Subject: [vmeet] Fwd: report on use of JITSI for the ROLL virtual Interim meeting

From: Russ Housley <[email-masked]>
Date: Wed, 11 Feb 2015 00:56:04 -0500

> From: Michael Richardson <[email-masked]>
> Date: February 10, 2015 1:23:10 PM EST
> To: Emil Ivov <[email-masked]>, [email-masked], Bernard Aboba <[email-masked]>
> Cc: Harald Alvestrand <[email-masked]>, [email-masked], Joe Hildebrand <[email-masked]>, Ines Robles <[email-masked]>
> Subject: report on use of JITSI for the ROLL virtual Interim meeting
> 
> 
> {this email is FYI, and doesn't really require any action.
> tl;dr>  it worked well enough for our meeting, thank you. }
> 
> The ROLL Virtual Interim meeting that occured today was done using the
> JITSI.tools interface.  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.
> 
> We published a 5 revisions of slides, with the first revision going out more
> than a week beforehand, and the last version about three hours before
> hand. They went out as PDF, and were uploaded to the proceedings site each
> time.  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.
> 
> 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... too bad, that would
> have been nice.  I will actually consider having my laptop host the "slide
> sharing" screen next time seperate from my own use, as that would let people
> see me and the slides. I understand that G+, webex and other systems are
> going in that direction as well.
> 
> 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.
> 
> Emil and I had previously set up a SIP bridge for audio, and it did work at
> one point, but SIP spammers/attackers on my PBX forced us to add some
> whitelist ACLs.  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.  While I was going to
> either call one or two participants from my SIP bridge, I also thought I
> might dial the bridge into a webex bridge, for those that I could not dial,
> could not dial me, or if my SOHO VDSL2 ran out of bandwidth.  I wanted to
> repoint the SIP bridge to a location with better connectivity, but I wanted
> it work first.
> 
> We had between 10 and 12 participants in the conference, and I think that we
> had the right set of people there.  It worked well once we got going.  Two
> participants were initially on Cisco VPNs, and they found they had to turn
> off the VPN, and then they had to either reload the page, or they had to
> restart the browser in order to get audio to work.  I know that Alvaro Retano
> tried to join from his cisco office, and that it failed because of his
> corporate firewall, which of course, he couldn't just "turn off". I've been
> told that this is a known pain point for webrtc, and I guess that the SEMI
> workshop talked about this too.
> 
> 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.
> 
> I want to say that today, that ietf.webex.com tells me again that I have a
> completely unsupported platform and browser; it did this to me last week for
> a day and then stopped. I can not even start the call which I could
> previously do (and then use the PSTN to connect).  This worked most of the
> time up to last week (but webex has NEVER worked for computer audio, and I
> can't share slides, the screen sharing, when it works is mostly a fail)  This
> almost totalled one nomcom call last week.  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 abandon it.
> 
> It would also be nice if we had an official SIP bridge that could get audio
> to the PSTN, ideally controlled from within JITSI.
> 
> {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}
> 
> --
> Michael Richardson <[email-masked]>, Sandelman Software Works
> -= IPv6 IoT consulting =-
> 
> 
>

---

From: John Leslie <[email-masked]>
Date: Wed, 11 Feb 2015 07:18:29 -0500

>> Cc: Harald Alvestrand <[email-masked]>, [email-masked], Joe Hildebrand <[email-masked]>, Ines Robles <[email-masked]>

   I note that Russ received this almost by accident (from the iesg list).

>> JITSI.tools interface.

   I actually meant to try to attend, though I really had nothing to
offer but limited experience with JITSI. Alas, I needed to do two errands,
and shoveling the drive killed my enthusiasm. :^(

>> That we were going to do JITSI was announced with the original
>> announcement (4 weeks ago), but of course, many did not bother

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

>>... We think that it is pretty important that the slides be out ahead of

   Absolutely!

>> that I could not just point at a PDF file somewhere...

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

>> microphone even work", and also "oops, I'm still muted" problems that occur

   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. :^(

>> Emil and I had previously set up a SIP bridge for audio,

   :^) :^)

>> and it did work at one point, but SIP spammers/attackers on my PBX
>> forced us to add some whitelist ACLs.

   :^(

>> 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?)

>>...
>> had the right set of people there. It worked well once we got going.

   :^) :^) :^)

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

   Do we understand why this keeps happening?

>> technology is critically important.  Dog food.

   These would have to be _other_than_ their regular calls.

   The learning-curve is just too steep -- often driven by things which
seem trivial _after_ you understand them. Though WebEx has problems
almost every telechat, they're problems which seem familiar...

>> previously do (and then use the PSTN to connect)...

   Indeed, that is a very common "Plan B"...

>> 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.

   But if we can get enough "virtual interims" using an alternative like
JITSI, we might develop a critical-mass used to working with it...

>> to the PSTN, ideally controlled from within JITSI.

   Hmm... Dunno what that would take.

>> 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.

--
John Leslie <[email-masked]>

---

From: Michael Richardson <[email-masked]>
Date: Wed, 11 Feb 2015 11:04:06 -0500

> 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...

    > 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.

    > 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.

    > 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.

    > 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.

    > (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...

    > 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)

    > These would have to be _other_than_ their regular calls.

Or a part of their regular call.

    > 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?

    > 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 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-masked]  http://www.sandelman.ca/        |   ruby on rails    [