SIP

This document specifies a framework that enables enterprise telephony

Session Initiation Protocol (SIP) networks to solicit and obtain a

capability set document from a SIP service provider. The capability

set document encodes a set of characteristics that enable easy

peering between enterprise and service provider SIP networks.

This document registers media types for Protocol Buffers, a common

extensible mechanism for serializing structured data.

The Third Generation Partnership Project (3GPP) has identified cases where different SIP private header extensions referred to as "P-" header fields, and defined in RFC 7315, need to be included in SIP requests and responses where they were not allowed according to RFC 7315. This document updates RFC 7315, in order to allow inclusion of the affected "P-" header fields in such requests and responses. This document obsoletes RFC 7976. The changes related to RFC 7976 involve the inclusion of the P-Visited-Network-ID header field in SIP responses.

This document also makes updates to RFC 7315 in order to fix misalignments that occurred when RFC 3455 was obsoleted by RFC 7315.

This document specifies a usage of the SIP Call-Info header field that incorporates Rich Call Data (RCD) associated with the identity of the originating party in order to provide to the terminating party a description of the caller (including details about the reason for the session). RCD includes information about the caller beyond the telephone number (such as a calling name, logo, photo, or jCard object representing the caller), which can help the called party decide how to handle the session request.

This document defines three new parameters 'call-reason', 'verified', and 'integrity' for the SIP Call-Info header field and also a new token ("jcard") for the 'purpose' parameter of the Call-Info header field. It also provides guidance on the use of the Call-Info 'purpose' parameter token, "icon".

The SIP-based Media Recording (SIPREC) protocol is defined by both "Session Initiation Protocol (SIP) Recording Metadata" (RFC 7865) and "Session Recording Protocol" (RFC 7866). Unfortunately, both RFCs contradict each other regarding how recording metadata is to be labeled. In addition, neither RFC registered the new media type. This document updates RFC 7866 to align with RFC 7865 when labeling recording metadata and also registers the media type.

This Informational document defines the 'sip-trunking-capability'

link relation type that may be used by an enterprise telephony

Session Initiation Protocol (SIP) network to retrieve a SIP trunking

capability set document, which contains the capabilities and

configuration requirements of an Internet Telephony Service Provider

(ITSP). These technical requirements allow for seamless peering

between SIP-based enterprise telephony networks and the ITSP.

The SIP Reason header field as defined in RFC 3326 allows only one Reason value per protocol value. Experience with more recently defined protocols shows it is useful to allow multiple values with the same protocol value. This document updates RFC 3326 to allow multiple values for an indicated registered protocol when that protocol defines what the presence of multiple values means.

Video Relay Service (VRS) is a term used to describe a method by which a hearing person can communicate with a sign language speaker who is deaf, deafblind, or hard of hearing (HoH) or has a speech disability using an interpreter (i.e., a Communications Assistant (CA)) connected via a videophone to the sign language speaker 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 sign language speakers. It is desirable that the videophones used by the sign language speaker conform to a standard so that any device may be used with any provider and that direct video calls between sign language speakers work. This document describes the interface between a videophone and a provider.

This document describes the registration of media types for the ECMAScript and JavaScript programming languages and conformance requirements for implementations of these types. This document obsoletes RFC 4329 ("Scripting Media Types)", replacing the previous registrations with information and requirements aligned with common usage and implementation experiences.

This document provides enhancements of real-time text (as specified in RFC 4103) suitable for mixing in a centralized conference model, enabling source identification and rapidly interleaved transmission of text from different sources. The intended use is for real-time text mixers and participant endpoints capable of providing an efficient presentation or other treatment of a multiparty real-time text session. The specified mechanism builds on the standard use of the Contributing Source (CSRC) list in the Real-time Transport Protocol (RTP) packet for source identification. The method makes use of the same "text/t140" and "text/red" formats as for two-party sessions.

Solutions using multiple RTP streams in the same RTP session are briefly mentioned, as they could have some benefits over the RTP-mixer model. The RTP-mixer model was selected to be used for the fully specified solution in this document because it can be applied to a wide range of existing RTP implementations.

A capability exchange is specified so that it can be verified that a mixer and a participant can handle the multiparty-coded real-time text stream using the RTP-mixer method. The capability is indicated by the use of a Session Description Protocol (SDP) (RFC 8866) media attribute, "rtt-mixer".

This document updates RFC 4103 ("RTP Payload for Text Conversation").

A specification for how a mixer can format text for the case when the endpoint is not multiparty aware is also provided.

This memo defines the Session Description Protocol (SDP). SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. This document obsoletes RFC 4566.

The Interactive Connectivity Establishment (ICE) protocol describes a Network Address Translator (NAT) traversal mechanism for UDP-based multimedia sessions established with the Offer/Answer model. The ICE extension for Incremental Provisioning of Candidates (Trickle ICE) defines a mechanism that allows ICE Agents to shorten session establishment delays by making the candidate gathering and connectivity checking phases of ICE non-blocking and by executing them in parallel.

This document defines usage semantics for Trickle ICE with the Session Initiation Protocol (SIP). The document also defines a new SIP Info Package to support this usage together with the corresponding media type. Additionally, a new Session Description Protocol (SDP) "end-of-candidates" attribute and a new SIP option tag "trickle-ice" are defined.

This document describes unknown key-share attacks on the use of Datagram Transport Layer Security for the Secure Real-Time Transport Protocol (DTLS-SRTP). Similar attacks are described on the use of DTLS-SRTP with the identity bindings used in Web Real-Time Communications (WebRTC) and SIP identity. These attacks are difficult to mount, but they cause a victim to be misled about the identity of a communicating peer. This document defines mitigation techniques that implementations of RFC 8122 are encouraged to deploy.

This document defines how to use the WebRTC data channel mechanism to realize a data channel, referred to as a Controlling Multiple Streams for Telepresence (CLUE) data channel, for transporting CLUE protocol messages between two CLUE entities.

Although the Session Initiation Protocol (SIP) includes a suite of security services that has been expanded by numerous specifications over the years, there is no single place that explains how to use SIP to establish confidential media sessions. Additionally, existing mechanisms have some feature gaps that need to be identified and resolved in order for them to address the pervasive monitoring threat model. This specification describes best practices for negotiating confidential media with SIP, including a comprehensive protection solution that binds the media layer to SIP layer identities.

This document defines the "Bearer" authentication scheme for the Session Initiation Protocol (SIP) and a mechanism by which user authentication and SIP registration authorization is delegated to a third party, using the OAuth 2.0 framework and OpenID Connect Core 1.0. This document updates RFC 3261 to provide guidance on how a SIP User Agent Client (UAC) responds to a SIP 401/407 response that contains multiple WWW-Authenticate/Proxy-Authenticate header fields.

There are some circumstances where a Geolocation header field may contain more than one locationValue. Knowing the identity of the node adding the locationValue allows the recipient more freedom in selecting the value to look at first rather than relying solely on the order of the locationValues. This document defines the "loc-src" parameter so that the entity adding the locationValue to the Geolocation header field can identify itself using its hostname. This document updates RFC 6442.

This document updates RFC 3261 by modifying the Digest Access Authentication scheme used by the Session Initiation Protocol (SIP) to add support for more secure digest algorithms, e.g., SHA-256 and SHA-512/256, to replace the obsolete MD5 algorithm.

Opportunistic Secure Real-time Transport Protocol (OSRTP) is an implementation of the Opportunistic Security mechanism, as defined in RFC 7435, applied to the Real-time Transport Protocol (RTP). OSRTP allows encrypted media to be used in environments where support for encryption is not known in advance and is not required. OSRTP does not require Session Description Protocol (SDP) extensions or features and is fully backwards compatible with existing implementations using encrypted and authenticated media and implementations that do not encrypt or authenticate media packets. OSRTP is not specific to any key management technique for Secure RTP (SRTP). OSRTP is a transitional approach useful for migrating existing deployments of real-time communications to a fully encrypted and authenticated state.

The SIP Reason header field is defined to carry ISUP (ISDN User Part) cause values as well as SIP response codes. Some services in SIP networks may need to know the ISUP location where the call was released in the PSTN (Public Switched Telephone Network) to correctly interpret the reason of release. This document updates RFC 3326 by adding a location parameter for this purpose.

This document describes how a Push Notification Service (PNS) can be used to wake a suspended Session Initiation Protocol (SIP) User Agent (UA) with push notifications, and it also describes how the UA can send binding-refresh REGISTER requests and receive incoming SIP requests in an environment in which the UA may be suspended. The document defines new SIP URI parameters to exchange PNS information between the UA and the SIP entity that will then request that push notifications be sent to the UA. It also defines the parameters to trigger such push notification requests. The document also defines new feature-capability indicators that can be used to indicate support of this mechanism.

Mobile messaging applications used with the Session Initiation Protocol (SIP) commonly use some combination of the SIP MESSAGE method and the Message Session Relay Protocol (MSRP). While these provide mechanisms for hop-by-hop security, neither natively provides end-to-end protection. This document offers guidance on how to provide end-to-end authentication, integrity protection, and confidentiality using the Secure/Multipurpose Internet Mail Extensions (S/MIME). It updates and provides clarifications for RFCs 3261, 3428, and 4975.

The P-Served-User header field was defined based on a requirement from the 3rd Generation Partnership Project (3GPP) IMS (IP Multimedia Subsystem) in order to convey the identity of the served user, his/ her registration state, and the session case that applies to that particular communication session and application invocation. A session case is metadata that captures the status of the session of a served user regardless of whether or not the served user is registered or the session originates or terminates with the served user. This document updates RFC 5502 by defining a new P-Served-User header field parameter, "orig-cdiv". The parameter conveys the session case used by a proxy when handling an originating session after Call Diversion (CDIV) services have been invoked for the served user. This document also fixes the ABNF in RFC 5502 and provides more guidance for using the P-Served-User header field in IP networks.

This document is published for the historical record. The Simple Internet Protocol was the basis for one of the candidates for the IETF's Next Generation (IPng) work that became IPv6.

The publication date of the original Internet-Draft was November 10, 1992. It is presented here substantially unchanged and is neither a complete document nor intended to be implementable.

The paragraph that follows is the Abstract from the original draft.

This document specifies a new version of IP called SIP, the Simple Internet Protocol. It also describes the changes needed to ICMP, IGMP, and transport protocols such as TCP and UDP, in order to work with SIP. A companion document [SIP-ADDR] describes the addressing and routing aspects of SIP, including issues of auto-configuration, host and subnet mobility, and multicast.

SIP networks use signaling monitoring tools to diagnose user-reported problems and to perform regression testing if network or user agent (UA) software is upgraded. As networks grow and become interconnected, including connection via transit networks, it becomes impractical to predict the path that SIP signaling will take between user agents and therefore impractical to monitor SIP signaling end to end.

This document describes an indicator for the SIP protocol that can be used to mark signaling as being of interest to logging. Such marking will typically be applied as part of network testing controlled by the network operator and is not used in normal user agent signaling. Operators of all networks on the signaling path can agree to carry such marking end to end, including the originating and terminating SIP user agents, even if a session originates and terminates in different networks.

This text documents the current usage of P-Charge-Info, an existing Session Initiation Protocol (SIP) private header field (P-Header) used to convey billing information about the party to be charged. This P-Header is currently used in production by several equipment vendors and carriers and has been in use since at least 2007. This document details the registration of this header field with IANA.

The "alert" namespace of Uniform Resource Names (URNs) can be used in the Alert-Info header field of Session Initiation Protocol (SIP) requests and responses to inform a voice over IP (VoIP) telephone (user agent) of the characteristics of the call that the user agent has originated or terminated. The user agent must resolve the URNs into a signal; that is, it must select the best available signal to present to its user to indicate the characteristics of the call.

RFC 7462 describes a non-normative algorithm for signal selection. This document describes a more efficient alternative algorithm: a user agent's designer can, based on the user agent's signals and their meanings, construct a finite state machine (FSM) to process the URNs to select a signal in a way that obeys the restrictions given in the definition of the "alert" URN namespace.

The functions of the Public Switched Telephone Network (PSTN) are rapidly migrating to the Internet. This is generating new requirements for many traditional elements of the PSTN, including Telephone Numbers (TNs). TNs no longer serve simply as telephone routing addresses: they are now identifiers that may be used by Internet-based services for a variety of purposes including session establishment, identity verification, and service enablement. This problem statement examines how the existing tools for allocating and managing telephone numbers do not align with the use cases of the Internet environment and proposes a framework for Internet-based services relying on TNs.

Users have various human (i.e., natural) language needs, abilities, and preferences regarding spoken, written, and signed languages. This document defines new Session Description Protocol (SDP) media- level attributes so that when establishing interactive communication sessions ("calls"), it is possible to negotiate (i.e., communicate and match) the caller's language and media needs with the capabilities of the called party. This is especially important for emergency calls, because it allows for a call to be handled by a call taker capable of communicating with the user or for a translator or relay operator to be bridged into the call during setup. However, this also applies to non-emergency calls (for example, calls to a company call center).

This document describes the need as well as a solution that uses new SDP media attributes.

This document specifies the Content-ID header field for usage in the Session Initiation Protocol (SIP). This document also updates RFC 5621, which only allows a Content-ID URL to reference a body part that is part of a multipart message-body. This update enables a Content-ID URL to reference a complete message-body and metadata provided by some additional SIP header fields.

This document updates RFC 5368 and RFC 6442 by clarifying their usage of the SIP Content-ID header field.

This document defines the 'multipart/multilingual' content type, which is an addition to the Multipurpose Internet Mail Extensions (MIME) standard. This content type makes it possible to send one message that contains multiple language versions of the same information. The translations would be identified by a language tag and selected by the email client based on a user's language settings.

This document describes methods for handling Unicode strings representing memorable, human-friendly names (called "nicknames", "display names", or "petnames") for people, devices, accounts, websites, and other entities. This document obsoletes RFC 7700.

RFC 3261 constrained several SIP header fields whose grammar contains the "name-addr / addr-spec" alternative to use name-addr when certain characters appear. Unfortunately, it expressed the constraints with prose copied into each header field definition, and at least one header field was missed. Further, the constraint has not been copied into documents defining extension headers whose grammar contains the alternative.

This document updates RFC 3261 to state the constraint generically and clarifies that the constraint applies to all SIP header fields where there is a choice between using name-addr or addr-spec. It also updates the RFCs that define extension SIP header fields using the alternative to clarify that the constraint applies (RFCs 3325, 3515, 3892, 4508, 5002, 5318, 5360, and 5502).

This document defines the 607 (Unwanted) SIP response code, allowing called parties to indicate that the call or message was unwanted. SIP entities may use this information to adjust how future calls from this calling party are handled for the called party or more broadly.

This document creates additional Session Initiation Protocol (SIP) Resource-Priority namespaces to meet the requirements of the 3GPP-defined Mission Critical Push To Talk (MCPTT) and places these namespaces in the corresponding IANA registry.

SIP networks use signaling monitoring tools to debug customer- reported problems and for regression testing if network or client software is upgraded. As networks grow and become interconnected, including connection via transit networks, it becomes impractical to predict the path that SIP signaling will take between clients and, therefore, impractical to monitor SIP signaling end-to-end.

This document describes the requirements for adding an indicator to the SIP Protocol Data Unit (PDU) or a SIP message that marks the PDU as a candidate for logging. Such a marking will typically be applied as part of network testing controlled by the network operator and not used in regular client signaling. However, such a marking can be carried end-to-end, including the SIP terminals, even if a session originates and terminates in different networks.

RFC 4458 (regarding SIP URIs for applications) defines a "cause" URI parameter, which may appear in the Request-URI of a SIP request, that is used to indicate a reason why the request arrived to the User Agent Server (UAS) receiving the message. This document updates RFC 4458 by creating a new predefined value for the "cause" URI parameter to cover service number translation for cases of retargeting due to specific service action leading to the translation of a called service access number. This document also provides guidance, which was missing in RFC 4458, for using the "cause" URI parameter within the History-Info header field, since this use is mandatory in some IP networks' implementations.

SIP Back-to-Back User Agents (B2BUAs) are often designed to also be on the media path, rather than just to intercept signalling. This means that B2BUAs often implement an RTP or RTP Control Protocol (RTCP) stack as well, thus leading to separate multimedia sessions that the B2BUA correlates and bridges together. If not disciplined, this behaviour can severely impact the communication experience, especially when statistics and feedback information contained in RTCP messages get lost because of mismatches in the reported data.

This document defines the proper behaviour B2BUAs should follow when acting on both the signalling plane and media plane in order to preserve the end-to-end functionality of RTCP.

This specification defines a new Session Initiation Protocol (SIP) Via header field parameter, 'received-realm', which allows a SIP entity acting as an entry point to a transit network to indicate from which adjacent upstream network a SIP request is received by using a network realm value associated with the adjacent network.

This document defines a bidirectional protocol mapping for the exchange of presence information between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP). This document obsoletes RFC 7248.

This document describes an end-to-end session identifier for use in IP-based multimedia communication systems that enables endpoints, intermediary devices, and management systems to identify a session end-to-end, associate multiple endpoints with a given multipoint conference, track communication sessions when they are redirected, and associate one or more media flows with a given communication session. While the identifier is intended to work across multiple protocols, this document describes its usage in the Session Initiation Protocol (SIP).

This document also describes a backwards-compatibility mechanism for an existing session identifier implementation (RFC 7329) that is sufficiently different from the procedures defined in this document.

This document obsoletes RFC 7329.

RFC 3263 defines how a Session Initiation Protocol (SIP) implementation, given a SIP Uniform Resource Identifier (URI), should locate the next-hop SIP server using Domain Name System (DNS) procedures. As SIP networks increasingly transition from IPv4-only to dual-stack, a quality user experience must be ensured for dual- stack SIP implementations. This document updates the DNS procedures described in RFC 3263 for dual-stack SIP implementations in preparation for forthcoming specifications for applying "Happy Eyeballs" principles to SIP.

RFC 5727 defined several processes for the former Real-time Applications and Infrastructure (RAI) area. These processes include the evolution of the Session Initiation Protocol (SIP) and related protocols, as well as the operation of the DISPATCH and SIPCORE working groups. This document updates RFC 5727 to allow flexibility for the area and working group structure, while preserving the SIP change processes. It also generalizes the DISPATCH working group processes so that they can be easily adopted by other working groups.

This document updates RFC 7315, by modifying the extension-access- info part of the P-Access-Network-Info header field Augmented Backus- Naur Form (ABNF), and by adding the following 'access-info' header field parameter values to the list of 'access-info' header field parameter values in the ABNF: 'operator-specific-GI' and 'utran-sai-3gpp'. The values are defined in the ABNF but are not included in the list.

Session Initiation Protocol (SIP) Back-to-Back User Agents (B2BUAs) exist on the signaling and media paths between the endpoints. This document describes the behavior of B2BUAs when Secure Real-time Transport (SRTP) security context is set up with the Datagram Transport Layer Security (DTLS) protocol.

This document defines a bidirectional protocol mapping for the exchange of instant messages in the context of a multi-party chat session among users of the Session Initiation Protocol (SIP) and users of the Extensible Messaging and Presence Protocol (XMPP). Specifically, this document defines a mapping between the SIP-based Message Session Relay Protocol (MSRP) and the XMPP Multi-User Chat (MUC) extension.

The SIP REFER method relies on the SIP-Specific Event Notification framework. That framework was revised by RFC 6665. This document highlights the implications of the requirement changes in RFC 6665, and updates the definition of the REFER method described in RFC 3515 to clarify and disambiguate the impact of those changes.

The Session Initiation Protocol (SIP) REFER request, as defined by RFC 3515, triggers an implicit SIP-Specific Event Notification framework subscription. Conflating the start of the subscription with handling the REFER request makes negotiating SUBSCRIBE extensions impossible and complicates avoiding SIP dialog sharing. This document defines extensions to REFER that remove the implicit subscription and, if desired, replace it with an explicit one.

Experience since the publication of the most recent SIP Events framework (in July 2012) has shown that there is room for interpretation around the use of Globally Routable User Agent URIs in that specification. This document clarifies the intended behavior.

This document updates RFC 6665.

Although the SIP History-Info header field described in RFC 7044 is the solution adopted in IETF, the non-standard Diversion header field described, as Historic, in RFC 5806 is nevertheless already implemented and used for conveying call-diversion-related information in Session Initiation Protocol (SIP) signaling.

RFC 7044 obsoletes the original RFC 4244 and redefines the History-Info header field for capturing the history information in requests.

Since the Diversion header field is used in existing network implementations for the transport of call diversion information, its interworking with the SIP History-Info standardized solution is needed. This document describes a recommended interworking guideline between the Diversion header field and the History-Info header field to handle call diversion information. This work is intended to enable the migration from non-standard implementations toward IETF specification-based implementations.

This document obsoletes RFC 6044, which describes the interworking between the Diversion header field defined in RFC 5806 and the obsoleted History-Info header field defined on RFC 4244.

Session Initiation Protocol (SIP) Back-to-Back User Agents (B2BUAs) are often designed to be on the media path rather than just intercepting signaling. This means that B2BUAs often act on the media path leading to separate media legs that the B2BUA correlates and bridges together. When acting on the media path, B2BUAs are likely to receive Session Traversal Utilities for NAT (STUN) packets as part of Interactive Connectivity Establishment (ICE) processing.

This document defines behavior for a B2BUA performing ICE processing. The goal of this document is to ensure that B2BUAs properly handle SIP messages that carry ICE semantics in Session Description Protocol (SDP) and STUN messages received as part of the ICE procedures for NAT and Firewall traversal of multimedia sessions.

This document defines a bidirectional protocol mapping for the exchange of single instant messages between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP).

This document defines a bidirectional protocol mapping for the exchange of instant messages in the context of a one-to-one chat session between a user of the Session Initiation Protocol (SIP) and a user of the Extensible Messaging and Presence Protocol (XMPP). Specifically for SIP text chat, this document specifies a mapping to the Message Session Relay Protocol (MSRP).

In 3GPP networks, the signaling path between a calling user and a called user can be partitioned into segments, referred to as traffic legs. Each traffic leg may span networks belonging to different operators and will have its own characteristics that can be different from other traffic legs in the same call. A traffic leg might be associated with multiple SIP dialogs, e.g., in case a Back-to-Back User Agent (B2BUA) that modifies the SIP dialog identifier is located within the traffic leg.

This document defines a new SIP URI parameter, 'iotl' (an abbreviation for Inter-Operator Traffic Leg). The parameter can be used in a SIP URI to indicate that the entity associated with the address, or an entity responsible for the host part of the address, represents the end of a specific traffic leg (or multiple traffic legs).

This document provides a terminology for benchmarking the Session Initiation Protocol (SIP) performance of devices. Methodology related to benchmarking SIP devices is described in the companion methodology document (RFC 7502). Using these two documents, benchmarks can be obtained and compared for different types of devices such as SIP Proxy Servers, Registrars, and Session Border Controllers. The term "performance" in this context means the capacity of the Device Under Test (DUT) to process SIP messages. Media streams are used only to study how they impact the signaling behavior. The intent of the two documents is to provide a normalized set of tests that will enable an objective comparison of the capacity of SIP devices. Test setup parameters and a methodology are necessary because SIP allows a wide range of configurations and operational conditions that can influence performance benchmark measurements. A standard terminology and methodology will ensure that benchmarks have consistent definitions and were obtained following the same procedures.

This document provides a methodology for benchmarking the Session Initiation Protocol (SIP) performance of devices. Terminology related to benchmarking SIP devices is described in the companion terminology document (RFC 7501). Using these two documents, benchmarks can be obtained and compared for different types of devices such as SIP Proxy Servers, Registrars, and Session Border Controllers. The term "performance" in this context means the capacity of the Device Under Test (DUT) to process SIP messages. Media streams are used only to study how they impact the signaling behavior. The intent of the two documents is to provide a normalized set of tests that will enable an objective comparison of the capacity of SIP devices. Test setup parameters and a methodology are necessary because SIP allows a wide range of configurations and operational conditions that can influence performance benchmark measurements.

The Session Initiation Protocol (SIP) supports the capability to provide a reference to a specific rendering to be used by the User Agent (UA) as an alerting signal (e.g., a ring tone or ringback tone) when the user is alerted. This is done using the Alert-Info header field. However, the reference (typically a URL) addresses only a specific network resource with specific rendering properties. There is currently no support for standard identifiers for describing the semantics of the alerting situation or the characteristics of the alerting signal, without being tied to a particular rendering. To overcome these limitations and support new applications, a new family of URNs for use in Alert-Info header fields (and situations with similar requirements) is defined in this specification.

This document normatively updates RFC 3261, which defines the Session Initiation Protocol (SIP). It changes the usage of the Alert-Info header field defined in RFC 3261 by additionally allowing its use in any non-100 provisional response to INVITE. This document also permits proxies to add or remove an Alert-Info header field and to add or remove Alert-Info header field values.

This document describes the requirements and implementation of a group telephony feature commonly known as Bridged Line Appearance (BLA) or Multiple Line Appearance (MLA), or Shared Call/Line Appearance (SCA). When implemented using the Session Initiation Protocol (SIP), it is referred to as shared appearances of an Address of Record (AOR) since SIP does not have the concept of lines. This feature is commonly offered in IP Centrex services and IP Private Branch Exchange (IPBX) offerings and is likely to be implemented on SIP IP telephones and SIP feature servers used in a business environment. This feature allows several user agents (UAs) to share a common AOR, learn about calls placed and received by other UAs in the group, and pick up or join calls within the group. This document discusses use cases, lists requirements, and defines extensions to implement this feature. This specification updates RFCs 3261 and 4235.

The prevalent use of the Session Initiation Protocol (SIP) in Next Generation Networks necessitates that SIP networks provide adequate control mechanisms to maintain transaction throughput by preventing congestion collapse during traffic overloads. A loss-based solution to remedy known vulnerabilities of the SIP 503 (Service Unavailable) overload control mechanism has already been proposed. Using the same signaling, this document proposes a rate-based control scheme to complement the loss-based control scheme.

There is a class of applications that benefit from using SIP to exchange User-to-User Information (UUI) data during session establishment. This information, known as call control UUI data, is a small piece of data inserted by an application initiating the session and utilized by an application accepting the session. The syntax and semantics for the UUI data used by a specific application are defined by a UUI package. This UUI data is opaque to SIP and its function is unrelated to any basic SIP function. This document defines a new SIP header field, User-to-User, to transport UUI data, along with an extension mechanism.

The motivation and use cases for interworking and transporting User- to-User Information (UUI) from the ITU-T Digital Subscriber Signalling System No. 1 (DSS1) User-user information element within SIP are described in RFC 6567. As networks move to SIP, it is important that applications requiring this data can continue to function in SIP networks as well as have the ability to interwork with this ISDN service for end-to-end transparency. This document defines a usage (a new package called the ISDN UUI package) of the User-to-User header field to enable interworking with this ISDN service.

This document covers interworking with both public ISDN and private ISDN capabilities, so the potential interworking with QSIG will also be addressed.

The package is identified by the new value "isdn-uui" of the "purpose" header field parameter.

SIP already provides the ability to perform hop-by-hop traceroute for SIP messages using the Max-Forwards header field to determine the reachability path of requests to a target. A mechanism for media-loopback calls has also been defined separately, which enables test calls to be generated that result in media being looped back to the originator. This document describes a means of performing hop-by-hop traceroute-style test calls using the media-loopback mechanism to test the media path when SIP sessions go through media-relaying back-to-back user agents (B2BUAs).

Overload occurs in Session Initiation Protocol (SIP) networks when SIP servers have insufficient resources to handle all the SIP messages they receive. Even though the SIP protocol provides a limited overload control mechanism through its 503 (Service Unavailable) response code, SIP servers are still vulnerable to overload. This document defines the behavior of SIP servers involved in overload control and also specifies a loss-based overload scheme for SIP.

RFC 7118 specifies a WebSocket subprotocol as a reliable real-time transport mechanism between Session Initiation Protocol (SIP) entities to enable usage of SIP in web-oriented deployments. This document updates the SIP Common Log Format (CLF), defined in RFC 6873, with a new "Transport Flag" for such SIP WebSocket transport.

This document specifies how the UDP Transport Layer (UDPTL) protocol, the predominant transport protocol for T.38 fax, can be transported over the Datagram Transport Layer Security (DTLS) protocol, how the usage of UDPTL over DTLS is indicated in the Session Description Protocol (SDP), and how UDPTL over DTLS is negotiated in a session established using the Session Initiation Protocol (SIP).

SIP Back-to-Back User Agents (B2BUAs) can cause unending SIP request routing loops because, as User Agent Clients, they can generate SIP requests with new Max-Forwards values. This document discusses the difficulties associated with loop detection for B2BUAs and the requirements for them to prevent infinite loops.

This document specifies the SIP P-Private-Network-Indication P-header used by the 3GPP. The P-Private-Network-Indication indicates that the message is part of the message traffic of a private network and identifies that private network. A private network indication allows nodes to treat private network traffic according to a different set of rules than the set applicable to public network traffic.

This document describes a set of private header (P-header) Session Initiation Protocol (SIP) fields used by the 3GPP, along with their applicability, which is limited to particular environments. The P-header fields are used for a variety of purposes within the networks that the partners implement, including charging and information about the networks a call traverses. This document obsoletes RFC 3455.

As a foundation for the definition of bidirectional protocol mappings between the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP), this document specifies the architectural assumptions underlying such mappings as well as the mapping of addresses and error conditions.

This specification defines how the Uniform Resource Name (URN) reserved for the Global System for Mobile Communications Association (GSMA) identities and its sub-namespace for the International Mobile station Equipment Identity (IMEI) can be used as an instance-id. Its purpose is to fulfill the requirements for defining how a specific URN needs to be constructed and used in the '+sip.instance' Contact header field parameter for outbound behavior.

This document creates the new Session Initiation Protocol (SIP) Resource Priority header field namespace 'esnet' and registers this namespace with IANA. The new header field namespace allows for local emergency session establishment to a public safety answering point (PSAP), between PSAPs, and between a PSAP and first responders and their organizations.

This document specifies the requirements for an end-to-end session identifier in IP-based multimedia communication networks. This identifier would enable endpoints, intermediate devices, and management and monitoring systems to identify a session end-to-end across multiple SIP devices, hops, and administrative domains.

This specification defines a load-control event package for the Session Initiation Protocol (SIP). It allows SIP entities to distribute load-filtering policies to other SIP entities in the network. The load-filtering policies contain rules to throttle calls from a specific user or based on their source or destination domain, telephone number prefix. The mechanism helps to prevent signaling overload and complements feedback-based SIP overload control efforts.

After an emergency call is completed (terminated either prematurely by the emergency caller or normally by the call taker), the call taker may feel the need for further communication. For example, the call may have been dropped by accident without the call taker having sufficient information about the current state of an accident victim. A call taker may trigger a callback to the emergency caller using the contact information provided with the initial emergency call. This callback could, under certain circumstances, be treated like any other call and, as a consequence, it may get blocked by authorization policies or may get forwarded to an answering machine.

The IETF emergency services architecture specification already offers a solution approach for allowing Public Safety Answering Point (PSAP) callbacks to bypass authorization policies in order to reach the caller without unnecessary delays. Unfortunately, the specified mechanism only supports limited scenarios. This document discusses shortcomings of the current mechanisms and illustrates additional scenarios where better-than-normal call treatment behavior would be desirable. We describe a solution based on a new header field value for the SIP Priority header field, called "psap-callback", to mark PSAP callbacks.

This document updates the registration guidance provided in Section 4.2 of RFC 5031, which allows the registration of service URNs with the 'sos' service type only for emergency services "that are offered widely and in different countries". This document updates those instructions to allow such registrations when, at the time of registration, those services are offered in only one country.

This document describes use cases and documents call flows that require the History-Info header field to capture the Request-URIs as a Session Initiation Protocol (SIP) Request is retargeted. The use cases are described along with the corresponding call flow diagrams and messaging details.

RFC 4412 defines the "Resource-Priority Namespaces" and "Resource-Priority Priority-values" registries. The management policy of these registries is "Standards Action". This document normatively updates RFC 4412 to change the management policy of these registries to "IETF Review".

"Music on hold" is one of the features of telephone systems that is most desired by buyers of business telephone systems. Music on hold means that when one party to a call has the call "on hold", that party's telephone provides an audio stream (often music) to be heard by the other party. Architectural features of SIP make it difficult to implement music on hold in a way that is fully standards-compliant. The implementation of music on hold described in this document is fully effective, is standards-compliant, and has a number of advantages over the methods previously documented. In particular, it is less likely to produce peculiar user interface effects and more likely to work in systems that perform authentication than the music-on-hold method described in Section 2.3 of RFC 5359.

This document defines a standard mechanism for capturing the history information associated with a Session Initiation Protocol (SIP) request. This capability enables many enhanced services by providing the information as to how and why a SIP request arrives at a specific application or user. This document defines an optional SIP header field, History-Info, for capturing the history information in requests. The document also defines SIP header field parameters for the History-Info and Contact header fields to tag the method by which the target of a request is determined. In addition, this specification defines a value for the Privacy header field that directs the anonymization of values in the History-Info header field. This document obsoletes RFC 4244.

The WebSocket protocol enables two-way real-time communication between clients and servers in web-based applications. This document specifies a WebSocket subprotocol as a reliable transport mechanism between Session Initiation Protocol (SIP) entities to enable use of SIP in web-oriented deployments.

This document defines and registers a value of "grouptextchat" ("Group Text Chat") for the URI <purpose> element of SIP's Conference Event Package.

In many SIP deployments, SIP entities exist in the SIP signaling path between the originating and final terminating endpoints, which go beyond the definition of a SIP proxy, performing functions not defined in Standards Track RFCs. The only term for such devices provided in RFC 3261 is for a Back-to-Back User Agent (B2BUA), which is defined as the logical concatenation of a SIP User Agent Server (UAS) and User Agent Client (UAC).

There are numerous types of SIP B2BUAs performing different roles in different ways; for example, IP Private Branch Exchanges (IPBXs), Session Border Controllers (SBCs), and Application Servers (ASs). This document identifies several common B2BUA roles in order to provide taxonomy other documents can use and reference.

This document suggests some strategies for the combined use of the Session Initiation Protocol (SIP) and the Extensible Messaging and Presence Protocol (XMPP) both in user-oriented clients and in deployed servers. Such strategies, which mainly consist of configuration changes and minimal software modifications to existing clients and servers, aim to provide a single, full-featured, real-time communication service by using complementary subsets of features from SIP and from XMPP. Typically, such subsets consist of telephony capabilities from SIP and instant messaging and presence capabilities from XMPP. This document does not define any new protocols or syntax for either SIP or XMPP and, by intent, does not attempt to standardize "best current practices". Instead, it merely aims to provide practical guidance to those who are interested in the combined use of SIP and XMPP for real-time communication.

This document defines and registers a value of "impp" ("instant messaging and presence protocol") for the "purpose" header field parameter of the Call-Info header field in the Session Initiation Protocol (SIP).

The "completion of calls" feature defined in this specification allows the caller of a failed call to be notified when the callee becomes available to receive a call.

For the realization of a basic solution without queuing, this document references the usage of the dialog event package (RFC 4235) that is described as 'Automatic Redial' in "Session Initiation Protocol Service Examples" (RFC 5359).

For the realization of a more comprehensive solution with queuing, this document introduces an architecture for implementing these features in the Session Initiation Protocol where "completion of calls" implementations associated with the caller's and callee's endpoints cooperate to place the caller's request for completion of calls into a queue at the callee's endpoint; when a caller's request is ready to be serviced, re-attempt of the original, failed call is then made.

The architecture is designed to interoperate well with existing completion of calls solutions in other networks.

This document defines and registers with IANA the new "fax" media feature tag for use with the Session Initiation Protocol (SIP). Currently, fax calls are indistinguishable from voice calls at call initiation. Consequently, fax calls can be routed to SIP user agents that are not fax capable. A "fax" media feature tag implemented in conjunction with caller preferences allows for more accurate fax call routing.

The IETF and other standards organizations have efforts targeted at standardizing various aspects of placing emergency calls on IP networks. This memo describes best current practice on how devices, networks, and services using IETF protocols should use such standards to make emergency calls.

This document defines a new IANA registry to keep track of the values defined for the Session Initiation Protocol (SIP) "Priority" header field. It updates RFC 3261.

Well-known web servers such as Apache and web proxies like Squid support event logging using a common log format. The logs produced using these de facto standard formats are invaluable to system administrators for troubleshooting a server and tool writers to craft tools that mine the log files and produce reports and trends. Furthermore, these log files can also be used to train anomaly detection systems and feed events into a security event management system. The Session Initiation Protocol (SIP) does not have a common log format, and, as a result, each server supports a distinct log format that makes it unnecessarily complex to produce tools to do trend analysis and security detection. This document describes a framework, including requirements and analysis of existing approaches, and specifies an information model for development of a SIP common log file format that can be used uniformly by user agents, proxies, registrars, and redirect servers as well as back-to-back user agents.

The SIPCLF working group has defined a Common Log Format (CLF) framework for Session Initiation Protocol (SIP) servers. This CLF mimics the successful event logging format found in well-known web servers like Apache and web proxies like Squid. This document proposes an indexed text encoding format for the SIP CLF that retains the key advantages of a text-based format while significantly increasing processing performance over a purely text-based implementation. This file format adheres to the SIP CLF information model and provides an effective encoding scheme for all mandatory and optional fields that appear in a SIP CLF record.

Proxy servers play a central role as an intermediary in the Session Initiation Protocol (SIP) as they define and impact policies on call routing, rendezvous, and other call features. This document specifies a framework for SIP session policies that provides a standard mechanism by which a proxy can define or influence policies on sessions, such as the codecs or media types to be used. It defines a model, an overall architecture and new protocol mechanisms for session policies. [STANDARDS-TRACK]

This specification defines a Session Initiation Protocol (SIP) event package for session-specific policies. This event package enables user agents (UAs) to subscribe to session policies for a SIP session and to receive notifications if these policies change. [STANDARDS-TRACK]

This specification defines an XML document format to describe the media properties of Session Initiation Protocol (SIP) sessions. Examples for media properties are the codecs or media types used in the session. This document also defines an XML document format to describe policies that limit the media properties of SIP sessions. [STANDARDS-TRACK]

This specification defines a new SIP header field, Feature-Caps. The Feature-Caps header field conveys feature-capability indicators that are used to indicate support of features and capabilities for SIP entities that are not represented by the Uniform Resource Identifier (URI) of the Contact header field.

SIP entities that are represented by the URI of the SIP Contact header field can convey media feature tags in the Contact header field to indicate support of features and capabilities.

This specification also defines feature-capability indicators and creates a new IANA registry, "Proxy-Feature Feature-Capability Indicator Trees", for registering feature-capability indicators. [STANDARDS-TRACK]

This document describes an extension to the Session Initiation Protocol (SIP) defined by RFC 3261. The purpose of this extension is to provide an extensible framework by which SIP nodes can request notification from remote nodes indicating that certain events have occurred.

Note that the event notification mechanisms defined herein are NOT intended to be a general-purpose infrastructure for all classes of event subscription and notification.

This document represents a backwards-compatible improvement on the original mechanism described by RFC 3265, taking into account several years of implementation experience. Accordingly, this document obsoletes RFC 3265. This document also updates RFC 4660 slightly to accommodate some small changes to the mechanism that were discussed in that document. [STANDARDS-TRACK]

This document introduces the transport of call control User-to-User Information (UUI) using the Session Initiation Protocol (SIP) and develops several requirements for a new SIP mechanism. Some SIP sessions are established by or related to a non-SIP application. This application may have information that needs to be transported between the SIP User Agents during session establishment. In addition to interworking with the Integrated Services Digital Network (ISDN) UUI Service, this extension will also be used for native SIP endpoints requiring application UUI. This document is not an Internet Standards Track specification; it is published for informational purposes.

This document describes a profile of the Sieve extension for notifications, to allow notifications to be sent over SIP MESSAGE. [STANDARDS-TRACK]

This document specifies mechanisms for adjusting the rate of Session Initiation Protocol (SIP) event notifications. These mechanisms can be applied in subscriptions to all SIP event packages. This document updates RFC 3265. [STANDARDS-TRACK]

This document describes filters that limit asynchronous location notifications to compelling events. These filters are designed as an extension to RFC 4661, an XML-based format for event notification filtering, and based on RFC 3856, the SIP presence event package. The resulting location information is conveyed in existing location formats wrapped in the Presence Information Data Format Location Object (PIDF-LO). [STANDARDS-TRACK]

This document defines an extension to the Session Initiation Protocol (SIP) to convey geographic location information from one SIP entity to another SIP entity. The SIP extension covers end-to-end conveyance as well as location-based routing, where SIP intermediaries make routing decisions based upon the location of the Location Target. [STANDARDS-TRACK]

Although the use of the SIP (Session Initiation Protocol) Reason header field in responses is considered in general in RFC 3326, its use is not specified for any particular response code. Nonetheless, existing deployments have been using Reason header fields to carry failure-related Q.850 cause codes in SIP responses to INVITE requests that have been gatewayed to Public Switched Telephone Network (PSTN) systems. This document normatively describes the use of the Reason header field in carrying Q.850 cause codes in SIP responses. [STANDARDS-TRACK]

Overload occurs in Session Initiation Protocol (SIP) networks when SIP servers have insufficient resources to handle all SIP messages they receive. Even though the SIP protocol provides a limited overload control mechanism through its 503 (Service Unavailable) response code, SIP servers are still vulnerable to overload. This document discusses models and design considerations for a SIP overload control mechanism. This document is not an Internet Standards Track specification; it is published for informational purposes.

The Session Initiation Protocol (SIP) utilizes the offer/answer model to establish and update multimedia sessions using the Session Description Protocol (SDP). The description of the offer/answer model in SIP is dispersed across multiple RFCs. This document summarizes all the current usages of the offer/answer model in SIP communication. This document is not an Internet Standards Track specification; it is published for informational purposes.

Traversal of the Session Initiation Protocol (SIP) and the sessions it establishes through Network Address Translators (NATs) is a complex problem. Currently, there are many deployment scenarios and traversal mechanisms for media traffic. This document provides concrete recommendations and a unified method for NAT traversal as well as documents corresponding flows. This document is not an Internet Standards Track specification; it is published for informational purposes.

This specification defines a new Session Initiation Protocol (SIP) response code, 199 Early Dialog Terminated, that a SIP forking proxy and a User Agent Server (UAS) can use to indicate to upstream SIP entities (including the User Agent Client (UAC)) that an early dialog has been terminated, before a final response is sent towards the SIP entities. [STANDARDS-TRACK]

This document describes how the IPv4 Session Initiation Protocol (SIP) user agents can communicate with IPv6 SIP user agents (and vice versa) at the signaling layer as well as exchange media once the session has been successfully set up. Both single- and dual-stack (i.e., IPv4-only and IPv4/IPv6) user agents are considered. [STANDARDS-TRACK]

This specification defines a new Session Initiation Protocol (SIP) Via header field parameter, "keep", which allows adjacent SIP entities to explicitly negotiate usage of the Network Address Translation (NAT) keep-alive mechanisms defined in SIP Outbound, in cases where SIP Outbound is not supported, cannot be applied, or where usage of keep-alives is not implicitly negotiated as part of the SIP Outbound negotiation. [STANDARDS-TRACK]

This document specifies how to establish a media session that represents a virtual private network using the Session Initiation Protocol for the purpose of on-demand media/application sharing between peers. It extends the protocol identifier of the Session Description Protocol (SDP) so that it can negotiate use of the Internet Key Exchange Protocol (IKE) for media sessions in the SDP offer/answer model. It also specifies a method to boot up IKE and generate IPsec security associations using a self-signed certificate. This document is not an Internet Standards Track specification; it is published for informational purposes.

This document shows example call flows demonstrating the use of Transport Layer Security (TLS), and Secure/Multipurpose Internet Mail Extensions (S/MIME) in Session Initiation Protocol (SIP). It also provides information that helps implementers build interoperable SIP software. To help facilitate interoperability testing, it includes certificates used in the example call flows and processes to create certificates for testing. This document is not an Internet Standards Track specification; it is published for informational purposes.

This document specifies a framework to enable configuration of Session Initiation Protocol (SIP) user agents (UAs) in SIP deployments. The framework provides a means to deliver profile data that user agents need to be functional, automatically and with minimal or no User and Administrative intervention. The framework describes how SIP user agents can discover sources, request profiles, and receive notifications related to profile modifications. As part of this framework, a new SIP event package is defined for notification of profile changes. The framework provides minimal data retrieval options to ensure interoperability. The framework does not include specification of the profile data within its scope. [STANDARDS-TRACK]

This document defines a mechanism by which a Session Initiation Protocol (SIP) server acting as a traditional Private Branch Exchange (PBX) can register with a SIP Service Provider (SSP) to receive phone calls for SIP User Agents (UAs). In order to function properly, this mechanism requires that each of the Addresses of Record (AORs) registered in bulk map to a unique set of contacts. This requirement is satisfied by AORs representing phone numbers regardless of the domain, since phone numbers are fully qualified and globally unique. This document therefore focuses on this use case. [STANDARDS-TRACK]

The procedures for handling SIP re-INVITEs are described in RFC 3261. Implementation and deployment experience has uncovered a number of issues with the original documentation, and this document provides additional procedures that update the original specification to address those issues. In particular, this document defines in which situations a UAS (User Agent Server) should generate a success response and in which situations a UAS should generate an error response to a re-INVITE. Additionally, this document defines further details of procedures related to target-refresh requests. [STANDARDS-TRACK]

This document defines a credential service that allows Session Initiation Protocol (SIP) User Agents (UAs) to use a SIP event package to discover the certificates of other users. This mechanism allows User Agents that want to contact a given Address-of-Record (AOR) to retrieve that AOR's certificate by subscribing to the credential service, which returns an authenticated response containing that certificate. The credential service also allows users to store and retrieve their own certificates and private keys. [STANDARDS-TRACK]

This document defines a set of metrics and their usage to evaluate the performance of end-to-end Session Initiation Protocol (SIP) for telephony services in both production and testing environments. The purpose of this document is to combine a standard set of common metrics, allowing interoperable performance measurements, easing the comparison of industry implementations. [STANDARDS-TRACK]

This document defines a method, INFO, for the Session Initiation Protocol (SIP), and an Info Package mechanism. This document obsoletes RFC 2976. For backward compatibility, this document also specifies a "legacy" mode of usage of the INFO method that is compatible with the usage previously defined in RFC 2976, referred to as "legacy INFO Usage" in this document. [STANDARDS-TRACK]

This document defines a Session Initiation Protocol (SIP) event package that enables the collection and reporting of metrics that measure the quality for Voice over Internet Protocol (VoIP) sessions. Voice call quality information derived from RTP Control Protocol Extended Reports (RTCP-XR) and call information from SIP is conveyed from a User Agent (UA) in a session, known as a reporter, to a third party, known as a collector. A registration for the application/ vq-rtcpxr media type is also included. [STANDARDS-TRACK]

This document describes private extensions to the Session Initiation Protocol (SIP) that enable a network of trusted SIP servers to assert the service of authenticated users. The use of these extensions is only applicable inside an administrative domain with previously agreed-upon policies for generation, transport, and usage of such information. This document does NOT offer a general service identification model suitable for use between different trust domains or for use in the Internet at large.

The document also defines a URN to identify both services and User Agent (UA) applications. This URN can be used within the SIP header fields defined in this document to identify services, and also within the framework defined for caller preferences and callee capabilities to identify usage of both services and applications between end UAs. This document is not an Internet Standards Track specification; it is published for informational purposes.

The Session Initiation Protocol (SIP) is increasingly being used in systems that are tightly coupled with Hypertext Transport Protocol (HTTP) servers for a variety of reasons. In many of these cases, applications can benefit from being able to discover, in near real- time, when a specific HTTP resource is created, changed, or deleted. This document proposes a mechanism, based on the SIP Event Framework, for doing so. [STANDARDS-TRACK]

This document defines procedures for how a SIP User Agent should locate, retrieve, and maintain current configuration information from a Configuration Service. This document is not an Internet Standards Track specification; it is published for informational purposes.

This document normatively updates RFC 3261, the Session Initiation Protocol (SIP), to address an error in the specified handling of success (2xx class) responses to INVITE requests. Elements following RFC 3261 exactly will misidentify retransmissions of the request as a new, unassociated request. The correction involves modifying the INVITE transaction state machines. The correction also changes the way responses that cannot be matched to an existing transaction are handled to address a security risk. [STANDARDS-TRACK]

This document states requirements for a standardized SIP registration mechanism for multiple addresses of record (AORs), the mechanism being suitable for deployment by SIP service providers on a large scale in support of small to medium sized Private Branch Exchanges (PBXs). The requirements are for a solution that can, as a minimum, support AORs based on E.164 numbers. This document is not an Internet Standards Track specification; it is published for informational purposes.

This document corrects the Augmented Backus-Naur Form (ABNF) production rule associated with generating IPv6 literals in RFC 3261. It also clarifies the rule for Uniform Resource Identifier (URI) comparison when the URIs contain textual representation of IP addresses. [STANDARDS-TRACK]

This document defines a new connectivity precondition for the Session Description Protocol (SDP) precondition framework. A connectivity precondition can be used to delay session establishment or modification until media stream connectivity has been successfully verified. The method of verification may vary depending on the type of transport used for the media. For unreliable datagram transports such as UDP, verification involves probing the stream with data or control packets. For reliable connection-oriented transports such as TCP, verification can be achieved simply by successful connection establishment or by probing the connection with data or control packets, depending on the situation. [STANDARDS-TRACK]

This document describes how to construct and interpret certain information in a PKIX-compliant (Public Key Infrastructure using X.509) certificate for use in a Session Initiation Protocol (SIP) over Transport Layer Security (TLS) connection. More specifically, this document describes how to encode and extract the identity of a SIP domain in a certificate and how to use that identity for SIP domain authentication. As such, this document is relevant both to implementors of SIP and to issuers of certificates. [STANDARDS-TRACK]

This document enables a pair of communicating proxies to reuse a congestion-controlled connection between themselves for sending requests in the forwards and backwards direction. Because the connection is essentially aliased for requests going in the backwards direction, reuse is predicated upon both the communicating endpoints authenticating themselves using X.509 certificates through Transport Layer Security (TLS). For this reason, we only consider connection reuse for TLS over TCP and TLS over Stream Control Transmission Protocol (SCTP). This document also provides guidelines on connection reuse and virtual SIP servers and the interaction of connection reuse and DNS SRV lookups in SIP. [STANDARDS-TRACK]

This memo documents an extended key usage (EKU) X.509 certificate extension for restricting the applicability of a certificate to use with a Session Initiation Protocol (SIP) service. As such, in addition to providing rules for SIP implementations, this memo also provides guidance to issuers of certificates for use with SIP. This document defines an Experimental Protocol for the Internet community.

In this specification, we define a framework to group "m" lines in the Session Description Protocol (SDP) for different purposes. This framework uses the "group" and "mid" SDP attributes, both of which are defined in this specification. Additionally, we specify how to use the framework for two different purposes: for lip synchronization and for receiving a media flow consisting of several media streams on different transport addresses. This document obsoletes RFC 3388. [STANDARDS-TRACK]

This document considers the problem of service identification in the Session Initiation Protocol (SIP). Service identification is the process of determining the user-level use case that is driving the signaling being utilized by the user agent (UA). This document discusses the uses of service identification, and outlines several architectural principles behind the process. It identifies perils when service identification is not done properly -- including fraud, interoperability failures, and stifling of innovation. It then outlines a set of recommended practices for service identification. This document is not an Internet Standards Track specification; it is published for informational purposes.

This document defines a framework and the requirements for call control and multi-party usage of the Session Initiation Protocol (SIP). To enable discussion of multi-party features and applications, we define an abstract call model for describing the media relationships required by many of these. The model and actions described here are specifically chosen to be independent of the SIP signaling and/or mixing approach chosen to actually set up the media relationships. In addition to its dialog manipulation aspect, this framework includes requirements for communicating related information and events such as conference and session state and session history. This framework also describes other goals that embody the spirit of SIP applications as used on the Internet such as the definition of primitives (not services), invoker and participant oriented primitives, signaling and mixing model independence, and others. This document is not an Internet Standards Track specification; it is published for informational purposes.

This document describes an "xcap-diff" SIP (Session Initiation Protocol) event package for the SIP Event Notification Framework, which clients can use to receive notifications of changes to Extensible Markup Language (XML) Configuration Access Protocol (XCAP) resources. The initial synchronization information exchange and document updates are based on the XCAP Diff format. [STANDARDS TRACK]

This document describes the framework, messages, and procedures for the Diameter Quality-of-Service (QoS) application. The Diameter QoS application allows network elements to interact with Diameter servers when allocating QoS resources in the network. In particular, two modes of operation, namely "Pull" and "Push", are defined. [STANDARDS TRACK]

This document specifies how to use the Session Initiation Protocol (SIP) to establish a Secure Real-time Transport Protocol (SRTP) security context using the Datagram Transport Layer Security (DTLS) protocol. It describes a mechanism of transporting a fingerprint attribute in the Session Description Protocol (SDP) that identifies the key that will be presented during the DTLS handshake. The key exchange travels along the media path as opposed to the signaling path. The SIP Identity mechanism can be used to protect the integrity of the fingerprint attribute from modification by intermediate proxies. [STANDARDS-TRACK]