Avtcore Workgroup RFCs

Browse Avtcore Workgroup RFCs by Number

RFC6263 - Application Mechanism for Keeping Alive the NAT Mappings Associated with RTP / RTP Control Protocol (RTCP) Flows
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]
RFC6284 - Port Mapping between Unicast and Multicast RTP Sessions
This document presents a port mapping solution that allows RTP receivers to choose their own ports for an auxiliary unicast session in RTP applications using both unicast and multicast services. The solution provides protection against denial-of-service or packet amplification attacks that could be used to cause one or more RTP packets to be sent to a victim client. [STANDARDS-TRACK]
RFC6354 - Forward-Shifted RTP Redundancy Payload Support
This document defines a simple enhancement to support RTP sessions with forward-shifted redundant encodings, i.e., redundant data sent before the corresponding primary data. Forward-shifted redundancy can be used to conceal losses of a large number of consecutive media frames (e.g., consecutive loss of seconds or even tens of seconds of media). [STANDARDS-TRACK]
RFC6562 - Guidelines for the Use of Variable Bit Rate Audio with Secure RTP
This memo discusses potential security issues that arise when using variable bit rate (VBR) audio with the secure RTP profile. Guidelines to mitigate these issues are suggested. [STANDARDS-TRACK]
RFC6642 - RTP Control Protocol (RTCP) Extension for a Third-Party Loss Report
In a large RTP session using the RTP Control Protocol (RTCP) feedback mechanism defined in RFC 4585, a feedback target may experience transient overload if some event causes a large number of receivers to send feedback at once. This overload is usually avoided by ensuring that feedback reports are forwarded to all receivers, allowing them to avoid sending duplicate feedback reports. However, there are cases where it is not recommended to forward feedback reports, and this may allow feedback implosion. This memo discusses these cases and defines a new RTCP Third-Party Loss Report that can be used to inform receivers that the feedback target is aware of some loss event, allowing them to suppress feedback. Associated Session Description Protocol (SDP) signaling is also defined. [STANDARDS-TRACK]
RFC6679 - Explicit Congestion Notification (ECN) for RTP over UDP
This memo specifies how Explicit Congestion Notification (ECN) can be used with the Real-time Transport Protocol (RTP) running over UDP, using the RTP Control Protocol (RTCP) as a feedback mechanism. It defines a new RTCP Extended Report (XR) block for periodic ECN feedback, a new RTCP transport feedback message for timely reporting of congestion events, and a Session Traversal Utilities for NAT (STUN) extension used in the optional initialisation method using Interactive Connectivity Establishment (ICE). Signalling and procedures for negotiation of capabilities and initialisation methods are also defined. [STANDARDS-TRACK]
RFC6792 - Guidelines for Use of the RTP Monitoring Framework
This memo proposes an extensible Real-time Transport Protocol (RTP) monitoring framework for extending the RTP Control Protocol (RTCP) with a new RTCP Extended Reports (XR) block type to report new metrics regarding media transmission or reception quality. In this framework, a new XR block should contain a single metric or a small number of metrics relevant to a single parameter of interest or concern, rather than containing a number of metrics that attempt to provide full coverage of all those parameters of concern to a specific application. Applications may then "mix and match" to create a set of blocks that cover their set of concerns. Where possible, a specific block should be designed to be reusable across more than one application, for example, for all of voice, streaming audio, and video. This document is not an Internet Standards Track specification; it is published for informational purposes.
RFC6904 - Encryption of Header Extensions in the Secure Real-time Transport Protocol (SRTP)
The Secure Real-time Transport Protocol (SRTP) provides authentication, but not encryption, of the headers of Real-time Transport Protocol (RTP) packets. However, RTP header extensions may carry sensitive information for which participants in multimedia sessions want confidentiality. This document provides a mechanism, extending the mechanisms of SRTP, to selectively encrypt RTP header extensions in SRTP.
This document updates RFC 3711, the Secure Real-time Transport Protocol specification, to require that all future SRTP encryption transforms specify how RTP header extensions are to be encrypted.
RFC7007 - Update to Remove DVI4 from the Recommended Codecs for the RTP Profile for Audio and Video Conferences with Minimal Control (RTP/AVP)
The RTP Profile for Audio and Video Conferences with Minimal Control (RTP/AVP) is the basis for many other profiles, such as the Secure Real-time Transport Protocol (RTP/SAVP), the Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF), and the Extended Secure RTP Profile for RTCP-Based Feedback (RTP/SAVPF). This document updates RFC 3551, the RTP/AVP profile (and by extension, the profiles that build upon it), to reflect changes in audio codec usage since that document was originally published.
RFC7022 - Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)
The RTP Control Protocol (RTCP) Canonical Name (CNAME) is a persistent transport-level identifier for an RTP endpoint. While the Synchronization Source (SSRC) identifier of an RTP endpoint may change if a collision is detected or when the RTP application is restarted, its RTCP CNAME is meant to stay unchanged, so that RTP endpoints can be uniquely identified and associated with their RTP media streams.
For proper functionality, RTCP CNAMEs should be unique within the participants of an RTP session. However, the existing guidelines for choosing the RTCP CNAME provided in the RTP standard (RFC 3550) are insufficient to achieve this uniqueness. RFC 6222 was published to update those guidelines to allow endpoints to choose unique RTCP CNAMEs. Unfortunately, later investigations showed that some parts of the new algorithms were unnecessarily complicated and/or ineffective. This document addresses these concerns and replaces RFC 6222.
RFC7164 - RTP and Leap Seconds
This document discusses issues that arise when RTP sessions span Coordinated Universal Time (UTC) leap seconds. It updates RFC 3550 by describing how RTP senders and receivers should behave in the presence of leap seconds.
RFC7201 - Options for Securing RTP Sessions
The Real-time Transport Protocol (RTP) is used in a large number of different application domains and environments. This heterogeneity implies that different security mechanisms are needed to provide services such as confidentiality, integrity, and source authentication of RTP and RTP Control Protocol (RTCP) packets suitable for the various environments. The range of solutions makes it difficult for RTP-based application developers to pick the most suitable mechanism. This document provides an overview of a number of security solutions for RTP and gives guidance for developers on how to choose the appropriate security mechanism.
RFC7202 - Securing the RTP Framework: Why RTP Does Not Mandate a Single Media Security Solution
This memo discusses the problem of securing real-time multimedia sessions. It also explains why the Real-time Transport Protocol (RTP) and the associated RTP Control Protocol (RTCP) do not mandate a single media security mechanism. This is relevant for designers and reviewers of future RTP extensions to ensure that appropriate security mechanisms are mandated and that any such mechanisms are specified in a manner that conforms with the RTP architecture.
RFC7272 - Inter-Destination Media Synchronization (IDMS) Using the RTP Control Protocol (RTCP)
This document defines a new RTP Control Protocol (RTCP) Packet Type and an RTCP Extended Report (XR) Block Type to be used for achieving Inter-Destination Media Synchronization (IDMS). IDMS is the process of synchronizing playout across multiple media receivers. Using the RTCP XR IDMS Report Block defined in this document, media playout information from participants in a synchronization group can be collected. Based on the collected information, an RTCP IDMS Settings Packet can then be sent to distribute a common target playout point to which all the distributed receivers, sharing a media experience, can synchronize.
Typical use cases in which IDMS is useful are social TV, shared service control (i.e., applications where two or more geographically separated users are watching a media stream together), distance learning, networked video walls, networked loudspeakers, etc.
RFC7273 - RTP Clock Source Signalling
NTP format timestamps are used by several RTP protocols for synchronisation and statistical measurements. This memo specifies Session Description Protocol (SDP) signalling that identifies timestamp reference clock sources and SDP signalling that identifies the media clock sources in a multimedia session.
RFC7667 - RTP Topologies
This document discusses point-to-point and multi-endpoint topologies used in environments based on the Real-time Transport Protocol (RTP). In particular, centralized topologies commonly employed in the video conferencing industry are mapped to the RTP terminology.
RFC7714 - AES-GCM Authenticated Encryption in the Secure Real-time Transport Protocol (SRTP)
This document defines how the AES-GCM Authenticated Encryption with Associated Data family of algorithms can be used to provide confidentiality and data authentication in the Secure Real-time Transport Protocol (SRTP).
RFC7983 - Multiplexing Scheme Updates for Secure Real-time Transport Protocol (SRTP) Extension for Datagram Transport Layer Security (DTLS)
This document defines how Datagram Transport Layer Security (DTLS), Real-time Transport Protocol (RTP), RTP Control Protocol (RTCP), Session Traversal Utilities for NAT (STUN), Traversal Using Relays around NAT (TURN), and ZRTP packets are multiplexed on a single receiving socket. It overrides the guidance from RFC 5764 ("SRTP Extension for DTLS"), which suffered from four issues described and fixed in this document.
This document updates RFC 5764.
RFC8035 - Session Description Protocol (SDP) Offer/Answer Clarifications for RTP/RTCP Multiplexing
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.
RFC8083 - Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions
The Real-time Transport Protocol (RTP) is widely used in telephony, video conferencing, and telepresence applications. Such applications are often run on best-effort UDP/IP networks. If congestion control is not implemented in these applications, then network congestion can lead to uncontrolled packet loss and a resulting deterioration of the user's multimedia experience. The congestion control algorithm acts as a safety measure by stopping RTP flows from using excessive resources and protecting the network from overload. At the time of this writing, however, while there are several proprietary solutions, there is no standard algorithm for congestion control of interactive RTP flows.
This document does not propose a congestion control algorithm. It instead defines a minimal set of RTP circuit breakers: conditions under which an RTP sender needs to stop transmitting media data to protect the network from excessive congestion. It is expected that, in the absence of long-lived excessive congestion, RTP applications running on best-effort IP networks will be able to operate without triggering these circuit breakers. To avoid triggering the RTP circuit breaker, any Standards Track congestion control algorithms defined for RTP will need to operate within the envelope set by these RTP circuit breaker algorithms.
RFC8108 - Sending Multiple RTP Streams in a Single RTP Session
This memo expands and clarifies the behavior of Real-time Transport Protocol (RTP) endpoints that use multiple synchronization sources (SSRCs). This occurs, for example, when an endpoint sends multiple RTP streams in a single RTP session. This memo updates RFC 3550 with regard to handling multiple SSRCs per endpoint in RTP sessions, with a particular focus on RTP Control Protocol (RTCP) behavior. It also updates RFC 4585 to change and clarify the calculation of the timeout of SSRCs and the inclusion of feedback messages.
RFC8269 - The ARIA Algorithm and Its Use with the Secure Real-Time Transport Protocol (SRTP)
This document defines the use of the ARIA block cipher algorithm within the Secure Real-time Transport Protocol (SRTP). It details two modes of operation (CTR and GCM) and the SRTP key derivation functions for ARIA. Additionally, this document defines DTLS-SRTP protection profiles and Multimedia Internet KEYing (MIKEY) parameter sets for use with ARIA.
RFC8285 - A General Mechanism for RTP Header Extensions
This document provides a general mechanism to use the header extension feature of RTP (the Real-time Transport Protocol). It provides the option to use a small number of small extensions in each RTP packet, where the universe of possible extensions is large and registration is decentralized. The actual extensions in use in a session are signaled in the setup information for that session. This document obsoletes RFC 5285.
RFC8759 - RTP Payload for Timed Text Markup Language (TTML)
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.
RFC8817 - RTP Payload Format for Tactical Secure Voice Cryptographic Interoperability Specification (TSVCIS) Codec
This document describes the RTP payload format for the Tactical Secure Voice Cryptographic Interoperability Specification (TSVCIS) speech coder. TSVCIS is a scalable narrowband voice coder supporting varying encoder data rates and fallbacks. It is implemented as an augmentation to the Mixed Excitation Linear Prediction Enhanced (MELPe) speech coder by conveying additional speech coder parameters to enhance voice quality. TSVCIS augmented speech data is processed in conjunction with its temporally matched Mixed Excitation Linear Prediction (MELP) 2400 speech data. The RTP packetization of TSVCIS and MELPe speech coder data is described in detail.
RFC8860 - Sending Multiple Types of Media in a Single RTP Session
This document specifies how an RTP session can contain RTP streams with media from multiple media types such as audio, video, and text. This has been restricted by the RTP specifications (RFCs 3550 and 3551), and thus this document updates RFCs 3550 and 3551 to enable this behaviour for applications that satisfy the applicability for using multiple media types in a single RTP session.
RFC8861 - Sending Multiple RTP Streams in a Single RTP Session: Grouping RTP Control Protocol (RTCP) Reception Statistics and Other Feedback
RTP allows multiple RTP streams to be sent in a single session but requires each Synchronization Source (SSRC) to send RTP Control Protocol (RTCP) reception quality reports for every other SSRC visible in the session. This causes the number of RTCP reception reports to grow with the number of SSRCs, rather than the number of endpoints. In many cases, most of these RTCP reception reports are unnecessary, since all SSRCs of an endpoint are normally co-located and see the same reception quality. This memo defines a Reporting Group extension to RTCP to reduce the reporting overhead in such scenarios.
RFC8872 - Guidelines for Using the Multiplexing Features of RTP to Support Multiple Media Streams
The Real-time Transport Protocol (RTP) is a flexible protocol that can be used in a wide range of applications, networks, and system topologies. That flexibility makes for wide applicability but can complicate the application design process. One particular design question that has received much attention is how to support multiple media streams in RTP. This memo discusses the available options and design trade-offs, and provides guidelines on how to use the multiplexing features of RTP to support multiple media streams.
RFC8888 - RTP Control Protocol (RTCP) Feedback for Congestion Control
An effective RTP congestion control algorithm requires more fine-grained feedback on packet loss, timing, and Explicit Congestion Notification (ECN) marks than is provided by the standard RTP Control Protocol (RTCP) Sender Report (SR) and Receiver Report (RR) packets. This document describes an RTCP feedback message intended to enable congestion control for interactive real-time traffic using RTP. The feedback message is designed for use with a sender-based congestion control algorithm, in which the receiver of an RTP flow sends back to the sender RTCP feedback packets containing the information the sender needs to perform congestion control.
RFC9071 - RTP-Mixer Formatting of Multiparty Real-Time Text
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.
RFC9134 - RTP Payload Format for ISO/IEC 21122 (JPEG XS)
This document specifies a Real-Time Transport Protocol (RTP) payload format to be used for transporting video encoded with JPEG XS (ISO/IEC 21122). JPEG XS is a low-latency, lightweight image coding system. Compared to an uncompressed video use case, it allows higher resolutions and video frame rates while offering visually lossless quality, reduced power consumption, and encoding-decoding latency confined to a fraction of a video frame.
RFC9328 - RTP Payload Format for Versatile Video Coding (VVC)
This memo describes an RTP payload format for the Versatile Video Coding (VVC) specification, which was published as both ITU-T Recommendation H.266 and ISO/IEC International Standard 23090-3. VVC was developed by the Joint Video Experts Team (JVET). The RTP payload format allows for packetization of one or more Network Abstraction Layer (NAL) units in each RTP packet payload, as well as fragmentation of a NAL unit into multiple RTP packets. The payload format has wide applicability in videoconferencing, Internet video streaming, and high-bitrate entertainment-quality video, among other applications.
RFC9335 - Completely Encrypting RTP Header Extensions and Contributing Sources
While the Secure Real-time Transport Protocol (SRTP) provides confidentiality for the contents of a media packet, a significant amount of metadata is left unprotected, including RTP header extensions and contributing sources (CSRCs). However, this data can be moderately sensitive in many applications. While there have been previous attempts to protect this data, they have had limited deployment, due to complexity as well as technical limitations.
This document updates RFC 3711, the SRTP specification, and defines Cryptex as a new mechanism that completely encrypts header extensions and CSRCs and uses simpler Session Description Protocol (SDP) signaling with the goal of facilitating deployment.
RFC9443 - Multiplexing Scheme Updates for QUIC
RFC 7983 defines a scheme for a Real-time Transport Protocol (RTP) receiver to demultiplex Datagram Transport Layer Security (DTLS), Session Traversal Utilities for NAT (STUN), Secure Real-time Transport Protocol (SRTP) / Secure Real-time Transport Control Protocol (SRTCP), ZRTP, and Traversal Using Relays around NAT (TURN) channel packets arriving on a single port. This document updates RFC 7983 and RFC 5764 to also allow QUIC packets to be multiplexed on a single receiving socket.