Internet DRAFT - draft-rosen-rue
draft-rosen-rue
Internet Engineering Task Force B. Rosen
Internet-Draft
Intended status: Standards Track J. Malloy
Expires: February 8, 2020 B. Henderson
The MITRE Corporation
August 7, 2019
Interoperability Profile for Relay User Equipment
draft-rosen-rue-01
Abstract
This document identifies a minimum set of standards and requirements
that must be supported by a Video Relay Service (VRS) Video Access
Technology Reference Platform (VATRP)-compliant client and United
States Telecommunications Relay Service providers required to be
VATRP compliant. This Relay User Equipment specification only
specifies a minimum set of requirements. It does not prohibit VRS
providers or endpoint developers from developing or deploying
additional capabilities, provided that doing so will not prevent
compliance with the requirements specified here.
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 February 8, 2020.
Copyright Notice
Copyright (c) 2019 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
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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. Scope . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
4. Requirements Language . . . . . . . . . . . . . . . . . . . . 6
5. General Requirements . . . . . . . . . . . . . . . . . . . . 6
6. SIP Signaling . . . . . . . . . . . . . . . . . . . . . . . . 6
6.1. Registration . . . . . . . . . . . . . . . . . . . . . . 7
6.2. Session Establishment . . . . . . . . . . . . . . . . . . 8
6.2.1. Normal Call Origination . . . . . . . . . . . . . . . 8
6.2.2. One-Stage Dial-Around Origination . . . . . . . . . . 9
6.2.3. RUE Contact Information . . . . . . . . . . . . . . . 10
6.2.4. Incoming Calls . . . . . . . . . . . . . . . . . . . 10
6.2.5. Emergency Calls . . . . . . . . . . . . . . . . . . . 11
6.3. Mid Call Signaling . . . . . . . . . . . . . . . . . . . 11
6.4. URI Representation of Phone Numbers . . . . . . . . . . . 12
6.5. Transport . . . . . . . . . . . . . . . . . . . . . . . . 12
7. Media . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
7.1. SRTP and SRTCP . . . . . . . . . . . . . . . . . . . . . 12
7.2. Text-Based Communication . . . . . . . . . . . . . . . . 13
7.3. Video . . . . . . . . . . . . . . . . . . . . . . . . . . 13
7.4. Audio . . . . . . . . . . . . . . . . . . . . . . . . . . 13
7.5. DTMF Digits . . . . . . . . . . . . . . . . . . . . . . . 13
7.6. Session Description Protocol . . . . . . . . . . . . . . 13
7.7. Privacy . . . . . . . . . . . . . . . . . . . . . . . . . 13
7.8. Negative Acknowledgment, Packet Loss Indicator, and Full
Intraframe Request Features . . . . . . . . . . . . . . . 13
8. Contacts . . . . . . . . . . . . . . . . . . . . . . . . . . 14
8.1. CardDAV Login and Synchronization . . . . . . . . . . . . 14
8.2. Contacts Import/Export Service . . . . . . . . . . . . . 14
9. Mail Waiting Indicator (MWI) . . . . . . . . . . . . . . . . 15
10. Provisioning and Provider Selection . . . . . . . . . . . . . 15
10.1. RUE Provider Selection . . . . . . . . . . . . . . . . . 15
10.2. RUE Configuration Service . . . . . . . . . . . . . . . 16
10.3. Schemas . . . . . . . . . . . . . . . . . . . . . . . . 19
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 22
12. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 22
13. Security Considerations . . . . . . . . . . . . . . . . . . . 22
14. Normative References . . . . . . . . . . . . . . . . . . . . 22
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 28
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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.
This RUE profile supports the requirements of relay services in the
United States, as described in 47 CFR 64.601 et seq., but may be
applicable to similar uses elsewhere.
2. Scope
This RUE Specification documents the standards and controls
associated with the Video Access Technology Reference Platform
(VATRP). This RUE specification identifies the minimum set of
standards for the interface between the VATRP and Providers' networks
to which the VATRP adheres. This RUE Specification does not prohibit
the implementation of additional features or functionality by any
Provider. It also contains some Provider-optional features. If a
Provider offers the feature described by the optional specification
on at least one endpoint, the Provider MUST supply the standardized
interface described in this document for that feature. This edition
of the RUE specification does not address Provider-to-Provider
communication (covered in the US VRS Provider Interface Profile) or
the user interface to the RUE.
3. Terminology
Communication Assistant (CA): The ASL interpreter stationed in a TRS-
registered call center working for a VRS Provider, acting as part of
the wire of a call to provide 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-
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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. A subscriber's default Provider provides the
VRS that handles incoming relay calls to the user. The default
Provider also handles outgoing relay calls by default.
Dial-around call: A relay call where the subscriber specifies the use
of a VRS provider other than one of the Providers with whom the
subscriber is registered. 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").
Full Intra Request (FIR): A request to a 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".
NANP: North America Numbering Plan (please refer to:
http://nationalnanpa.org).
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. Please refer to FCC-VRS-GUIDE.
Relay number database (RND): The iTRS Relay Number Database (RND)
functions as a 10-digit NANP phone number lookup for SIP and H.323
URLs for TRS subscribers.
Relay-to-relay call: A call between two subscribers each using
different forms of relay (video relay, IP relay, TTY), each with a
separate CA to assist in relaying the conversation.
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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 assigned
to the RUE in ITU-T E.164 format.
Relay user equipment (RUE): A SIP user agent (UA) enhanced with extra
features to support a subscriber in requesting 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.
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.
Telecommunications relay services (TRS): Telephone transmission
services that provide the ability for an individual who has a hearing
impairment or speech impairment to engage in communication by wire or
radio with a hearing individual in a manner that is functionally
equivalent to the ability of an individual who does not have a
hearing impairment or speech impairment to communicate using voice
communication services by wire or radio. TRS includes services that
enable two-way communication between an individual who uses a
Telecommunications Device for the Deaf (TDD) or other non-voice
terminal device and an individual who does not use such a device.
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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.
4. Requirements Language
The keywords "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119]
5. General Requirements
All HTTP/HTTPS connections specified throughout this document MUST
use HTTPS. Both HTTPS and all SIP connections MUST use TLA
conforming to [RFC7525]
During the establishment of secure connections with a provider, the
RUE MAY be asked by the server for a client certificate. In that
case it SHOULD provide the provisioned client certificate (See
Section 10.2. Providers MAY reject requests that fail to provide a
recognized certificate.
All text data payloads not otherwise constrained by a specification
in another standards document MUST be encoded as Unicode UTF/8.
6. SIP Signaling
The RUE and Providers MUST conform to the following core SIP
standards [RFC3261] (Base SIP) [RFC3263] (Locating SIP Servers),
[RFC3264] (Offer/Answer), [RFC3840] (User Agent Capabilities),
[RFC5626] (Outbound), [RFC4566] (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), and [RFC6442] (Geolocation Header).
In addition, the RUE MUST, and Providers MAY, conform to [RFC3327]
(Path), [RFC5245] (ICE), [RFC3326] (Reason header), [RFC3515] (REFER
Method), [RFC3891] (Replaces Header), [RFC3892] (Referred-By).
RUEs 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.
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6.1. Registration
The RUE MUST register with a SIP registrar, following [RFC3261] and
[RFC5626]. If the configuration (please refer to Section 11)
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 13, 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.netv" 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 MD5 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. Because MD5 is considered
insecure, [I-D.yusef-sipcore-digest-scheme] SHOULD be implemented by
both the RUE and Providers and SHA-based digest algorithms SHOULD be
used for digest authentication.
If the registration request fails with an indication that credentials
from the configuration are invalid, then the RUE SHOULD 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 SHOULD inform the RUE User of the
problem.
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Support for multiple simultaneous registrations by Providers is
OPTIONAL, as described in Section 2.
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.
6.2. Session Establishment
6.2.1. Normal Call Origination
After initial SIP registration, the RUE adheres to SIP [RFC3261]
basic call flows, as documented in [RFC3665].
The RUE MUST route all calls through the outbound proxy of the
default Provider.
INVITE requests used to initiate calls SHOULD NOT contain Route
headers. Route headers MAY be included in one-stage dial-around
calls and emergency calls. The SIP URIs in the To field and the
Request-URI MUST be formatted as specified in subsection 6.4 using
the destination phone number. The domain field of the URIs SHOULD be
the "provider-domain" from the configuration (e.g.,
sip:+13115552368@red.example.com;user=phone). The same exceptions
apply, including anonymous calls.
Anonymous calls MUST be supported by both the RUE and Providers. An
anonymous call is signaled per [RFC3323].
The From-URI MUST be formatted as specified in Section 6.4, 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 10.2.)
Negotiated media MUST follow the guidelines specified in Section 7 of
this document.
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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.
6.2.2. One-Stage Dial-Around Origination
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
URI of the dial-around Provider and the domain of the URI is the
Provider domain from the configuration.
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 6.2.1 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 10.1.
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:
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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>
Route: sip:green.example.net
[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
One Stage Dial-Around
6.2.3. 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 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 an 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").
6.2.4. Incoming Calls
The RUE MUST accept inbound calls sent to it by the 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.
6.2.5. Emergency Calls
The RUE MUST comply with [RFC6881] for handling of emergency calls.
Providers MAY comply with RFC6881 for handling of emergency calls.
In addition, they MUST:
o Accept RUE emergency calls complying with the specifications in
this document;
o Recognize such calls as emergency calls and properly handle them
as such;
o Address other behavior not specified by RFC6881 as specified in
Section 6.2.
Specifically, if the emergency call is to be handled using E9-1-1
(VPC) procedures, the Provider is responsible for modifying the
INVITE to conform to the VPC requirements. In this case, location
MAY be extracted from the RFC6881 conformant INVITE and used to
propagate it to the VPC where possible with the emergency call.
Because the RUE may have a more accurate and timely location of the
device than the typical manual entry location for nomadic RUE
devices, the RUE MUST send a Geolocation header containing its
location in the REGISTER request if the configuration specifies it.
The Provider MAY use that information to populate the location of the
device in the VPC before any emergency call.
6.3. Mid Call Signaling
The RUE and Providers MUST support re-INVITE to renegotiate media
session parameters (among other uses). Per Section 7.1, the RUE
MUST, and providers SHOULD, be able to support an INFO request for
full frame refresh for devices in a call with the RUE that do not
support RTCP mechanisms (please refer to Section 7.8). The RUE MUST
support an in-dialog REFER ([RFC3515] updated by [RFC7647] and
including support for norefersub per [RFC4488]) with the Replaces
header [RFC3891] to enable call transfer.
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6.4. 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.
A dial string that can be written as an E.164 formatted 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.
Dial strings that cannot be written 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 (using resolve (in accord with [RFC3263]) to globally
routable IPv4 addresses. The AoRs MAY also resolve to IPv6
addresses.
6.5. Transport
The RUE and providers MUST conform to [I-D.ietf-rtcweb-transports]
with the understanding that this specification does not use the
WebRTC data channel.
The RUE and providers MUST support SIP outbound [RFC5626] (please
also refer to Section 6.1).
7. Media
This specification adopts the media specifications for WebRTC
([I-D.ietf-rtcweb-overview]). 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 following sections specify the WebRTC documents to
which conformance is required.
7.1. SRTP and SRTCP
The RUE and Providers MUST support [I-D.ietf-rtcweb-rtp-usage] with
the understanding that RUE does not specify an API and therefore
MediaStreamTracks are not used. Implementations MUST conform to
Section 6.4 of [I-D.ietf-rtcweb-security-arch].
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7.2. Text-Based Communication
The RUE MUST and Providers 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. Note that this 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.
7.3. Video
The RUE and Providers MUST conform to [RFC7742].
7.4. Audio
The RUE and Providers MUST conform to [RFC7874].
7.5. DTMF Digits
The RUE and Providers 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.
7.6. Session Description Protocol
The SDP offers and answers MUST conform [I-D.ietf-rtcweb-jsep] with
the understanding that the RUE uses SIP transport for SDP.
7.7. Privacy
The RUE MUST be able to control privacy of the user by implementing a
one-way mute of audio and or video, without signaling, locally, but
MUST maintain any NAT bindings by periodically sending media packets
on all active media sessions containing silence/comfort noise/black
screen/etc. per [RFC6263].
7.8. Negative Acknowledgment, Packet Loss Indicator, and Full
Intraframe Request Features
NACK SHOULD be used when negotiated and conditions warrant its use.
Signaling picture losses as Packet Loss Indicator (PLI) SHOULD be
preferred, as described in [RFC5104].
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.
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For backwards compatibility with calling devices that do not support
the foregoing methods, the RUE MUST implement and use SIP INFO
messages to send and receive XML encoded Picture Fast Update messages
according to [RFC5168].
8. Contacts
8.1. CardDAV Login and Synchronization
Support of CardDAV by Providers is OPTIONAL, as described in
Section 2.
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 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 attempt to 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.
8.2. Contacts Import/Export Service
Each Provider MUST supply a standard xCard import/export interface
and the RUE 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.
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9. Mail Waiting Indicator (MWI)
Support of MWI by Providers is OPTIONAL, as described in Section 2
The RUE MUST and Providers SHOULD support subscriptions to "message-
summary" events [RFC3842] to the URI specified in the configuration
if the Provider supports message waiting indicator on any endpoint.
In notification bodies, videomail messages SHOULD be reported using
"message-context-class multimedia-message" defined in [RFC3458].
10. Provisioning and Provider Selection
10.1. RUE Provider Selection
To allow the user to select a relay service, the RUE MAY obtain, on
startup, a list of Providers from a configured accessible URL.
The provider list, formatted as JSON, contains:
o Version: Specifies the version number of the Provider list format.
A new version number SHOULD only be used if the new version is not
backwards-compatible with the older version. A new version number
is not needed if new elements are optional and can be ignored by
older implementations.
o Providers: 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.
* domain: The domain parameter is used for configuration purposes
by the RUE (as discussed in Section 10.2) and as the domain to
use when targeting one-stage dial-around calls to this Provider
(as discussed in Section 6.2.2).
* operator: (OPTIONAL) The operator parameter is a SIP URL that
identifies the operator "front-door" that VRS users may contact
for manual (two-stage) dial-around calls.
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.
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{
"version": 1,
"providers": [
{
"name": "Red",
"domain": "red.example.net",
"operator": "sip:operator@red.example.net"
},
{
"name": "Green",
"domain": "green.example.net",
"operator": "sip:+18885550123@green.example.net;user=phone"
},
{
"name": "Blue",
"domain": "blue.example.net"
}
]
}
Example of a Provider list JSON object
10.2. RUE Configuration Service
The RUE is provisioned with one or more URIs that may be queried for
configuration with HTTPS.
The data returned will include a set of key/value configuration
parameters to be used by the RUE, formatted as a JSON object and
identified by the associated [RFC7159] "application/json" MIME type,
to allow for other formats in the future.
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.
o version: Identifies the version of the configuration data format.
A new version number SHOULD only be used if the new version is not
backwards-compatible with the older version. A new version number
is not needed if new elements are optional and can be ignored by
older implementations.
o lifetime: Specifies how long (in seconds) the RUE MAY cache the
configuration values. Values may not be valid when lifetime
expires. Emergency Calls MUST continue to work.
o display-name: (OPTIONAL) A user-friendly name to identify the
subscriber when originating calls.
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o 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.
o provider-domain: The DNS domain name of the default Provider
servicing this subscriber.
o outbound-proxies: (OPTIONAL) A URI of a SIP proxy to be used when
sending requests to the Provider.
o mwi: (OPTIONAL) A URI identifying a SIP event server that
generates "message-summary" events for this subscriber.
o videomail: (OPTIONAL) A SIP URI that can be called to retrieve
videomail messages.
o contacts: An HTTPS URI that may be used to export (retrieve) the
subscriber's complete contact list managed by the Provider.
o carddav: (OPTIONAL) A username and domain name (separated by
""@"") identifying a "CardDAV" server and user name that can be
used to synchronize the RUE's contact list with the contact list
managed by the Provider.
o sendLocationWithRegistration: True if the RUE should send a
Geolocation Header with REGISTER, false if it should not.
Defaults to false if not present.
o turn-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.
o credentials: (OPTIONAL) TBD
{
"version": 1,
"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" ,
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"sendLocationWithRegistration": false,
"ice-servers": [
{"stun:stun.l.google.com:19302" },
{"turn:turn.red.example.net:3478"}
],
"credentials": [
{
"realm": "red.example.net",
"username": "bob",
"password": "reg-pw"
},
{
"realm": "proxies.red.example.net",
"username": "bob",
"password": "proxy-pw"
},
{
"realm": "cd.red.example.net",
"username": "bob",
"password": "cd-pw"
},
{
"realm": "vm.red.example.net",
"username": "bob",
"password": "vm-pw"
},
{
"realm": "stun-turn.red.example.net",
"username": "bob",
"password": "stun-turn-pw"
}
]
}
Example JSON configuration payload
The wire format of the data is in keeping with the standard JSON
description in RFC7159.
The "lifetime" parameter in the configuration indicates how long the
RUE MAY cache the configuration values. 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.
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Note: In some cases, the RUE may successfully retrieve a fresh copy
of the configuration using digest credentials cached from the prior
retrieval. If this is not successful, then the RUE will need to ask
the user for the username and password. Unfortunately, this
authentication step might occur when the user is not present,
preventing SIP registration and thus incoming calls. To avoid this
situation, the RUE MAY retrieve a new copy of the configuration when
it knows the user is present, even if there is time before the
lifetime expires.
10.3. Schemas
The following JSON schemas are for the Provider List and the RUE
Configuration. These are represented using the JSON Content Rules
[JCR] schema notation.
{
"version": 1,
"providers": [
1*
{
"name": string,
"domain": fqdn,
?"operator": ; "front-door" access to provider
uri, ; (sip uri)
* /^.*$/ : any ; (allow future extensions)
}
] ,
* /^.*$/ : any ; (allow future extensions)
}
Provider List JSON Schema
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{
"version": 1, ; Interface version
"lifetime": integer, ; Deadline (in seconds) for
; refreshing this config without
; user input.
"phone-number": /^\+[0-9]+$/ , ; E.164 phone number
; for this user
?"display-name" : string,; display name for From: header
"provider-domain": fqdn, ; SHOULD match that in Provider List
?"outbound-proxies": [ 1* : uri ], ; sip URIs
?"mwi": uri , ; sip URI for MWI subscriptions
?"videomail": uri , ; sip URI for videomail retrieval
"contacts": uri , ; https URI for contact list retrieval
?"carddav": /^[^@]+@[^@]+$/ , ; for contact list synch
?"sendLocationWithRegistration": boolean , ; send location y/n
?"ice-servers": ; (Required for ICE use)
[ 1* : uri ], ; (stun[s] & turn[s] URIs
?"credentials": ; for digest authentication
[ 1* {
"realm": string,
"username": string,
"password": string
} ],
* /^.*$/ : any ; (allow future extensions)
}
RUE Configuration JSON Schema
The following illustrates the message flow for retrieving a RUE
automatic configuration using HTTPS Digest Authentication:
,-.
`-'
/|\ ,---. ,---. ,------------. ,----------------. ,---.
| |RUE| |DNS| |HTTPS Server| | Provider | |CRM|
/ \ | | | | | | |Global Settings | | |
RUE User `-+-' `-+-' `-----+------' `--------+-------' `-+-'
| | | | | |
[1] Select a VRS Provider name | | |
| ------->| | | | |
| | | | | |
[2] NAPTR "SFUA.CFG" red.example.net | |
| |----->| | | |
| | | | | |
[3] NAPTR "!.*!https://server.red.example.net/!" | |
| |<-----| | | |
| | | | | |
[4] If NAPTR found, query DNS server.red.example.net |
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| |----->| | | |
| | | | | |
[5] If NXDOMAIN, query DNS config.red.example.net | |
| | - - >| | | |
| | | | | |
[6] IP Address of Config Server | | |
| |<-----| | | |
| | | | | |
[7] Establish TLS connection | | |
| |<--------------->| | |
| | | | | |
[8] HTTP: https://config.red.example.net/v1 | |
| |---------------->| | |
| | | | | |
[9] HTTP: 401 Unauthorized | | |
WWW-Authenticate Digest realm="Y" qop="auth,auth-int" nonce=|
| |<----------------| | |
| | | | | |
[10] Query for userid/pw | | |
|<--------| | | | |
| | | | | |
[11] User="bob", pw="bob's global provider pw" | |
|-------->| | | | |
| | | | | |
[12] HTTP: https://config.red.example.net/v1 | |
| Authorization Digest username="bob" realm="Y" qop="auth" |
| nonce=... response="..." ... | |
| |---------------->| | |
| | | | | |
| [13] Find subscriber information for username="bob" |
| | | |----------------------------->|
| | | | | |
| [14] Subscriber specific configuration information |
| | | |<-----------------------------|
| | | | | |
| [15] Retrieve provider specific settings |
| | | |---------------->| |
| | | | | |
| [16] Provider configuration information | |
| | | |<----------------| |
| | | | | |
[17] 200 OK + JSON merge subscriber + provider configs |
| |<----------------| | |
| | | | | |
RUE User ,---. ,---. ,------------. ,----------------. ,---.
,-. |RUE| |DNS| |HTTPS Server| | Provider | |CRM|
`-' | | | | | | |Global Settings | | |
/|\ `-+-' `-+-' `-----+------' `--------+-------' `-+-'
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|
/ \
RUE Configuration Retrieval
11. Acknowledgements
12. IANA Considerations
This memo includes no request to IANA.
13. 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.
14. Normative References
[I-D.ietf-rtcweb-jsep]
Uberti, J., Jennings, C., and E. Rescorla, "JavaScript
Session Establishment Protocol", draft-ietf-rtcweb-jsep-26
(work in progress), February 2019.
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", draft-ietf-rtcweb-overview-19
(work in progress), November 2017.
[I-D.ietf-rtcweb-rtp-usage]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP",
draft-ietf-rtcweb-rtp-usage-26 (work in progress), March
2016.
[I-D.ietf-rtcweb-security-arch]
Rescorla, E., "WebRTC Security Architecture", draft-ietf-
rtcweb-security-arch-20 (work in progress), July 2019.
[I-D.ietf-rtcweb-transports]
Alvestrand, H., "Transports for WebRTC", draft-ietf-
rtcweb-transports-17 (work in progress), October 2016.
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[I-D.yusef-sipcore-digest-scheme]
Shekh-Yusef, R., "The Session Initiation Protocol (SIP)
Digest Authentication Scheme", draft-yusef-sipcore-digest-
scheme-07 (work in progress), April 2019.
[pip] SIPForum, "VRS US Providers Profile TWG-6-1.0", 2015,
<https://www.sipforum.org/download/
vrs-us-providers-profile-twg-6-1-0-pdf/#>.
[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>.
[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>.
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[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>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <https://www.rfc-editor.org/info/rfc3550>.
[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>.
[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>.
[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>.
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[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>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <https://www.rfc-editor.org/info/rfc4566>.
[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>.
[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007,
<https://www.rfc-editor.org/info/rfc4961>.
[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>.
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[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>.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
DOI 10.17487/RFC5245, April 2010,
<https://www.rfc-editor.org/info/rfc5245>.
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246,
DOI 10.17487/RFC5246, August 2008,
<https://www.rfc-editor.org/info/rfc5246>.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
DOI 10.17487/RFC5389, October 2008,
<https://www.rfc-editor.org/info/rfc5389>.
[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>.
[RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
for Establishing a Secure Real-time Transport Protocol
(SRTP) Security Context Using Datagram Transport Layer
Security (DTLS)", RFC 5763, DOI 10.17487/RFC5763, May
2010, <https://www.rfc-editor.org/info/rfc5763>.
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[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010,
<https://www.rfc-editor.org/info/rfc5764>.
[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766,
DOI 10.17487/RFC5766, April 2010,
<https://www.rfc-editor.org/info/rfc5766>.
[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>.
[RFC6184] Wang, Y., Even, R., Kristensen, T., and R. Jesup, "RTP
Payload Format for H.264 Video", RFC 6184,
DOI 10.17487/RFC6184, May 2011,
<https://www.rfc-editor.org/info/rfc6184>.
[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., "SIP-Specific Event Notification", RFC 6665,
DOI 10.17487/RFC6665, July 2012,
<https://www.rfc-editor.org/info/rfc6665>.
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[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>.
[RFC7159] Bray, T., Ed., "The JavaScript Object Notation (JSON) Data
Interchange Format", RFC 7159, DOI 10.17487/RFC7159, March
2014, <https://www.rfc-editor.org/info/rfc7159>.
[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. 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., "WebRTC Video Processing and Codec
Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016,
<https://www.rfc-editor.org/info/rfc7742>.
[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>.
Authors' Addresses
Brian Rosen
Mars, PA
US
Phone: +1 724 382 1051
Email: br@brianrosen.net
Rosen, et al. Expires February 8, 2020 [Page 28]
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Jim Malloy
The MITRE Corporation
McLean, VA
US
Phone: +1 703 983 2835
Email: jmalloy@mitre.org
Brett Henderson
The MITRE Corporation
McLean, VA
US
Phone: +1 619 758 6071
Email: brhenderson@mitre.org
Rosen, et al. Expires February 8, 2020 [Page 29]
