John Leslie <[email protected]> wrote:
    > I note that Russ received this almost by accident (from the iesg list).

I also forgot about vmeet, or even that I'm on the list...

    >>> That we were going to do JITSI was announced with the original
    >>> announcement (4 weeks ago), but of course, many did not bother
    >>> testing their environment until the last moment.

    > This is inevitable. I'm not even sure earlier testing is justifiable,
    > since so many of these tools change without warning. :^(

Some problems remain, some can be fixed with more than 1 hours notice, and
sometimes just knowing that it doesn't work at location X, means that one can
be in location Y.

    >>> ... We think that it is pretty important that the slides be out ahead of
    >>> time so people know what to expect, and that they go up in PDF, so that
    >>> people can read them on any platform.

    > Absolutely!

I mention this because there continue to be WG chairs that post vendor
proprietary formats to the proceedings site, which the secretatiat has to
convert.  This means that people have to leave their browser to see the
slides, which really can be an annoyance, even on supported platforms.

    >>> I did try to the "Share Prezi" link in JITSI (which I had not seen 
before),
    >>> but I did not realize that Prezi is a cloud based slide creation 
system, and
    >>> that I could not just point at a PDF file somewhere...

    > Perhaps this is documented somewhere; but I've never seen it documented.

I don't know what the Prezi protocol is or how it interacts with JISTI. I
played with the Prezi site trying to figure out if I could upload the PDF
there, or some other format, but failed to see anything.  Maybe there is
another site that would do it for me.

    >>> A few people had to reload the page; and there was the usual "does my
    >>> microphone even work", and also "oops, I'm still muted" problems that 
occur
    >>> on all teleconference.

    > We've become used to this. It's a flaw, IMHO; but nobody seems to want
    > to fix the audio problems which lead to this. :^(

There are three issues:
      1) lack of ability to do a local loopback test easily.
      2) too many mute buttons which do not communicate with each other.
      3) poor options for local noise elimination which would permit people
            to leave their MIC's live, and direct their audio at the
            place/application that actually needs to get it.

    >>> Despite dropping those, the SIP bridge did not work again, and we
    >>> did not re-establish communication. The lack of an available audio
    >>> bridge was an issue for a number of participants.

    > (Why does this always come as a surprise?)

I wasn't surprised that people would want it; I'm annoyed that so many have
to resort to doing this, 20 years after we started on SIP technology...

    >>> Two participants were initially on Cisco VPNs, and they found they
    >>> had to turn off the VPN...

    > Do we understand why this keeps happening?

I assume that it's RTP packets that can not get through firewalls.
It's not the VPN technology (which is piss-poor) so much that the IT
departments of even major technology companies still get in the way of IT.
"Mordack, preventer of IT". (See _The Phoenix Project_. It's a must-read)

    >>> I would like to encourage the IESG and IAB to hold a few calls a year on
    >>> JITSI, in order to make it clear to respective IT departments that this
    >>> technology is critically important.  Dog food.

    > These would have to be _other_than_ their regular calls.

Or a part of their regular call.

    >>> Joe and I have talked in the past about webex' lack of support for
    >>> common technical user platforms, and I wish the IETF would move to
    >>> formerly (formally?) abandon it.

    > That's not going to happen.

So, why is it okay for some people to scream at me because *they* can't do
JITSI, and how dare I use it, but when *I* scream that webex doesn't work, I
get ignored?

    >>> It would also be nice if we had an official SIP bridge that could get 
audio
    >>> to the PSTN, ideally controlled from within JITSI.

    > Hmm... Dunno what that would take.

1) some low-latency bandwidth in a DC for a machine, and some PSTN voip
   credits.
2) someone to set it up.
3) some work with JITSI to make it easier to set this up.
   Really, JITSI needs to be able to advertise a SIP: URL, and then
   we need other things to be able to connect to it.

    >>> {The 2013/2014 nomcom used G+ Hangouts, and we were able to use the 
Google
    >>> internal one which permitted >9 participants as we had a Google person 
who
    >>> had access to it.  The 2014/2015 Nomcom tried a call or two with JITSI, 
but
    >>> the audio bridge problem was more severe}

    > I've had generally good experiences with G+ Hangouts -- though of course
    > the details change, seemingly every time I use it.

I also have good experiences, and I use it regularly.

--
]               Never tell me the odds!                 | ipv6 mesh networks [
]   Michael Richardson, Sandelman Software Works        | network architect  [
]     [email protected]  http://www.sandelman.ca/        |   ruby on rails    [


--
Michael Richardson <[email protected]>, Sandelman Software Works
 -= IPv6 IoT consulting =-



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