W3C home > Mailing lists > Public > public-webrtc@w3.org > June 2018

Re: RTT implementation = Real-Time Text implementation

From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
Date: Tue, 19 Jun 2018 19:44:48 +0200
Cc: public-webrtc@w3.org, Gunnar Hellström <gunnar.hellstrom@omnitor.se>
Message-Id: <EE31094D-0758-4DC8-9AC7-765416A54335@lurchi.franken.de>
To: Lennart Grahl <lennart.grahl@gmail.com>
> On 19. Jun 2018, at 18:36, Lennart Grahl <lennart.grahl@gmail.com> wrote:
> 
> This may benefit from having more knobs to control SCTP-specific timers
> such as the retransmission interval. It might actually be the first use
> case I've encountered that could make some use out of `maxRetransmits:
> <some carefully chosen value above 0>` along with a specific
> retransmission interval.
Are there timing/performance requirements for RTT?

Best regards
Michael
> 
> Cheers
> Lennart
> 
> 
> On 19.06.2018 11:27, Gunnar Hellström wrote:
>> This discussion started with RTT meaning "Real-Time Text" which is
>> time-sampled text used in conversational sessions, often together with
>> audio and video.
>> 
>> The time sampling is traditionally done in about 300 ms samples in order
>> to not cause a lot of load. So any new text entered within 300 ms is
>> transmitted in a chunk, regardless if the user has indicated any end of
>> message or not. This way of sending text implies a much better sense of
>> being connected between users in intensive conversations than the
>> transmission in completed messages does.
>> 
>> Today, when bandwidth and processing is less expensive, it could be
>> worth while decreasing the sample time, so that latencies close to what
>> is used for audio and video in conversational sessions is achieved.
>> 
>> It should be possible to use WebRTC data channel for Real-Time Text.
>> 
>> The synchronism requirements versus video and audio are mild. Users
>> barely notice an asynchronism of 500 ms or 1 second. Some applications,.
>> like speech-to-text transmit in complete words, and that is also allowed.
>> 
>> So, I do not know if the implementation needs to build on the
>> synchronized media use case. It may just be sufficient to use regular
>> WebRTC data channels with suitable characteristics. I like the
>> simplicity of the "reliable" transfer in data channels, but not the risk
>> for long delays in case of transmission problems.
>> 
>> Since the topic is mentioned in the initial functionality goals for Data
>> Channels, but not mentioned in RFC 7478, I suggest that it is included
>> in the NV discussions.
>> 
>> /Gunnar
>> 
>> 
>> 
>> Den 2018-06-19 kl. 10:07, skrev Harald Alvestrand:
>>> 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.
>>>>>> 
>>>> 
>> 
>> 
> 
Received on Tuesday, 19 June 2018 17:45:16 UTC

This archive was generated by hypermail 2.4.0 : Friday, 17 January 2020 19:18:42 UTC