Re: [Rum] [EXT] Re: Real-time text in WebRTC is discussed in mmusic - a topic closely telated to rum
Brian Rosen <br@brianrosen.net> Thu, 29 August 2019 17:17 UTC
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From: Brian Rosen <br@brianrosen.net>
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Date: Thu, 29 Aug 2019 13:17:45 -0400
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Cc: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rum@ietf.org" <rum@ietf.org>
To: Jim Malloy <jmalloy@mitre.org>
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Subject: Re: [Rum] [EXT] Re: Real-time text in WebRTC is discussed in mmusic - a topic closely telated to rum
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I think we want a rum device to be able to be a good conference participant, but like nearly all conference participants, they only see a 2 party call and the bridge does the magic. I don’t think we need to require anything else. I think we want a bridge to be able to host a rum-compliant endpoint. Brian > On Aug 29, 2019, at 12:20 PM, Malloy, Jim <jmalloy@mitre.org> wrote: > > Do we intend for the RUE to accommodate multiple party calls (more than three)? I don't recall any language related to anything but a point to point call. For emergency calls, conferencing in the PSAP (three parties) may be appropriate. Are we looking for more than that? > > --Jim Malloy > > -----Original Message----- > From: Rum <rum-bounces@ietf.org <mailto:rum-bounces@ietf.org>> On Behalf Of Paul Kyzivat > Sent: Thursday, August 29, 2019 12:05 PM > To: rum@ietf.org <mailto:rum@ietf.org> > Subject: [EXT] Re: [Rum] Real-time text in WebRTC is discussed in mmusic - a topic closely telated to rum > > I question how well RTT text is suited to multiparty conferences. > > If you have messages on your screen from multiple parties, and many of them are updating in real-time, are you going to be able to perceive what is going on? > > And while you can have a column per person for two-party and maybe 3-party conversations, that doesn't scale up. With many parties, some typing may scroll off the screen before it is complete. > > Perhaps for conferences it is better to just use line-at-a-time chat. If necessary, I presume there could be gateways between RTT and chat. A RUE could have the capability to negotiate down from RTT to chat. > > Thanks, > Paul > > On 8/28/19 2:15 AM, Gunnar Hellström wrote: >> Den 2019-08-27 kl. 23:15, skrev Brian Rosen: >>> At least by centralizing the problem at a “mixer”, Alice and Bob will >>> see the same thing. >>> >>> You don’t have the problem in Instant Messaging, because you can’t >>> backspace or delete a sent message. Of course if multiple people are >>> typing simultaneously in such systems, message order will be >>> confusing in that instant. >> Right, it is a similar kind of problem that text appears in an >> unexpected order. There is also at least one instant messaging service >> that allows modification in already sent message. But I think it has >> limitations to only accept that in the last message sent. it is >> convenient anyway. >>> >>> Anyway, we need to specify the mixer for RTT so it receives each of >>> the RTT streams and produces a single composite stream for each >>> participant. >> >> Yes, right, and there is an effort in that direction in: >> >> http://www.realtimetext.org/sites/default/files/Files_and_Documents/Sp >> ecifications/multiparty-real-time-text-mixer-2011-04-30.pdf >> >> It is written for conference-unaware user devices. >> >> The goals are specified as follows: >> >> The procedures are intended to make best efforts to present a >> multi-party text conversation on a terminal that has no awareness of >> multi-party calls. There are some obvious drawbacks, and a terminal >> designed with multi-party awareness will be able to present >> multi-party call contents in a more flexible way. Only two parties at >> a time will be allowed to display added text in real-time, while the other parties’ >> produced text will need to be stored in the multi-party server for a >> moment awaiting a suitable occasion to be displayed. There are also >> some cases of erasure that will not be performed on the target text >> but only indicated in another way. Even with these drawbacks, the >> procedure provides an opportunity to display text from more than two >> parties in a smooth and readable way. >> >> ---------------------------------------------------------------------- >> ----------------------------- >> >> I see such mixer procedures as a fall-back for cases without >> conference awareness, but want to see support for conference-aware >> terminals, where text from more than two parties can be presented in >> real-time, and the end user or app can have influence over the presentation style - e.g. >> select between the multiple column view and the one-column-with-labels >> view. A mixer for that case would only need to assure that the >> receiver has the right kind of multi-party awareness and send RTT text >> with source information attached, and let the receiving terminal sort >> out the presentation. This is already possible with CSRC and CNAME >> when using RTP, but we lose that possibility natively when using the >> WebRTC data channel to transport RTT, and would need to specify a way >> to include the source also for that case. >> >> ------------------------------------------------------ >> >> By the way, what is your current view of how to transport RTT for RUM, >> now when you say that you will use WebRTC transports for media? >> >> Regards >> >> Gunnar >> >>> >>>> On Aug 27, 2019, at 4:43 PM, Gunnar Hellström >>>> <gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se> <mailto:gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se>>> wrote: >>>> >>>> Den 2019-08-27 kl. 21:48, skrev Brian Rosen: >>>>> The problem of conference 4103 RTT is high on my list of work I >>>>> need to get done. So, I’m motivated to help out. >>>> Thanks, great. >>>>> The basic problem is that we’re going to get very inconsistent UI >>>>> doing it that way, because of how systems will handle backspace of >>>>> one party that extends beyond responses from other parties: >>>> (well, for me the currently most basic problem is to have a reliable >>>> way to append received text to the already presented text of the >>>> right participant. And that is getting worse in WebRTC than it was >>>> in RFC 4103. But we will sort it out.) >>>>> >>>>> Alice: I waited for you >>>>> Bob: I didn’t see you >>>>> Alice: sorry >>>>> >>>>> And then Alice types 12 backspaces. >>>>> >>>>> What should happen? >>>> >>>> You are right that there are a number of ways to handle the RTT UI. >>>> And just as inconsistencies are common with a message oriented UI, >>>> where messages show up in a confusing order because two users >>>> completed messages in an unexpected time order, it is possible that >>>> RTT text gets displayed in a strange order after erasure and >>>> retyping. It is better for RTT than for message oriented >>>> presentation, and user get used to it in both cases. With the >>>> labelled style in one column you have in the example, I would >>>> recommend that first 5 backspaces erase "sorry", next backspace >>>> erases the line separator, and pulls down "I waited for you" to be >>>> shown last, as an uncompleted text. Then the next 6 backspaces erase >>>> so that only "I waited f" is displayed. When Alice adds text and end >>>> with a new line, the corrected sentence is allowed to flow up when >>>> new text is added from any participant. That causes a bit strange >>>> order, but it is just as manageable as when text in messaging >>>> applications appear in an unexpected order so that one message seems >>>> to be a respone on something totally else than what was intended. >>>> >>>> A sophisticated UI may mark text that is moved and modified. >>>> >>>> We want to keep sentences or at least phrases from each participant >>>> together in a readable unit. Already that causes a design decision >>>> on where to place the completed chunk of text once the user has >>>> completed it. The start of the chunk may be older than completed >>>> text from other participants which would motivate to move it up a >>>> bit in the presentation. But the end of it is at that moment the >>>> latest text to present. I think it is best to let the finished text >>>> be presented last on the display, but let others' newer text push >>>> everything up and be displayed last. >>>> >>>> >>>> T.140 has information on how to handle erasure: >>>> >>>> -------------------From T.140--------------------------- >>>> >>>> 8.2 Erase last character >>>> Purpose: Erase the last character sent from the display at the >>>> receiving end. >>>> Code: BS: 0008. >>>> Procedure: On the receiving end: Move the insertion point to the >>>> last character and erase it. >>>> Combined characters are erased as a unit, with one BS erasing the >>>> whole character even if it is combined from more than one component. >>>> Control sequences (like CR LF) are erased in one operation. >>>> NOTE – The same action shall be taken on the local display. >>>> >>>> ------------------------------------------------------------ >>>> >>>> /Gunnar >>>> >>>>> >>>>> Brian >>>>> >>>>>> On Aug 27, 2019, at 9:52 AM, Gunnar Hellström >>>>>> <gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se> <mailto:gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se>>> >>>>>> wrote: >>>>>> >>>>>> Hi, >>>>>> >>>>>> A topic is currently discussed in mmusic that is closely related >>>>>> to rum. it is WebRTC transport of real-time text. >>>>>> >>>>>> The draft is draft-holmberg-mmusic-t140-usage-data-channel . >>>>>> >>>>>> A good point to start reading could be: >>>>>> >>>>>> https://mailarchive.ietf.org/arch/browse/mmusic/?gbt=1&q=draft-hol <https://mailarchive.ietf.org/arch/browse/mmusic/?gbt=1&q=draft-hol> >>>>>> mberg-mmusic-t140-usage-data-channel >>>>>> >>>>>> Please check if the current state of the discussion suits rum! >>>>>> >>>>>> The only issue that seems to be remaining is how to transport RTT >>>>>> data to and from a conference server that combines all traffic per >>>>>> media in a meeting in one data stream. That is not very elegantly >>>>>> specified for RFC 4103 transport of RTT in RTP either, so we might >>>>>> want to do a rapid action together to solve the multi-party RTT >>>>>> MCU case in a general and consistent way. >>>>>> >>>>>> Regards >>>>>> >>>>>> Gunnar >>>>>> >>>>>> -- >>>>>> ----------------------------------------- >>>>>> Gunnar Hellström >>>>>> Omnitor >>>>>> gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se> >>>>>> +46 708 204 288 >>>>>> >>>>>> >>>> -- >>>> ----------------------------------------- >>>> Gunnar Hellström >>>> Omnitor >>>> gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se> <mailto:gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se>> >>>> +46 708 204 288 >>> >>> >> -- >> ----------------------------------------- >> Gunnar Hellström >> Omnitor >> gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se> >> +46 708 204 288 >> >> > > -- > Rum mailing list > Rum@ietf.org <mailto:Rum@ietf.org> > https://www.ietf.org/mailman/listinfo/rum <https://www.ietf.org/mailman/listinfo/rum> > -- > Rum mailing list > Rum@ietf.org <mailto:Rum@ietf.org> > https://www.ietf.org/mailman/listinfo/rum <https://www.ietf.org/mailman/listinfo/rum>
- [Rum] Real-time text in WebRTC is discussed in mm… Gunnar Hellström
- Re: [Rum] Real-time text in WebRTC is discussed i… Brian Rosen
- Re: [Rum] Real-time text in WebRTC is discussed i… Gunnar Hellström
- Re: [Rum] Real-time text in WebRTC is discussed i… Brian Rosen
- Re: [Rum] Real-time text in WebRTC is discussed i… Gunnar Hellström
- Re: [Rum] Real-time text in WebRTC is discussed i… Paul Kyzivat
- Re: [Rum] [EXT] Re: Real-time text in WebRTC is d… Malloy, Jim
- Re: [Rum] [EXT] Re: Real-time text in WebRTC is d… Paul Kyzivat
- Re: [Rum] Real-time text in WebRTC is discussed i… Brian Rosen
- Re: [Rum] [EXT] Re: Real-time text in WebRTC is d… Brian Rosen
- Re: [Rum] [EXT] Re: Real-time text in WebRTC is d… Paul Kyzivat
- Re: [Rum] [EXT] Re: Real-time text in WebRTC is d… Brian Rosen
- Re: [Rum] [EXT] Re: Real-time text in WebRTC is d… Paul Kyzivat
- Re: [Rum] [EXT] Re: Real-time text in WebRTC is d… Brian Rosen
- Re: [Rum] [EXT] Re: Real-time text in WebRTC is d… Malloy, Jim
- Re: [Rum] [EXT] Re: Real-time text in WebRTC is d… Gunnar Hellström
- Re: [Rum] [EXT] Re: Real-time text in WebRTC is d… John Martin
- Re: [Rum] [EXT] Re: Real-time text in WebRTC is d… Paul Kyzivat
- Re: [Rum] [EXT] Re: Real-time text in WebRTC is d… Gunnar Hellström
- Re: [Rum] [EXT] Re: Real-time text in WebRTC is d… Gunnar Hellström
- Re: [Rum] [EXT] Re: Real-time text in WebRTC is d… Brian Rosen
- Re: [Rum] [EXT] Re: Real-time text in WebRTC is d… Gunnar Hellström