Re: [dispatch] Request DISPATCH of RUM
Paul Kyzivat <pkyzivat@alum.mit.edu> Mon, 04 February 2019 20:05 UTC
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To: Gunnar Hellström <gunnar.hellstrom@omnitor.se>, Brian Rosen <br@brianrosen.net>
Cc: dispatch@ietf.org
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Date: Mon, 04 Feb 2019 15:05:42 -0500
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Subject: Re: [dispatch] Request DISPATCH of RUM
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On 2/4/19 2:32 PM, Gunnar Hellström wrote: > Thanks, I think all your answers are good. > About the WebRTC issue, some clarification in wording seems to be needed. > > About the addressing issue, I also think your response is good. > > You started that topic with "To my knowledge, there are no Video > Interpretation Services that provide interpreters for telephone calls > that do not use E.164s as the address." > That is right. My request for support of other addressing forms was > rather from the observation that an enormous amount of calls nowadays > are set up with other addressing forms than E.164 numbers. With such > calls lacking VRS support we are seeing a rapidly decreasing > accessibility level in the communication area and a need for improvements. I have been thinking for a long time that it would be good to define a new interface for VRS provision, that only provides the interpretation services, not the telephony origination/termination service. Anybody who wanted to provide audio/video connectivity (regardless of the form of addressing) could then "bridge" in the VRS service when it is needed. Thanks, Paul > Your conclusion is sufficient. Good. > > What wording do you propose for the emergency service support? > > Regards > > Gunnar > > > > Gunnar Hellström > gunnar.hellstrom@omnitor.se > +46 708 204 288 > > > -------- Originalmeddelande -------- > Från: Brian Rosen <br@brianrosen.net> > Datum: 2019-02-04 16:46 (GMT+01:00) > Till: Gunnar Hellström <gunnar.hellstrom@omnitor.se> > Kopia: Paul Kyzivat <pkyzivat@alum.mit.edu>, dispatch@ietf.org > Rubrik: Re: [dispatch] Request DISPATCH of RUM > > Inline > >> On Feb 3, 2019, at 5:12 PM, Gunnar Hellström >> <gunnar.hellstrom@omnitor.se <mailto:gunnar.hellstrom@omnitor.se>> wrote: >> >> Brian, >> >> A. About technology and WebRTC >> At the moment it seems most important to get the scope clear. And for >> that, the issue if this is about a WebRTC RUM interface or a native >> SIP RUM interface or both. Another implementation environment could be >> 3GPP IMS RUM. Is that also in scope? Is it really realistic to write a >> RUM spec that fulfilles the goal to allow the users to move a device >> between providers regardless if they use native SIP or WebRTC (or even >> IMS )? ( if that is what is meant by the section about WebRTC). >> Maybe a list of supported call cases is needed to make us understand >> and accept an intended scope. > I proposed scope that defined a SIP interface where it would be possible > to build a WebRTC implementation that ultimately provided the client > side of that interface. That means that the WebRTC server created the > client side SIP interface towards the provider. > > I would like it to be possible to do that, but I don’t think that is the > design center. I do think re-using the media specs from WebRTC is the > right set of choices to make for this profile, independent of whether > the user has a WebRTC client or a native SIP client. >> >> B. About Paul's call cases: >> The statement: "For a p2p call with another WebRTC user a gateway >> won't be >> needed." is in conflict with a statement by the end of the charter >> saying: "Although the interface between providers also >> requires standardization to enable multi-provider >> point-to-point calls, that interface has already been defined in a >> SIP Forum document and is thus out of scope for RUM." >> If the users are registered by WebRTC to different providers, and want >> to set up a p2p call, then the charter says that the providers would >> use the provider-to-provider interface specified by SIP Forum. But >> that is a native SIP interface and requires gateways from any WebRTC >> RUM. So it must be decided, are we specifying for WebRTC, and do we >> support multi-provider WebRTC calls and do we assume the >> provider-to-provider interface to be the SIP FORUM spec? > We are specifying a native SIP interface. It may be possible to > interwork some other client to that interface. > >> >> C. Number calling and other forms of calling >> I think the scope regarding addressing should be clarified in the >> charter. Only one sentence mentions addressing: "The hearing person >> can also reach D-HOH-SI individuals by in the same manner as calling >> any other phone number." >> Nowadays a large portion of calls are made with other addressing than >> phone numbers, and we must prepare to enable VRS use also in such >> calls. It would be best if the RUM spec is agnostic to what kind of >> addressing is used. That may be possible, because the number handling >> and conversion between numbers and SIP URI is handled elsewhere. I >> think anyway that the scope of addressing variants is worth a >> paragraph in the charter. > > To my knowledge, there are no Video Interpretation Services that provide > interpreters for telephone calls that do not use E.164s as the address. > While the actual address of the SIP device is a SIP URI, and we could > say that we support that, the interwork for dialing needs to be > explicit, so that whenever telephone numbers are used, the device > interface uses them consistently. I’ll propose language that allows > SIP URIs for the device address. >> >> Proposal: Delete the mentioning of number, and include the following >> paragraph: "Addressing of participants in the calls are supposed to be >> based on general addressing conventions in SIP. Any conversion needed >> between this form and other addressing forms (e.g. phone numbers) >> required for completion of the calls are assumed to take place in >> other parts of the networks." >> >> D: Do you also want to discuss minor edits in the charter? If so, here >> are a couple of hints: >> >> D.1. This phrase: "The deaf, hard- of- hearing or speech-impaired >> person (D-HOH-SI) uses a SIP-based video phone to connect with an >> interpreter, and the interpreter places a phone call on the PSTN to >> the hearing person." should be changed. PSTN is fading away. The call >> might also be automated. Maybe it is sufficient to modify it to: "The >> deaf, hard- of- hearing or speech-impaired person (D-HOH-SI) uses a >> SIP-based video phone to connect with a provider, and the provider >> conveyes the call to the hearing person and includes an interpreter in >> the call." >> >> (this wording would also support the implementation when the >> interpreter actually is involved in placing the leg of the call to the >> hearing person) > I will adjust the wording >> >> D.2. There are a number of places where it is obvious that the wording >> is copied from a document with changemarks, and both the original and >> the modification happened to be included in the charter. The first >> example of this kind is on line 4, where "VRSwhich" has apparently >> history in a modification from "which" to "VRS" and should be just >> "VRS". I can help to sort them out if you want. > Yes, sorry, I didn’t notice that. I will fix it in the next version > >> >> >> Regards >> >> Gunnar >> >> >> >> >> >> >> -------- Originalmeddelande -------- >> Från: Paul Kyzivat <pkyzivat@alum.mit.edu <mailto:pkyzivat@alum.mit.edu>> >> Datum: 2019-02-03 02:29 (GMT+01:00) >> Till:dispatch@ietf.org <mailto:dispatch@ietf.org> >> Rubrik: Re: [dispatch] Request DISPATCH of RUM >> >> On 2/2/19 6:18 PM, Brian Rosen wrote: >> > Hmmm, yeah, we really need to keep RTT in RTP. >> > We’ll have to use a gateway for that >> >> In the case of a call with an interpreter the provider can provide the >> gateway, or provide the interpreter with native text over data channel >> support. For a p2p call with another WebRTC user a gateway won't be >> needed. The only special case is a p2p call with one WebRTC endpoint and >> one native SIP endpoint. >> >> Thanks, >> Paul >> >> > Yes emergency calls are in scope. >> > >> > Brian >> > >> > On Sat, Feb 2, 2019 at 5:48 PM Gunnar Hellström >> > <gunnar.hellstrom@omnitor.se >> <mailto:gunnar.hellstrom@omnitor.se><mailto:gunnar.hellstrom@omnitor.se>> >> wrote: >> > >> > So far, in tradiitional SIP based VRS, real-time text has been >> > implemented with RTP, while in WebRTC it is supposed to use the data >> > channel. How would you specify that interop without a media gateway? >> > >> > >> > Another issue: should possibility to interop with emergency services >> > be mentioned in the charter? I assume that such calls need to pass >> > through the provider, and can be gatewayed ther, but there is a >> > desire that all media is conveyed between the emergency service and >> > the RUM and there might therefore be a need to consider this >> > requirement when specifying the RUM interface. >> > >> > >> > Regards >> > Gunnar >> > >> > >> > >> > -------- Originalmeddelande -------- >> > Från: Christer Holmberg <christer.holmberg@ericsson.com <mailto:christer.holmberg@ericsson.com> >> > <mailto:christer.holmberg@ericsson.com>> >> > Datum: 2019-02-02 00:10 (GMT+01:00) >> > Till: Brian Rosen <br@brianrosen.net <mailto:br@brianrosen.net><mailto:br@brianrosen.net>> >> > Kopia: DISPATCH list <dispatch@ietf.org <mailto:dispatch@ietf.org><mailto:dispatch@ietf.org>> >> > Rubrik: Re: [dispatch] Request DISPATCH of RUM >> > >> > >> > Hi, >> > >> > >Yes, that’s the idea. I will work on some wording. I don’t want >> > the charter to have a >> > >list of such features. >> > >> > You could say that the profile will mandate all features needed in >> > order to interoperate with WebRTC without having to use a media >> > gateway, or something like that. >> > >> > Regards, >> > >> > Christer >> > >> > >> > >> > Brian >> > >> > On Fri, Feb 1, 2019 at 5:30 PM Christer Holmberg >> > <christer.holmberg@ericsson.com <mailto:christer.holmberg@ericsson.com> >> > <mailto:christer.holmberg@ericsson.com>> wrote: >> > >> > Hi, >> > >> > >Can you suggest a wording change? >> > >> > Not at the moment, I first want to understand exactly what the >> > scope and purpose is. >> > >> > >It now says "A WebRTC- based RUM could create a SIP interface >> > (using, e.g., SIP over >> > > WebSockets) towards the provider that conformed to the >> > document RUM will produce. Such >an implementation should be >> > possible, ideally without requiring a media gateway.” That >> > >seems to me to be clear that the wg won’t do any work beyond >> > making sure it’s possible to >create a WebRTC based RUM without >> > a media gateway. >> > >> > If the WG is going to "make sure" that it works without a media >> > gateway, does that mean that you would also mandate e.g., ICE, >> > continuous consent, DTLS, and whatever other media level >> > features might be mandated to support by WebRTC? If so, I think >> > it needs to be very clear. >> > >> > Regards, >> > >> > Christer >> > >> > >> > Brian >> > >> > >> > >> >> On Feb 1, 2019, at 4:57 PM, Christer Holmberg >> >> <christer.holmberg@ericsson.com <mailto:christer.holmberg@ericsson.com> >> >> <mailto:christer.holmberg@ericsson.com>> wrote: >> >> >> >> >> >> Hi, >> >> >> >> >We want to make sure the mechanisms interact reasonably. We >> >> suspect this is mostly codec >> >> >choices, etc. >> >> >> >> Then you should say that a goal is interoperability with >> >> WebRTC when it comes to codecs etc. >> >> >> >> The way I read the text now is that the WG is about writing >> >> WebRTC SIP clients, which I assume is outside the scope😊 >> >> >> >> Regards, >> >> >> >> Christer >> >> >> >> >> >>> On Feb 1, 2019, at 4:11 PM, Christer Holmberg >> >>> <christer.holmberg@ericsson.com <mailto:christer.holmberg@ericsson.com> >> >>> <mailto:christer..holmberg@ericsson.com>> wrote: >> >>> >> >>> Hi, >> >>> >> >>> If the purpose of the WG is to define a SIP profile, I assume >> >>> it does not matter if the SIP UAs are implemented using >> >>> WebRTC or something else, so why does the charter need to >> >>> talk about WebRTC? >> >>> >> >>> If you want to look at some of the specific network >> >>> technologies used by WebRTC, e.g., the data channel, I think >> >>> should talk about that instead. >> >>> >> >>> Regards, >> >>> >> >>> Christer >> >>> >> >>> >> >>> ------------------------------------------------------------------------ >> >>> *From:*dispatch <dispatch-bounces@ietf.org <mailto:dispatch-bounces@ietf.org> >> >>> <mailto:dispatch-bounces@ietf.org>> on behalf of Brian Rosen >> >>> <br@brianrosen.net <mailto:br@brianrosen.net><mailto:br@brianrosen.net>> >> >>> *Sent:*Friday, February 1, 2019 10:50:53 PM >> >>> *To:*DISPATCH list >> >>> *Subject:*[dispatch] Request DISPATCH of RUM >> >>> I would like dispatch to consider spinning up a mini-work >> >>> group to create a sip device profile for use with Video Relay >> >>> Services. >> >>> >> >>> >> >>> The proposed charter is: >> >>> >> >>> Relay User Machine (rum) Working Group Proposed Charter >> >>> ART Area >> >>> >> >>> Many current instances of Video Relay Service (VRS), >> >>> (sometimes called Video Interpretation Service.), use the >> >>> Session Initiation Protocol (SIP) and other IETF multimedia >> >>> protocols. VRSwhich is used bya service that deaf and hard- >> >>> of- hearing persons and person with speech impairments use to >> >>> communicate with hearing persons. The deaf, hard- of- >> >>> hearing or speech-impaired person (D-HOH-SI) uses a SIP- >> >>> based video phone to connect with an interpreter, and the >> >>> interpreter places a phone call on the PSTN to the hearing >> >>> person. The hearing person can also reach D-HOH-SI >> >>> individuals by in the same manner as calling any other phone >> >>> number. The D-HOH-SI person uses sign language and possibly >> >>> real-time text with the interpreter and the interpreter uses >> >>> spoken language with the hearing person, providing on- line, >> >>> real- time, two- way communication. VRS services are often >> >>> government- supported. In some countries, private companies >> >>> provide the interpreters, and compete with one another. >> >>> Often these companies use proprietary implementations for >> >>> user devices as a means of vendor lock in. >> >>> >> >>> Having a standard interface between the end- user device and >> >>> the VRS provider allows vendors and open-source developers to >> >>> build devices that work with multiple service providers; >> >>> devices can also be retained when changing providers. In >> >>> this instance, “device” could be a purpose-built videophone >> >>> or could be downloadable software on a general purpose >> >>> computing platform or mobile phone. The SIP protocol is >> >>> complex enough that some form of profiling is needed to >> >>> achieve interoperability between devices and providers. To >> >>> ensure interoperability of the key features of this service, >> >>> certain aspects (e.g., codecs, media transport, and SIP >> >>> features) must be specified as mandatory-to-implement for >> >>> SIP-based VRS devices. These specified features effectively >> >>> form a profile for SIP and the media it negotiates. >> >>> >> >>> This working group will produce a single document: a profile >> >>> of SIP and media features for use with video relay services >> >>> (which includes video, real time text, and audio), and other >> >>> similar interpretation services that require multimedia. It >> >>> will reference the IETF’s current thinking on multimedia >> >>> communicationsuch devices, including references to work >> >>> beyond SIP (e.g., WebRTC and SLIM). No protocol changes are >> >>> anticipated by this work. >> >>> >> >>> While WebRTC could be used to implement a RUM, the group’s >> >>> work will focusis on the device-to-provider interface. A >> >>> WebRTC- based RUM couldwould create a SIP interface (using, >> >>> e.g., SIP over WebSockets) towards the provider that >> >>> conformed to the document RUMrum will produce. Such an >> >>> implementation should be possible, ideally without requiring >> >>> a media gateway. >> >>> >> >>> The scope of the work includes mechanisms to provision the >> >>> user’s device with common features such as speed dial lists, >> >>> provider to contact, videomail service interface point and >> >>> similar items. These features allow users to more easily >> >>> switch providers temporarily (a feature known as “dial >> >>> around”) or permanently, while retaining their data. >> >>> >> >>> Devices used in VRS can be used to place point- to- point >> >>> calls, i.e., where both communicating parties use sign >> >>> language. When used for point-to-point calling where the >> >>> participants are not served by the same VRS provider, or when >> >>> one provider provides the originating multimedia transport >> >>> environment, but another provides the interpreter >> >>> (“dial-around call”), the call traverses two providers. Both >> >>> of these uses impose additional requirements on a RUMrum >> >>> device and are in scope for this work. >> >>> >> >>> Although the interface between providers also requires >> >>> standardization to enable multi-provider point-to-point >> >>> calls, that interface has already been defined in a SIP >> >>> Forum document and is thus out of scope for RUM. >> >>> _______________________________________________ >> >>> dispatch mailing list >> >>> dispatch@ietf.org <mailto:dispatch@ietf.org><mailto:dispatch@ietf.org> >> >>> https://www.ietf.org/mailman/listinfo/dispatch >> > >> > >> > _______________________________________________ >> > dispatch mailing list >> >dispatch@ietf.org <mailto:dispatch@ietf.org> >> >https://www.ietf.org/mailman/listinfo/dispatch >> > >> >> _______________________________________________ >> dispatch mailing list >> dispatch@ietf.org <mailto:dispatch@ietf.org> >> https://www.ietf.org/mailman/listinfo/dispatch >> _______________________________________________ >> dispatch mailing list >> dispatch@ietf.org <mailto:dispatch@ietf.org> >> https://www.ietf.org/mailman/listinfo/dispatch >
- Re: [dispatch] Request DISPATCH of RUM Christer Holmberg
- [dispatch] Request DISPATCH of RUM Brian Rosen
- Re: [dispatch] Request DISPATCH of RUM Brian Rosen
- Re: [dispatch] Request DISPATCH of RUM Christer Holmberg
- Re: [dispatch] Request DISPATCH of RUM Richard Shockey
- Re: [dispatch] Request DISPATCH of RUM Brian Rosen
- Re: [dispatch] Request DISPATCH of RUM Brian Rosen
- Re: [dispatch] Request DISPATCH of RUM Christer Holmberg
- Re: [dispatch] Request DISPATCH of RUM Brian Rosen
- Re: [dispatch] Request DISPATCH of RUM Christer Holmberg
- Re: [dispatch] Request DISPATCH of RUM Gunnar Hellström
- Re: [dispatch] Request DISPATCH of RUM Brian Rosen
- Re: [dispatch] Request DISPATCH of RUM Paul Kyzivat
- Re: [dispatch] Request DISPATCH of RUM Gunnar Hellström
- Re: [dispatch] Request DISPATCH of RUM Brian Rosen
- Re: [dispatch] Request DISPATCH of RUM Brian Rosen
- Re: [dispatch] Request DISPATCH of RUM Brian Rosen
- Re: [dispatch] Request DISPATCH of RUM Gunnar Hellström
- Re: [dispatch] Request DISPATCH of RUM Paul Kyzivat
- Re: [dispatch] Request DISPATCH of RUM Brian Rosen
- Re: [dispatch] Request DISPATCH of RUM Paul Kyzivat
- Re: [dispatch] Request DISPATCH of RUM Brian Rosen
- Re: [dispatch] Request DISPATCH of RUM Gunnar Hellström
- Re: [dispatch] Request DISPATCH of RUM Gunnar Hellström
- Re: [dispatch] Request DISPATCH of RUM Brian Rosen
- [dispatch] Fwd: Request DISPATCH of RUM Mary Barnes
- Re: [dispatch] Fwd: Request DISPATCH of RUM Eric Burger
- Re: [dispatch] Fwd: Request DISPATCH of RUM Paul Kyzivat
- Re: [dispatch] Fwd: Request DISPATCH of RUM Roni Even (A)
- [dispatch] DISPATCH of JCS Anders Rundgren
- Re: [dispatch] DISPATCH of JCS Dale R. Worley
- Re: [dispatch] DISPATCH of JCS Anders Rundgren
- Re: [dispatch] DISPATCH of JCS Dale R. Worley
- Re: [dispatch] Fwd: Request DISPATCH of RUM Adam Roach
- Re: [dispatch] Request DISPATCH of RUM Brian Rosen