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Re: RTT implementation

From: Gunnar Hellström <gunnar.hellstrom@omnitor.se>
Date: Tue, 19 Jun 2018 09:06:03 +0200
To: public-webrtc@w3.org
Message-ID: <8fa3efe1-9606-3e24-ded6-6d3b9df951b5@omnitor.se>
Den 2018-06-19 kl. 08:46, skrev Bernard Aboba:

> In practice, the requirement for "synchronized data" can be supported by allowing applications to fill in the payload format defined in RFC 4103.
>
> This enables RTT to be implemented in Javascript on top of an "RTP data channel" transport, utilizing the existing RTCDataChannel interface.
>
> So in practice the need for RTT support can be included in a "synchronized data" requirement, if properly implemented.
Yes, it can be specified with current mechanisms, it is just a matter of 
selecting some properties and values and getting it specified. A 
standard is needed so that gateways and bridges can be developed 
separately from user agents, and so that, as you say, it all gets 
"properly implemented". So far, the latency requirements have been 
slightly lower than for audio and video in conversational sessions, when 
the user is typing the text, but now, with automatic speech to text 
becoming useful, the requirement for short delays is becoming more strict .

/Gunnar
>
>
> ________________________________________
> From: Peter Thatcher [pthatcher@google.com]
> Sent: Monday, June 18, 2018 10:49 PM
> To: Gunnar Hellström
> Cc: public-webrtc@w3.org
> Subject: Re: WebRTC NV Use Cases
>
> Thanks, I added that as a new requirement to the conferencing use case.
>
> On Mon, Jun 18, 2018 at 11:18 PM Gunnar Hellström <gunnar.hellstrom@omnitor.se<mailto:gunnar.hellstrom@omnitor.se>> wrote:
> I suggest to include real-time text (= text transmitted in the same rate
> as it is created so that it can be used for real conversational
> purposes) in the NV work.
>
> It is not included in RFC 7478, but included a U-C 5 in section 3.2 of
> https://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-13<https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ftools.ietf.org%2Fhtml%2Fdraft-ietf-rtcweb-data-channel-13&data=02%7C01%7CBernard.Aboba%40microsoft.com%7C4ecd480c191a456ac73d08d5d5a89c6f%7C72f988bf86f141af91ab2d7cd011db47%7C1%7C0%7C636649842519679581&sdata=fEZV7O6vIb1m3bi6mIBmi%2Bbf6PeJCtKx3Jb3WeFjWbA%3D&reserved=0>
>
>
> It could possibly be done by continuing the work started in
>
> https://datatracker.ietf.org/doc/draft-schwarz-mmusic-t140-usage-data-channel/<https://na01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fdatatracker.ietf.org%2Fdoc%2Fdraft-schwarz-mmusic-t140-usage-data-channel%2F&data=02%7C01%7CBernard.Aboba%40microsoft.com%7C4ecd480c191a456ac73d08d5d5a89c6f%7C72f988bf86f141af91ab2d7cd011db47%7C1%7C0%7C636649842519689589&sdata=KXNSeVQPxSLMa0%2FmzSQRio1W2p7Wgmn2oet%2FAoJTHjA%3D&reserved=0>
>
> Use cases are e.g.
>
> 1. conversational two-party sessions with video, audio and real-time text.
>
> 2. conversational multi-party sessions with video, audio and real-time text.
>
> 3. sessions with automatic speech - to - real-time text conversion in
> one or both directions.
>
> 4. interworking WebRTC with audio, video and real-time text and legacy
> SIP with audio, video and real-time text.
>
> /Gunnar
>
>
> Den 2018-05-09 kl. 21:29, skrev Bernard Aboba:
>> On June 19-20 the WebRTC WG will be holding a face-to-face meeting in Stockholm, which will focus largely on WebRTC NV.
>>
>> Early on in the discussion, we would like to have a discussion of the use cases that WebRTC NV will address.
>>
>> Since the IETF has already published RFC 7478, we are largely interested in use cases that are either beyond those articulated in RFC 7478, or use cases in the document that somehow can be done better with WebRTC NV than they could with WebRTC 1.0.
>>
>> As with any successful effort, we are looking for volunteers to develop a presentation for the F2F, and perhaps even a document.
>>
>
Received on Tuesday, 19 June 2018 07:06:40 UTC

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