> From: Michael Richardson <[email protected]> > Date: February 10, 2015 1:23:10 PM EST > To: Emil Ivov <[email protected]>, [email protected], Bernard Aboba > <[email protected]> > Cc: Harald Alvestrand <[email protected]>, [email protected], Joe Hildebrand > <[email protected]>, Ines Robles <[email protected]> > 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 protected]>, Sandelman Software Works > -= IPv6 IoT consulting =- > > >
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