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Re: [Rum] Real-time text in WebRTC is discussed in mmusic - a topic closely telated to rum

Brian Rosen <br@brianrosen.net> Tue, 27 August 2019 19:49 UTC

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From: Brian Rosen <br@brianrosen.net>
In-Reply-To: <9a14addd-9a1c-6130-3880-a814be717323@omnitor.se>
Date: Tue, 27 Aug 2019 15:48:59 -0400
Cc: rum@ietf.org
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To: Gunnar Hellström <gunnar.hellstrom@omnitor.se>
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Subject: Re: [Rum] Real-time text in WebRTC is discussed in mmusic - a topic closely telated to rum
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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.
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:

Alice: I waited for you
Bob: I didn’t see you
Alice: sorry

And then Alice types 12 backspaces.

What should happen?

Brian

> On Aug 27, 2019, at 9:52 AM, Gunnar Hellström <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-holmberg-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
> +46 708 204 288
> 
>

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