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Re: RTT implementation

From: Harald Alvestrand <harald@alvestrand.no>
Date: Tue, 19 Jun 2018 10:07:16 +0200
To: public-webrtc@w3.org
Message-ID: <f6c24a05-8e29-0d50-d1d6-d529471c7dc1@alvestrand.no>
Existing RTT measurements:

https://w3c.github.io/webrtc-stats/webrtc-stats.html#dom-rtcremoteinboundrtpstreamstats-roundtriptime

https://w3c.github.io/webrtc-stats/webrtc-stats.html#dom-rtcicecandidatepairstats-totalroundtriptime


On 06/19/2018 09:06 AM, Gunnar Hellström wrote:
> 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.
>>>
>>
>
>

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Received on Tuesday, 19 June 2018 08:07:45 UTC

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