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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.
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.