could
be
Clips,
this
is
setting
a
maximum
and
it
can
go
underneath
it
for
be
clear
that
you
know
that
this
is
setting
a
cap
and
not
necessarily
you
know,
things
can
go
underneath
it.
For
other
reasons,
right.
D
I
All
right
next
slice
piece
yeah,
so
so
we
asked
for
the
the
picture
adoption.
So
that
concludes
my
presentation.
Thanks.
B
Yeah
I
mean
as
I
would
say:
yes,
I
think
you
know
the
the
issues
that
were
raised
are
all
things
can
be
done.
You
know
addressed
successfully
during
you
know
the
I.
You
know
the
working
group
development
of
the
draft
and
I
think
that
this
is
probably
fine
to
be
adopted
up
with
the
possible.
We
should,
you
know,
get
possibly
get
consensus
on
the
title.
Just
because
that
probably
the
that
the
documents.
B
Before
we
so
before,
we
got
to
decide
what
the
draft
iatf
ABT
core
documents
going
to
be
named.
So
let's
have
a
little
discussion
of
that
on
the
list
and
then.
C
Options
so
the
conclusion
for
the
minutes
is
to
debate
the
title
on
the
list
and
then
once
the
title
is
settled,
to
submit
a
working
group
draft
with
that
with
that
title
reflected
in
the
name.
K
C
Okay,
this
is
the
next
topic
and
to
be
clear,
this
is
related
to
rtb
over
quick
and
it's
a
subject
that
came
up
both
in
some
experiments,
I've
been
doing
and
also
came
up
on
the
muck
list.
C
So
on
the
mock
list
there
was
a
posting
from
Christian.
Wiedema
who's
also
been
implementing
media
over
quick,
and
he
discovered
something
which
I
think
I've
also
seen,
which
is
that
it's
possible,
in
some
circumstances,
for
quick
to
run
out
of
them
to
hit
What's
called
the
max
streams
limit,
and
it
depends
depends
on
the
use
case
and
also
if
the
application
is
doing,
perhaps
not
cleaning
up
after
itself
correctly
leaving
streams
open.
C
So
the
question
is:
is
this
a
potential
issue
with
a
quick
or
a
web
transport
implementation?
There
do
seem
to
be
some
use
cases
where
a
default,
Max
streams
limit
could
be
exceeded.
We'll
talk
about
that
in
a
moment
like
conferencing,
I
have
filed
an
issue
on
the
web
transport
API
because
at
least
currently
there's
no
way
for
a
client
to
set
a
Max
streams
limit
that
might
be
higher
than
the
default.
C
So
there's
a
potential
API
issue
there
and
but
I've
done
a
few
experiments
and
if
you're
just
sending
a
single
RTP
stream
and
you're
cleaning
up
after
yourself
closing
streams
after
they're
they're
used,
then
you
probably
won't
hit
this
limit
unless,
for
some
reason
your
implementation
has
a
very
small
but
I've
encountered
situations
where,
if,
if
you
don't
clean
up
after
yourself,
you
can
leak
resources,
leave
streams
open
for
some
reason,
then
you
will
hit
it
and
there's
all
kinds
of
bad
things
that
happen
like
the
classical
Legos
goes
very,
very
high.
C
So
we'll
we'll
talk
about
a
little
bit
more,
whether
it's
something
we
actually
need
to
address
in
a
document
or
just
be
aware
of
for
implementers.
So
a
little
bit
about
the
frame
per
stream
model.
You
know
I
think
we
know
how
it
works,
which
is
in
the
send
pipeline
to
Sender,
opens
the
unit
directional
stream.
C
Perhaps
if
they
want
partial
relatability
they'll
set
a
timer
and
if
the
timer
fires
they
reset
the
Stream,
otherwise
they'll
send
they'll,
write
the
RTP
header
in
the
payload
to
the
unidirectional
stream
and
then
close
the
Stream
and
then
receive
side.
C
Receiver
is
notified
of
an
incoming
uni-directional
stream
and
they
read
from
it
until
the
stream
is
closed
or
it's
reset,
and
sometimes
the
lane
field
is,
is
useful
to
know
if
it
was
completely
received
now,
so
I
did
a
little
experiment
and
with
frame
for
stream
transport,
it
turns
out
how
this
performs
and
whether
you
hit
the
limit
depends
a
lot
on
what
what
I
would
call
concurrency.
C
As
you
know,
iframes
are
a
lot
larger
than
your
average
P
frame,
sometimes
as
much
as
10
times
the
size,
and
what
concurrency
means
is
that
on
the
quick
stream
you're
sending
the
iframe
at
the
same
time
as
potentially
multiple
keyframes,
it's
at
least
in
JavaScript.
You
have
to
be
aware
of
blocking
if
you're
doing
that,
and
in
particular
JavaScript
awaits
will
block.
So
you
have
to
get
rid
of
that
stuff.
If
you
want
to
make
sure
that
you
can
do
concurrent
sending
of
iframes
and
B
frames.
So
it's
a
good
sign.
C
J
C
So
I've
been
going
through
my
example:
implementation
just
trying
to
eliminate
all
the
awaits,
and
after
doing
that,
you
do
see
a
lot
of
real.
You
do
see
reordering
about
three
to
six
events
per
experiment,
mostly
in
the
initial
iframe
and
B
frame
and
a
few
other
nice
effects,
because
the
iframe
is
closer
to
transmission
line
still
made.
C
C
So
here's
an
example
of
what
you
see
after
you've
optimized
it
a
bit.
This
is
a
run
with
av1
at
full
HD
resolution
about
418
kilobits
average
bit
rate
30
frames,
a
second
gap
size
about
3
000
frames
between
keyframes.
So
almost
all
P
frames,
with
maybe
one
p
frame
in
an
experiment
with
three
temporal
layers.
So
you
can
see
the
largest
iframe
here
is
about
12
590
octets.
The
P
frame
is
much
smaller,
a
little
bit
bigger
than
the
average
packet
at
15
23.
C
and
with
concurrency.
The
iframe
is
much
closer
to
the
transmission
line.
So
essentially
you
have
an
FC
win
to
send
the
iframe
in
a
single
sea
wit.
D
C
So,
overall,
the
question
is
this:
is
this
a
problem
and
for
a
single
rtb
stream?
The
answer
is
almost
certainly
no
I
haven't
what
in
when
I
was
doing.
This
I
only
saw
a
maximum
of
five
concurrent
open
Quick
streams
for
a
single
ssrc.
Assuming
you
don't
have
a
lot
of
loss,
I
guess
if
you
have
loss,
you
might
end
up
with
more,
but
I
would
note
that
I
did
see.
C
I
did
hit
the
max
stream
limit
once,
and
that
was
because
there
were
issues,
closing
streams
and
releasing
all
resources.
So
if
you
have
leaks,
then
you
could
conceivably
hit
the
limit,
but
if
you're
doing
a
good
job,
probably
not
now,
as
I
mentioned,
if
you
have
lost
it,
the
limit
could
be
higher
if
you're
doing
priority.
C
Typically,
what
people
will
do
is
they'll
say,
send
the
iframe
and
don't
do
concurrency
with
the
p
frames
and
so
you'll
you'll
end
up
with
lower
concurrent
streams,
but
on
the
other
hand,
your
glass
delay
will
be
higher,
so
that
might
not
be
the
best
idea.
C
C
So
in
that
kind
of
a
circumstance
you
know,
if
you
had
say
five
quick,
simultaneous
quick
streams
and
then
50.
You
know
video
streams,
you
could
end
up
with
250
or
500
or
something
Max
streams.
Well,.
E
I
thought
the
max
streams
limit
was
just
the
numeric
number
of
stream
IDs,
the
maximum
stream
ID
numeric
value
that
it
can
be
not
the
number
of
actual
concurrent
open
streams
right
now,
so
I
didn't
think
there
was
any
accounting
that
that
closed
streams
would
would
somehow
alter
the
max
streams
limit.
I
thought
just
the
fixed
limit
that
that
number,
a
stream
ID
can't
go
above
that
number.
This
suggests
that
the
the
max
streams
is
the
number
of
active,
open.
Concurrent
streams.
Is
that.
E
C
E
E
C
Okay,
so
anyway,
there's
also
an
interaction
with
priority
in
that
priority
can
be
used
to
limit
the
number
of
concurrent
streams,
and,
but
you
know,
people
talk
about
waiting
for
the
iPhone
to
be
finished
before
sending
any
P
frames.
C
That's
actually
a
little
bit
problematic
because
it
reintroduces
head
of
line
blocking
it's
a
little
bit
better
is
to
allow
kind
of
a
finite
number
of
P
frames
to
be
sent
along
with
your
iframe,
but
she
probably
shouldn't
be
infinite,
and
you
know
in
terms
of
what
implementations
do,
at
least
in
chromium
and
portions
bandwidth
equally
between
the
open
streams
by
default.
That's
another
good
reason
not
to
leave
streams
open
to
close
them,
because
otherwise,
you'll
kind
of
slow
everybody
down
and
the
issue
of
partial
reliability
is
independent,
a
priority.
C
So
if
stuff
is
taking
too
long,
you
can
always
you
can
always
send
a
reset
stream.
You
can
always
call
a
board
or
something
and
get
rid
of
it
anyway.
I
think
I
think
we
have
a
good
action
item.
Just
is
to
look
at
this
a
little
bit
further.
It
strikes
me
it's
not.
It
is
probably
something
maybe
worth
a
note
in
the
RTP
over
quick
spec
I.
Don't
know
that
it
deserves
any
more
than
that,
but
just
something
to
be
aware
of.
C
F
Okay,
I
think
this
is
I,
think
this
I
think
this
is
me
methods.
Do
you
think
this
is
me
yeah.
F
I
wanted
to
say
too,
okay,
cool
and
I'm
just
finishing
up
one
sentence
in
the
notes:
real
quick.
F
F
Or
or
okay
never
mind
so
we
talked
a
little
bit
and
very
rapidly
and
at
the
end
of
the
session,
about
whether
we
actually
had
a
an
agreed
scope
for
RTP
over
quick
during
our
session
in
London
and
and
York
and
I
agreed
that
it
was
a
good
thing
for
us
to
work
on
that
and
we
spent
some
time
having
a
session
working
session.
F
While
we
were
still
in
London-
and
we
wanted
to
talk
about
where
we
think
we
are
on
scoping
just
level
setting
with
the
working
group
and
making
sure
that
the
RTP
over
quick
and
the
corresponding
sdp
specification
are
going
to
do
what
the
working
group
thinks
they
is
expecting
them
to
do
so.
Early
discussions
on
RTP
over
quick
started
out
small.
This
is
going
to
be
a
drop-in
replacement
for
RTP
over
UDP.
That's
an
unrealistic
idea
for
a
variety
of
reasons
that
we
talked
about
in
previous
sessions.
F
We
want
to
do
the
right
thing
with
RTP
over
quick,
and
the
proposed
goal
is
to
do
enough
for
it
this
to
be
useful,
and
the
proposed
non-goal
is
don't
do
everything
that
we
can
imagine
so
now.
So
for
now,
let's
focus
on
what
we
know.
We
will
need
now
and
go
don't
do
more
than
what
we
need
know
that
we
need
now,
but
try
not
to
break
anything
that
we
might
need
later
I'm
going
to
be
asking.
F
Does
that
sound
about
right
so
far
on
each
slide,
and
this
is
a
slide,
so
I
wanted
to
give
people
a
chance
to
talk
about
that.
If
any.
If
anyone
has
something
to
say.
B
Yeah
I
think
that's
about
right,
I!
Think
it's
of
course.
You
know
more
specifically,
I
think
that
you
know
the
this
is
designed
to
be
something
comparatively
simple,
as
opposed
to
mock,
which
is
sort
of
doing
the
re-architect
the
world
to
you
know
for
exactly
what
we
want.
So
I
think
this
is
I'm.
F
Sorry,
yeah,
sorry
and
it
looks
like
it
looks
like
Sam-
is
going
to
correct
at
least
me.
Okay,.
H
Yeah,
fantastic
yeah,
some
lspp
r
d,
not
sure
if
this
is
going
to
be
a
correction,
more
understanding
I
might
be
preempting
what
you're
about
to
say
on
a
later
slide.
But
what
exactly?
Do
you
mean
by
enough
for
RTP
over
quick
to
be
useful.
F
There
is,
there
is
actually
a
hive
mind,
group,
think
statement.
So,
basically,
if
people
in
the
working
group
jump
up
and
down
and
say
you
didn't
do
too
much
or
did
you
didn't
do
enough
or
you
did
or
you
did
too
much
we'd
like
to
know
that
before
you
know
going
in
rather
than
coming
out,
so
the
the
basically
this
presentation
is
intended
to
be
a
significant
chunk
of
that
discussion,
although
not
all
of
it
and
we'll
talk
about
that
during
during
the
presentation.
H
Yeah
yeah
it
does
my
only
input
on
on
that
would
be
I.
Think
the
enough
for
it
to
be
useful
is
to
Define
some
way
of
carrying
RTP
frames
over
quick,
and
then
you
can
scope
that
on
multicast
is
problematic.
Maybe
don't
do
that
just
yet.
F
Yeah
yeah
yeah
dude,
you're
re
you're,
reading
the
issues
and
GitHub
good
job.
F
I'm
waiting
for
the
next
slide
to
pop
up.
So
in
the
1980s
there
was
a
discussion
on
the
history
and
history
of
programming.
Languages
about
some
languages
were
like
I
order,
a
pizza
with
the
toppings
that
I
want
things
like
APL
and
Pascal
that
were
designed
by
one
person
for
one
person's
use
and
then
adopted
that
sometimes
we
say
we
we
will
order
a
pizza
with
the
toppings
we
want.
This
was
the
COBOL
committee
and
people
like
that.
F
Where
you
get
everybody
in
a
room
and
say
it's
got
to
do
at
least
this,
and
so
some
people
order
a
pizza
with
all
the
toppings
that
anybody
could
ever
want,
and
that
was
what
the
share
committee
on
pl1
did
on.
Npl
and
po1
did
so
fast
forwarding
to
the
2020s
discussion
on
RTP
over
quick.
My
goal
is
that
we
order
a
quick,
RTP
pizza
with
only
the
toppings
that
we
need
next
slide.
F
Please
and
I
apologize
for
anybody,
who's
suddenly
hungry
in
the
middle
of
lunch,
so
the
planned
usage
was
Bernard.
Are
you?
Are
you
in
queue.
F
You
want
me
to
finish:
you're
finished,
okay,
so
I'm
finishing
this
slide,
yeah
go
ahead,
Okay
cool,
so
the
planned
usages
that
we
know
about,
and
we
probably
started
out
with
the
sip
VoIP
environments
that
are
currently
using
RTP
and
that
kind
of
led
us
into
the
land
of
a
drop
in
replacement
for
RTP
over
UDP
we've
been
talking
about
webrtc
environments
that
are
currently
using
RTP
and
but
we've
also
been
talking
about
web
transport
environments
carrying
real-time
media.
F
There's
medicine
has
more
about
details
about
this
on
in
the
next
section.
So
let's
defer
discussion
about
whether
that's
a
bad
idea
or
a
good
idea
for
now
to
to
measure
this
deck
slot
and
yeah,
but
I
think
I
want
to
I
wanted
to
make
sure
that
I.
We
were
not
missing,
planned
usages
that
somebody
knew
about
now.
F
C
Start
with
yeah
I'd
be
interested
if
if
there
are
folks
who've
dealt
with
pbxs
or
handsets,
but
I
haven't
really
seen
any
interest
in
this
from
that
Community
like
if
you're
thinking
that
rtb
ever
quick
would
be
used
in
a
SIP
trunk
to
a
carrier.
I
I
haven't
really
seen
interest
in
that
or
in
a
handset
or
in
a
PBX.
So
I
I
do
think
that's
curious
with
revitc.
C
You
know
there
is
some
history
appear
to
peer
quick
that
went
through
An
Origin
trial,
but
there's
no
official
w3c
activity
so
that
one
also
seems
kind
of
curious.
So
the
first
two
I'm
not
sure
where
people
got
the
idea
that
there
was
interest
there
but
I'm
not
seeing
it
at
least.
F
Excellent,
so,
are
you
pointing
us
towards
web
transport
as
the
planned
usage.
C
F
Yeah
yeah
you're
right
right,
right,
right,
I,
would
say.
I
was
saying
on
this
slide
so
far,
and
this
this
is
definitely
something
that
would
be
whoever
is
taking
notes.
F
It
would
be
awesome
to
capture
this
as
an
action
for
the
mailing
list
discussion,
but
just
basically,
especially
especially
the
webrtc
one,
but
just
we're
just
asking
that
I'm
seeing
people
in
like
3gppsa4
who
are
wanting
to
do
webrtc
for
media
and
in
the
you
know
in
the
same
the
same
technical
specification
committee
they
are,
they
are
doing
other
things
with
quick.
So
there's
not
that
much
of
a
leap
for
them.
You
know
for
me
to
imagine
them
being
interested.
C
Yeah
I
guess
there.
There
is
a
little
bit
of
a
difference.
Spencer
you
have
to
be
clear,
there's
webrtc
and
then
there's
the
web
right
right
and
those
could
be
different,
like
they
could
be
doing
webrtc
stuff
with
a
webrtc
API,
and
then
they
could
be
doing
something
else
with
the
web
which,
like
web
transport
or
something
so.
C
To
be
clear
anyway,
Peter
Peter
probably
has
more
comments
on
that
hi
Peter.
D
D
B
Oh
yeah
I
was
going
to
comment
I
feel,
like
you
know,
the
one
thing
about
support
environments.
There
was
and
forget
all
the
details,
but
several
ideas
back
now
we
hear
you
Peter,
but.
B
Okay,
that
there
was
I
know
there
was
a
thing
that
Jonathan
Rosenberg
brought
to
dispatch
several
idfs
back.
Talking
about.
You
know
the
idea
of
having
a
sip
architecture
that
integrated
better
into
the
web.
B
Model
which
I
think
that
was
you
know
there
was
some
interest
in
having
and
that
would
I
think
presumably
use
something
like
r2p
over
quick
as
it's
media
plane,
potentially
so
I
think
that
might
be
I
mean
I,
don't
think!
That's
not
anybody
who's
actively
working
on
it,
but
there's
some
interest
in
it
eventually
too
right,
wow.
F
F
Right
I
mean
yes,
that's
the
other
thing
it
was.
It
was
really
early
and
we
spent
a
lot
of
time
talking
about
how
much
of
the
web
infrastructure
was
HTTP.
1.1.
F
So
so
I
think
that
that's
that's
something
to
look
back
on
if
somebody
wanted
to
add
a
if
somebody
wanted
to
add
an
action
item
for
me
to
chase
that,
that
would
be.
That
would
be
lovely.
F
So
got
Jonathan
have
I,
have
I
done
you
yeah,
yes,.
B
F
Cool
Peter,
so
Peter.
L
Okay,
so
first
I
think
Bernard's
right
that
you
should
probably
just
speak
in
terms
of
web
environments
right
now,
someone
might
use
webrtc,
but
then
maybe
one
or
integrate
rtpa
for
quick
and
that,
as
far
as
I
know,
could
not
be
done
with
webrtc
unless
extensions
to
it
were
made.
But
the
only
extensions
I
know
of
are
kind
of
a
bring
your
own
transport,
and
in
that
case
it
would
be
a
web
transport.
L
So
you
might
have
a
situation
where
webrtc
is
emitting,
something
that
looks
like
packets
and
then
whoever
web
transport
actually
carries
them.
But,
inter
in
terms
of
this
slide
in
planned
usage
is
I
think
you
should
just
speak
in
terms
of
web
clients,
but
web
transport
would
have
to
be
a
part
of
that.
I
think
the
other
comment
I
have
is
that
I
think
you're
missing
use
case
here,
which
is
just
say,
mobile
applications
that
do
real-time
communication
using
RTP
and
they
might
want
to
use
quick
instead.
L
F
F
Excellent
and
can
can
we
and
we've
we've
added
that
as
in
the
minutes
of
something
for
us
to
chase
right,
it
looks
yes,
it
looks
like
someone
is
hopefully
typing
still
in
the
minutes
cool.
F
Thank
you.
Peter
Stefan.
F
You
may
have
just
cut
on
well
I'm,
not
sure
you
just
popped
up
and
anyway
Stefan.
We
are
not
hearing
you
yet.
F
F
Stefan
is
is
planning
violence
on
his
computer
in
in
in
zulup
and
Stefan
I
am
going
to
be
looking
at
the
at
the
at
the
chat
for
this.
If,
if
it's
helpful
for
you
to
put
a
comment
in
there,
I
will
I
will
definitely
see
that
as
well.
B
F
B
F
Yeah
Zoom,
so
for
topologies,
we
know
that
we
can't
support
all
the
7667
topologies,
because
section
3.3
of
that
specification
does
point
to
point
point
to
multi-point
using
multicast
and
quick
doesn't
have
multicast
yet
so
we
know
we
want
to
support
some
of
those
apologies,
but
we
haven't
discussed
at
which
the
polities
in
detail,
yet
the
dash
zero
one
version
of
the
RTP
over
quick
specification
describes
supported
RTP
Technologies
in
general
terms,
and
the
proposal
was
to
analyze
the
rest
of
the
unicast,
the
rest
of
the
unicast
apologies
and
I
added
issue.
F
47
for
this
in
in
GitHub.
Is
this
about
right?.
D
F
J
Okay,
so
good
morning,
oh
good
afternoon,
oh
good
morning,
the
rest
of
the
morning,
we
should
be
very
careful
with
the
apologies,
because
there
are
a
lot
of
apologies
there,
which
would
unnecessarily
limit
stuff
like,
for
instance,
like
we,
because
we're
going
to
end
up
with
SFU
of
a
link
or
TP
and
all
the
other
wonderful
things
which,
if
we
yeah
and
seriously
limit
the
functionality
and
implementation
options
which
we
have
right.
F
F
Okay
fold,
this
full
disclosure,
I
added
issue,
52
about
multicast
and
I'm,
not
presenting
it
now,
and
we
will
ignore
that
that
possible.
We
will
ignore
that
possibility
for
the
foreseeable
future.
Yes,.
J
Because
it
seriously
messed
up
RTP
and
can
seriously
limit
what
we
can
do
with
like
our
RTC
over
quick
one
thing
which
kept
coming
up
in
topology
with
RTP
as
an
issue
is
that
and
even
if
it's
a
peer-to-peer
topology
because
of
the
different
negotiation
glare,
you
have
multiple
streams
going
or
basically
multiple
RTP
streams
going
over
the
same
pre-negotiated
channel.
J
There
was
in
traditional
RTP,
it's
not
always
easy
to
the
max
them
like,
especially
like,
for
instance,
if
you
have
a
dtls
srtp
right.
D
J
A
problem
and
some
of
the
quick
parameters
characteristics
make
it
easier
because
you
can
just
start
new
stream
and
the
multiplex
by
that,
but
we
might
have
some
like
it's.
It's
probably
gonna
go
into
the
sdp
negotiation
like
we
need
to
make
sure
that
if
we
negotiated
this
over
sdp
that
we
take
have
a
way
of
the
marketing
multiple
streams
in
the
in
the
process
of
renegotiation
and
I,
don't
think
it's
actually
covered
in
the
topology
section
right
now.
So
that
might
be
something
to
be.
F
Cool
any
anybody
else
and
queue
on
topologies.
B
Sorry
that
didn't
give
us
a
few,
but
I
think
you
know
mention
them
all
I
suspect
some
of
them
will
be
mentioned.
For
you
know,
we
don't
think
this
is
important
for
this
use
case,
but
you
know
at
least
being
able
to
make
that
decision
explicitly
is
probably
a
good
idea.
F
Yeah
and
so.
F
Obviously,
you
know
obviously
and
I
I.
Well,
okay,
it's
obvious
to
me.
It
may
not
be
obvious
to
everybody,
because
I
could
be
wrong,
but
it's
obvious
to
me
that
we
may
end
up
with
a
certain
amount
of
text
on
this
and
that
certain
amount
of
text
on
this
may
be
longer
than
what's
in
the
in
the
current
draft,
and
that
could
be
that
could
appear
that
that
text
could
appear
in
the
draft.
It
could
appear
in
its
own
draft.
F
It
could
appear
in
which
one
or
which
might
update
67th
sorry
76
67,
or
it
could
end
up
in
its
own
draft.
That
stands
alone.
If
it
turns
out
to
be
different
enough
from
the
things
that
have
been
done
with
with
with
RTP
in
the
past,
so
I
think
the
I
think
the
right
answer
now
is
to
write
down
the
text
and
then
figure
out
what
to
do
with
it.
F
If
that,
if
that
sounds,
if
that
sounds
fair,
and
is
that,
does
that
sound,
fair,
Jonathan.
B
F
Excellent,
so
I
think
that
may
take
us
through
the
next
slide.
Relevant,
quick
extensions.
So
datagrams
is
a
quick
extension
that
has
been
part
of
the
RTP
over
quick
plan,
basically
forever
and
dash
zero.
One
of
the
RTP
over
quick
specification
assumes
that
the
use
of
datagrams
and
please
we
have
had
conversations
but
not
a
lot
of
conversations
about
the
possibility
of
using
quick
timestamps,
which
is
currently
a
individual
draft,
targeting
the
quick
working
group
to
allow
better
loss,
detection
and
congestion
control
in
some
in
some
situations.
F
So
those
those
are
the
two
extensions
that
I
think
that
I
think
the
avt
core
Community
has
spent
the
most
time
talking
about
at
least
at
least
where
I
could
hear
them.
Bernard
I
see
you
in
queue.
C
C
Can
we
make
this
mandatory
for
web
transport
for
for
the
reasons
that
were
discussed
in
ABT
core?
The
general
response?
Was
you
know
that
many
quick
implementations
don't
support
time
stamps
so
that
that
wasn't
recommended
as
an
approach?
I
don't
know
if
this
working
group
will
feel
differently
that
quick
timestamps
is
so
important
that
there
has
to
be
there,
but
in
general
at
least
at
the
moment,
it's
not
widely
implemented
by
any
means.
F
Yeah
well
I
mean
I,
know,
I,
know
that
people
Implement
individual
drafts,
but
I
mean
this
hasn't
been
adopted
by
the
working
group.
Yet
so
people
who
have
no,
who
have
not
implemented
it
yet
have
a
pretty
plausible
excuse
for
not
having
done
so,
but
yeah
I
I,
like
I,
say-
and
this
is
I'm
asking
these
questions
now
before
we
do
a
whole
lot
of
work
with
quick,
timestamps
and
then
say:
oh
never,
mind
yeah.
C
F
But
yeah
and
and
I
agree
and
I'm,
not
I'm,
not
trying
to
be
the
ABC
Corporal
Whisperer,
but
I
I
suspect
that
VZ
core
would
feel
better
about
pursuing
this.
If
it
is
moving
forward
and
the
quick
working
group
Peter
hi
hi.
L
I'm
interested
to
see
quick,
quick
timestamps
happen
so
that
congestion
control
algorithms
that
need
them
can
use
them.
Is
there
any
summary
of
a
status
of
how
well
this
particular
draft
is
going.
F
Spencer's
recollection,
without
clicking
on
the
data
tracker,
is
it's
been
through
something
like
nine
revisions?
If
I'm,
if
I'm
remembering
the
right
draft
but
I,
don't
you
know,
I
I
have
not
seen
a
call
for
adoption.
F
Does
anyone
does
anyone?
Is
anyone
smarter
about
this
than
me
and
I?
See
I,
see
York
popped
up
in
the
queue
as
I
was
asking
a
question.
I
don't
know
if
York
was
asking
was
had
a
thought
about
that
or
not.
F
Oh
okay,
I
am
seeing
in
I
am
seeing
a
zulup,
York
saying
limited
response
at
this
point
and
that's
like
I,
say
I
think
I
think
it's
just
too
early
in
the
in
the
discussions
about
quick
time
spans
in
detail
for
for
a
lot
of
people
to
have
really
focused
on
that.
F
So
this
this,
this
could
be
a
at
a,
we
will
add
a
issue
and
that
the
pro
the
appropriate
time
we
would
see
if
we
can
move
it
forward.
A
A
Doesn't
help
anyway,
so
yeah
Christian
was
wondering
whether
he
should
be
putting
more
Cycles
into
this
specification.
A
A
So
Christian
was
wondering
what
how
much
support
or
how
much
interest
there
would
be
into
solving
this
timestamp
problem.
There
was
limited
response
inside
the
quick
working
group,
and
so
no
it
is
no
enthusiasm
that
would
make
it
that
he
would
put
effort
into
taking
this
further
right
away.
A
F
A
F
And,
like
I
said
that
all
makes
perfect
sense
could
I
ask
the
chairs
if
it's
okay,
if
we
have
an
action
for
working
group
participants
who
are
interested
in
the
possibility
of
using
quick
timestamps
for
RTP
over
quick
to
focus
on
that
and
provide
some
feedback.
C
No,
that
that
sounds
good
yeah
you
mean
provide
feedback
to
the
quick
working
group
or
on
the
list.
F
Or
you
you
guys
tell
you
guys,
tell
me
I
mean
it.
It
could
be
that
it
could
be
that
it's
helpful
for
us
to
discuss
it
like
whether
whether
it
would
be
useful
and
if,
if
we
can't
convince
ourselves
that
it
would
be
useful
having
some
people
spend
Christian
up
before
we
decide
not
we
don't
care,
it
doesn't
seem
hope,
doesn't
seem
friendly.
B
I
mean
I
feel
like
it
depends
somewhat
on
what
exactly
the
congestion
control
coupling
model
is.
Where
you
know,
if
we're
expecting
to
get.
You
know
tight
contest
control
coupling
with
the
quick
stack,
then.
Yes,
if
it's
loose
coupling-
and
you
know
you
have
to
do
your
own-
you
know
higher
level
congestion
for
low
low
latency
at
user
level,
which
is
not
entirely
clear
to
me
yet
either
so.
C
C
F
Yeah
yeah,
you
and,
and
so
I
think
the
if
I'm,
if
I'm,
quoting
the
draft
correctly,
you
know
the
issues
are
either
it
is
asymmetric
or
at
that
the
congestion
on
the
ACT
path
is
significantly
higher.
But
but,
like
I
said,
don't
quote
me
on
that.
That's
what
I'm
remembering,
but
you
know
like
I,
say
for
us
for
us
to
just
get
more
experience
with
what
we're
doing
without
time,
stamps
and
then
figure
out
how
badly
we
need.
It
would
be
one
strategy
going
forward.
E
Yes,
I
just
wanted
to
agree
with
Peter
that
I'm
also
interested
in
this,
for
congestion,
control
improvements
and,
and
also
for
the
same
reasons
that
Bernard
is
interested
in
one
way,
delay
that
often
feeds
back
into
both
congestion
control
and
some
other
decisions
at
the
application.
Level.
So
I
would.
A
E
See
this
progress
for
for
the
notes,
I
wasn't
clear
what
the
action
was.
Is
it
an
action
for
the
chairs
to
ask
something
on
the
ABT
core
list
soliciting
feedback,
or
is
it
an
action
for
for
the
authors
for
for
Spencer
to
to
solicit
something
on
the
list.
F
Foreign
chairs,
could
you
could
you
Express
a
thought.
F
F
C
About,
if
you
send
it
Spencer
action
for.
F
F
Perfect,
and
and
thank
you
and
thank
you
for
thank
you
for
asking
for
that
clarification,
hi
Harold.
How
was
your
day.
M
It's
a
beautiful
day
and
or
night
at
the
moment,
but
they
all
said.
We
wanted
to
add
a
thought
about
quick
timestamps
that
timestamps
aren't
strictly
needed
for
congestion
control,
but
they're
awfully
useful
for
figuring
out
what
what
to
send
next.
M
Yes,
so
if
it,
if
we
get
around
to
detangling
this
question
of
of
protect
the
internet
against
those
evil
bits,
and
what
do
we
send
next
now
that
we
know
what
we
can
send
right,
I
think
yeah,
we're
going
to
find
that
timestamps
are
extremely
useful
and
defining
the
timestamps
is
sometimes
a
hard
job.
First
packet
last
packet,
assemble
packets,.
F
Right
I
think
I
think
that
I
think
that
I
think
it
makes
perfect
sense
to
me
that
people
who
have
been
doing
H3
may
not
care
because
you
know
because
they're
they
know
what
they're
going
to
send
next
roughly,
since
this
is
reliable.
F
But
it's
maybe
more
of
an
interesting
question
for
us
than
it
is
people
who've
been
doing
age
three
actually.
M
People
who
people
who,
who
are
still
the
people
who
invented
the
six
parallel
HTTP
gets
for
for
the
first
browsers,
are
probably
still
interested
in
what
they
can
do
and
do
in
parallel
and
yeah
next
sure.
F
Yeah
cool
that
gets
us
to
Jonathan.
B
Yeah
so
I
mean
I.
Think
part
of
the
question
here
also
is:
what
do
we
expect?
How
much
information
do
we
expect
the
RTP
player
is
getting
from
the
quick
stack
and
if
we
will
talk
about
useful,
quick
timestamps
are
I.
Think
presume
that
the
you
know,
whatever
the
application
that's
sending
RTP
actually
gets
to
know
them,
which.
D
B
Think
has
implications
for
what
information
is
getting
passed
and
forward
what
what
you
know
affect
API.
Do
we
assume
there
so
I
think
that's
also
something
that
we're
going
to
need
to
think
about
to
some
extent,
and
that's
also
probably
has
implications
also
for
the
web
transport
discussions,
because
that's
actually
a
explicit
API
being
defined.
F
Yeah,
exactly
okay,
cool
I,
think
that
cleared
the
queue
MBT
profiles
and
I
apologize
for
my
rant
about
this
in
London,
but
I
don't
rent
very
often,
we've
had
a
we've
had
proposals
that
have
ranged
for
from
avt
PDF
to
whether
you're,
adding
where
you're,
whether
you're
going
to
be
doing
secure,
RTP,
whether
you're
going
to
be
using
the
the.
F
Extended
rtpc
rtcp
feedback
and
things
like
that,
and
so
we've
had
some
proposals
for
full
protocol
definition
and
actually
I
just
I'm
leaving
no,
but
basically
basically.
F
Filling
out
the
definitions
for,
for
you
know
as
full
Pro
definitions
that
we
would
put
in
a
specification
and
that
that
included
the
point
Roman
made
a
bit
back
a
while
back
about.
F
If
you
are
doing
ice
and
you
get
all
your
ice
candidates
come
back
as
TCP
candidates,
whether
you
want
to
proceed
with
that
or
try
and
do
something
else,
because
if
you
can't
do
ice
for
you
for
UDP
for
quick
to
use,
you
probably
can't
do
eyes
for
UDP
for
RTP
to
use
and
then
steam
streams,
datagrams
or
both
as
attributes
to
limit
the
combinatorialness
of
the
of
the
registration
and
like
I,
said
this
is
I
I'm
not
typing
this
now,
but
but
that's
I,
think
that's
what
I'm.
I
F
Think
that's
where
I'm
headed
towards
the
next
time
I
do
a
revision
of
the
STP
specification.
Jonathan.
B
Yeah,
so
my
only
comment
on
the
SCP
at
spdf
on
you
know,
you
say
it's
for
ease
of
relaying
and
forwarding
to
udprtp,
but
the
problem
is
that
once
you
talk
about
asset
that
layer,
you
talk
about
what
that
layer
is
keying
looks
like
and
I,
don't
think
we're
going
to
be
able
to
Gateway
btls
srtp
to
Quick,
unless
you
want
a
tunnel
dtls
over
quick,
which
sounds
terrible
and
so
and
I.
B
Don't
think
you
want
to
use
security
descriptions
because
that's
generally
terrible,
so
I
feel
like
I
would
say
the
the
problem
with
the
asses.
Is
you
need
to
find
what
the
keying
looks
like,
which
might
mean
exporting
from
quick
but
I'm,
not
sure
how
useful
that
is,
if
you're,
relaying
and
forwarding
so
I
would
say.
Maybe
don't.
F
So
so
let
me
let
me
try
another
thing
like
I
said,
these
are
things
that
we
have
talked
about
in
in
the
in
the
in
discussions
in
the
past,
but
not
in
a
huge
amount
of
detail.
One
thing
we
said
for
relay
and
forwarding
is:
it
would
be
easier
for
us
to
write
a
BCP
saying
if
you're
doing
RTP
over
quick
and
on
one
side
and
RTP
over
UDP.
F
On
the
other
side,
you
need
to
do
secure
RTP
over
UDP
on
the
on
the
other
side,
and
rather
than
rather
than
saying,
oh
we're
going
to
do
double
encryption
so
that
what
the
middle
box
gets
is
something
that
it
can
forward
using
srtp
on
the
other
side.
F
F
And
that
would
be.
That
would
also
be
a
fine
way
to
get
off
the
way
to
get
get
through
this.
Yes,
Rob.
Thank
you
for
that
Roman.
What's
up.
J
J
It's
not
ideal,
but
it's
we're
talking
about
like
three
or
four
extra
packets
during
the
session
setup
so
and
people
who
make
it
very
easier
to
build
essentially
gateways
which
will
run
in
front
of
the
existing
details,
srtp
infrastructure
and
Gateway
to
a
30p
or
essentially
30p.
Over
quick
with
bypass
of
again,
it's
going
to
be
double
encrypted,
we're
going
to
end
up
with
more
overhead,
and
we
want
a
slightly
more
overhead,
but
it
will
work
and
it
will
work
very
quickly
for
any
sort
of
like
prototyping
or
buildups.
F
Yeah
right
right,
I,
I,
think.
Another
thing
that
we're
talking
about
here
is
conversations
that
will
be
good
for
us
to
have
after
we've
looked
more
closely
at
topologies.
B
Yeah
I
mean
sorry
just
to
say
to
respond
to
Roman
I
feel
like
yes,
there
probably
are
use
cases
for
what
you're
essentially
doing
is
tunneling
a
dtls
srtp
over
quick
but
I.
Think
that's
mask
and
I.
Think
for
the
right
answer.
There
is,
you
know,
use
a
mask.
You
know
UDP
forwarding
session
they're
talking.
People
are
already
talking
about
using
masks.
B
And
I
don't
feel
like
that's
a
use
case.
We
need
to
just
discuss
here.
D
D
J
F
Excellent
quick
adjust
your
control,
so
so
we
we've
had
again
a
range
of
thoughts
about
what
to
do,
how
much
to
say
about
quick
congestion
control
and
what
I'm
talking
about
here
is
I
think
the
least
visible
approach
to
Quick
adjustment
control,
which
is
to
Define
an
alpn.
That's
not
H3,
currently
in
zero
one,
it's
RTP,
Muk,
quick,
that
that
would
avoid
the
most
awkward
aspects
of
nested,
RTP
and
H3
controls.
F
F
C
C
The
aopm
issue
really
is
distinct
from
congestion
control,
so
I
don't
know
that
I
would
mix
those
two
things,
but
I
think
the
alpn
proposal
you
have
is
fine,
but
it's
it's
really
not
doesn't
relate
to
congestion
control.
That's
a
separate,
separate
thing.
F
I
think
so
I'm
allow
me
to
misquote
Lucas,
but
I
believe
that
I
believe
the
discussions
about
this.
Probably
in
the
quick
working
group,
because
it's
been
a
while
he
was,
he
was
saying
that
that
was
the
easiest
way
to
make
sure
that
the
person
you
were
talking
to
knew
you
were
not
doing
http
3..
F
So
so
what.
C
C
Yeah
so
I'm,
going
to
take
an
example
in
web
transfer.
They've
been
talking
about
adding
to
the
Constructor
some
hint
about
the
congestion
control
you
want,
but
that
doesn't
affect
the
alpn
at
all.
It
just
affects
what
what
congestion
control
is
implemented
anyway,
just.
C
F
C
F
And
and
I
I
think
that
I
think
that
you're
thinking
that
they're,
more
orthogonal
than
I
was
so
and
the
reason
we're
having
these
discussions
is
to
give
me
something
to
think
about,
but
the
the
the
other
thing
was
we
we've.
F
You
know
these
proposals
have
extended
all
the
way
into
saying:
I
want
to
tell
you
what
the
congestion
control
is
going
to
be,
and
you
know
to
Define
certain
aspects
of
the
congestion
control
and,
and
that
was
what
we
were
trying
to
talk
ourselves
out
of.
If
there's
a
Constructor
that
says,
do
something
for
that's
reasonable
for
for
real-time
media,
that
that
would
be
fine
with
me
too.
F
Okay,
Peter.
L
I
agree
with
Bernard
that
it
would
be
sad
if
there
were
implementations.
That
said,
if
aopn
is
RTP,
mux
quick
use,
real-time
congestion
control,
algorithm
else
views
not
real
time.
I
think
it'd
be
much
better.
If
we
just
had
to
fit
something,
I
would
like
congestion
controlling
to
them,
and
then
we
didn't
have
some
case
statement
in
implementations
based
on
aopm.
F
J
G
Was
just
say
that
yeah.
B
It
was
sort
of
hard
to
hear,
but
I
think
she
was
agreeing
that
we
shouldn't
have
the
congestion
control
tied
to
the
alpn.
But.
F
You
know
cool,
okay,
excellent,
yeah,
and-
and
thank
you
thank
you
for
that.
Is
there
anything
else
we
may
move
on
to
the
next
slide?
F
If
we
finish
all
of
those
then
doing
them,
then
I've
got
I've,
got
issues
as
of
last
night
for
doing
the
mapping
between
quick
feedback
and
what
we
might
want
to
get
out
of
extended
RTC
rtcp
feedback
and
working
out
the
details
of
quick
connection
interconnection.
You
know,
sorry,
quick
connection,
interaction
with
ice,
and
so
so
those
are
new
new.
D
F
In
the
get
in
the
GitHub
and
I
saw,
Bernard
has
put
in
a
couple,
and
so
people
were
also
visibly
typing
so,
but
that
that's
the
way
it
looked
to
me
and
I
wanted
to
thank
people
a
lot
for
this
session.
I
know
it
was
very
helpful
for
me
and
I
hope
it
was
for
York
and
Mattis
as
well.
F
And
with
that
I
well,
I
think
I
have
one
more
slide
saying.
Thank
you
for
sharing
your
thoughts
and
see
you
in
Yokohama.
Next.
B
B
B
K
Yeah
thanks
I
only
have
one
topic
to
talk
about
today.
Next
slide,
please.
K
K
We
have
talked
about
multiplexing
kind
of
related
to
aopn
because
when
we
Define
an
ilpn,
we
probably
or
maybe
also
have
to
define
the
multiplexing
or
also
the
other
way
around.
If
we
Define
some
multiplexing,
we
may
also
already
have
a
decision
on
my
LPN
and
we
also
discussed
about
how
much
needs
to
be
defined
in
this
document
and
yeah.
That
kind
of
extends
the
scoping
discussion
from
earlier.
K
K
The
current
graph
defines
a
flow
ID
or
session
identifier
to
do
this,
but
we
are
not
sure
how
compatible
this
will
be
with
other
formats
and
then,
depending
on
what
multiplexing
we
use.
This
might
have
implications
on
RDP
over
quick,
because
we
also
want
to
try
lightweight
and
independent
as
much
as
possible,
but
of
course,
we
also
want
to
build
something.
That's
usable
by
most
of
us
for
all
of
us
next
slide.
K
So
to
discuss
multiplexing
to
get
everyone
on
the
same
page
first,
the
easiest
thing
we
could
do
is
to
say
we
just
put
RDP
packets
for
one
RTP
session
and
one
quick
connection,
and
if
you
want
to
do
multiple
r2p
sessions
open
another
quick
connection
that
is
kind
of
easy
and
it's
similar
to
what
we
could
do
over
UDP,
where
we
could
put
different
RTP
sessions
using
different
UDP
ports
and
do
the
multiplexing
on
those
parts.
And
then
we
would
similar
to
that
use.
K
Quick
connection
IDs
instead,
but
that's
also,
maybe
not
the
best
thing
we
can
do,
because
maybe
we
can
use
one
quick
connection
for
multiple
sessions
and
maybe
also
even
further
things.
So
the
next
point
is
to
how
can
we
do
multiplexing
with
multiple
RTP.
K
Quick
connections,
accurate
connection
and
these
RTP
packets
can
be
encapsulated
in
streams
or
datagrams
or
both,
and
because
we
are
using
the
stream
per
frame
model
that
Bernard
talked
about
earlier.
We
can't
use
the
quick
stream
IDE
to
do
multiplexing,
because
one
RTP
session
was
streamed
with
a
lot
of
different
streams
and
also
the
datagrams
don't
have
any
stream
ID
is
there.
K
That
would
also
help
us
to
do
our
gcp
packets
in
the
same
quick
connection,
and
these
rtcp
packets
could
then
be
multiplexed
with
part
of
the
versions
of
different
RTP
sessions
and
then
the
last
point.
If
we
want
to
do
multiplexing
with
other
protocols,
what
we
would
do
over
UDP
is,
for
example,
using
RC
7983
this
to
Multiplex
these
different
protocols.
B
Yeah
so
I
think
one
of
the
use
cases
that
a
lot
of
people
I
think
when
we're
looking
at
doing
you
know
various
media
over
quick
type
Solutions
are
interested
in
is
the
use
case
where
your
call
control
and
your
media
are
sharing
the
same
quick
connection,
and
the
use
case
here
is
for
like
when
you're
going
to
a
load
balancer,
you
know
and
you've
got
something
that
spreads
out
to
a
lot
of
back-end
systems,
and
it
makes
the
problem
of
making
sure
that
your
secure
call,
control
and
your
media
go
to
the
same
box
on
the
back
end
much
easier.
B
If
it
actually
goes,
it's
a
quick
connection,
so
I
feel
like
looking
at
Multiplex,
make
finding
something
where
we
can
Multiplex.
Actually,
our
call
control,
whether
that's
something
proprietary
or
ship,
or
you
know,
whatever
with
the
media.
It's
probably
something
we
should
look
at
as
an
important
use
case,
because
I
think
that's
been
something.
A
lot
of
people
have
been
interested
in
looking
at.
B
B
H
Yeah
I
certainly
can
do
it
so
H3
for
uni-directional
streams
has
a
stream
type.
I
was
wondering
if
there
was
maybe
more
of
a
a
general
applicability
to
multiplexing.
So
so
you
could
start
a
stream
as
just
RTP
over
quick
in
the
alpn,
but
then
there's
a
transport
extension
that
says
I
support
some
other
frame
type
that
allows
you
to
negotiate
adding
a
stream
type
in
the
from
when
you
open
a
new
stream,
so
you
can
identify
which
protocol
this
is
actually
for.
I,
don't
know
if
that
would
be
potentially
more
helpful.
K
Maybe
I
think
it
may
become
helpful
if
we
go
to
the
next
slide.
We
have
a
couple
of
options
which
we
will
would
like
to
talk
about.
One
of
them
is
web
transport.
The
second
is
some
part
of
web
transport,
and
the
last
one
is
something
I
talked
about
in
the
last
interview
meeting
and
I
think
when
I
go
through
these
different
options,
we
will
also
talk
about
how
to
bring
in
other
protocols
like
that,
and
maybe
that
would
already
be
helpful
is
that
okay,
other.
L
Say
that
if
we
want
to
make
a
generic
multiplexing
system,
I
mean
that
is
what
web
transport
is,
so
it'd
be
kind
of
Reinventing
it
so
I'm
glad
to
see.
That's
the
first
bullet
point
on
your
slide
here.
K
Okay,
so
the
three
options
options
we
have
is
web
transport,
which
was
brought
up
in
the
last
meeting
or
actually
was
brought
up
as
some
more
generic
multiplexing
layer
which
we
could
introduce
between
the
quick
connection
and
RTP
and
web
transport
could
be
one
instance
of
this.
K
The
second
option
is
to
use
partial
web
transport,
which
would
be
kind
of
more
generic,
using
our
own
abstraction
layer
in
between
to
do
multiplexing,
but
maybe
reuse,
parts
of
web
transport.
For
that
and
the
last
option,
I
will
talk
about
it
later
on
a
dedicated
slide
for
only
doing
RTP
and
rtcp
first,
but
try
to
be
extensible
in
the
future
for
other
protocols.
K
So,
let's
start
with
web
transport
on
the
next
slide
down,
so
web
transport
defines
a
couple
of
Transport
features,
properties
and
requirements.
K
These
trans
features
and
properties
and
requirements
can
be
implemented
by
different
instances
of
Transport
protocols
below
and
that
are
currently
defined
for
HTTP
2
and
https
3..
The
features
would
be
very
much
what
we
would
like
to
use
for
RTP
over
quick.
We
would
have
different
sessions
which
could
be
mapped
to
RTP
sessions
or
multiple
RTP
sessions.
K
In
one
quick
connection
or
even
multiple
web
transport
sessions
to
have
one
for
RTP
and
one
for
something
else,
membrok
transport
defines
datagrams
and
uni
and
bidirectional
streams,
which
is
exactly
what
we're
using
in
quick
for
RDP
over
quick
and
the
transport
properties
include
some
nodes
about
different
transport
protocols.
Implementing
these
features
because
for
HTTP
2
we
don't
have,
for
example,
unreliable
datagrams,
so
the
transport
properties
are,
or
some
of
them
are
optional.
K
For
example,
unreliable
data
delivery,
aborting
screens
may
result
in
re-transmissions,
which
would
be
the
case
for
HTTP
2,
which
runs
over
TCP
and
datagrams
may
also
be
retransmitted
in
that
case,
but
for
the
web
transport
mapping
to
http
3,
we
would
pretty
much
get
exactly
the
features
we
have
in
quick.
K
Then
there
are
a
couple
of
requirements.
The
two
first
of
them
are
I
think
they
are
clear.
We
don't
need
to
talk
about
them.
Much
I
also
already
mentioned
the
multiple
sessions
which
we
could
use
for
multiplexing,
so
that
would
solve
our
multiplexing
problem,
but
then
there
are
also
two
things
which
I
don't
really
know
how
to
work
around
them,
how
to
or
what
to
do
with
them.
K
First
of
them
is
that
a
client
needs
to
be
able
to
indicate
An
Origin
to
the
server,
and
the
second
is
to
be
able
to
describe
a
server
endpoint
location
using
our
URI
and
I
have
some
questions
about
these
points
on
the
next
slide.
So,
but
not
if
you
can
write
one
more
slide,
that
would
be
nice
or
is
it
specific
to
the
slide.
C
Some
yeah,
just
one
thing,
is
the
way
web
transport
sessions
are
used.
Is
they're
typically
used
to
delimit
browser
tag
tabs.
C
So
you
can't
really
use
a
web
transport
session
to
have
multiple
different
RTP
sessions
within
a
single
tab,
and
you
could
use
a
separate,
quick
connection
for
that.
But
I,
don't
think
that's
what
you
want
so
yeah
I
think
I
think
you're
misinterpreting
any
what
a
web
transport
session
is.
K
C
Well,
you,
you
could
either
just
use
bundle,
in
which
case
it
wouldn't
matter,
I
just
throw
everything
there,
but
if
you
wanted
to
actually
be
able
to
do
different
RTP
sessions
within
a
web
transport
session,
you'd
need
to
have
some
kind
of
multiplexing.
For
that.
K
C
B
C
Can
open
a
set,
a
different
connection
and
potentially
have
it
be
pooled
right
and
then
end
up
with
different
sessions?
But
it's
it's
unpredictable.
You
don't
know
whether
you
you
can't
really.
You
can't
really
control
it.
Pooling
is
not
it's
not
easily
controlled.
It's
something!
The
browser
decides,
not
your
application.
K
Mm-Hmm,
okay,
then,
maybe
to
continue
what
questions
we
also
had
about
rock
transport.
So
that's
an
important
point.
I
think
that
we
still
would
need
to
Define
our
multiplexing.
On
top
of
that,
then,
additional
things
we
noted
is
the
two
points
I
noted
before
we
don't
really
know
what
to
do
within
with
a
client
origin
in
RTP,
real,
quick
and
server
endpoint
URI,
because
we
don't
really
have
these
Concepts.
K
Maybe
we
just
ignore
them.
I,
don't
really
know
if
it
would
be
a
problem
for
us.
We
just
wanted
to
notice
here.
C
You
don't
have
to
deal
with
either
these
things
in
your
respect,
basically
a
way
to
think
about
the
URI.
It's
like
web
sockets.
You
need
a
URI
destination
to
establish
the
connection.
C
K
Them
down
here
for
RTP,
then
for
sdp.
We
might
need
to
do
something
else,
and
then
the
last
point
is
probably
the
one
which
is
most
important
about
when
we
think
about
using
web
transport
for
multiplexing
for
rgp
over
quick
is
that
it
would
be
defined
for
HTTP,
2
and
3,
and
we
were
wondering
whether
it
really
makes
sense
to
depend
on
rep
transport
and
HTTP,
2
and
3
to
get
RTP
over
quick
done
and
I.
K
Think
that's
the
you
know
the
most
important
question
we
would
like
to
answer
here
either
now
or
after
they
show
the
other
two
options.
Maybe
the
next
slide.
K
But
we
are
also
not
really
sure
if
we
can
simply
reference
things
from
web
transport
without
saying
or
depending
on
http,
and
we're
also
wondering
how
much
different
is
this
from
the
current
approach
using
flow
IDs,
because
we
would
probably
rebuild
something
similar
to
The
Sessions
in
web
transport
using
our
own
abstraction
layer,
and
that
would
probably
look
similar
to
what
we
are
using
with
flow
IDs
already
and.
K
What
I,
already
presented
in
October
I
think
is
that
we
limit
ourselves
to
RTP
and
rtcp
for
now
keep
using
the
flow
identifier.
So
we
can
have
multiple
RTP
sessions
and
rtcp.
All
in
the
same
quick
connection
then
register
the
aopn
for
RTP
click,
because
it's
not
multiplexing
with
anything
else
anymore
and
then
allow
other
documents
that
could
be
described
or
could
be
written
in
the
future
to
specify
new,
LP
and
tokens,
which
will
also
then
have
to
specify
the
multiplexing
between
RDP
over
quit
and
other
protocols.
K
How
the
atmos
flexing
looks
like
it's
hard
to
describe
in
our
document
now,
because
we
don't
know
which
other
protocols
we
would
need
and
new
documents
could
knowing
which
protocols
to
Multiplex
with
RDP
over
quick,
for
example,
we
use
our
flow
identifier
to
assign
different
fluid
interfires
or
different
protocols,
or
build
some
other
kind
of
multiplexing
between
those
yeah.
That's
other
three
options
we
currently
see
and
would
be
I'm
happy
to
hear
your
opinions
on
these
foreign.
C
Yeah
I
don't
understand
what
partial
web
transport
is
like
when
you
say
a
void.
A
dependency
on
HTTP
kind
of
web
transport
is
negotiated
with
nhdp3,
so
I
mean
in
your
implementation.
You
don't
need
all
of
the
HTTP
3
headers.
So
in
that
respect,
you
don't
need
a
full
hap3
implementation
to
do
web
transport,
so
your
implementation
can
largely
avoid
it,
but
but
the
protocol
spec
has
kind
of
been
meshed
with
http
3.
C
and
you
also
don't
want
to
create
your
own
version
of
web
transport,
so
it
can't
communicate
with
the
ones
and
browsers.
It's
largely
a
browser
specific
thing.
So
you
want
to
you
want
to
maintain
browser,
compatibility,
I,
think
or
it
doesn't
make
a
lot
less.
Yeah.
K
I
think
maybe
maybe
partial
web
transport
is
a
bad
naming
here.
What
we
were
thinking
was
since
web
transport
is
doing
sessions
and
multiplexing
between
different
sessions.
We
can
look
at
how
it's
been
done
in
web
transport
and
I.
Think
it's
using
session
IDs
and
reference
these
parts
from
the
spec,
so
we
don't
have
to
invent
our
own
multiplexing,
which
we
currently
do
with
multi-abits
with
our
flow
identifiers.
K
K
C
Yeah
and
if
you
don't
want
to
do
HTTP
2,
you
don't
really
have
to
I
mean
the
API
allows
you
to
say:
I,
don't
wanna
I,
don't
want
to
drop
down
to
hdb2.
It
has
no,
you
know,
there's
no
point
or.
C
For
some
reason
that
you
want
to
do
it
and
then
obviously
you
will
get
these
weird
effects
for
the
datagrams,
but
presumably
you're
ready
for
it.
Yeah.
K
I
agree
that
HTTP
2
is
probably
an
optional
for
us
anyway,
because
I
mean
there's
also
actually
RTP
over
TCP,
and
if
you
really
want
to
do
HTTP-
and
you
know
what
you're
doing
that's
fine,
but
probably
most
of
us
would
use
HTTP
3
and
the
question
here
is:
rather,
how
can
we
or
do
we
want
to
avoid
this
dependency
on
HTTP
3?
Does
it
have
implications
that
we
do
want
to
avoid
or
not.
B
B
My
inclination
would
be
to
say,
do
Define
sort
of
a
raw
single
RTP
session
over
quick
and
Define.
Our
you
know.
Web
transport
and
the
web
transport
can
have
should
be
able
to
handle
all
the
dmux
cases
and
all
the
browser
cases
and
the
other
one
can
handle
the
well
I.
Don't
want
that
complexity,
but
don't
sort
of
that
you
know
don't
do
anything
else
unless
we
feel
like
we
understand,
there's
a
use
case
for
it,
but
maybe
there
is
a
use
case
for
it.
I'm
not
seeing
it.
K
K
But
since
Bernard
mentioned
this
earlier,
would
that
be
not
given
that
we
can't
have
multiple
or
can't
decide
which
reference,
what
sessions
go
into?
One
click
connection.
D
B
B
Yeah
so
yeah
so
yeah
I
think,
like
I,
said
I
think
you
know
if
I
guess,
the
question
is:
do
you
actually
need
the
two,
the
two
RTP
sessions
or
whatever,
to
go
over
the
same
put
connection
as
opposed
to
share
a
quick
connection,
as
opposed
to
parallel
quick
connections?
Does
that
how
much
does
that
matter
to
you
and
I
guess:
they're!
Quite
the
you
know
the
this
thing
would
be.
B
You
know
the
load
balancer
thing,
but
given
that
web
transport
is
designed
for
load
balancer
cases,
it
isn't
guaranteeing
that
generally
people
will
have
solutions.
They
understand
for
how
to
do
that.
K
Yeah
good
to
know
how
what
what
people
think
about
this,
but
I
think
it
might
be
useful
to
have
multiple
RDP
sessions
in
one
pre-connection,
even
if
it's
just
to
make
congestion
control
between
them
easier.
Foreign.
C
C
B
C
Little
bit
different
and
requiring
web
transport
and
all
that
Machinery
might
not
there's
a
lot
of
Machinery
in
web
transfer,
which
isn't
necessary
for
peer-to-peer
quick
like
the
the
client,
IDs
and
stuff,
like
that.
That's.
C
C
But
the
question
is:
what
are:
how
do
we
move
forward
on
some
of
these
questions?
There's
something
we
should
note
for
the
minutes.
B
I
would
say
we
don't
yet
have
consensus
and
it
probably
makes
sense
to
for
the
authors
to
move
forward
on
defining
the
various
things
and
maybe
like,
as
you
get
more
into
the
details,
you
decide
hey.
This
is
really
hard
or
hey.
This
is
really
easy
and
you
know
that
might
give
us
more
information.
C
K
C
N
You
hear
me
yeah,
we
have
yes,
yeah
I'll,
try
to
I'll
try
to
do
it,
quick,
maybe
over
quick.
We
will
we'll
see
next
slide,
so
this
presentation
is
actually
stemming
from
a
side
meeting
that
that
was
organized
during
the
last
ietf
meeting
and
the
goal
was
to
look
at
as
frame
and
RTP
packetization
and,
generally
speaking,
the
the
goal
in
that
right
is
to
enable
end-to-end
encryption
of
media
over
RTP
and
I.
N
Think
in
that
meaning
Bernard,
you
you
mentioned
skip,
which
is
solving
basically
that
so
skip
is
defining
a
payload
format,
also
media
subtypes,
to
allow
uncrypted
content
over
RTP.
It's
it's
very
in
the
focus
of
a
skip
session.
So
it's
not
really
applicable
here,
but
yeah.
N
The
idea
is
to
to
do
the
same
as
what
what
was
done
for
skip,
but
in
with
a
different
scope
and
the
scope
is
webrtc
and
they
have
the
the
slide
is
slightly
misleading
because
we're
not
trying
to
solve
end-to-end
encryption
of
media
variety
we're
just
trying
to
to
solve
it
in
the
context
of
webrtc,
where
there
are
some
webrtc
clients
and
sfus
in
the
middle,
and
so
during
that
meeting
it
was
food
good
to
Define,
an
architecture
of
webrtc
clients,
sfus
and
so
on
how
they
would
interact
with
and
try
to
solve
the
end-to-end
encryption
thing
so
that
we
can
identify
requirements
and
also
maybe
provide
hints
at
how
we
could
do
it.
N
So
that's
what
we
started
doing
in
in
the
draft
the
encrypted
media
rtb
architecture
document.
So
you
can
click
on
the
link
it's
on
GitHub,
so
you
can
provide
commands
PRS
or
file
issues
as
well.
If
that's,
if
you
have
some
time
so
next
slide
so
webrtc,
so
we
have
like
three
main
nodes.
Usually
we
have
senders,
which
will
be
a
webrtc
clients,
usually
browsers.
Then
we
have
sfus
that
will
do
the
package
for
forwarding
thing
and
then
receivers,
which
might
also
be
a
web
browsers.
N
The
typical
architecture
and
it's
no
different
than
generic
RTP
is
a
the
packetizer,
is
generating
packets
from
from
metadata
and
and
and
the
encoded
frame
content,
and
basically
whenever
intermediaries
need
to
get
information
of
what
the
encoded
frame
is
and
so
on.
It
can
just
report
the
encoded
frame
content,
because
it's
in
the
packet
payloads
and
then
do
processing
based
on
that.
In
our
case
where
content
is
encrypted
end-to-end,
the
packetizer
cannot
really
extract
information
from
the
Android
frame
content
because
it's
encrypted.
N
So
this
content
is
really
opaque
and
the
same
issues
happens
for
intermediate
intermediaries.
We
cannot
extract
information
from
from
the
content,
so
it
has
to
be
replaced
somehow
by
providing
more
encodified
metadata
to
these
instruments.
So
in
the
architecture
the
encryptor
is
between
the
encoder
and
the
packetizer
next
slide
to
yeah.
Traditionally,
the
packetizer
is
tied
with
the
encoder,
but
now
we
have
the
encryptor
that
is
in
between
and
we
think
that
now
it
kind
of
makes
sense
for
bucketizer
to
become
tied
to
a
media.
N
Encrypter
is
trying
to
define
the
relationship
between
the
two
and
try
to
Define
some
kind
of
interfaces
that
we
between
the
encrypter
and
the
packetizer.
Ideally
because
we
are
in
webrtc
territory,
changes.
Small
changes
to
Media
encrypter
should
not
require
a
new
packetizer.
We
want
to
Define
enough
requirements
on
the
media
encrypter
on
the
packetizer,
so
that,
whether
you
use
as
frame
format,
V1
or
preet
stream
format,
or
even
s
packet,
basically,
the
architecture
Remains,
the
Same
and
the
packetizer-
could
be
a
roughly
the
same.
N
So
since
we
are
in
webrtc
territory
we
can,
and
since
we
want
to
Define
some
kind
of
interfaces,
we
have
web
contacts
and
web
codex
is
a
web
API.
That
is
defining
interfaces
for
media
decoders
and
encoders,
and
it's
it's
a
really
nice
abstraction
that
we
can
use
to
model
encrypt
the
packetizer
relationship,
so
next
slide
will
focus
on
on
video,
because
yeah
video
is
a
bit
more
difficult
as
well.
Since
we
are
like
it
more.
N
So,
for
instance,
video
frame
members,
so
video
frame
is,
is
representing
a
video
frame
in
webcodex,
and
it
has
some
members
like
color
space
orientation,
and
all
of
these
is
metadata
that
the
media
encrypter
would
pass
through
to
to
the
packetizer
there's
also
some
metadata
related
to
the
entire
video
check,
like
the
codec
type,
the
frame
type,
the
timestamp,
the
decoder,
config
vs,
similar
data
and
so
on.
So
it's
a
another
bunch
of
metadata.
N
That
is
that
encrypto
could
share
with
the
media
applicatizer
and
finally,
the
mega
and
crypto
is
generated
generating
encrypted
content.
It
will
not
provide
the
encoded
content
to
the
packetizer.
It
will
instead
provide
the
encrypted
content
and
from
that,
the
media
application
must
do
its
job
of
generating
packets,
and
the
idea
here
is
that,
since
it
has
a
lot
of
content
metadata,
it
can
generate
HP
headers
like
CPU
from
video
frame
orientation
and
so
on,
and
it
can
also
generate
RTD
pillows.
N
N
So
that's
working
pretty
well
in
the
context
of
a
video
Codec
that
is
not
doing
SVC
now
with
SVC.
It's
a
it's
a
bit
different,
because
now
you
really
you're
not
only
caring
about
a
video
frame,
but
you
you
will
care
about
even
individually
decodable
units
that
together
will
form.
N
We
will
allow
you
to
decode
a
video
frame
and
since
we
are
in
Weber
security,
where
the
SFU
might
want
to
send
some
decodable
units
to
some
clients
and
some
some
over
the
colorable
units
to
some
of
the
clients,
it's
up
to
the
media
encrypter
to
get
the
decodable
units
and
to
encrypt
them
separately,
and
then
it
can
provide
to
the
to
the
packetizer
the
encrypted
decodable
unit
content
and
plus
the
midday
data.
That
is
identifying
what
the
decodable
unit
is.
N
So
it's
not
only
the
media
data
related
to
the
video
frame,
but
also
limited
data
that
is
identified
identified
in
the
decodable
unit
and
again
there.
The
packetizer
will
need
to
transmit
this
additional
metadata.
The
good
thing
is
that
there
we
have
an
interface
as
well
that
is
available
in
the
slides,
so
you
can
see
the
frame
number
and,
and
so
on
and
webcodex
is
providing
that.
N
So
we
already
have
an
interface
and
abstraction
that
we
can
use
to
clearly
Define
the
boundary
between
and
the
the
contract
between
the
encryptor
and
the
packetizer.
N
Next
slide
going
now
to
packet
forwarding
unit,
so
the
packet
forwarding
unit
is
receiving
packets
sent
by
the
sender
and
we'll
send
some
of
them
to
other
clients
and
when
it
receives
a
a
packet
in,
it
needs
to
be
able
to
compute,
to
which
decodable
unit
the
packet
is
belonging
to,
and
for
that
we
are
thinking
that
the
underlying
connect
type
is
very
useful,
as
well
as
the
marker
bit
and
timestamp,
and
to
actually
decide
whether
this
RTP
packet
needs
to
be
forwarded
or
not.
N
That's
where
the
decodable
unit
metadata
is
really
important,
and
one
possibility
here
is
to
put
that
metadata
in
in
the
dependency
descriptor,
and
one
additional
potential
requirement
is
that
some
sfus
might
want
to
record
a
call
and
they
might
want
to
record
the
encrypted
content.
So
it
should
be
visible
for
an
SFU
to
get
back
from
the
packets,
the
actual
encrypted
content
and
store
it
somewhere
and
then
send
it
later
on,
if
need
be,
going
very
quickly
to
the
reserver
side
architecture
now
next
slide.
N
So
now
we
we
are
at
the
level
of
the
unrecipient,
the
robotic
client
it
has
a
key.
So
it
can
decrypt,
but
still
before
that
it's
receiving
packets
and
it
has
very
similar
requirements
as
the
packet
forwarding
unit,
because
it
needs
to
reconstruct
the
encrypted
content.
But
it
also
needs
to
do
early
processing
on
the
packets
like
if
it
seems
that
it
it
needs
a
a
keyframe,
then
send
a
fear
and
and
so
on
and
yeah.
It
also
needs
to
be
able
to
use
existing
rtp-based
redundancy
mechanisms.
N
Yep
next
slide
will
slightly
over
time.
I.
Guess
it's
fine.
So
so,
basically,
the
document
is
defining
the
architecture
and
some
API
contracts
that
they
that
we
could
set.
J
N
The
different
breaks
of
the
pipeline-
and
we
we
are
thinking
that
there
is
a
proof
of
feasibility
of
the
overall
implementation
in
the
reality
world.
Basically,
you
you,
you
apply
as
frame
on
each
encoded.
Video
chunk
as
frame
would
be
a
password
to
all
the
related
metadata
that
is
received
from
web
codecs
and
the
RTP
packetizer
would
generate
the
dependency
descriptor
rtph
extension
from
the
metadata,
and
it
will
encrypt
the
content.
It
will
split.
N
The
uncomplete
content
based
on
MTU
and
the
payload
type
itself
could
be
a
prepended
in
the
payload,
for
instance,
we
could
think
of
which
is
very
similar
to
what
skip
has
done.
Adding
some
subtypes
like
audio
and
Cryptid
or
video
encrypted,
for
instance,
and
and
use
that
for
negotiation.
N
Next
slide
and
I
will
be
close
to
to
the
end.
So
Peter
mentioned
that
sfus
might
want
to
repacketize,
and
that's
something
that
is
interesting.
It's
not
really
in
the
draft
now
and
that's
something
that
I
think
we
would
welcome
feedback
there,
since
there
may
be
different
mtus
or
there
might
be
different
protocols.
N
Maybe
some
packets
need
to
be
split
or
merged
and
I
wonder
whether
it
should
be
in
scope
of
this
work
and
whoever
there
are
some
cases
that
are
more
important,
like
large
to
small
functions
might
be
more
important
than
small
to
large
or
so
so
I'm,
not
sure.
We
will
have
a
lot
of
time
to
discuss
this,
but
yeah
feedback
is
welcome
either
here
or
in
the
mailing
list
or
on
GitHub,
and
the
last
slide.
N
So
my
my
temporary
conclusion
is
that
in
the
context
of
webrtc,
we
we
can
its
desired
to
support
end-to-end
encrypted
content,
and
we
can
actually
do
that
and
it
can
be
done
even
though
the
practically
and
encrypted
format
changes
like
it
can
be
as
frame
as
packet.
It
can
be
something
else.
We
can
still
Define
the
same
architecture
and
hopefully
reuse
very,
very
similar
packetas,
so
the
current
architecture
provides
a
basis
for
what
and
how
we
want
to
support
packetization
foreign
content.
N
It's
still
a
draft,
so
I'm
wondering
whether
at
some
point
we
want
to
submit
it
to
the
working
group
or
whether
we
should
think
of
starting
Solutions
instead,
so
votes
from
the
working
group,
I
only
have
like
a
few
minutes
left,
because
I
need
to
jump
in
in
some
of
the
meetings.
But
if
you
have
feedback,
it's
most
welcome.
M
Just
say
that
I'm
also
working
on
this
area
in
this
area,
for
other
reasons,
the
handling
of
encoded
media
without
having
to
decode
it.
In
order
to
understand
what
to
do
with,
it
is
quite
important
in
a
couple
of
scenarios
we
have
where
we
have
looked
at
I
mean
enter
any
crypto.
Content
is
one
important
point,
but
I
think
we
need
an
architecture
for
for
doing
a
little
more
than
that,
but
this
is
also
being
discussed
in
webrtc
working
group
in
wgse.
So
the
initial
corporate.
L
If
it
wasn't
already
obvious
I'm
in
support
of
doing
this,
and
starting.
N
So
what
would
be
the
next
steps
when
Bernard
Jonathan
should
I
try
to
finalize
the
document
and
try
to
submit
it,
or
should
we
focus
more
on
Solutions.
B
I
think
having
the
architecture
just
sort
of
as
the
background
for
things
is
useful.
I'd
say
you
know,
maybe
submit
that
as
a
I
mean
submitted.
You
know
it
doesn't
have
to
be
complete
it
just
you
know
as
a
zero
zero
digital
draft.
It
doesn't
have
to
be
perfect,
obviously,
but
I'd
say,
but
probably
also
start
working
on
Solutions
in
parallel
just
so,
we
can
sort
of
see
okay.
What
do
you
have
in
mind
and
how
are
you
thinking
of
doing
it?
Just
you
know
again,
sort
of
you
know
broad
outlines.
B
You
know
with
some
straw
bands,
if
need
be
just
so,
we
can
start
having
something
up
a
place
to
start
working
on
it,
but
I
think
there's.
Clearly
there
is
a
lot
of
interest
in
this
I
think
it
would
certainly
be
something
that
the
working
group
is
interested
in
doing
sort
of
has
a
problem
to
solve.
C
Progressing
this
I
know,
there's
there's
a
bunch
of
things
that
that
you've
done
that
are
new
and
I
think
represent
progress
from
where
we
were
so
I
just
want
to
encourage
you
to
keep
keep
at
it.
N
Okay,
I'll
try
to
submit
it
before
the
next
ITF
meeting
and
I
guess,
maybe
with
Peter
and
maybe
Sergio
as
well.
We
we
might
want
to
advance
Solutions,
then
in
in
the
scope
of
this,
this
architecture
graft
and
webrtc.
J
N
Them
yeah
good
point:
I
will
so
maybe
around
I
will
put
you
in
the
room
as
well.
C
Okay,
I,
don't
know
Peter.
If
you
want
to
go
over
any
of
the
remaining
slides,
you
got
I
think
two
slides.
L
Basically,
if
we're
going
to
have
a
solution,
I
think
basically
we're
trying
to
decide
what
to
go
into
that
and
as
far
as
I
know,
there's
the
things
that
could
go
in
the
biggest
one
is
the
payload
type,
but
because
it
including
that
helps
avoid
Associated
payload
exclusion
and
then
within
that
we
can
decide
if
it's
optional
or
required,
because
it
might
be
cases
where
you
only
have
one
audio
codecs.
L
What's
the
point
of
burning
a
byte,
but
then
there
are
other
things
like
frame
number
or
offset
or
index
that
are
really
just
helpful
for
repacketization
by
an
SFU
for
out
of
order
packets,
and
this
is
something
that
you
and
talked
about
briefly
I.
Don't
think
we
have
time
to
go
into
why
those
are
useful.
But
I
just
wanted
to
point
out
that
a
major
design
decision
is
really
just
what
is
included
in
the
format
that
we
come
up
with
and
you
know
things
can
be
optional.
D
B
And
this
this
might
be
something
we're
just
having
like
a
quick
and
dirty.
You
know,
like
you,
know,
proof
of
concept,
implementation
just
to
play
with
it
and
see
how
hard
is
this
implementation?
How
much
do
I
actually
use
this
information
in
my
implementation
would
be
useful,
so
just
like
toss
something
together
in
JavaScript
or
something.
C
Okay,
I
think
we've
reached
the
end
of
the
meeting.
I
wanted
to
thank
everybody
for
coming
and
I
guess.
We
will,
as
mentioned,
discuss
a
bunch
of
things
on
the
mailing
list
and
also
we
will
of
course
have
our
normal
meeting
at
itf116..