Interoperability Profile for Relay User Equipment
draft-ietf-rum-rue-11
| Document | Type | Active Internet-Draft (rum WG) | |
|---|---|---|---|
| Author | Brian Rosen | ||
| Last updated | 2022-04-01 (Latest revision 2022-02-17) | ||
| Replaces | draft-rosen-rue | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Intended RFC status | Proposed Standard | ||
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| Stream | WG state | Submitted to IESG for Publication | |
| Document shepherd | Paul Kyzivat | ||
| Shepherd write-up | Show Last changed 2022-01-21 | ||
| IESG | IESG state | RFC Ed Queue | |
| Action Holders |
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| Consensus boilerplate | Yes | ||
| Telechat date | (None) | ||
| Responsible AD | Murray Kucherawy | ||
| Send notices to | pkyzivat@alum.mit.edu | ||
| IANA | IANA review state | Version Changed - Review Needed | |
| IANA action state | RFC-Ed-Ack | ||
| RFC Editor | RFC Editor state | RFC-EDITOR | |
| Details |
draft-ietf-rum-rue-11
rum B. Rosen
Internet-Draft 17 February 2022
Intended status: Standards Track
Expires: 21 August 2022
Interoperability Profile for Relay User Equipment
draft-ietf-rum-rue-11
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/hard of hearing/
hearing impaired 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/hard of hearing/hearing impaired 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
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
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 21 August 2022.
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Copyright Notice
Copyright (c) 2022 IETF Trust and the persons identified as the
document authors. All rights reserved.
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 Revised BSD License text as
described in Section 4.e of the Trust Legal Provisions and are
provided without warranty as described in the Revised BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Requirements Language . . . . . . . . . . . . . . . . . . . . 6
4. General Requirements . . . . . . . . . . . . . . . . . . . . 6
5. SIP Signaling . . . . . . . . . . . . . . . . . . . . . . . . 7
5.1. Registration . . . . . . . . . . . . . . . . . . . . . . 8
5.2. Session Establishment . . . . . . . . . . . . . . . . . . 10
5.2.1. Normal Call Origination . . . . . . . . . . . . . . . 10
5.2.2. Dial-Around Origination . . . . . . . . . . . . . . . 10
5.2.3. RUE Contact Information . . . . . . . . . . . . . . . 12
5.2.4. Incoming Calls . . . . . . . . . . . . . . . . . . . 12
5.2.5. Emergency Calls . . . . . . . . . . . . . . . . . . . 12
5.3. Mid-Call Signaling . . . . . . . . . . . . . . . . . . . 13
5.4. URI Representation of Phone Numbers . . . . . . . . . . . 13
5.5. Transport . . . . . . . . . . . . . . . . . . . . . . . . 14
6. Media . . . . . . . . . . . . . . . . . . . . . . . . . . . . 14
6.1. SRTP and SRTCP . . . . . . . . . . . . . . . . . . . . . 14
6.2. Text-Based Communication . . . . . . . . . . . . . . . . 15
6.3. Video . . . . . . . . . . . . . . . . . . . . . . . . . . 15
6.4. Audio . . . . . . . . . . . . . . . . . . . . . . . . . . 15
6.5. DTMF Digits . . . . . . . . . . . . . . . . . . . . . . . 15
6.6. Session Description Protocol . . . . . . . . . . . . . . 15
6.7. Privacy . . . . . . . . . . . . . . . . . . . . . . . . . 16
6.8. Negative Acknowledgment, Packet Loss Indicator, and Full
Intraframe Request Features . . . . . . . . . . . . . . . 16
7. Contacts . . . . . . . . . . . . . . . . . . . . . . . . . . 16
7.1. CardDAV Login and Synchronization . . . . . . . . . . . . 16
7.2. Contacts Import/Export Service . . . . . . . . . . . . . 17
8. Video Mail . . . . . . . . . . . . . . . . . . . . . . . . . 17
9. Provisioning and Provider Selection . . . . . . . . . . . . . 18
9.1. RUE Provider Selection . . . . . . . . . . . . . . . . . 18
9.2. RUE Configuration Service . . . . . . . . . . . . . . . . 20
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9.2.1. Provider Configuration . . . . . . . . . . . . . . . 21
9.2.2. RUE Configuration . . . . . . . . . . . . . . . . . . 21
9.2.3. Versions . . . . . . . . . . . . . . . . . . . . . . 23
9.2.4. Examples . . . . . . . . . . . . . . . . . . . . . . 24
9.2.5. Using the Provider Selection and RUE Configuration
Services Together . . . . . . . . . . . . . . . . . . 25
9.3. OpenAPI Interface Descriptions . . . . . . . . . . . . . 26
9.3.1. Provider List . . . . . . . . . . . . . . . . . . . . 26
9.3.2. Configuration . . . . . . . . . . . . . . . . . . . . 27
10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 33
10.1. RUE Provider List Registry . . . . . . . . . . . . . . . 33
10.2. Registration of rue-owner Value of the purpose
Parameter . . . . . . . . . . . . . . . . . . . . . . . 33
11. Security Considerations . . . . . . . . . . . . . . . . . . . 34
12. Normative References . . . . . . . . . . . . . . . . . . . . 34
13. Informative References . . . . . . . . . . . . . . . . . . . 40
Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . . 41
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 41
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.
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The following illustrates a typical relay call. The RUE device and
the Commincations Assistant (sign language interpretter) have
videophones. The Hearing User has a telephone (mobile or fixed).
||== RUE Interface (this document)
||
\/
+----+ +------+ - +--------+ - +-------+
|Deaf| |RUE | ( ) |Provider| ( ) |Hearing|
|User|<->|Device|<-(Internet)->| |<-( PSTN )->|User |
+----+ +------+ -------- +--------+ ------ +-------+
^
|
+--------------+
|Communications|
|Assistant |
+--------------+
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
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
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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 conventional voice
services with the aid of a communication assistant (CA) to relay the
communication.
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 smartphone; or proprietary equipment
connected to a server that provides the RUE interface.
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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
conventional 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.
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.
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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)
* [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
RFC6665 (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)
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* [RFC3326] (Reason header field)
* [RFC3515] (REFER Method)
* [RFC3891] (Replaces Header field)
* [RFC3892] (Referred-By)
Implementations MUST implement full ICE, although they MAY interwork
with User Agents that implement ICE-lite.
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.
The SIP signaling for registration and placing/receiving calls
depends on configuration of various values into the RUE device.
Section 9.2 describes the configuration mechanism which provides
values that are used in the signaling. When the device starts, the
configuration mechanism is run which retrieves the configuration
data, and then SIP registration occurs using the values from the
configuration. After registration, calls may be sent or received by
the RUE device.
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" in
"RueConfigurationData", 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 follows:
* if "user-name" is provided:"username@provider-domain";
* if "user-name" is not provided: as specified in Section 5.4, using
"phone-number" and "provider-domain" from the configuration.
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The RUE determines the URI to resolve by initially determining if one
or more outbound proxies are+ configured. If there are, the URI will
be that of one of the "outbound-proxies". If no "outbound-proxies"
are 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 the RUE using SIP digest
authentication. The credentials to be used MUST come from the
configuration Section 9.2: "user-name" if provided or "phone-number"
if user-name is not provided, and "sip-password". This "user-
name"/"sip-password" combination SHOULD NOT be the same as that used
for other purposes, except as expressly described below, 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).
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.
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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:+15551234567@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 requirements specified in Section 6
of this document.
To allow time to time out 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.
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
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"frontDoor" URI (see Section 9.2) in "ProviderConfigurationData" of
the dial-around provider. The RUE Provider Selection service
(Section 9.1) can be used by the RUE to obtain a list of providers
and then the Provider Configuration (Section 9.2.1) 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
,-+-. ,----+----. ,-----+-----.
|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:+15552220001@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:+15552220001@red.example.net;user=phone
P-Asserted-Identity: sip:+15552220001@red.example.net
Figure 1
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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 the RUE uses to accept a call, 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-ID 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"). The RUE owner is the entity that has
local control over the device which is not necessarily the legal
owner of the equipment. It often is the user, but that is not
necessarily true. While no minimum fields in the xCard are
specified, the name, address, phone number and email address of the
RUE owner are expected to be supplied.
5.2.4. Incoming Calls
The RUE MUST only accept inbound calls sent to it by a proxy
mentioned in the configuration.
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
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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 Information 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.
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.
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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. Various RTP, RTCP, SDP and
specific media requirements specified for WebRTC are adopted for this
document. Explicit requirements from the WebRTC suite of documents
are described below .
To use WebRTC with this document, a gateway that presents a WebRTC
server interface towards a browser, and a RUE client interface
towards a provider is assumed. The gateway would interwork
signaling, and as noted below, interwork at least any real time text
media, in order to allow a standard browser based WebRTC client to be
a VRS client. The combination of the browser client and the gateway
would be a RUE user. This document does not specify the gateway.
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
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. Conformance to this document allows end-to-end RTCP and
media congestion control for audio and video.
6.1. SRTP and SRTCP
Implementations MUST support [RFC8834] except that MediaStreamTracks
are not used. Implementations MUST conform to Section 6.4 of
[RFC8827].
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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 channel. 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].
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].
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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 states in the network (e.g. 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]).
The configuration (see Section Section 9.2) RueConfigurationData MAY
supply a "carddav-username" and "carddav-domain" identifying a
CardDAV server and address book for this account, plus an optional
"carddav-password".
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To access the CardDAV server and address book, the RUE MUST follow
Section 6 of RFC6764, using the configured carddav-username and
carddav-domain in place of an email address. If the request triggers
a challenge for digest authentication credentials, the RUE MUST
continue using matching carddav-username and carddav-password from
the configuration. If no carddav-username and carddav-password are
configured, the RUE MUST use the SIP user-name and sip-password 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.
The RUE MAY support other CardDAV services.
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 the "contacts-username" and "contacts-password" from
the configuration. If no contacts-username is configured, the sip
user-name from the configuration is used, and if the sip user-name is
not configured, the phone-number is used. If user-name or phone-
number is used, the sip-password is used to authenticate to the
contact list server.
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 mechanisms MUST be
supported. RUE devices MUST support both. See Section 9.2 for how
the URI to reach the retrieval interface is obtained.
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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.
The "videomail" and "mwi" properties in the configuration (see
RueConfigurationData in Section 9.2.2) gives the URIs for message
retrieval and "message-summary" subscription.
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 (See
Section 10.1). The entry point, when used with the following OpenAPI
interface, returns a list of provider names for a country code
suitable for display, with a corresponding entry point to obtain
information about that provider. No mechanism to determine the
country the RUE is located is specified in this document. Typically
the country is the home country of the user, but may be a local
country while traveling. Some countries allow support from their
home country when traveling abroad. Regardless, the RUE device will
need to allow the user to choose the country.
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 includes an occurance of
"localhost". The value from the registry of the "list entry point"
string for the desired country is substituted for "localhost" in the
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"servers" component to obtain the server URI prefix of the interface
to be used to obtain the list of providers for that country. The
"Providers" path then specifies the rest of the URI used to obtain
the list. For example, if the list entryPoint is "example.com/api",
the provider list would be obtained from
https://example.com/api/rum/v1/Providers.
The V1.0 "ProviderList" 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.
* providerEntryPoint: A string used for configuration purposes by
the RUE (as discussed in Section 9.2). The string MUST start with
a domain, but MAY include other URI path elements after the
domain.
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 ProviderList JSON object
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9.2. RUE Configuration Service
A RUE device may retrieve a provider configuration using a simple
HTTPs web service. There are two entry points. One is used without
user credentials, and the response includes configuration data for
new user sign up and dial around. The other uses locally stored
username and password that results from a new user sign up 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 string includes an occurence of "localhost". The
entry point string 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 "red.example.net", the provider configuration would be
obtained from https://red.example.net/rum/V1/ProviderConfig and the
RUE configuration would be obtained from
https://red.example.net/rum/V1/RueConfig.
In both the queries, an optional parameter may be provided to the
interface which is an API Key (apiKey). The implementation MAY have
an apiKey obtained from the provider and specific to the
implementation. The method used to obtain the apiKey is not
specified in this document. The provider MAY refuse to provide
service to an implementation presenting an apiKey it does not
recognize.
Also in both queries, the RUE device provides a client provided,
required parameter, which contains an instance identifier
(instanceId). This parameter MUST be the same value each time this
instance (same implementation on same device) queries the interface.
This MAY be used by the provider, for example, to associate a
location with the instance for emergency calls. This should be
globally unique. A UUID is suggested.
For example, a query for the RUE configuration could be
https://red.example.net/rum/V1/RueConfig?apiKey="t65667Ajjss90uuuDisK
t8999"&instanceId="5595b5a3-0687-4b8e-9913-a7f2a04fb7bd"
The data returned is a JSON object consisting of key/value
configuration parameters to be used by the RUE.
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.
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9.2.1. Provider Configuration
* signup: (OPTIONAL) an array of JSON objects consisting of:
- language: entry from the IANA language subtag registry
(https://www.iana.org/assignments/language-subtag-registry/
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 account credentials (e.g. username and password),
which would be manually entered into the RUE device which form
the authentication parameters for the RUE configuration service
described below in Section 9.2.2.
* 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:
- 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
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* 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 against unauthorized disclosure
(e.g. by other applications on the platform the RUE is built on).
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 it MUST
cryptographically protect against unauthorized disclosure (e.g. by
other applications on the platform the RUE is built on). 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, as well as STUN and TURN
servers mentioned in this configuration.
* 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.
* user-name: (optional) a username used for authenticating to the
provider. If not provided, the phone-number is used.
* display-name: (optional) a human readable display name for the
subscriber
* provider-domain: the domain for the provider. This becomes the
server portion of the SIP URI identifying the subscriber.
* outbound-proxies: (optional) An array of URIs of SIP proxies 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 used to
retrieve video mail messages.
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* contacts: (optional) An HTTPS URI ("contacts-uri"), (optional)
"contacts-username" and "contacts-password" that may be used to
export (retrieve) the subscriber's complete contact list managed
by the provider. At least the URI MUST be provided if the
subscriber has contacts. If contact-username and contacts-
password are not supplied, the sip credentials are used. If the
contacts-username is provided, contacts-password MUST be provided.
If contacts-password is provided, contacts-username MUST be
provided.
* carddav: (optional) An address ("carddav-domain"), (optional)
"carddav-username" and "carddav-password" 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
carddav-username and carddav-password are not supplied, the sip
credentials are used. If the carddav-username is provided,
carddav-password MUST be provided. If carddav-password is
provided, carddav-username MUST be provided.
* 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 server types and URLs
identifying STUN and TURN servers available for use by the RUE for
establishing media streams in calls via the provider. If the same
URL provides both STUN and TURN services, it MUST be listed twice,
each with different server types.
9.2.3. Versions
Both web services also have 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. Backwards
compatibility applies to both the server and the client. Either may
have any higher or lower minor revision and interoperate with its
counterpart with the same major version. 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 non-mandatory-to use
functions and SHALL NOT delete or change 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 version mechanism returns
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an array of supported versions, one for each major version supported,
with the minor version listed being the highest supported minor
version.
Unless the per-country provider list service is operated by a
provider at the same base URI as that provider's configuration
service, the version of the configuration service MAY be different
from the version of the provider list service.
{
"versions": [
{
"major": 1,
"minor": 6
},
{
"major": 2,
"minor": 13
},
{
"major": 3,
"minor": 2
}
]
}
Figure 3: Example of a Version JSON object
9.2.4. Examples
Example JSON provider configuration payload
{
"signUp": [
{ "language" : "en", "uri" : "https:hello-en.example.net"},
{ "language" : "es", "uri" : "https:hello-es.example.net"} ] ,
"dialAround": [
{ "language" : "en", "frontDoor" : "sip:fd-en.example.net",
"oneStage" : "sip:1stg-eng.example.com" } ,
{ "language" : "es", "frontDoor" : "sip:fd-es.example.net",
"oneStage" : "sip:1stg-spn.example.com" } ] ,
"helpDesk": [
{ "language" : "en", "uri" : "sip:help-en.example.net"} ,
{ "language" : "es", "uri" : "sip:help-es.example.net"} ]
}
Figure 4
Example JSON RUE configuration payload
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{
"lifetime": 86400,
"display-name" : "Bob Smith",
"phone-number": "+15551234567",
"provider-domain": "red.example.net",
"outbound-proxies": [
"sip:p1.red.example.net",
"sip:p2.red.example.net" ],
"mwi": "sip:+15551234567@red.example.net;user=phone",
"videomail": "sip:+15551234567@vm.red.example.net;user=phone",
"contacts": {
"contacts-uri":
"https://red.example.net:443/c/3617b719-2c3a-46f4-9c13",
"contacts-username": "bob",
"contacts-password": "XhOT4ch@ZEi&3u2xEYQNMO^5UGb"
},
"carddav": {
"carddav-domain": "carddav.example.com",
"carddav-username": "bob",
"carddav-password": "sj887%dd*jJty%87hyys5hHT"
},
"sendLocationWithRegistration": false,
"ice-servers": [
{"stun":"stun.red.example.com:19302" },
{"turn":"turn.red.example.com:3478"}
]
}
Figure 5
9.2.5. 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, if
requested by the user, use the ProviderConfig path without
credentials to obtain signup, dial around and helpdesk
information.
- If the RUE has credentials for that provider, use the RueConfig
path with the locally stored credentials to configure the RUE
for that provider.
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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.
The OpenAPI description below is normative. If there is any conflict
between the text or examples and this section, the OpenAPI
description takes precedence.
9.3.1. Provider List
openapi: 3.0.1
info:
title: RUM Provider List API
version: "1.0"
servers:
- url: https://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'
/Versions:
servers:
- url: https://localhost/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
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required:
- providers
properties:
providers:
type: array
items:
type: object
required:
- name
- providerEntryPoint
properties:
name:
type: string
description: Human readable provider name
providerEntryPoint:
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
Figure 6: Provider List OpenAPI description (RueProviderList.yaml)
9.3.2. Configuration
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openapi: 3.0.1
info:
title: RUM Configuration API
version: "1.0"
servers:
- url: https://localhost/rum/v1
paths:
/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:
- 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
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on this device
responses:
'200':
description: configuration object
content:
application/json:
schema:
$ref: '#/components/schemas/RueConfigurationData'
/Versions:
servers:
- url: https://localhost/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:
ProviderConfigurationData:
type: object
required:
- dialAround
properties:
signup:
type: array
items:
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
this language
dialAround:
type: array
items:
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type: object
required:
- language
- frontDoor
- 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: array
items:
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
- provider-domain
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 in
E.164 format
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user-name:
type: string
description: a username assigned this subscriber.
display-name:
type: string
description: display name for the subscriber
provider-domain:
type: string
description: domain of the provider for this subscriber
outbound-proxies:
type: array
items:
type: string
format: uri
description: SIP URI of a proxy to be used when sending
requests to the provider
mwi:
type: string
format: uri
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 used to
retrieve video mail messages.
contacts:
type: object
description: server and credentials for contact
import/export
required:
- contacts-uri
properties:
contacts-uri:
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.
contacts-username:
type: string
description: username for authentication with CardDAV
server. Use sip user-name if not provided
contacts-password:
type: string
description: password for authentication. Use provider
sip-password if not provided
carddav:
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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.
required:
- carddav-domain
properties:
carddav-domain:
type: string
description: CardDAV server address
carddav-username:
type: string
description: username for authentication with CardDAV
server. Use sip user-name if not provided
carddav-password:
type: string
description: password for authentication to the CardDAV
server. Use provider sip-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: object
required:
- server-type
- uri
properties:
server-type:
type: string
description: server type ("stun" or "turn")
uri:
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.
VersionsArray:
type: object
required:
- versions
properties:
versions:
type: array
items:
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type: object
required:
- major
- minor
properties:
major:
type: integer
format: int32
description: Version major number
minor:
type: integer
format: int32
description: Version minor number
Figure 7: Configuration OpenAPI description (RueConfiguration.yaml)
10. IANA Considerations
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. The
use for rue-owner is defined in Section 5.2.3. 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
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* 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.
This document requires implementation and use of a number of other
specifications in order to fulfill the RUE profile; the security
considerations described in those documents apply accordingly to the
RUE interactions.
When a CA participates in a conversation they have access to the
content of the conversation even though it is nominally a
conversation between the two endpoints. There is an expectation that
the CA will keep the communication contents in confidence. This is
usually defined by contractual or legal requirements.
Since different providers (within a given country) may have different
policies, RUE implementations MUST include a user interaction step to
select from available providers before proceeding to actually
register with any given provider.
12.
Normative References
[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>.
[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>.
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[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>.
[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>.
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[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>.
[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>.
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[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>.
[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>.
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[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>.
[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>.
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[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>.
[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>.
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[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
[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>.
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Acknowledgements
Brett Henderson and Jim Malloy provided many helpful edits to prior
versions of this document. Paul Kyzivat provided extensive reviews
and comments.
Author's Address
Brian Rosen
470 Conrad Dr
Mars, PA 16046
United States of America
Email: br@brianrosen.net
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