- From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
- Date: Tue, 19 Jun 2018 19:44:48 +0200
- To: Lennart Grahl <lennart.grahl@gmail.com>
- Cc: public-webrtc@w3.org, Gunnar Hellström <gunnar.hellstrom@omnitor.se>
> 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