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art rum rue 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.

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. 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). |Device|<-(Internet)->| |<-( PSTN )->|User | +----+ +------+ -------- +--------+ ------ +-------+ ^ | +--------------+ |Communications| |Assistant | +--------------+ ]]>

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 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. 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.

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 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.

General Requirements All HTTP/HTTPS and connections specified throughout this document MUST use HTTPS. Both HTTPS and all SIP connections MUST use TLS conforming to at least and MUST support . 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 () is used, in which case candidates may explicitly offer IPv4, IPv6 or both for any media stream.

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

• (Base SIP)
• (Locating SIP Servers)
• (Offer/Answer)
• (User Agent Capabilities)
• (Outbound)
• (Session Description Protocol)
• (Privacy)
• (RTCP Attribute in SDP)
• (SIP Events)
• (UPDATE Method)
• (Loop-Fix)
• (Record Route fix)
• (ABNF fix)
• (Early Media)
• (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 ), which is optional and providers not supporting video mail need not support RFC6665. In addition, implementations MUST conform to:

• (Path)
• and (ICE)
• (Reason header field)
• (REFER Method)
• (Replaces Header field)
• (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 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. 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.

Registration The RUE MUST register with a SIP registrar, following and at a provider it has an account with. If the configuration (see ) 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 , using "phone-number" and "provider-domain" from the configuration.

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: 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 : "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. 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. 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.

Session Establishment

Normal Call Origination After initial SIP registration, the RUE adheres to SIP basic call flows, as documented in . 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 using the destination phone number, or as SIP URIs, as provided in the configuration (). 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 . The From-URI MUST be formatted as specified in , 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 .) Negotiated media MUST follow the requirements specified in 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.

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 "frontDoor" URI (see ) in "ProviderConfigurationData" of the dial-around provider. The RUE Provider Selection service () can be used by the RUE to obtain a list of providers and then the Provider Configuration () 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 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 . 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 | [2] INVITE | | |-------------->| Message Details: [1] INVITE Rue -> Default Provider INVITE sip:+15551234567@green.example.net;user=phone SIP/2.0 To: From: "Bob Smith" [2] INVITE Default Provider -> Dial-Around Provider INVITE sip:+15551234567@green.example.net;user=phone SIP/2.0 To: From: "Bob Smith" sip:+15552220001@red.example.net;user=phone P-Asserted-Identity: sip:+15552220001@red.example.net ]]>

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 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 . Please refer to 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.

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.

Emergency Calls Implementations MUST conform to for handling of emergency calls, except that if the device is unable to determine its own location, it MAY send the emergency call without a Geolocation header field and without a Route header field (since it would be unable to query the LoST server for a route per RFC6881). If an emergency call arrives at the provider without a Geolocation header field, the provider MUST supply location by adding the Geolocation header field, and MUST supply the route by querying the LoST server with that location. If the emergency call is to be handled using existing country specific procedures, the provider is responsible for modifying the INVITE to conform to the country-specific requirements. In this case, location MAY be extracted from the RFC6881 conformant INVITE and used to propagate it to the appropriate country-specific entities. If the configuration specifies it, an implementation of a RUE device MAY send a Geolocation header field containing its location in the REGISTER request. If implemented, users MUST be offered an opt-out. Country-specific procedures might require the location to be pre-loaded in some entity prior to placing an emergency call; however, the RUE may have a more accurate and timely device location than the manual, pre-loaded entry. That information MAY be used to populate the location to appropriate country-specific entities. Re-registration SHOULD be used to update the location, so long as the rate of re-registration is limited if the device is moving. Implementations MUST implement Additional Data, . RUE devices MUST implement Data Provider, Device Information and Owner/Subscriber Information blocks.

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

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 . The user part MUST NOT contain any 'visual-separator' characters, as defined in . Dial strings that cannot be expressed as E.164 numbers MUST be represented as dialstring URIs, as specified by , 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 ) to globally routable IPv4 and/or IPv6 addresses.

Transport Implementations MUST conform to except for its guidance on the WebRTC data channel, which this specification does not use. See for how RUE supports real-time text without the data channel. Implementations MUST support SIP outbound (please also refer to ).

Media This specification adopts the media specifications for WebRTC (). 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.

SRTP and SRTCP Implementations MUST support except that MediaStreamTracks are not used. Implementations MUST conform to Section 6.4 of .

Text-Based Communication Implementations MUST support real-time text ( and ) via T.140 media. One original and two redundant generations MUST be transmitted and supported, with a 300 ms transmission interval. Implementations MUST support 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 , 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 .

Video Implementations MUST conform to with following exceptions: only H.264, as specified in , 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.

Audio Implementations MUST conform to .

DTMF Digits Implementations MUST support the "audio/telephone-event" 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.

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

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 .

Negative Acknowledgment, Packet Loss Indicator, and Full Intraframe Request Features The NACK, FIR and PLI features as described in and 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 .

Contacts

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 ( and ). The configuration (see Section ) RueConfigurationData MAY supply a "carddav-username" and "carddav-domain" identifying a CardDAV server and address book for this account, plus an optional "carddav-password". 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.

Contacts Import/Export Service Implementations MUST be able to export/import the list of contacts in xCard 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.

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 for how the URI to reach the retrieval interface is obtained. Implementations MUST support subscriptions to "message-summary" events 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 ) 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 .

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.

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 ). 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 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 "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 ). 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.

Example of a ProviderList JSON object

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 interface description. In that interface description, the "servers" component string includes an occurence of "localhost". The entry point string obtained from the provider list () 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="t65667Ajjss90uuuDisKt8999"&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.

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 .
• 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 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.

RUE Configuration

• lifetime: (optional) Specifies how long (in seconds) the RUE MAY cache the configuration values. Values may not be valid when lifetime expires. If the RUE caches configuration values, it MUST cryptographically protect them 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.
• 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.

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 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.

Example of a Version JSON object

Examples

Example JSON provider configuration payload

Example JSON RUE configuration payload

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.

OpenAPI Interface Descriptions The interfaces in and are formally specified with OpenAPI 3.0 () 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.

Provider List

Provider List OpenAPI description (RueProviderList.yaml)

Configuration

Configuration OpenAPI description (RueConfiguration.yaml)

IANA Considerations

RUE Provider List Registry IANA has created the "RUE Provider List" registry. The management policy for this registry is "Expert Review" . 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

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

• Header Field: Call-Info
• Parameter Name: purpose
• Predefined Values: Yes

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.

Normative References Informative References

Acknowledgements Brett Henderson and Jim Malloy provided many helpful edits to prior versions of this document. Paul Kyzivat provided extensive reviews and comments.