rum                                                             B. Rosen
Internet-Draft                                           11 October 2021
Intended status: Standards Track
Expires: 14 April 2022


           Interoperability Profile for Relay User Equipment
                         draft-ietf-rum-rue-09

Abstract

   Video Relay Service (VRS) is a term used to describe a method by
   which a hearing person can communicate with a deaf, hard of hearing
   or hearing impaired user using an interpreter ("Communications
   Assistant") connected via a videophone to the deaf/HoH user and an
   audio telephone call to the hearing user.  The CA interprets using
   sign language on the videophone link and voice on the telephone link.
   Often the interpreters may be employed by a company or agency termed
   a "provider" in this document.  The provider also provides a video
   service that allows users to connect video devices to their service,
   and subsequently to CAs and other deaf/HoH users.  It is desirable
   that the videophones used by the deaf, hard of hearing or hearing
   impaired user conform to a standard so that any device may be used
   with any provider and that direct video calls between deaf, hard of
   hearing or hearing impaired users work.  This document describes the
   interface between a videophone and a provider.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
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   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on 14 April 2022.

Copyright Notice

   Copyright (c) 2021 IETF Trust and the persons identified as the
   document authors.  All rights reserved.



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   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents (https://trustee.ietf.org/
   license-info) in effect on the date of publication of this document.
   Please review these documents carefully, as they describe your rights
   and restrictions with respect to this document.  Code Components
   extracted from this document must include Simplified BSD License text
   as described in Section 4.e of the Trust Legal Provisions and are
   provided without warranty as described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Requirements Language . . . . . . . . . . . . . . . . . . . .   6
   4.  General Requirements  . . . . . . . . . . . . . . . . . . . .   6
   5.  SIP Signaling . . . . . . . . . . . . . . . . . . . . . . . .   6
     5.1.  Registration  . . . . . . . . . . . . . . . . . . . . . .   8
     5.2.  Session Establishment . . . . . . . . . . . . . . . . . .   9
       5.2.1.  Normal Call Origination . . . . . . . . . . . . . . .   9
       5.2.2.  Dial-Around Origination . . . . . . . . . . . . . . .  10
       5.2.3.  RUE Contact Information . . . . . . . . . . . . . . .  11
       5.2.4.  Incoming Calls  . . . . . . . . . . . . . . . . . . .  11
       5.2.5.  Emergency Calls . . . . . . . . . . . . . . . . . . .  12
     5.3.  Mid Call Signaling  . . . . . . . . . . . . . . . . . . .  12
     5.4.  URI Representation of Phone Numbers . . . . . . . . . . .  13
     5.5.  Transport . . . . . . . . . . . . . . . . . . . . . . . .  13
   6.  Media . . . . . . . . . . . . . . . . . . . . . . . . . . . .  13
     6.1.  SRTP and SRTCP  . . . . . . . . . . . . . . . . . . . . .  14
     6.2.  Text-Based Communication  . . . . . . . . . . . . . . . .  14
     6.3.  Video . . . . . . . . . . . . . . . . . . . . . . . . . .  14
     6.4.  Audio . . . . . . . . . . . . . . . . . . . . . . . . . .  14
     6.5.  DTMF Digits . . . . . . . . . . . . . . . . . . . . . . .  15
     6.6.  Session Description Protocol  . . . . . . . . . . . . . .  15
     6.7.  Privacy . . . . . . . . . . . . . . . . . . . . . . . . .  15
     6.8.  Negative Acknowledgment, Packet Loss Indicator, and Full
           Intraframe Request Features . . . . . . . . . . . . . . .  15
   7.  Contacts  . . . . . . . . . . . . . . . . . . . . . . . . . .  15
     7.1.  CardDAV Login and Synchronization . . . . . . . . . . . .  15
     7.2.  Contacts Import/Export Service  . . . . . . . . . . . . .  16
   8.  Video Mail  . . . . . . . . . . . . . . . . . . . . . . . . .  16
   9.  Provisioning and Provider Selection . . . . . . . . . . . . .  17
     9.1.  RUE Provider Selection  . . . . . . . . . . . . . . . . .  17
     9.2.  RUE Configuration Service . . . . . . . . . . . . . . . .  19
       9.2.1.  Provider Configuration  . . . . . . . . . . . . . . .  20
       9.2.2.  RUE Configuration . . . . . . . . . . . . . . . . . .  21
       9.2.3.  Examples  . . . . . . . . . . . . . . . . . . . . . .  22
       9.2.4.  Using the Provider Selection and RUE Configuration
               Services Together . . . . . . . . . . . . . . . . . .  23



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     9.3.  OpenAPI Interface Descriptions  . . . . . . . . . . . . .  23
   10. IANA Considerations . . . . . . . . . . . . . . . . . . . . .  28
     10.1.  RUE Provider List Registry . . . . . . . . . . . . . . .  29
     10.2.  Registration of rue-owner Value of the purpose
            Parameter  . . . . . . . . . . . . . . . . . . . . . . .  29
   11. Security Considerations . . . . . . . . . . . . . . . . . . .  29
   12. Normative References  . . . . . . . . . . . . . . . . . . . .  29
   13. Informative References  . . . . . . . . . . . . . . . . . . .  35
   Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .  36
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  36

1.  Introduction

   Video Relay Service (VRS) is a form of Telecommunications Relay
   Service (TRS) that enables persons with hearing disabilities who use
   sign language, such as American Sign Language (ASL), to communicate
   with voice telephone users through video equipment.  These services
   also enable communication between such individuals directly in
   suitable modalities, including any combination of sign language via
   video, real-time text (RTT), and speech.

   This Interoperability Profile for Relay User Equipment (RUE) is a
   profile of the Session Initiation Protocol (SIP) and related media
   protocols that enables end-user equipment registration and calling
   for VRS calls.  It specifies the minimal set of call flows, Internet
   Engineering Task Force (IETF) and ITU-T standards that must be
   supported, provides guidance where the standards leave multiple
   implementation options, and specifies minimal and extended
   capabilities for RUE calls.

   Both deaf/HoH to provider (interpreted) and direct deaf/HoH to deaf/
   HoH calls are supported on this interface.  While there are some
   accommodations in this document to maximize backwards compatibility
   with other devices and services that are used to provide VRS service,
   backwards compatibility is not a requirement, and some interwork may
   be required to allow direct video calls to older devices.  This
   document only describes the interface between the device and the
   provider, and not any other interface the provider may have.

2.  Terminology

   Communication Assistant (CA): A sign-language interpreter working for
   a VRS provider, providing functionally equivalent phone service.

   Communication modality (modality): A specific form of communication
   that may be employed by two users, e.g., English voice, Spanish
   voice, American Sign Language, English lip-reading, or French real-
   time-text.  Here, one communication modality is assumed to encompass



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   both the language and the way that language is exchanged.  For
   example, English voice and French voice are two different
   communication modalities.

   Default video relay service: The video relay service operated by a
   subscriber's default VRS provider.

   Default video relay service provider (default provider): The VRS
   provider that registers, and assigns a telephone number to a specific
   subscriber, and by default provides the VRS for incoming voice calls
   to the user.  The default provider also by default provides VRS for
   outgoing relay calls.  The user can have more than one telephone
   number and each has a default provider.

   Outbound Dial-around call: A relay call where the subscriber
   specifies the use of a VRS provider other than the default VRS
   provider.  This can be accomplished by the user dialing a "front-
   door" number for a VRS provider and signing or texting a phone number
   to call ("two-stage").  Alternatively, this can be accomplished by
   the user's RUE software instructing the server of its default VRS
   provider to automatically route the call through the alternate
   provider to the desired public switched telephone network (PSTN)
   directory number ("one-stage").  Dial-around is per-call -- for any
   call, a user can use the default VRS provider or any dial-around VRS
   provider.

   Full Intra Request (FIR): A request to a video media sender,
   requiring that media sender to send a Decoder Refresh Point at the
   earliest opportunity.  FIR is sometimes known as "instantaneous
   decoder refresh request", "video fast update request", or "fast
   update request".

   Point-to-Point Call (P2P Call): A call between two RUEs, without
   including a CA.

   Relay call: A call that allows persons with hearing or speech
   disabilities to use a RUE to talk to users of traditional voice
   services with the aid of a communication assistant (CA) to relay the
   communication.












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   Relay service (RS): A service that allow a registered subscriber to
   use a RUE to make and receive relay calls, point-to-point calls, and
   relay-to-relay calls.  The functions provided by the relay service
   include the provision of media links supporting the communication
   modalities used by the caller and callee, and user registration and
   validation, authentication, authorization, automatic call distributor
   (ACD) platform functions, routing (including emergency call routing),
   call setup, mapping, call features (such as call forwarding and video
   mail), and assignment of CAs to relay calls.

   Relay service provider (provider): An organization that operates a
   relay service.  A subscriber selects a relay service provider to
   assign and register a telephone number for their use, to register
   with for receipt of incoming calls, and to provide the default
   service for outgoing calls.

   Relay user: Please refer to "subscriber".

   Relay user E.164 Number (user E.164): The telephone number (in ITU-T
   E.164 format) assigned to the user.

   Relay user equipment (RUE): A SIP user agent (UA) enhanced with extra
   features to support a subscriber in requesting, receiving and using
   relay calls.  A RUE may take many forms, including a stand-alone
   device; an application running on a general-purpose computing device
   such as a laptop, tablet or smart phone; or proprietary equipment
   connected to a server that provides the RUE interface.

   RUE Interface: the interfaces described in this document between a
   RUE and a VRS provider who supports it

   Sign language: A language that uses hand gestures and body language
   to convey meaning including, but not limited to, American Sign
   Language (ASL).

   Subscriber: An individual who has registered with a provider and who
   obtains service by using relay user equipment.  This is the
   traditional telecom term for an end-user customer, which in our case
   is a relay user.  A user may be a subscriber to more than one VRS
   provider.

   Video relay service (VRS): A relay service for people with hearing or
   speech disabilities who use sign language to communicate using video
   equipment (video RUE) with other people in real time.  The video link
   allows the CA to view and interpret the subscriber's signed
   conversation and relay the conversation back and forth with the other
   party.




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3.  Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
   "OPTIONAL" in this document are to be interpreted as described in BCP
   14 [RFC2119] [RFC8174] when, and only when, they appear in all
   capitals, as shown here.  Lower- or mixed-case uses of these key
   words are not to be interpreted as carrying special significance.

4.  General Requirements

   All HTTP/HTTPS [RFC7230] and [RFC2818] connections specified
   throughout this document MUST use HTTPS.  Both HTTPS and all SIP
   connections MUST use TLS conforming to at least [RFC7525] and MUST
   support [RFC8446].

   All text data payloads not otherwise constrained by a specification
   in another standards document MUST be encoded as Unicode UTF-8.

   Implementations MUST support IPv4 and IPv6.  Dual stack support is
   NOT required and provider implementations MAY support separate
   interfaces for IPv4 and IPv6 by having more than one server in the
   appropriate SRV record where there is either an A or AAAA record in
   each server DNS record but not both.  The same version of IP MUST be
   used for both signaling and media of a call unless ICE ([RFC8445]) is
   used, in which case candidates may explicitly offer IPv4, IPv6 or
   both for any media stream.

5.  SIP Signaling

   Implementations of the RUE Interface MUST conform to the following
   core SIP standards:

   *  [RFC3261] (Base SIP)

   *  [RFC3263] (Locating SIP Servers)

   *  [RFC3264] (Offer/Answer)

   *  [RFC3840] (User Agent Capabilities)

   *  [RFC5626] (Outbound)

   *  [RFC8866] (Session Description Protocol)

   *  [RFC3323] (Privacy)

   *  [RFC3605] (RTCP Attribute in SDP)



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   *  [RFC6665] (SIP Events)

   *  [RFC3311] (UPDATE Method)

   *  [RFC5393] (Loop-Fix)

   *  [RFC5658] (Record Route fix)

   *  [RFC5954] (ABNF fix)

   *  [RFC3960] (Early Media)

   *  [RFC6442] (Geolocation header field)

   In the above documents the RUE device conforms to the requirements of
   a SIP user Agent, and the provider conforms to the requirements of
   Registrar and Proxy Server where the document specifies different
   behavior for different roles.  The only requirement on providers for
   RFC6655 (Events) is support for the Message Waiting Indicator (See
   Section 8), which is optional and providers not supporting video mail
   need not support RFC6665.

   In addition, implementations MUST conform to:

   *  [RFC3327] (Path)

   *  [RFC8445] and [RFC8839] (ICE)

   *  [RFC3326] (Reason header field)

   *  [RFC3515] (REFER Method)

   *  [RFC3891] (Replaces Header field)

   *  [RFC3892] (Referred-By)

   Implementations MUST include a "User-Agent" header field uniquely
   identifying the RUE application, platform, and version in all SIP
   requests, and MUST include a "Server" header field with the same
   content in SIP responses.

   Implementations intended to support mobile platforms MUST support
   [RFC8599] and MUST use it as at least one way to support waking up
   the client from background state.







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5.1.  Registration

   The RUE MUST register with a SIP registrar, following [RFC3261] and
   [RFC5626] at a provider it has an account with.  If the configuration
   (see Section 9.2) contains multiple "outbound-proxies", then the RUE
   MUST use them as specified in [RFC5626] to establish multiple flows.

   The Request-URI for the REGISTER request MUST contain the "provider-
   domain" from the configuration.  The To-URI and From-URI MUST be
   identical URIs, formatted as specified in Section 5.4, using the
   "phone-number" and "provider-domain" from the configuration.

   The RUE determines the URI to resolve by initially determining if an
   outbound proxy is configured.  If it is, the URI will be that of the
   outbound proxy.  If no outbound proxy is configured, the URI will be
   the Request-URI from the REGISTER request.  The RUE extracts the
   domain from that URI and consults the DNS record for that domain.
   The DNS entry MUST contain NAPTR records conforming to RFC3263.  One
   of those NAPTR records MUST specify TLS as the preferred transport
   for SIP.  For example, a DNS NAPTR query for "sip:
   p1.red.example.net" could return:

         IN NAPTR 50  50 "s" "SIPS+D2T" "" _sips._tcp.p1.red.example.net
         IN NAPTR 90  50 "s" "SIP+D2T"  "" _sip._tcp.p1.red.example.net

   If the RUE receives a 439 (First Hop Lacks Outbound Support) response
   to a REGISTER request, it MUST re-attempt registration without using
   the outbound mechanism.

   The registrar MAY authenticate using SIP digest authentication.  The
   credentials to be used (username and password) MUST be supplied
   within the credentials section of the configuration and identified by
   the realm the registrar uses in a digest challenge.  This username/
   password combination SHOULD NOT be the same as that used for other
   purposes, such as retrieving the RUE configuration or logging into
   the Provider's customer service portal.  [RFC8760] MUST be supported
   by all implementations and SHA-512 digest algorithms MUST be
   supported.

   If the registration request fails with an indication that credentials
   from the configuration are invalid, then the RUE MUST retrieve a
   fresh version of the configuration.  If credentials from a freshly
   retrieved configuration are found to be invalid, then the RUE MUST
   cease attempts to register and inform the RUE User of the problem.

   Support for multiple simultaneous registrations with multiple
   providers by the RUE is OPTIONAL for the RUE (and providers do not
   need any support for this option).



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   Multiple simultaneous RUE SIP registrations from different RUE
   devices with the same SIP URI SHOULD be permitted by the provider.
   The provider MAY limit the total number of simultaneous
   registrations.  When a new registration request is received that
   results in exceeding the limit on simultaneous registrations, the
   provider MAY then prematurely terminate another registration;
   however, it SHOULD NOT do this if it would disconnect an active call.

   If a provider prematurely terminates a registration to reduce the
   total number of concurrent registrations with the same URI, it SHOULD
   take some action to prevent the affected RUE from automatically re-
   registering and re-triggering the condition.

5.2.  Session Establishment

5.2.1.  Normal Call Origination

   After initial SIP registration, the RUE adheres to SIP [RFC3261]
   basic call flows, as documented in [RFC3665].

   A RUE device MUST route all outbound calls through an outbound proxy
   if configured.

   The SIP URIs in the To field and the Request-URI MUST be formatted as
   specified in subsection Section 5.4 using the destination phone
   number, or as SIP URIs, as provided in the configuration
   (Section 9.2).  The domain field of the URIs SHOULD be the "provider-
   domain" from the configuration (e.g.,
   sip:+13115552368@red.example.com;user=phone) except that an anonymous
   call would not use the provider domain.

   Anonymous calls MUST be supported by all implementations.  An
   anonymous call is signaled per [RFC3323].

   The From-URI MUST be formatted as specified in Section 5.4, using the
   phone-number and "provider-domain" from the configuration.  It SHOULD
   also contain the display-name from the configuration when present.
   (Please refer to Section 9.2.)

   Negotiated media MUST follow the guidelines specified in Section 6 of
   this document.

   To allow time to timeout an unanswered call and direct it to a
   videomail server, the User Agent Client MUST NOT impose a time limit
   less than the default SIP Invite transaction timeout of 3 minutes.






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5.2.2.  Dial-Around Origination

   Providers and RUE devices MUST support both One-Stage and Two-Stage
   dial-around

   Outbound dial-around calls allow a RUE user to select any provider to
   provide interpreting services for any call.  "Two-stage" dial-around
   calls involve the RUE calling a telephone number that reaches the
   dial-around provider and using signing or DTMF to provide the called
   party telephone number.  In two-stage dial-around, the To URI is the
   front door URI (see Section 9.2) of the dial-around provider and the
   domain of the URI is the provider domain from the configuration.  The
   provider list service (Section 9.1) can be used by the RUE to obtain
   a list of providers and then the configuration service
   (Section 9.2.1) without credentials can be used to find the front
   door URI for each of these providers.

   One-stage dial-around is a method where the called party telephone
   number is provided in the To URI and the Request-URI, using the
   domain of the dial-around provider.

   For one-stage dial-around, the RUE MUST follow the procedures in
   Section 5.2.1 with the following exception: the domain part of the
   SIP URIs in the To field and the Request-URI MUST be the domain of
   the dial-around provider, discovered according to Section 9.1.

   The following is a partial example of a one-stage dial-around call
   from VRS user +1-555-222-0001 hosted by red.example.com to a hearing
   user +1-555-123-4567 using dial-around to green.example.com for the
   relay service.  Only important details of the messages are shown and
   many header fields have been omitted:

   One-Stage Dial-Around


















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     ,-+-.        ,----+----.    ,-----+-----.
     |RUE|        |Default  |    |Dial-Around|
     |   |        |Provider |    | Provider  |
     `-+-'        `----+----'    `-----+-----'
       |               |               |
       | [1] INVITE    |               |
       |-------------->| [2] INVITE    |
       |               |-------------->|

     Message Details:

     [1] INVITE Rue -> Default Provider

     INVITE sip:+15551234567@green.example.net;user=phone SIP/2.0
     To: <sip:+15551234567@green.example.net;user=phone>
     From: "Bob Smith" <sip:+18135551212@red.example.net;user=phone>


     [2] INVITE Default Provider -> Dial-Around Provider

     INVITE sip:+15551234567@green.example.net;user=phone SIP/2.0
     To: <sip:+15551234567@green.example.net;user=phone>
     From: "Bob Smith" sip:+18135551212@red.example.net;user=phone
     P-Asserted-Identity: sip:+18135551212@red.example.net

                                  Figure 1

5.2.3.  RUE Contact Information

   To identify the owner of a RUE, the initial INVITE for a call from a
   RUE, or the 200 OK accepting a call by a RUE, identifies the owner by
   sending a Call-Info header field with a purpose parameter of "rue-
   owner".  The URI MAY be an HTTPS URI or Content-Indirect URL.  The
   latter is defined by [RFC2392] to locate message body parts.  This
   URI type is present in a SIP message to convey the RUE ownership
   information as a MIME body.  The form of the RUE ownership
   information is a xCard [RFC6351].  Please refer to [RFC6442] for an
   example of using Content-Indirect URLs in SIP messages.  Note that
   use of the Content-Indirect URL usually implies multiple message
   bodies ("mime/multipart").

5.2.4.  Incoming Calls

   The RUE MUST only accept inbound calls sent to it by a proxy
   mentioned in the configuration.






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   If Multiple simultaneous RUE SIP registrations from different RUE
   devices with the same SIP URI exist, the provider MUST parallel fork
   the call to all registered RUEs so that they ring at the same time.
   The first RUE to reply with a 200 OK answers the call and the
   provider MUST CANCEL other call branches.

5.2.5.  Emergency Calls

   Implementations MUST conform to [RFC6881] for handling of emergency
   calls, except that if the device is unable to determine its own
   location, it MAY send the emergency call without a Geolocation header
   field and without a Route header field (since it would be unable to
   query the LoST server for a route per RFC6881).  If an emergency call
   arrives at the provider without a Geolocation header field, the
   provider MUST supply location by adding the Geolocation header field,
   and MUST supply the route by querying the LoST server with that
   location.

   If the emergency call is to be handled using existing country
   specific procedures, the provider is responsible for modifying the
   INVITE to conform to the country-specific requirements.  In this
   case, location MAY be extracted from the RFC6881 conformant INVITE
   and used to propagate it to the appropriate country-specific
   entities.  If the configuration specifies it, an implementation of a
   RUE device MAY send a Geolocation header field containing its
   location in the REGISTER request.  If implemented, users MUST be
   offered an opt-out.  Country-specific procedures might require the
   location to be pre-loaded in some entity prior to placing an
   emergency call; however, the RUE may have a more accurate and timely
   device location than the manual, pre-loaded entry.  That information
   MAY be used to populate the location to appropriate country-specific
   entities.  Re-registration SHOULD be used to update the location, so
   long as the rate of re-registration is limited if the device is
   moving.

   Implementations MUST implement Additional Data, [RFC7852].  RUE
   devices MUST implement Data Provider, Device Implementation and
   Owner/Subscriber Information blocks.

5.3.  Mid Call Signaling

   Implementations MUST support re-INVITE to renegotiate media session
   parameters (among other uses).  Per Section 6.1, implementations MUST
   be able to support an INFO request for full frame refresh for devices
   that do not support RTCP mechanisms (please refer to Section 6.8).
   Implementations MUST support an in-dialog REFER ([RFC3515] updated by
   [RFC7647] and including support for norefersub per [RFC4488]) with
   the Replaces header field [RFC3891] to enable call transfer.



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5.4.  URI Representation of Phone Numbers

   SIP URIs constructed from non-URI sources (dial strings) and sent to
   SIP proxies by the RUE MUST be represented as follows, depending on
   whether they can be represented as an E.164 number.  In this section
   "expressed as an E.164 number" includes numbers such as toll free
   numbers that are not actually E.164 numbers, but have the same
   format.

   A dial string that can be expressed as an E.164 phone number MUST be
   represented as a SIP URI with a URI ";user=phone" tag.  The user part
   of the URI MUST be in conformance with 'global-number' defined in
   [RFC3966].  The user part MUST NOT contain any 'visual-separator'
   characters, as defined in [RFC3966].

   Dial strings that cannot be expressed as E.164 numbers MUST be
   represented as dialstring URIs, as specified by [RFC4967], e.g.,
   sip:411@red.example.net;user=dialstring.

   The domain part of Relay Service URIs and User Address of Records
   (AoR) MUST resolve (per [RFC3263]) to globally routable IPv4 and/or
   IPv6 addresses.

5.5.  Transport

   Implementations MUST conform to [RFC8835] except for its guidance on
   the WebRTC data channel, which this specification does not use.  See
   Section 6.2 for how RUE supports real-time text without the data
   channel.

   Implementations MUST support SIP outbound [RFC5626] (please also
   refer to Section 5.1).

6.  Media

   This specification adopts the media specifications for WebRTC
   ([RFC8825]).  Where WebRTC defines how interactive media
   communications may be established using a browser as a client, this
   specification assumes a normal SIP call.  The RTP, RTCP, SDP and
   specific media requirements specified for WebRTC are adopted for this
   document.  The RUE is a WebRTC "non-browser" endpoint, except as
   noted expressly below.

   The following sections specify the WebRTC documents to which
   conformance is required.  "Mandatory to Implement" means a conforming
   implementation must implement the specified capability.  It does not
   mean that the capability must be used in every session.  For example,
   OPUS is a mandatory to implement audio codec, and all conforming



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   implementations must support OPUS.  However, implementation
   presenting a call across the RUE Interface where the call originates
   in the Public Switched Telephone Network, or an older, non-RUE-
   compatible device, which only offers G.711 audio, does not need to
   include the OPUS codec in the offer, since it cannot be used with
   that call.

6.1.  SRTP and SRTCP

   Implementations MUST support [RFC8834] except that MediaStreamTracks
   are not used.  Implementations MUST conform to Section 6.4 of
   [RFC8827].

6.2.  Text-Based Communication

   Implementations MUST support real-time text ([RFC4102] and [RFC4103])
   via T.140 media.  One original and two redundant generations MUST be
   transmitted and supported, with a 300 ms transmission interval.
   Implementations MUST support [RFC9071] especially for emergency
   calls.  Note that RFC4103 is not how real-time text is transmitted in
   WebRTC and some form of transcoder would be required to interwork
   real-time text in the data channel of WebRTC to RFC4103 real-time
   text.

   Transport of T.140 real-time text in WebRTC is specified in
   [RFC8865], using the WebRTC data chanel.  RFC 8865 also has some
   advice on how gateways between RFC 4103 and RFC 8865 should operate.
   It is RECOMMENDED that RFC 8865 including multiparty support is used
   for communication with browser-based WebRTC implementations.
   Implementations MUST support [RFC9071].

6.3.  Video

   Implementations MUST conform to [RFC7742] with following exceptions:
   only H.264, as specified in [RFC7742], is Mandatory to Implement, and
   VP8 support is OPTIONAL at both the device and providers.  This is
   because backwards compatibility is desirable, and older devices do
   not support VP8.

6.4.  Audio

   Implementations MUST conform to [RFC7874].









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6.5.  DTMF Digits

   Implementations MUST support the "audio/telephone-event" [RFC4733]
   media type.  They MUST support conveying event codes 0 through 11
   (DTMF digits "0"-"9", "*","#") defined in Table 7 of [RFC4733].
   Handling of other tones is OPTIONAL.

6.6.  Session Description Protocol

   The SDP offers and answers MUST conform to the SDP rules in [RFC8829]
   except that the RUE interface uses SIP transport for SDP.  The SDP
   for real-time text MUST specify the T.140 payload type [RFC4103].

6.7.  Privacy

   The RUE MUST provide for user privacy by implementing a local one-way
   mute, without signaling, for both audio and video.  However, RUE MUST
   maintain any NAT bindings by periodically sending media packets on
   all active media sessions containing silence/comfort noise/black
   screen/etc. per [RFC6263].

6.8.  Negative Acknowledgment, Packet Loss Indicator, and Full
      Intraframe Request Features

   The NACK, FIR and PLI features as described in [RFC4585] and
   [RFC5104] MUST be implemented.  Availability of these features MUST
   be announced with the "ccm" feedback value.  NACK should be used when
   negotiated and conditions warrant its use and the other end supports
   it.  Signaling picture losses as Packet Loss Indicator (PLI) should
   be preferred.  FIR should be used only in situations where not
   sending a decoder refresh point would render the video unusable for
   the users, as per RFC5104 subsection 4.3.1.2.

   For backwards compatibility with calling devices that do not support
   the foregoing methods, implementations MUST implement SIP INFO
   messages to send and receive XML encoded Picture Fast Update messages
   according to [RFC5168].

7.  Contacts

7.1.  CardDAV Login and Synchronization

   Support of CardDAV by providers is OPTIONAL.

   The RUE MUST and providers MAY be able to synchronize the user's
   contact directory between the RUE endpoint and one maintained by the
   user's VRS provider using CardDAV ([RFC6352] and [RFC6764]).




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   The configuration MAY supply a username and domain identifying a
   CardDAV server and address book for this account.

   To access the CardDAV server and address book, the RUE MUST follow
   Section 6 of RFC6764, using the chosen username and domain in place
   of an email address.  If the request triggers a challenge for digest
   authentication credentials, the RUE MUST continue using matching
   "credentials" from the configuration.  If no matching credentials are
   configured, the RUE MUST use the SIP credentials from the
   configuration.  If the SIP credentials fail, the RUE MUST query the
   user.

   Synchronization using CardDAV MUST be a two-way synchronization
   service, with proper handling of asynchronous adds, changes, and
   deletes at either end of the transport channel.

7.2.  Contacts Import/Export Service

   Implementations MUST be able to export/import the list of contacts in
   xCard [RFC6351] xml format.

   The RUE accesses this service via the "contacts" URI in the
   configuration.  The URL MUST resolve to identify a web server
   resource that imports/exports contact lists for authorized users.

   The RUE stores/retrieves the contact list (address book) by issuing
   an HTTPS POST or GET request.  If the request triggers a challenge
   for digest authentication credentials, the RUE MUST attempt to
   continue using matching "credentials" from the configuration.  If no
   credentials are configured, the RUE MUST query the user.

8.  Video Mail

   Support for video mail includes a retrieval mechanism and a Message
   Waiting Indicator (MWI).  Message storage is not specified by this
   document.  RUE devices MUST support message retrieval using a SIP
   call to a specified SIP URI using DTMF to manage the mailbox, as well
   as a browser based interface reached at a specified HTTPS URI.  If a
   provider supports video mail at least one of these mechansism MUST be
   supported.  RUE devices MUST support both.  See Section 9.2 for how
   the URI to reach the retrieval interface is obtained.

   Implementations MUST support subscriptions to "message-summary"
   events [RFC3842] to the URI specified in the configuration.
   Providers MUST support MWI if they support video mail.  RUE devices
   MUST support MWI.





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   In notification bodies, if detailed message summaries are available,
   messages with video MUST be reported using "message-context-class
   multimedia-message" defined in [RFC3458] .

9.  Provisioning and Provider Selection

   To simplify how users interact with RUE devices, the RUE interface
   separates provisioning into two parts.  One provides a directory of
   providers so that a user interface can allow easy provider selection
   either for registering or for dial-around.  The other provides
   configuration data for the device for each provider.

9.1.  RUE Provider Selection

   To allow the user to select a relay service, the RUE MAY at any time
   obtain (typically on startup) a list of Providers that provide
   service in a country.  IANA has established a registry that contains
   a two letter country code and an entry point string.  The entry
   point, when used with the following interface, returns a list of
   provider names for a country code suitable for display, with a
   corresponding a entry point to obtain information about that
   provider.

   Each country that supports video relay service using this
   specification MAY support the provider list.  This document does not
   specify who maintains the list.  Some possibilities are a regulator
   or entity designated by a regulator, an agreement among providers to
   provide the list, or a user group.

   The interface to obtain the list of providers is described by an
   OpenApi [OpenApi] interface description.  In that interface
   description, the "servers" component is specified as "localhost".
   The entry point in the registry is substituted for "localhost" to
   obtain the server prefix of the interface.  The "Providers" path then
   specifies the rest of the URI used to obtain the list.  For example,
   if the entryPoint is "example.com", the provider list would be
   obtained from https://example.com/rum/V1/Providers.

   The web service also has a simple version mechanism that returns a
   list of versions of the web service it supports.  This document
   describes version 1.0.  Versions are described as a major version,
   the period "." and a minor version, where major and minor versions
   are integers.  A backwards compatible change within a major version
   MAY increment only the minor version number.  A non-backwards
   compatible change MUST increment the major version number.  To
   achieve backwards compatibility, implementations MUST ignore any
   object members they do not implement.  Minor version definitions
   SHALL only add objects, non-required members of existing objects, and



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   non-mandatory-to use functions and SHALL NOT delete any objects,
   members of objects or functions.  This means an implementation of a
   specific major version and minor version is backwards compatible with
   all minor versions of the major version.  The versions mechanism
   returns an array of supported versions, one for each major version
   supported, with the minor version listed being the highest supported
   minor version.

   The V1.0 provider list is a json object consisting of an array where
   each entry describes one provider.  Each entry consists of the
   following items:

   *  name: This parameter contains the text label identifying the
      provider and is meant to be displayed to the human VRS user.

   *  entryPoint: A string used for configuration purposes by the RUE
      (as discussed in Section 9.2)

   The VRS user interacts with the RUE to select from the provider list
   one or more providers with whom the user has already established an
   account, wishes to establish an account, or wishes to use the
   provider for a one-stage dial around.

     {
       "providers": [
         {
           "name": "Red",
           "entryPoint": "red.example.net"
         },
         {
           "name": "Green",
           "entryPoint": "green.example.net"
         },
         {
           "name": "Blue",
           "entryPoint": "blue.example.net"
         }
       ]
     }

              Figure 2: Example of a provider list JSON object










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     {
       "versions": [
         {
           "major": 1,
           "minor": 6
         },
         {
           "major": 2,
           "minor": 13
         },
         {
           "major": 3,
           "minor": 2
         }
       ]
     }

                 Figure 3: Example of a Version JSON object

9.2.  RUE Configuration Service

   A RUE device may retrieve a provider configuration the using a simple
   HTTPs web service.  There are two entry points.  One is used without
   user credentials or parameters, the response includes configuration
   data for new user sign up and dial around.  The other uses the
   userid/password to authenticate to the interface and returns
   configuration data for the RUE.

   The interface to obtain configuration data is described by an OpenApi
   [OpenApi] interface description.  In that interface description, the
   "servers" component is specified as "localhost".  The entry point
   obtained from the provider list (Section 9.1) is substituted for
   "localhost" to obtain the server prefix of the interface.  The path
   then specifies the rest of the URI used to obtain the list.  For
   example, if the entryPoint from the provider list is
   "redexample.net", the provider configuration would be obtained from
   https://red.example.net/rum/V1/ProviderConfig.

   In both the queries, an optional parameter may be provided to the
   interface which is an API Key.  The implementation MAY have an API
   Key obtained from the provider and specific to the implementation.
   The method the API Key is obtained is not specified in this document.
   The provider MAY refuse to provide service to an implementation
   presenting an API Key it does not recognize.

   Also in both queries, the RUE device provides a required parameter
   which contains an instance identifier.  This parameter MUST be the
   same value each time this instance (same implementation on same



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   device) queries the interface.  This MAY be used by the provider, for
   example, to associate a location with the instance for emergency
   calls.

   The data returned is a json object consisting of an array of key/
   value configuration parameters to be used by the RUE.

   The configuration API also provides the same version mechanism as
   specified above in Section 9.1.  The version of the configuration
   service MAY be different than the version of the provider list
   service.

   The configuration data payload includes the following data items.
   Items not noted as (OPTIONAL) are REQUIRED.  If other unexpected
   items are found, they MUST be ignored.

9.2.1.  Provider Configuration

   *  signup: (OPTIONAL) an array of json objects consisting of:

      -  language: entry from the IANA language subtag registry.
         Normally, this would be a written language tag.

      -  uri: a URI to the website for creating a new account in the
         supported language.  The new user signup URI may only initiate
         creation of a new account.  Various vetting, approval and other
         processes may be needed, which could take time, before the
         account is established.  The result of creating a new account
         would be a username and password, which would be manually
         entered into the RUE device to allow connection to the
         provider.

   *  dialAround: an array of json objects consisting of:

      -  language: entry from the IANA language subtag registry.
         Normally, this would be a sign language tag.

      -  frontDoor: a URI to a queue of interpreters supporting the
         specified language for a two-stage dial-around

      -  oneStage: a URI that can be used with a one-stage dial-around
         Section 5.2.2 using an interpreter supporting the specified
         language

   *  helpDesk: (OPTIONAL) an array of json objects consisting of:






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      -  language: entry from the IANA language subtag registry.
         Normally this would be a sign language tag, although it could
         be a written language tag if the help desk only supports a chat
         interface

      -  uri: URI that reaches a help desk for callers supporting the
         specified language.  The URI MAY be a SIP URI for help provided
         with a SIP call, or MAY be an HTTPS URI for help provided with
         a browser interface.

      A list is specified so that the provider can offer multiple
      choices to users for language and interface styles.

9.2.2.  RUE Configuration

   *  lifetime: (optional) Specifies how long (in seconds) the RUE MAY
      cache the configuration values.  Values may not be valid when
      lifetime expires.  If the RUE caches configuration values, it MUST
      cryptographically protect them.  The RUE SHOULD retrieve a fresh
      copy of the configuration before the lifetime expires or as soon
      as possible after it expires.  The lifetime is not guaranteed: the
      configuration may change before the lifetime value expires.  In
      that case, the Provider MAY indicate this by generating
      authorization challenges to requests and/or prematurely
      terminating a registration.  Emergency Calls MUST continue to
      work.  If not specified, the RUE MUST fetch new configuration data
      every time it starts.

   *  sip-password: (optional) a password used for SIP, STUN and TURN
      authentication.  The RUE device retains this data, which must be
      stored securely.  If it is not supplied, but was supplied on a
      prior invocation of this interface, the most recently supplied
      password MUST be used.  If it was never supplied, the password
      used to authenticate to the configuration service is used for SIP,
      STUN and TURN.

   *  phone-number: The telephone number (in E.164 format) assigned to
      this subscriber.  This becomes the user portion of the SIP URI
      identifying the subscriber.

   *  outbound-proxy: (optional) A URI of a SIP proxy to be used when
      sending requests to the provider.

   *  mwi: (optional) A URI identifying a SIP event server that
      generates "message-summary" events for this subscriber.

   *  videomail: (optional) An SIP or HTTPS URI that can be called to
      retrieve video mail messages.



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   *  contacts: (optional) An HTTPS URI that may be used to export
      (retrieve) the subscriber's complete contact list managed by the
      provider.  MUST be provided if the subscriber has contacts.

   *  carddav: (optional) A username, password and domain name
      (separated by ""@"") identifying a "CardDAV" server and account
      that can be used to synchronize the RUE's contact list with the
      contact list managed by the provider.  If username or password are
      not supplied, the main account credentials are used.

   *  sendLocationWithRegistration: (optional) True if the RUE should
      send a Geolocation header field with REGISTER, false if it should
      not.  Defaults to false if not present.

   *  ice-servers: (optional) An array of URLs identifying STUN and TURN
      servers available for use by the RUE for establishing media
      streams in calls via the provider.

9.2.3.  Examples

   Example JSON provider configuration payload
     {
       "signUp": [
          { "language" : "en", "uri" : "welcome-en.example.net"} ,
          { "language" : "es", "uri" : "welcome-es.example.net"} ] ,
       "dialAround": [
          { "language" : "en", "frontDoor" : "fd-en.example.net",
               "oneStage" : "1stg-eng.example.com" } ,
          { "language" : "es", "frontDoor" : "fd-es.example.net",
               "oneStage" : "1stg-spn.example.com" } ] ,
       "helpDesk": [
          { "language" : "en", "uri" : "help-en.example.net"} ,
          { "language" : "es", "uri" : "help-es.example.net"} ]
     }

                                  Figure 4

   Example JSON RUE configuration payload













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     {
       "lifetime": 86400,
       "display-name" : "Bob Smith",
       "phone-number": "+18135551212",
       "provider-domain": "red.example.net",
       "outbound-proxies": [
         "sip:p1.red.example.net",
         "sip:p2.red.example.net" ],
       "mwi": "sip:+18135551212@red.example.net",
       "videomail": "sip:+18135551212@vm.red.example.net",
       "contacts": "https://red.example.net:443/contacts/1dess45awd",
       "carddav": "bob@red.example.com" ,
       "sendLocationWithRegistration": false,
       "ice-servers": [
          {"stun":"stun.l.google.com:19302" },
          {"turn":"turn.red.example.net:3478"}
       ]
     }

                                  Figure 5

9.2.4.  Using the Provider Selection and RUE Configuration Services
        Together

   One way to use these two services is:

   *  At startup, the RUE retrieves the provider list for the country it
      is located in.

   *  For each provider in the list:

      -  If the RUE does not have credentials for that provider, use the
         configuration service without credentials to obtain signup,
         dial around and helpdesk information.

      -  If the RUE has credentials for that provider, use the
         configuration service with credentials to obtain all
         configuration data.

9.3.  OpenAPI Interface Descriptions

   The interfaces in Section 9.1 and Section 9.2 are formally specified
   with OpenAPI 3.0 ([OpenApi]) descriptions in yaml form.








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  openapi: 3.0.1
  info:
    title: RUM API
    version: "1.0"
  servers:
    - url: http://localhost/rum/v1
  paths:
    /Providers:
      get:
        summary: Get a list of providers and domains to get
                 config data from
        operationId: GetProviderList
        responses:
          '200':
            description: List of providers for a country
            content:
              application/json:
                schema:
                  $ref: '#/components/schemas/ProviderList'
    /ProviderConfig:
      get:
        summary: Configuration data for one provider
        operationId: GetProviderConfiguration
        parameters:
          - in: query
            name: apiKey
            schema:
              type: string
            description: API Key assigned to this implementation
          - in: query
            name: instanceId
            schema:
              type: string
            required: true
            description: Unique string for this implementation
                         on this device
        responses:
          '200':
            description: configuration object
            content:
              application/json:
                schema:
                  $ref: '#/components/schemas/ProviderConfigurationData'
    /RueConfig:
      get:
        summary: Configuration data for one RUE
        operationId: GetRueConfiguration
        parameters:



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          - in: query
            name: apiKey
            schema:
              type: string
            description: API Key assigned to this implementation
          - in: query
            name: instanceId
            schema:
              type: string
            required: true
            description: Unique string for this implementation
                         on this device
        responses:
          '200':
            description: configuration object
            content:
              application/json:
                schema:
                  $ref: '#/components/schemas/RueConfigurationData'
    /Versions:
      servers:
        - url: https://api.example.com/rum
          description: Override base path for Versions query
      get:
        summary: Retrieves all supported versions
        operationId: RetrieveVersions
        responses:
          '200':
            description: Versions supported
            content:
              application/json:
                schema:
                  $ref: '#/components/schemas/VersionsArray'
  components:
    schemas:
      ProviderList:
        type: object
        required:
          - providers
        properties:
          providers:
            type: array
            items:
              type: object
              required:
                - name
                - domain
              properties:



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                name:
                  type: string
                  description: Human readable provider name
                entryPoint:
                  type: string
                  description: provider entry point for interface
      VersionsArray:
        type: object
        required:
          - versions
        properties:
          versions:
            type: array
            items:
              type: object
              required:
                - major
                - minor
              properties:
                major:
                  type: integer
                  format: int32
                  description: Version major number
                minor:
                  type: integer
                  format: int32
                  description: Version minor number
      ProviderConfigurationData:
        type: object
        properties:
          signup:
            type: object
            required:
              - language
              - uri
            properties:
              language:
                type: string
                description: entry from IANA language-subtag-registry
              uri:
                type: string
                format: uri
                description: URI to signup website supporting language
          dialAround:
            type: object
            required:
              - language
              - frontDoor



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              - oneStage
            properties:
              language:
                type: string
                description: entry from IANA language-subtag-registry
              frontDoor:
                type: string
                format: uri
                description: SIP URI for two-stage dial around
              oneStage:
                type: string
                format: uri
                description: SIP URI for one-stage dial around
          helpDesk:
            type: object
            required:
              - language
              - uri
            properties:
              language:
                type: string
                description: entry from IANA language-subtag-registry
              uri:
                type: string
                format: uri
                description: SIP URI of helpdesk supporting language
      RueConfigurationData:
        type: object
        required:
          - phone-number
        properties:
          lifetime:
            type: integer
            description: how long (in seconds) the RUE MAY cache the
                         configuration values
          sip-password:
            type: string
          phone-number:
            type: string
            description: telephone number assigned this subscriber
          outbound-proxy:
            type: string
            format: uri
            description: SIP URI of proxy to be used when sending
                         requests to the provider
          mwi:
            type: string
            format: uri



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            description: A URI identifying a SIP event server that
                generates "message-summary" events for this subscriber.
          videomail:
            type: string
            format: uri
            description: An HTTPS or SIP URI that can be called to
                         retrieve video mail messages.
          contacts:
            type: string
            format: uri
            description: An HTTPS URI that may be used to export
               (retrieve) the subscriber's complete contact list
               managed by the provider.
          carddav:
            type: object
            description: CardDAV server and user information that can be
                 used to synchronize the RUE's contact list with the
                 contact list managed by the provider.
            properties:
              domain:
                type: string
                description: CardDAV server address
              username:
                type: string
                description: username for authentication with CardDAV
                   server.  Use provider username if not provided
              password:
                type: string
                description: password for authentication to the CardDAV
                   server. Use provider password if not provided
              sendLocationWithRegistration:
                type: boolean
                description: True if the RUE should send a Geolocation
                    header field with REGISTER, false if it should not.
                    Defaults to false if not present.
          ice-servers:
            type: array
            items:
              type: string
              format: uri
              description: URIs identifying STUN and TURN servers
                 available for use by the RUE for establishing
                 media streams in calls via the provider.

               Figure 6: Provider List OpenAPI description

10.  IANA Considerations




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10.1.  RUE Provider List Registry

   IANA has created the "RUE Provider List" registry.  The management
   policy for this registry is "Expert Review" [RFC8126].  The expert
   should prefer a regulator operated or designated list interface
   operator.  Otherwise, evidence that the proposed list interface
   operator will provide a complete list of providers is required to add
   the entry to the registry.  Updates to the registry are permitted if
   the expert judges the new proposed URI to provide a more accurate
   list than the existing entry.  Each entry has two fields, values for
   both of which MUST be provided when registering or updating an entry:

   *  country code: a two letter ISO93166 country code

   *  list entry point: a string is used to compose the uri to the
      provider list interface for that country

10.2.  Registration of rue-owner Value of the purpose Parameter

   This document defines the new predefined value "rue-owner" for the
   "purpose" header field parameter of the Call-Info header field.  This
   modifies the "Header Field Parameters and Parameter Values"
   subregistry of the "Session Initiation Protocol (SIP) Parameters"
   registry by adding this RFC as a reference to the line for the header
   field "Call-Info" and parameter name "purpose"

   *  Header Field: Call-Info

   *  Parameter Name: purpose

   *  Predefined Values: Yes

11.  Security Considerations

   The RUE is required to communicate with servers on public IP
   addresses and specific ports to perform its required functions.  If
   it is necessary for the RUE to function on a corporate or other
   network that operates a default-deny firewall between the RUE and
   these services, the user must arrange with their network manager for
   passage of traffic through such a firewall in accordance with the
   protocols and associated SRV records as exposed by the provider.
   Because VRS providers may use different ports for different services,
   these port numbers may differ from provider to provider.

12.
Normative References





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   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <https://www.rfc-editor.org/info/rfc2119>.

   [RFC2392]  Levinson, E., "Content-ID and Message-ID Uniform Resource
              Locators", RFC 2392, DOI 10.17487/RFC2392, August 1998,
              <https://www.rfc-editor.org/info/rfc2392>.

   [RFC2818]  Rescorla, E., "HTTP Over TLS", RFC 2818,
              DOI 10.17487/RFC2818, May 2000,
              <https://www.rfc-editor.org/info/rfc2818>.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              DOI 10.17487/RFC3261, June 2002,
              <https://www.rfc-editor.org/info/rfc3261>.

   [RFC3263]  Rosenberg, J. and H. Schulzrinne, "Session Initiation
              Protocol (SIP): Locating SIP Servers", RFC 3263,
              DOI 10.17487/RFC3263, June 2002,
              <https://www.rfc-editor.org/info/rfc3263>.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              DOI 10.17487/RFC3264, June 2002,
              <https://www.rfc-editor.org/info/rfc3264>.

   [RFC3311]  Rosenberg, J., "The Session Initiation Protocol (SIP)
              UPDATE Method", RFC 3311, DOI 10.17487/RFC3311, October
              2002, <https://www.rfc-editor.org/info/rfc3311>.

   [RFC3323]  Peterson, J., "A Privacy Mechanism for the Session
              Initiation Protocol (SIP)", RFC 3323,
              DOI 10.17487/RFC3323, November 2002,
              <https://www.rfc-editor.org/info/rfc3323>.

   [RFC3326]  Schulzrinne, H., Oran, D., and G. Camarillo, "The Reason
              Header Field for the Session Initiation Protocol (SIP)",
              RFC 3326, DOI 10.17487/RFC3326, December 2002,
              <https://www.rfc-editor.org/info/rfc3326>.

   [RFC3327]  Willis, D. and B. Hoeneisen, "Session Initiation Protocol
              (SIP) Extension Header Field for Registering Non-Adjacent
              Contacts", RFC 3327, DOI 10.17487/RFC3327, December 2002,
              <https://www.rfc-editor.org/info/rfc3327>.




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   [RFC3458]  Burger, E., Candell, E., Eliot, C., and G. Klyne, "Message
              Context for Internet Mail", RFC 3458,
              DOI 10.17487/RFC3458, January 2003,
              <https://www.rfc-editor.org/info/rfc3458>.

   [RFC3515]  Sparks, R., "The Session Initiation Protocol (SIP) Refer
              Method", RFC 3515, DOI 10.17487/RFC3515, April 2003,
              <https://www.rfc-editor.org/info/rfc3515>.

   [RFC3605]  Huitema, C., "Real Time Control Protocol (RTCP) attribute
              in Session Description Protocol (SDP)", RFC 3605,
              DOI 10.17487/RFC3605, October 2003,
              <https://www.rfc-editor.org/info/rfc3605>.

   [RFC3840]  Rosenberg, J., Schulzrinne, H., and P. Kyzivat,
              "Indicating User Agent Capabilities in the Session
              Initiation Protocol (SIP)", RFC 3840,
              DOI 10.17487/RFC3840, August 2004,
              <https://www.rfc-editor.org/info/rfc3840>.

   [RFC3842]  Mahy, R., "A Message Summary and Message Waiting
              Indication Event Package for the Session Initiation
              Protocol (SIP)", RFC 3842, DOI 10.17487/RFC3842, August
              2004, <https://www.rfc-editor.org/info/rfc3842>.

   [RFC3891]  Mahy, R., Biggs, B., and R. Dean, "The Session Initiation
              Protocol (SIP) "Replaces" Header", RFC 3891,
              DOI 10.17487/RFC3891, September 2004,
              <https://www.rfc-editor.org/info/rfc3891>.

   [RFC3892]  Sparks, R., "The Session Initiation Protocol (SIP)
              Referred-By Mechanism", RFC 3892, DOI 10.17487/RFC3892,
              September 2004, <https://www.rfc-editor.org/info/rfc3892>.

   [RFC3960]  Camarillo, G. and H. Schulzrinne, "Early Media and Ringing
              Tone Generation in the Session Initiation Protocol (SIP)",
              RFC 3960, DOI 10.17487/RFC3960, December 2004,
              <https://www.rfc-editor.org/info/rfc3960>.

   [RFC3966]  Schulzrinne, H., "The tel URI for Telephone Numbers",
              RFC 3966, DOI 10.17487/RFC3966, December 2004,
              <https://www.rfc-editor.org/info/rfc3966>.

   [RFC4102]  Jones, P., "Registration of the text/red MIME Sub-Type",
              RFC 4102, DOI 10.17487/RFC4102, June 2005,
              <https://www.rfc-editor.org/info/rfc4102>.





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   [RFC4103]  Hellstrom, G. and P. Jones, "RTP Payload for Text
              Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,
              <https://www.rfc-editor.org/info/rfc4103>.

   [RFC4488]  Levin, O., "Suppression of Session Initiation Protocol
              (SIP) REFER Method Implicit Subscription", RFC 4488,
              DOI 10.17487/RFC4488, May 2006,
              <https://www.rfc-editor.org/info/rfc4488>.

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
              DOI 10.17487/RFC4585, July 2006,
              <https://www.rfc-editor.org/info/rfc4585>.

   [RFC4733]  Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
              Digits, Telephony Tones, and Telephony Signals", RFC 4733,
              DOI 10.17487/RFC4733, December 2006,
              <https://www.rfc-editor.org/info/rfc4733>.

   [RFC4967]  Rosen, B., "Dial String Parameter for the Session
              Initiation Protocol Uniform Resource Identifier",
              RFC 4967, DOI 10.17487/RFC4967, July 2007,
              <https://www.rfc-editor.org/info/rfc4967>.

   [RFC5104]  Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
              "Codec Control Messages in the RTP Audio-Visual Profile
              with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
              February 2008, <https://www.rfc-editor.org/info/rfc5104>.

   [RFC5168]  Levin, O., Even, R., and P. Hagendorf, "XML Schema for
              Media Control", RFC 5168, DOI 10.17487/RFC5168, March
              2008, <https://www.rfc-editor.org/info/rfc5168>.

   [RFC5393]  Sparks, R., Ed., Lawrence, S., Hawrylyshen, A., and B.
              Campen, "Addressing an Amplification Vulnerability in
              Session Initiation Protocol (SIP) Forking Proxies",
              RFC 5393, DOI 10.17487/RFC5393, December 2008,
              <https://www.rfc-editor.org/info/rfc5393>.

   [RFC5626]  Jennings, C., Ed., Mahy, R., Ed., and F. Audet, Ed.,
              "Managing Client-Initiated Connections in the Session
              Initiation Protocol (SIP)", RFC 5626,
              DOI 10.17487/RFC5626, October 2009,
              <https://www.rfc-editor.org/info/rfc5626>.






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   [RFC5658]  Froment, T., Lebel, C., and B. Bonnaerens, "Addressing
              Record-Route Issues in the Session Initiation Protocol
              (SIP)", RFC 5658, DOI 10.17487/RFC5658, October 2009,
              <https://www.rfc-editor.org/info/rfc5658>.

   [RFC5954]  Gurbani, V., Ed., Carpenter, B., Ed., and B. Tate, Ed.,
              "Essential Correction for IPv6 ABNF and URI Comparison in
              RFC 3261", RFC 5954, DOI 10.17487/RFC5954, August 2010,
              <https://www.rfc-editor.org/info/rfc5954>.

   [RFC6263]  Marjou, X. and A. Sollaud, "Application Mechanism for
              Keeping Alive the NAT Mappings Associated with RTP / RTP
              Control Protocol (RTCP) Flows", RFC 6263,
              DOI 10.17487/RFC6263, June 2011,
              <https://www.rfc-editor.org/info/rfc6263>.

   [RFC6351]  Perreault, S., "xCard: vCard XML Representation",
              RFC 6351, DOI 10.17487/RFC6351, August 2011,
              <https://www.rfc-editor.org/info/rfc6351>.

   [RFC6352]  Daboo, C., "CardDAV: vCard Extensions to Web Distributed
              Authoring and Versioning (WebDAV)", RFC 6352,
              DOI 10.17487/RFC6352, August 2011,
              <https://www.rfc-editor.org/info/rfc6352>.

   [RFC6442]  Polk, J., Rosen, B., and J. Peterson, "Location Conveyance
              for the Session Initiation Protocol", RFC 6442,
              DOI 10.17487/RFC6442, December 2011,
              <https://www.rfc-editor.org/info/rfc6442>.

   [RFC6665]  Roach, A.B., "SIP-Specific Event Notification", RFC 6665,
              DOI 10.17487/RFC6665, July 2012,
              <https://www.rfc-editor.org/info/rfc6665>.

   [RFC6764]  Daboo, C., "Locating Services for Calendaring Extensions
              to WebDAV (CalDAV) and vCard Extensions to WebDAV
              (CardDAV)", RFC 6764, DOI 10.17487/RFC6764, February 2013,
              <https://www.rfc-editor.org/info/rfc6764>.

   [RFC6881]  Rosen, B. and J. Polk, "Best Current Practice for
              Communications Services in Support of Emergency Calling",
              BCP 181, RFC 6881, DOI 10.17487/RFC6881, March 2013,
              <https://www.rfc-editor.org/info/rfc6881>.

   [RFC7230]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
              Protocol (HTTP/1.1): Message Syntax and Routing",
              RFC 7230, DOI 10.17487/RFC7230, June 2014,
              <https://www.rfc-editor.org/info/rfc7230>.



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   [RFC7525]  Sheffer, Y., Holz, R., and P. Saint-Andre,
              "Recommendations for Secure Use of Transport Layer
              Security (TLS) and Datagram Transport Layer Security
              (DTLS)", BCP 195, RFC 7525, DOI 10.17487/RFC7525, May
              2015, <https://www.rfc-editor.org/info/rfc7525>.

   [RFC7647]  Sparks, R. and A.B. Roach, "Clarifications for the Use of
              REFER with RFC 6665", RFC 7647, DOI 10.17487/RFC7647,
              September 2015, <https://www.rfc-editor.org/info/rfc7647>.

   [RFC7742]  Roach, A.B., "WebRTC Video Processing and Codec
              Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016,
              <https://www.rfc-editor.org/info/rfc7742>.

   [RFC7852]  Gellens, R., Rosen, B., Tschofenig, H., Marshall, R., and
              J. Winterbottom, "Additional Data Related to an Emergency
              Call", RFC 7852, DOI 10.17487/RFC7852, July 2016,
              <https://www.rfc-editor.org/info/rfc7852>.

   [RFC7874]  Valin, JM. and C. Bran, "WebRTC Audio Codec and Processing
              Requirements", RFC 7874, DOI 10.17487/RFC7874, May 2016,
              <https://www.rfc-editor.org/info/rfc7874>.

   [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
              2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
              May 2017, <https://www.rfc-editor.org/info/rfc8174>.

   [RFC8445]  Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive
              Connectivity Establishment (ICE): A Protocol for Network
              Address Translator (NAT) Traversal", RFC 8445,
              DOI 10.17487/RFC8445, July 2018,
              <https://www.rfc-editor.org/info/rfc8445>.

   [RFC8446]  Rescorla, E., "The Transport Layer Security (TLS) Protocol
              Version 1.3", RFC 8446, DOI 10.17487/RFC8446, August 2018,
              <https://www.rfc-editor.org/info/rfc8446>.

   [RFC8599]  Holmberg, C. and M. Arnold, "Push Notification with the
              Session Initiation Protocol (SIP)", RFC 8599,
              DOI 10.17487/RFC8599, May 2019,
              <https://www.rfc-editor.org/info/rfc8599>.

   [RFC8760]  Shekh-Yusef, R., "The Session Initiation Protocol (SIP)
              Digest Access Authentication Scheme", RFC 8760,
              DOI 10.17487/RFC8760, March 2020,
              <https://www.rfc-editor.org/info/rfc8760>.





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   [RFC8825]  Alvestrand, H., "Overview: Real-Time Protocols for
              Browser-Based Applications", RFC 8825,
              DOI 10.17487/RFC8825, January 2021,
              <https://www.rfc-editor.org/info/rfc8825>.

   [RFC8827]  Rescorla, E., "WebRTC Security Architecture", RFC 8827,
              DOI 10.17487/RFC8827, January 2021,
              <https://www.rfc-editor.org/info/rfc8827>.

   [RFC8829]  Uberti, J., Jennings, C., and E. Rescorla, Ed.,
              "JavaScript Session Establishment Protocol (JSEP)",
              RFC 8829, DOI 10.17487/RFC8829, January 2021,
              <https://www.rfc-editor.org/info/rfc8829>.

   [RFC8834]  Perkins, C., Westerlund, M., and J. Ott, "Media Transport
              and Use of RTP in WebRTC", RFC 8834, DOI 10.17487/RFC8834,
              January 2021, <https://www.rfc-editor.org/info/rfc8834>.

   [RFC8835]  Alvestrand, H., "Transports for WebRTC", RFC 8835,
              DOI 10.17487/RFC8835, January 2021,
              <https://www.rfc-editor.org/info/rfc8835>.

   [RFC8839]  Petit-Huguenin, M., Nandakumar, S., Holmberg, C., Keränen,
              A., and R. Shpount, "Session Description Protocol (SDP)
              Offer/Answer Procedures for Interactive Connectivity
              Establishment (ICE)", RFC 8839, DOI 10.17487/RFC8839,
              January 2021, <https://www.rfc-editor.org/info/rfc8839>.

   [RFC8865]  Holmberg, C. and G. Hellström, "T.140 Real-Time Text
              Conversation over WebRTC Data Channels", RFC 8865,
              DOI 10.17487/RFC8865, January 2021,
              <https://www.rfc-editor.org/info/rfc8865>.

   [RFC8866]  Begen, A., Kyzivat, P., Perkins, C., and M. Handley, "SDP:
              Session Description Protocol", RFC 8866,
              DOI 10.17487/RFC8866, January 2021,
              <https://www.rfc-editor.org/info/rfc8866>.

   [RFC9071]  Hellström, G., "RTP-Mixer Formatting of Multiparty Real-
              Time Text", RFC 9071, DOI 10.17487/RFC9071, July 2021,
              <https://www.rfc-editor.org/info/rfc9071>.

13.
Informative References







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   [OpenApi]  Miller, D., Whitlock, J., Gardiner, M., Ralpson, M.,
              Ratovsky, R., and U. Sarid, "OpenAPI Specification
              v3.0.1", December 2017,
              <https://spec.openapis.org/oas/v3.0.1>.

   [RFC3665]  Johnston, A., Donovan, S., Sparks, R., Cunningham, C., and
              K. Summers, "Session Initiation Protocol (SIP) Basic Call
              Flow Examples", BCP 75, RFC 3665, DOI 10.17487/RFC3665,
              December 2003, <https://www.rfc-editor.org/info/rfc3665>.

   [RFC8126]  Cotton, M., Leiba, B., and T. Narten, "Guidelines for
              Writing an IANA Considerations Section in RFCs", BCP 26,
              RFC 8126, DOI 10.17487/RFC8126, June 2017,
              <https://www.rfc-editor.org/info/rfc8126>.

Acknowledgements

   Brett Henderson and Jim Malloy provided many helpful edits to prior
   versions of this document.

Author's Address

   Brian Rosen
   470 Conrad Dr
   Mars, PA 16046
   United States of America

   Phone: +1 724 382 1051
   Email: br@brianrosen.net






















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