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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.