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From YouTube: Jitsi Community Call - November 19, 2018
Description
The Jitsi Community Call is held every other Monday at 10:30AM Central US time at https://meet.jit.si/TheCall. The Jitsi team provides updates and community members can ask the team questions.
See https://community.jitsi.org in the mean time for questions.
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B
People
have
been
asking
about
the
effects
of
octo
and
cascading
video
bridges
and
in
coming
weeks,
then
this
is
what
I
said
on
the
last
go
as
well:
I'll
be
looking
at
the
data
that
we
have
and
trying
to
see
what
the
effects
are
and
we'll
eventually
post
about
it.
But
right
now,
I
don't
have
any
actual
updates.
C
Yeah,
you
don't
have
many
updates
as
well
I'm
here
mostly
to
just
in
case
I'm
needed
to
answer
questions
to
people
having
questions
they're
there.
There
were
some
fixes
for
the
bridge
that
didn't
get
merged
because
we
have
some.
The
piers
were
failing
that
I'm
expecting
that
will
be
fixed
either
today
or
tomorrow.
C
So
we
should
have
them
in
by
tomorrow.
The
fixes
are
related
to
upscaling
to
the
HD
resolution,
and
sometimes
that
doesn't
happen
when
there
is
there's
somewhat
limited
bandwidth
and
then
then
we're
going
to
start
working
on
fixing
some
artifacts
in
the
video
streams
we're
going
to
resume
that
work
actually
and
hopefully
we'll
be
done
pretty
quickly,
and
we
will
have
some
good
efforts
in
the
next
community.
Go.
That's
for
me,
cool.
D
E
F
A
G
I
have
one
quick
question.
This
is
kind
of
a
follow-up
from
from
last
week,
where
we're
trying
to
leverage
the
recorder
RTP
in
poll
in
sigasi
and
I
posted
a
couple.
Questions
on
community
I
was
having
trouble
at
account,
switch
back
to
the
audio
mixer
approach,
but
wasn't
clear
the
best
approach
for
getting
the
RTP
translator,
I've
sort
of
used,
Pacific
gateway
as
a
as
an
example.
G
At
what
point
is
the
TV
translator
kind
of
getting
hooked
in
in
the
transcription
gateway
kicks?
So
in
the
in
the
mixer
case,
you
override
you've
got
a
new
media
called
conference
and
you
override
get
get
default
device
and
you
create
the
audio
silence
mixer,
but
in
the
case
of
translator,
wasn't
clear,
there's
two
things
or
having
the
suit
gateway
side.
B
B
We
had
some
discussions
about
sending
opus
or
all
you
in
opus
format.
We
do
instead
of
decoding
and
sending
raw
PCM
and
last
time
I
talked
with
Nick.
He
had
finished
his
other
tasks
and
he
was
thinking
about
switching
to
this
one.
So
in
the
future
that
might
change
the
future,
we
might
have
transcription
using
the
translator
mode
instead
of.
G
It
doesn't
seem
the
SIP
in
the
transcription
again
why
the
transcription
is
not
using
translation
is,
are
to
be
translators
because
sort
of
the
way
it's
set
up,
but
is
it
not
possible
to
just
create
an
order
to
be
translator
in
the
new
media
called
conference
or
or
what
is
the
approach
that
Nick's
using
for
getting
opus
audio?
It's
as
opposed
to
the
PCM.
B
G
User
so
yeah,
it
sort
of
seems
yeah.
That
makes
sense,
which
is
you
know,
so
that
should
be
possible
if
I
just
create
a
in
the
new
media
call
conference
that
gets
created
eyes,
I
create
a
translator.
The
media
service
is
there
anything
else.
I
need
to
do
to
be
able
to
sort
of
intercept
those
RTP
streams
evident.
So
that's
sort
of
the
only
missing
piece
that
wasn't
clear
on
its
like
how
there
must
be
some.
So
in
the
SiC
gateway
case,
it's
it's
using
this
SSRC
rewriter.
It
wasn't
exactly
sure
why.
B
Yeah,
that
should
that
should
be
pretty
much
everything
that
you
need
you
you
need
to
attach
somehow
to
the
translator.
Now,
if
you
look
at
the
recorder
or
the
import
code,
I
think
that's
that's
where
I
think
it
that
just
to
get
translator,
yeah.
G
It
takes
the
translator
as
a
constructor
yeah
argument:
okay,
okay,
so
you
know
that
yeah.
So
just
wasn't
clear
like
what
else
I
need
to
do
at
the
call
level
to
get
the
to
ensure
that
that
RTP
translator,
so
is
it
enough
to
just
create
RTP
translator
and
override
that
at
the
get
default
device?
So
right
now
we're
the
transcriber
gateway
session?
Is
you
know
it's
creating
an
audio
mixer
using
sort
of
the
audio
silent
stream
at
the
and
overriding
get
default
device
should
I
be
doing
the
same
thing.
They're
in
the
RTP
translator
side.
G
Virtual,
a
Richard
I,
you
know
I
posted
something
to
the
community,
but
I
yeah
right
right.
That's
what
I
mean
how
about
Union?
Okay,
I'll,
try
to
take
a
look:
okay,
yeah
I'll,
be
really
helpful.
I
served
rolling
back
to
the
to
be
big
surveys,
but
it
would
just
be
a
lot
more
efficient
if
we
could
just
go
straight
RTP
and.
G
You
know,
I
think,
would
also
be
more
consistent
between
that
way.
You
can
invoke
the
RTP
translator
for
both
sip
gateway
or
for
corporate
transcriber
gateway
anyway,
I
was
using
I
think
it
was
your
recommendation
last
last
call
overriding
in
the
RGB
translator,
I'm,
starting
the
record
RTP
in
poll
overriding
or
leveraging.
The
audio
silence
effect
to
get
the
to
get
the
audio
trans
like
the
regular
audio
to
pass
into
it
into
the
transcriber.
Also
so,
but
you
can
have
the
best
cup
it'll
be
that
much
compatible
so
to
speak.
C
H
C
C
So
the
insides
are
I
guess
on.
If
it's
an
inside
question
it's
more
about,
when
safaris
vp8
support
is
going
to
become
mainstream,
which
I
think
should
be
in
the
following
months.
Okay,
there
would
be
very
little
on
our
site
to
do
other
than
just
stop
disabling
it
and
I
guess
that's
a
very
likely
path.
C
C
C
H
C
C
So
I'm
wiring,
not
saying
this
yeah
I
was
I
was
about
to
say
that
there
are
some
peculiar
peculiarities
in
the
in
declined.
As
far
as
I
remember
and
the
Safari
code
as
well
for
Safari,
we
use
a
different
code
paths
that
specifically
supports
only
edge
audio
only
its
mode
if
Lenny
were
Brian,
is
in
a
call,
and
maybe
they
can
correct
me-
maybe
I'm
wrong,
but
I
don't
think
it's
only
bridge.
C
C
C
Probably
get
it
for
me
with
just
a
little
and
so
we're
looking
into
that
last
and
I'm
sorry,
I'm,
not
I'm,
not
able
to
be
more
helpful,
maybe
try
to
play
with
it,
and
then
there
are
more
people
on
this
call.
We
have
another
discussion
or
you
can
send
the
meeting
too.
You
can
send
a
note
to
community
and
see
if
someone
is
able
to
jump
and
help
you
there
yep.