Re: [dispatch] Request DISPATCH of RUM

Gunnar Hellström <gunnar.hellstrom@omnitor.se> Sat, 02 February 2019 22:49 UTC

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From: =?ISO-8859-1?Q?Gunnar_Hellstr=F6m?= <gunnar.hellstrom@omnitor.se>
To: Christer Holmberg <christer.holmberg@ericsson.com>, Brian Rosen <br@brianrosen.net>
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Subject: Re: [dispatch] Request DISPATCH of RUM
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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.RegardsGunnar

-------- Originalmeddelande --------Från: Christer Holmberg <christer.holmberg@ericsson.com> Datum: 2019-02-02  00:10  (GMT+01:00) Till: Brian Rosen <br@brianrosen.net> Kopia: DISPATCH list <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> 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> 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> 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>
 on behalf of Brian Rosen <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.
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