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.