SDP

This document describes the mechanisms for allowing a JavaScript application to control the signaling plane of a multimedia session via the interface specified in the W3C RTCPeerConnection API and discusses how this relates to existing signaling protocols.

This specification obsoletes RFC 8829.

This specification defines a new Session Description Protocol (SDP) Grouping Framework extension called 'BUNDLE'. The extension can be used with the SDP offer/answer mechanism to negotiate the usage of a single transport (5-tuple) for sending and receiving media described by multiple SDP media descriptions ("m=" sections). Such transport is referred to as a "BUNDLE transport", and the media is referred to as "bundled media". The "m=" sections that use the BUNDLE transport form a BUNDLE group.

This specification defines a new RTP Control Protocol (RTCP) Source Description (SDES) item and a new RTP header extension.

This specification updates RFCs 3264, 5888, and 7941.

This specification obsoletes RFC 8843.

In this specification, we define a framework for specifying restrictions on RTP streams in the Session Description Protocol (SDP). This framework defines a new "rid" ("restriction identifier") SDP attribute to unambiguously identify the RTP streams within an RTP session and restrict the streams' payload format parameters in a codec-agnostic way beyond what is provided with the regular payload types.

This specification updates RFC 4855 to give additional guidance on choice of Format Parameter (fmtp) names and their relation to the restrictions defined by this document.

In some application scenarios, it may be desirable to send multiple differently encoded versions of the same media source in different RTP streams. This is called simulcast. This document describes how to accomplish simulcast in RTP and how to signal it in the Session Description Protocol (SDP). The described solution uses an RTP/RTCP identification method to identify RTP streams belonging to the same media source and makes an extension to SDP to indicate that those RTP streams are different simulcast formats of that media source. The SDP extension consists of a new media-level SDP attribute that expresses capability to send and/or receive simulcast RTP streams.

This document defines the Session Description Protocol (SDP) offer/answer procedures for negotiating and establishing Binary Floor Control Protocol (BFCP) streams.

This document obsoletes RFC 4583.

Data channel setup can be done using either the in-band Data Channel Establishment Protocol (DCEP) or some out-of-band non-DCEP protocol. This document specifies how the SDP (Session Description Protocol) offer/answer exchange can be used to achieve an out-of-band non-DCEP negotiation for establishing a data channel.

This document specifies how a Web Real-Time Communication (WebRTC) data channel can be used as a transport mechanism for real-time text using the ITU-T Protocol for multimedia application text conversation (Recommendation ITU-T T.140) and how the Session Description Protocol (SDP) offer/answer mechanism can be used to negotiate such a data channel, referred to as a T.140 data channel. This document updates RFC 8373 to specify its use with WebRTC data channels.

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.

This document specifies a Session Description Protocol (SDP) grouping mechanism for RTP media streams that can be used to specify relations between media streams.

This mechanism is used to signal the association between the SDP concept of "media description" and the Web Real-Time Communication (WebRTC) concept of MediaStream/MediaStreamTrack using SDP signaling.

This document describes Session Description Protocol (SDP) Offer/Answer procedures for carrying out Interactive Connectivity Establishment (ICE) between the agents.

This document obsoletes RFCs 5245 and 6336.

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.

The Stream Control Transmission Protocol (SCTP) is a transport protocol used to establish associations between two endpoints. RFC 8261 specifies how SCTP can be used on top of the Datagram Transport Layer Security (DTLS) protocol, which is referred to as SCTP-over-DTLS.

This specification defines the following new Session Description Protocol (SDP) protocol identifiers (proto values): "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP". This specification also specifies how to use the new proto values with the SDP offer/answer mechanism for negotiating SCTP-over-DTLS associations.

This document defines the Session Description Protocol (SDP) offer/answer procedures for negotiating and establishing a Datagram Transport Layer Security (DTLS) association. The document also defines the criteria for when a new DTLS association must be established. The document updates RFCs 5763 and 7345 by replacing common SDP offer/answer procedures with a reference to this specification.

This document defines a new SDP media-level attribute, "tls-id".

This document also defines how the "tls-id" attribute can be used for negotiating and establishing a Transport Layer Security (TLS) connection, in conjunction with the procedures in RFCs 4145 and 8122.

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.

This document defines a new Session Description Protocol (SDP) media-level attribute, 'rtcp-mux-only', that can be used by an endpoint to indicate exclusive support of RTP and RTP Control Protocol (RTCP) multiplexing. The document also updates RFC 5761 by clarifying that an offerer can use a mechanism to indicate that it is not able to send and receive RTCP on separate ports.

The purpose of this specification is to provide a framework for analyzing the multiplexing characteristics of Session Description Protocol (SDP) attributes when SDP is used to negotiate the usage of a single 5-tuple for sending and receiving media associated with multiple media descriptions.

This specification also categorizes the existing SDP attributes based on the framework described herein.

This document specifies how a Web Real-Time Communication (WebRTC) data channel can be used as a transport mechanism for the Message Session Relay Protocol (MSRP) and how the Session Description Protocol (SDP) offer/answer mechanism can be used to negotiate such a data channel, referred to as an MSRP data channel. Two network configurations are supported: the connection of two MSRP data channel endpoints; and a gateway configuration, which connects an MSRP data channel endpoint with an MSRP endpoint that uses either TCP or TLS. This document updates RFC 4975.

This memo describes a Real-time Transport Protocol (RTP) payload format for Timed Text Markup Language (TTML), an XML-based timed text format from W3C. This payload format is specifically targeted at streaming workflows using TTML.

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.

The WebSocket protocol enables bidirectional real-time communication between clients and servers in web-based applications. This document specifies extensions to Session Description Protocol (SDP) for application protocols using WebSocket as a transport.

This document specifies how to establish secure connection-oriented media transport sessions over the Transport Layer Security (TLS) protocol using the Session Description Protocol (SDP). It defines the SDP protocol identifier, 'TCP/TLS'. It also defines the syntax and semantics for an SDP 'fingerprint' attribute that identifies the certificate that will be presented for the TLS session. This mechanism allows media transport over TLS connections to be established securely, so long as the integrity of session descriptions is assured.

This document obsoletes RFC 4572 by clarifying the usage of multiple fingerprints.

This document defines a solution for Network Address Translation (NAT) traversal for datagram-based media streams set up and controlled with the Real-Time Streaming Protocol version 2 (RTSP 2.0). It uses Interactive Connectivity Establishment (ICE) adapted to use RTSP as a signaling channel, defining the necessary RTSP extensions and procedures.

This memorandum defines the Real-Time Streaming Protocol (RTSP) version 2.0, which obsoletes RTSP version 1.0 defined in RFC 2326.

RTSP is an application-layer protocol for the setup and control of the delivery of data with real-time properties. RTSP provides an extensible framework to enable controlled, on-demand delivery of real-time data, such as audio and video. Sources of data can include both live data feeds and stored clips. This protocol is intended to control multiple data delivery sessions; provide a means for choosing delivery channels such as UDP, multicast UDP, and TCP; and provide a means for choosing delivery mechanisms based upon RTP (RFC 3550).

This document updates RFC 5761 by clarifying the SDP offer/answer negotiation of RTP and RTP Control Protocol (RTCP) multiplexing. It makes it clear that an answerer can only include an "a=rtcp-mux" attribute in a Session Description Protocol (SDP) answer if the associated SDP offer contained the attribute.

The Real-time Transport Protocol (RTP) specification establishes a registry of profile names for use by higher-level control protocols, such as the Session Description Protocol (SDP), to refer to the transport methods. This specification describes the following new SDP transport protocol identifiers for transporting RTP Media over TCP: 'TCP/RTP/AVPF', 'TCP/RTP/SAVP', 'TCP/RTP/SAVPF', 'TCP/DTLS/RTP/SAVP', 'TCP/DTLS/RTP/SAVPF', 'TCP/TLS/RTP/AVP', and 'TCP/TLS/RTP/AVPF'.

This document describes several Network Address Translator (NAT) traversal techniques that were considered to be used for establishing the RTP media flows controlled by the Real-Time Streaming Protocol (RTSP). Each technique includes a description of how it would be used, the security implications of using it, and any other deployment considerations it has. There are also discussions on how NAT traversal techniques relate to firewalls and how each technique can be applied in different use cases. These findings were used when selecting the NAT traversal for RTSP 2.0, which is specified in a separate document.

This document describes the behavior of signaling intermediaries in Real-Time Communication (RTC) deployments, sometimes referred to as Session Border Controllers (SBCs), when performing Hosted NAT Traversal (HNT). HNT is a set of mechanisms, such as media relaying and latching, that such intermediaries use to enable other RTC devices behind NATs to communicate with each other.

This document is non-normative and is only written to explain HNT in order to provide a reference to the Internet community and an informative description to manufacturers and users.

Latching, which is one of the HNT components, has a number of security issues covered here. Because of those, and unless all security considerations explained here are taken into account and solved, the IETF advises against use of the latching mechanism over the Internet and recommends other solutions, such as the Interactive Connectivity Establishment (ICE) protocol.

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).

This document provides the offer/answer considerations for the annexa parameter of G723 and the annexb parameter of G729, G729D, and G729E when the value of the annexa or annexb parameter does not match in the Session Description Protocol (SDP) offer and answer.

This memo describes use cases, requirements, and protocol extensions for using the Session Description Protocol (SDP) offer/answer model for establishing audio and video media streams over circuit-switched bearers in the Public Switched Telephone Network (PSTN).

A straightforward approach to provide protection against packet losses due to network outages with a longest duration of T time units is to duplicate the original packets and send each copy separated in time by at least T time units. This approach is commonly referred to as "time-shifted redundancy", "temporal redundancy", or simply "delayed duplication". This document defines an attribute to indicate the presence of temporally redundant media streams and the duplication delay in the Session Description Protocol.

Packet loss is undesirable for real-time multimedia sessions, but it can occur due to congestion or other unplanned network outages. This is especially true for IP multicast networks, where packet loss patterns can vary greatly between receivers. One technique that can be used to recover from packet loss without incurring unbounded delay for all the receivers is to duplicate the packets and send them in separate redundant streams. This document defines the semantics for grouping redundant streams in the Session Description Protocol (SDP). The semantics defined in this document are to be used with the SDP Grouping Framework. Grouping semantics at the Synchronization Source (SSRC) level are also defined in this document for RTP streams using SSRC multiplexing.

The Session Description Protocol (SDP) has been extended with a capability negotiation mechanism framework that allows the endpoints to negotiate transport protocols and attributes. This framework has been extended with a media capabilities negotiation mechanism that allows endpoints to negotiate additional media-related capabilities. This negotiation is embedded into the widely used SDP offer/answer procedures.

This memo extends the SDP capability negotiation framework to allow endpoints to negotiate three additional SDP capabilities. In particular, this memo provides a mechanism to negotiate bandwidth ("b=" line), connection data ("c=" line), and session or media titles ("i=" line for each session or media).

This document proposes a mechanism that allows the same SDP offer to carry multiple IP addresses of different address families (e.g., IPv4 and IPv6). The proposed attribute, the "altc" attribute, solves the backward-compatibility problem that plagued Alternative Network Address Types (ANAT) due to their syntax.

The proposed solution is applicable to scenarios where connectivity checks are not required. If connectivity checks are required, Interactive Connectivity Establishment (ICE), as specified in RFC 5245, provides such a solution.

Session Description Protocol (SDP) capability negotiation provides a general framework for indicating and negotiating capabilities in SDP. The base framework defines only capabilities for negotiating transport protocols and attributes. This documents extends the framework by defining media capabilities that can be used to negotiate media types and their associated parameters.

This document updates the IANA Considerations of RFC 5939.

The wide deployment of Voice over IP (VoIP), real-time text, and Video over IP services has introduced new challenges in managing and maintaining real-time voice/text/video quality, reliability, and overall performance. In particular, media delivery is an area that needs attention. One method of meeting these challenges is monitoring the media delivery performance by looping media back to the transmitter. This is typically referred to as "active monitoring" of services. Media loopback is especially popular in ensuring the quality of transport to the edge of a given VoIP, real-time text, or Video over IP service. Today, in networks that deliver real-time media, short of running 'ping' and 'traceroute' to the edge, administrators are left without the necessary tools to actively monitor, manage, and diagnose quality issues with their service. The extension defined herein adds new Session Description Protocol (SDP) media types and attributes that enable establishment of media sessions where the media is looped back to the transmitter. Such media sessions will serve as monitoring and troubleshooting tools by providing the means for measurement of more advanced VoIP, real-time text, and Video over IP performance metrics.

Interactive Connectivity Establishment (ICE) defines a mechanism for NAT traversal for multimedia communication protocols based on the offer/answer model of session negotiation. ICE works by providing a set of candidate transport addresses for each media stream, which are then validated with peer-to-peer connectivity checks based on Session Traversal Utilities for NAT (STUN). ICE provides a general framework for describing candidates but only defines UDP-based media streams. This specification extends ICE to TCP-based media, including the ability to offer a mix of TCP and UDP-based candidates for a single stream. [STANDARDS-TRACK]

This document describes the usage of the 'image' media type and registers it with IANA as a top-level media type for the Session Description Protocol (SDP). This media type is primarily used by SDP to negotiate and establish T.38 media streams. [STANDARDS-TRACK]

This document specifies the use of the Session Description Protocol (SDP) to describe the parameters required to signal the Forward Error Correction (FEC) Framework Configuration Information between the sender(s) and receiver(s). This document also provides examples that show the semantics for grouping multiple source and repair flows together for the applications that simultaneously use multiple instances of the FEC Framework. [STANDARDS-TRACK]

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.

This document lists the different mechanisms that enable applications using the Real-time Transport Protocol (RTP) and the RTP Control Protocol (RTCP) to keep their RTP Network Address Translator (NAT) mappings alive. It also makes a recommendation for a preferred mechanism. This document is not applicable to Interactive Connectivity Establishment (ICE) agents. [STANDARDS-TRACK]

This document proposes a new generic session setup attribute to make it possible to negotiate different image attributes such as image size. A possible use case is to make it possible for a \%low-end \%hand- held terminal to display video without the need to rescale the image, something that may consume large amounts of memory and processing power. The document also helps to maintain an optimal bitrate for video as only the image size that is desired by the receiver is transmitted. [STANDARDS-TRACK]

This document describes an RTP payload format for the Reduced- Complexity Decoding Operation (RCDO) for H.264 Baseline profile bitstreams, as specified in ITU-T Recommendation H.241. RCDO reduces the decoding cost and resource consumption of the video processing. The RCDO RTP payload format is based on the H.264 RTP payload format. [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.

The Packet-switched Streaming Service (PSS) and the Multimedia Broadcast/Multicast Service (MBMS) defined by 3GPP use the Session Description Protocol (SDP) and Real Time Streaming Protocol (RTSP) with some extensions. This document provides information about these extensions and registers the RTSP and SDP extensions with IANA. This document is not an Internet Standards Track specification; it is published for informational purposes.

This document defines the semantics for grouping the associated source and FEC-based (Forward Error Correction) repair flows in the Session Description Protocol (SDP). The semantics defined in this document are to be used with the SDP Grouping Framework (RFC 5888). These semantics allow the description of grouping relationships between the source and repair flows when one or more source and/or repair flows are associated in the same group, and they provide support for additive repair flows. SSRC-level (Synchronization Source) grouping semantics are also defined in this document for Real-time Transport Protocol (RTP) streams using SSRC multiplexing. [STANDARDS-TRACK]

The Session Description Protocol (SDP) was intended to describe multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. SDP was not intended to provide capability indication or capability negotiation; however, over the years, SDP has seen widespread adoption and as a result it has been gradually extended to provide limited support for these, notably in the form of the offer/answer model defined in RFC 3264. SDP does not define how to negotiate one or more alternative transport protocols (e.g., RTP profiles) or attributes. This makes it difficult to deploy new RTP profiles such as Secure RTP or RTP with RTCP-based feedback, negotiate use of different security keying mechanisms, etc. It also presents problems for some forms of media negotiation.

The purpose of this document is to address these shortcomings by extending SDP with capability negotiation parameters and associated offer/answer procedures to use those parameters in a backwards compatible manner.

The document defines a general SDP Capability Negotiation framework. It also specifies how to provide attributes and transport protocols as capabilities and negotiate them using the framework. Extensions for other types of capabilities (e.g., media types and media formats) may be provided in other documents. [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]

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 memo defines semantics that allow for signaling the decoding dependency of different media descriptions with the same media type in the Session Description Protocol (SDP). This is required, for example, if media data is separated and transported in different network streams as a result of the use of a layered or multiple descriptive media coding process.

A new grouping type "DDP" -- decoding dependency -- is defined, to be used in conjunction with RFC 3388 entitled "Grouping of Media Lines in the Session Description Protocol". In addition, an attribute is specified describing the relationship of the media streams in a "DDP" group indicated by media identification attribute(s) and media format description(s). [STANDARDS-TRACK]

The Session Description Protocol (SDP) provides mechanisms to describe attributes of multimedia sessions and of individual media streams (e.g., Real-time Transport Protocol (RTP) sessions) within a multimedia session, but does not provide any mechanism to describe individual media sources within a media stream. This document defines a mechanism to describe RTP media sources, which are identified by their synchronization source (SSRC) identifiers, in SDP, to associate attributes with these sources, and to express relationships among sources. It also defines several source-level attributes that can be used to describe properties of media sources. [STANDARDS-TRACK]

Speex is an open-source voice codec suitable for use in VoIP (Voice over IP) type applications. This document describes the payload format for Speex-generated bit streams within an RTP packet. Also included here are the necessary details for the use of Speex with the Session Description Protocol (SDP). [STANDARDS-TRACK]

This document provides a mechanism to negotiate the transfer of one or more files between two endpoints by using the Session Description Protocol (SDP) offer/answer model specified in RFC 3264. SDP is extended to describe the attributes of the files to be transferred. The offerer can describe either the files it wants to send or the files it would like to receive. The answerer can either accept or reject the offer separately for each individual file. The transfer of one or more files is initiated after a successful negotiation. The Message Session Relay Protocol (MSRP) is defined as the default mechanism to actually carry the files between the endpoints. The conventions on how to use MSRP for file transfer are also provided in this document. [STANDARDS-TRACK]

The offer/answer model for the Session Description Protocol (SDP) assumes that endpoints somehow establish the Quality of Service (QoS) required for the media streams they establish. Endpoints in closed environments typically agree out-of-band (e.g., using configuration information) regarding which QoS mechanism to use. However, on the Internet, there is more than one QoS service available. Consequently, there is a need for a mechanism to negotiate which QoS mechanism to use for a particular media stream. This document defines such a mechanism. [STANDARDS-TRACK]

This document provides descriptions of Session Description Protocol (SDP) attributes used by the Open Mobile Alliance's Broadcast Service and Content Protection specification. This memo provides information for the Internet community.

This document defines a new security precondition for the Session Description Protocol (SDP) precondition framework described in RFCs 3312 and 4032. A security precondition can be used to delay session establishment or modification until media stream security for a secure media stream has been negotiated successfully. [STANDARDS-TRACK]

This document defines a new Session Description Protocol (SDP) media- level attribute, 'content'. The 'content' attribute defines the content of the media stream to a more detailed level than the media description line. The sender of an SDP session description can attach the 'content' attribute to one or more media streams. The receiving application can then treat each media stream differently (e.g., show it on a big or small screen) based on its content. [STANDARDS-TRACK]

This document describes a scheme to packetize an H.263 video stream for transport using the Real-time Transport Protocol (RTP) with any of the underlying protocols that carry RTP.

The document also describes the syntax and semantics of the Session Description Protocol (SDP) parameters needed to support the H.263 video codec.

The document obsoletes RFC 2429 and updates the H263-1998 and H263-2000 MIME media type in RFC 3555. [STANDARDS-TRACK]

This document defines a new Session Description Protocol (SDP) media-level attribute: "label". The "label" attribute carries a pointer to a media stream in the context of an arbitrary network application that uses SDP. The sender of the SDP document can attach the "label" attribute to a particular media stream or streams. The application can then use the provided pointer to refer to each particular media stream in its context. [STANDARDS-TRACK]

This memo describes a scheme to packetize an H.261 video stream for transport using the Real-time Transport Protocol, RTP, with any of the underlying protocols that carry RTP.

The memo also describes the syntax and semantics of the Session Description Protocol (SDP) parameters needed to support the H.261 video codec. A media type registration is included for this payload format.

This specification obsoletes RFC 2032. [STANDARDS-TRACK]

This document defines the MIME sub-type audio/t38. The usage of this MIME type, which is intended for use within Session Description Protocol (SDP), is specified within ITU-T Recommendation T.38. This memo defines a Historic Document for the Internet community.

This document defines general extensions for Session Description Protocol (SDP) and Real Time Streaming Protocol (RTSP) to carry messages, as specified by a key management protocol, in order to secure the media. These extensions are presented as a framework, to be used by one or more key management protocols. As such, their use is meaningful only when complemented by an appropriate key management protocol.

General guidelines are also given on how the framework should be used together with SIP and RTSP. The usage with the Multimedia Internet KEYing (MIKEY) key management protocol is also defined. [STANDARDS-TRACK]

This document defines a Session Description Protocol (SDP) cryptographic attribute for unicast media streams. The attribute describes a cryptographic key and other parameters that serve to configure security for a unicast media stream in either a single message or a roundtrip exchange. The attribute can be used with a variety of SDP media transports, and this document defines how to use it for the Secure Real-time Transport Protocol (SRTP) unicast media streams. The SDP crypto attribute requires the services of a data security protocol to secure the SDP message. [STANDARDS-TRACK]

This document describes how to adapt the Session Description Protocol (SDP) to express one or more source addresses as a source filter for one or more destination "connection" addresses. It defines the syntax and semantics for an SDP "source-filter" attribute that may reference either IPv4 or IPv6 address(es) as either an inclusive or exclusive source list for either multicast or unicast destinations. In particular, an inclusive source-filter can be used to specify a Source-Specific Multicast (SSM) session. [STANDARDS-TRACK]

This memo specifies requirements for a framework and protocols for accessing and updating Internet Media Guide (IMG) information for media-on-demand and multicast applications. These requirements are designed to guide choice and development of IMG protocols for efficient and scalable delivery. This memo provides information for the Internet community.

This document defines a framework for the delivery of Internet Media Guides (IMGs). An IMG is a structured collection of multimedia session descriptions expressed using the Session Description Protocol (SDP), SDPng, or some similar session description format. This document describes a generalized model for IMG delivery mechanisms, the use of existing protocols, and the need for additional work to create an IMG delivery infrastructure. This memo provides information for the Internet community.

This document gives examples of Session Description Protocol (SDP) offer/answer exchanges. Examples include codec negotiation and selection, hold and resume, and addition and deletion of media streams. The examples show multiple media types, bidirectional, unidirectional, inactive streams, and dynamic payload types. Common Third Party Call Control (3pcc) examples are also given. This memo provides information for the Internet community.

This document describes the RTP payload format for the BroadVoice(R) narrowband and wideband speech codecs. The narrowband codec, called BroadVoice16, or BV16, has been selected by CableLabs as a mandatory codec in PacketCable 1.5 and has a CableLabs specification. The document also provides specifications for the use of BroadVoice with MIME and the Session Description Protocol (SDP). [STANDARDS-TRACK]

This document describes how to express media transport over TCP using the Session Description Protocol (SDP). It defines the SDP 'TCP' protocol identifier, the SDP 'setup' attribute, which describes the connection setup procedure, and the SDP 'connection' attribute, which handles connection reestablishment. [STANDARDS-TRACK]

This document defines a Session Description Protocol (SDP) Transport Independent Application Specific Maximum (TIAS) bandwidth modifier that does not include transport overhead; instead an additional packet rate attribute is defined. The transport independent bit-rate value together with the maximum packet rate can then be used to calculate the real bit-rate over the transport actually used.

The existing SDP bandwidth modifiers and their values include the bandwidth needed for the transport and IP layers. When using SDP with protocols like the Session Announcement Protocol (SAP), the Session Initiation Protocol (SIP), and the Real-Time Streaming Protocol (RTSP), and when the involved hosts has different transport overhead, for example due to different IP versions, the interpretation of what lower layer bandwidths are included is not clear. [STANDARDS-TRACK]

This document defines the Extended Report (XR) packet type for the RTP Control Protocol (RTCP), and defines how the use of XR packets can be signaled by an application if it employs the Session Description Protocol (SDP). XR packets are composed of report blocks, and seven block types are defined here. The purpose of the extended reporting format is to convey information that supplements the six statistics that are contained in the report blocks used by RTCP's Sender Report (SR) and Receiver Report (RR) packets. Some applications, such as multicast inference of network characteristics (MINC) or voice over IP (VoIP) monitoring, require other and more detailed statistics. In addition to the block types defined here, additional block types may be defined in the future by adhering to the framework that this document provides.

The Session Description Protocol (SDP) is used to describe the parameters of media streams used in multimedia sessions. When a session requires multiple ports, SDP assumes that these ports have consecutive numbers. However, when the session crosses a network address translation device that also uses port mapping, the ordering of ports can be destroyed by the translation. To handle this, we propose an extension attribute to SDP.

This document defines an extension to the Session Description Protocol (SDP) to specify two additional modifiers for the bandwidth attribute. These modifiers may be used to specify the bandwidth allowed for RTP Control Protocol (RTCP) packets in a Real-time Transport Protocol (RTP) session. [STANDARDS-TRACK]

This document defines an extension to the Session Description Protocol (SDP) grouping framework. It allows requesting a group of media streams to be mapped into a single resource reservation flow. The SDP syntax needed is defined, as well as a new "semantics" attribute called Single Reservation Flow (SRF). [STANDARDS-TRACK]

The Session Initiation Protocol (SIP) is a text-based protocol for initiating and managing communication sessions. The protocol can be compressed by using Signaling Compression (SigComp). Similarly, the Session Description Protocol (SDP) is a text-based protocol intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. This memo defines the SIP/SDP-specific static dictionary that SigComp may use in order to achieve higher efficiency. The dictionary is compression algorithm independent. [STANDARDS-TRACK]

In wireless environments and especially in cellular systems, e.g. GSM

(Global System for Mobile communications) and UMTS (Universal Mobile

Telecommunications System), there is a need to maximize the transport

efficiency for data over the radio interface. With the introduction

of SIP/SDP (Session Initiation Protocol/Session Description Protocol)

to cellular devices, compression of the signaling messages should be

considered in order to improve both service availability and quality,

mainly by reducing the user idle time, e.g. at call setup. The

purpose of this document is to outline requirements and motivations

for development of a scheme for compression and decompression of

messages from signaling protocols.

This document defines a set of Session Description Protocol (SDP) attributes that enables SDP to provide a minimal and backwards compatible capability declaration mechanism. Such capability declarations can be used as input to a subsequent session negotiation, which is done by means outside the scope of this document. This provides a simple and limited solution to the general capability negotiation problem being addressed by the next generation of SDP, also known as SDPng. [STANDARDS-TRACK]

This document defines a mechanism by which two entities can make use of the Session Description Protocol (SDP) to arrive at a common view of a multimedia session between them. In the model, one participant offers the other a description of the desired session from their perspective, and the other participant answers with the desired session from their perspective. This offer/answer model is most useful in unicast sessions where information from both participants is needed for the complete view of the session. The offer/answer model is used by protocols like the Session Initiation Protocol (SIP). [STANDARDS-TRACK]

The local Message Bus (Mbus) is a light-weight message-oriented coordination protocol for group communication between application components. The Mbus provides automatic location of communication peers, subject based addressing, reliable message transfer and different types of communication schemes. The protocol is layered on top of IP multicast and is specified for IPv4 and IPv6. The IP multicast scope is limited to link-local multicast. This document specifies the Mbus protocol, i.e., message syntax, addressing and transport mechanisms. This memo provides information for the Internet community.

This document describes conventions for using the Session Description Protocol (SDP) described in RFC 2327 for controlling ATM Bearer Connections, and any associated ATM Adaptation Layer (AAL). The AALs addressed are Type 1, Type 2 and Type 5. [STANDARDS-TRACK]

This document describes version 2 of the multicast session directory announcement protocol, Session Announcement Protocol (SAP), and the related issues affecting security and scalability that should be taken into account by implementors. This memo defines an Experimental Protocol for the Internet community.

This document contains the specification of the PINT Service Protocol 1.0, which defines a protocol for invoking certain telephone services from an IP network. [STANDARDS-TRACK]