RTCP

This memo describes the RTCP Payload-Specific Feedback Message Layer Refresh Request (LRR), which can be used to request a state refresh of one or more substreams of a layered media stream. This document also defines its use with several RTP payloads for scalable media formats.

This memo discusses the rate at which congestion control feedback can be sent using the RTP Control Protocol (RTCP) and the suitability of RTCP for implementing congestion control for unicast multimedia applications.

The Real-Time Transport Protocol (RTP) is used to transmit media in telephony and video conferencing applications. This document describes the guidelines to evaluate new congestion control algorithms for interactive point-to-point real-time media.

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.

Congestion control is needed for all data transported across the Internet, in order to promote fair usage and prevent congestion collapse. The requirements for interactive, point-to-point real-time multimedia, which needs low-delay, semi-reliable data delivery, are different from the requirements for bulk transfer like FTP or bursty transfers like web pages. Due to an increasing amount of RTP-based real-time media traffic on the Internet (e.g., with the introduction of the Web Real-Time Communication (WebRTC)), it is especially important to ensure that this kind of traffic is congestion controlled.

This document describes a set of requirements that can be used to evaluate other congestion control mechanisms in order to figure out their fitness for this purpose, and in particular to provide a set of possible requirements for a real-time media congestion avoidance technique.

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.

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.

This document describes monitoring features related to media streams in Web real-time communication (WebRTC). It provides a list of RTP Control Protocol (RTCP) Sender Report (SR), Receiver Report (RR), and Extended Report (XR) metrics, which may need to be supported by RTP implementations in some diverse environments. It lists a set of identifiers for the WebRTC's statistics API. These identifiers are a set of RTCP SR, RR, and XR metrics related to the transport of multimedia flows.

Content splicing is a process that replaces the content of a main multimedia stream with other multimedia content and that delivers the substitutive multimedia content to the receivers for a period of time. The splicer is designed to handle RTP splicing and needs to know when to start and end the splicing.

This memo defines two RTP/RTCP extensions to indicate the splicing-related information to the splicer: an RTP header extension that conveys the information "in band" and an RTP Control Protocol (RTCP) packet that conveys the information out of band.

SIP Back-to-Back User Agents (B2BUAs) are often designed to also be on the media path, rather than just to intercept signalling. This means that B2BUAs often implement an RTP or RTP Control Protocol (RTCP) stack as well, thus leading to separate multimedia sessions that the B2BUA correlates and bridges together. If not disciplined, this behaviour can severely impact the communication experience, especially when statistics and feedback information contained in RTCP messages get lost because of mismatches in the reported data.

This document defines the proper behaviour B2BUAs should follow when acting on both the signalling plane and media plane in order to preserve the end-to-end functionality of RTCP.

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 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.

This document defines an RTP Control Protocol (RTCP) Extended Report (XR) block that allows the reporting of burst/gap discard metrics independently of the burst/gap loss metrics for use in a range of RTP applications.

Source Description (SDES) items are normally transported in the RTP Control Protocol (RTCP). In some cases, it can be beneficial to speed up the delivery of these items. The main case is when a new synchronization source (SSRC) joins an RTP session and the receivers need this source's identity, relation to other sources, or its synchronization context, all of which may be fully or partially identified using SDES items. To enable this optimization, this document specifies a new RTP header extension that can carry SDES items.

This document defines a new RTP Control Protocol (RTCP) Extended Report (XR) block that allows the reporting of loss concealment metrics for video applications of RTP.

With the increased popularity of real-time multimedia applications, it is desirable to provide good control of resource usage, and users also demand more control over communication sessions. This document describes how a receiver in a multimedia conversation can pause and resume incoming data from a sender by sending real-time feedback messages when using the Real-time Transport Protocol (RTP) for real- time data transport. This document extends the Codec Control Message (CCM) RTP Control Protocol (RTCP) feedback package by explicitly allowing and describing specific use of existing CCMs and adding a group of new real-time feedback messages used to pause and resume RTP data streams. This document updates RFC 5104.

This document defines an RTP Control Protocol (RTCP) Extended Report (XR) block that allows reporting of a post-repair loss count metric for a range of RTP applications. In addition, another metric, repaired loss count, is also introduced in this report block for calculating the pre-repair loss count when needed, so that the RTP sender or a third-party entity is able to evaluate the effectiveness of the repair methods used by the system.

An MPEG2 Transport Stream (TS) is a standard container format used in the transmission and storage of multimedia data. Unicast/multicast MPEG2 TS over RTP is widely deployed in IPTV systems. This document defines an RTP Control Protocol (RTCP) Extended Report (XR) block that allows the reporting of MPEG2 TS decodability statistics metrics related to transmissions of MPEG2 TS over RTP. The metrics specified in the RTCP XR block are related to Program Specific Information (PSI) carried in MPEG TS.

This document defines two RTP Control Protocol (RTCP) Extended Report (XR) blocks that allow the reporting of concealment metrics for audio applications of RTP.

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.

This document defines an RTP Control Protocol (RTCP) Extended Report (XR) Block including two new segment types and associated Session Description Protocol (SDP) parameters that allow the reporting of mean opinion score (MOS) Metrics for use in a range of RTP applications.

This document defines two RTP Control Protocol (RTCP) Extended Report (XR) blocks that allow the reporting of initial synchronization delay and synchronization offset metrics for use in a range of RTP applications.

The RTP Control Protocol (RTCP) is used in conjunction with the Real-time Transport Protocol (RTP) to provide a variety of short-term and long-term reception statistics. The available reporting may include aggregate information across longer periods of time as well as individual packet reporting. This document specifies a report computing the bytes discarded from the de-jitter buffer after successful reception.

The RTP Control Protocol (RTCP) is used in conjunction with the Real- time Transport Protocol (RTP) in order to provide a variety of short- term and long-term reception statistics. The available reporting may include aggregate information across longer periods of time as well as individual packet reporting. This document specifies a per-packet report metric capturing individual packets discarded from the de- jitter buffer after successful reception.

This document defines an RTP Control Protocol (RTCP) Extended Report (XR) block that allows the reporting of a simple discard count metric for use in a range of RTP applications.

This document defines an RTP Control Protocol (RTCP) Extended Report (XR) block that allows the reporting of burst and gap discard metrics for use in a range of RTP applications.

This document defines three RTP Control Protocol (RTCP) Extended Report (XR) blocks that allow the reporting of loss, duplication, and discard summary statistics metrics in a range of RTP applications.

This document defines an RTP Control Protocol (RTCP) Extended Report (XR) block that allows the reporting of de-jitter buffer metrics for a range of RTP applications.

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.

An MPEG-2 Transport Stream (TS) is a standard container format used in the transmission and storage of multimedia data. Unicast/ multicast MPEG-2 TS over RTP is widely deployed in IPTV systems. This document defines an RTP Control Protocol (RTCP) Extended Report (XR) block that allows the reporting of MPEG-2 TS decodability statistics metrics related to transmissions of MPEG-2 TS over RTP. The metrics specified in the RTCP XR block are not dependent on Program Specific Information (PSI) carried in MPEG-2 TS.

This document defines an RTP Control Protocol (RTCP) Extended Report (XR) Block that allows the reporting of burst and gap loss metrics for use in a range of RTP applications.

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.

This document defines an RTP Control Protocol (RTCP) Extended Report (XR) block that allows the reporting of delay metrics for use in a range of Real-time Transport Protocol (RTP) applications. [STANDARDS-TRACK]

This document defines an RTP Control Protocol (RTCP) Extended Report (XR) block that allows the reporting of packet delay variation metrics for a range of RTP applications. [STANDARDS-TRACK]

This document defines an RTP Control Protocol (RTCP) Source Description (SDES) item and an RTCP Extended Report (XR) block carrying parameters that identify and describe a measurement period to which one or more other RTCP XR blocks may refer. [STANDARDS-TRACK]

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]

The Session PEERing for Multimedia INTerconnect (SPEERMINT) working group (WG) provides a peering framework that leverages the building blocks of existing IETF-defined protocols such as SIP and ENUM for the interconnection between SIP Service Providers (SSPs). The objective of this document is to identify and enumerate SPEERMINT- specific threat vectors and to give guidance for implementers on selecting appropriate countermeasures. Security requirements for SPEERMINT that have been derived from the threats detailed in this document can be found in RFC 6271; this document provides concrete countermeasures to meet those SPEERMINT security requirements. In this document, the different security threats related to SPEERMINT are classified into threats to the Lookup Function (LUF), the Location Routing Function (LRF), the Signaling Function (SF), and the Media Function (MF) of a specific SIP Service Provider. Various instances of the threats are briefly introduced inside the classification. Finally, existing security solutions for SIP and RTP/RTCP (Real-time Transport Control Protocol) are presented to describe countermeasures currently available for such threats. Each SSP may have connections to one or more remote SSPs through peering or transit contracts. A potentially compromised remote SSP that attacks other SSPs is out of the scope of this document; this document focuses on attacks on an SSP from outside the trust domain such an SSP may have with other SSPs. This document is not an Internet Standards Track specification; it is published for informational purposes.

In most RTP-based multicast applications, the RTP source sends inter- related data. Due to this interdependency, randomly joining RTP receivers usually cannot start consuming the multicast data right after they join the session. Thus, they often experience a random acquisition delay. An RTP receiver can use one or more different approaches to achieve rapid acquisition. Yet, due to various factors, performance of the rapid acquisition methods usually varies. Furthermore, in some cases, the RTP receiver can do a simple multicast join (in other cases, it is compelled to do so). For quality reporting, monitoring, and diagnostic purposes, it is important to collect detailed information from the RTP receivers about their acquisition and presentation experiences. This document addresses this issue by defining a new report block type, called the Multicast Acquisition (MA) report block, within the framework of RTP Control Protocol (RTCP) Extended Reports (XRs) (RFC 3611). This document also defines the necessary signaling of the new MA report block type in the Session Description Protocol (SDP). [STANDARDS-TRACK]

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]

The Session Description Protocol (SDP) has an attribute that allows RTP applications to specify an address and a port associated with the RTP Control Protocol (RTCP) traffic. In RTP-based source-specific multicast (SSM) sessions, the same attribute is used to designate the address and the RTCP port of the Feedback Target in the SDP description. However, the RTCP port associated with the SSM session itself cannot be specified by the same attribute to avoid ambiguity, and thus, is required to be derived from the "m=" line of the media description. Deriving the RTCP port from the "m=" line imposes an unnecessary restriction. This document removes this restriction by introducing a new SDP attribute. [STANDARDS-TRACK]

This document defines a Session Initiation Protocol (SIP) event package that enables the collection and reporting of metrics that measure the quality for Voice over Internet Protocol (VoIP) sessions. Voice call quality information derived from RTP Control Protocol Extended Reports (RTCP-XR) and call information from SIP is conveyed from a User Agent (UA) in a session, known as a reporter, to a third party, known as a collector. A registration for the application/ vq-rtcpxr media type is also included. [STANDARDS-TRACK]

This memo outlines how RTP sessions are synchronised, and discusses how rapidly such synchronisation can occur. We show that most RTP sessions can be synchronised immediately, but that the use of video switching multipoint conference units (MCUs) or large source-specific multicast (SSM) groups can greatly increase the synchronisation delay. This increase in delay can be unacceptable to some applications that use layered and/or multi-description codecs.

This memo introduces three mechanisms to reduce the synchronisation delay for such sessions. First, it updates the RTP Control Protocol (RTCP) timing rules to reduce the initial synchronisation delay for SSM sessions. Second, a new feedback packet is defined for use with the extended RTP profile for RTCP-based feedback (RTP/AVPF), allowing video switching MCUs to rapidly request resynchronisation. Finally, new RTP header extensions are defined to allow rapid synchronisation of late joiners, and guarantee correct timestamp-based decoding order recovery for layered codecs in the presence of clock skew. [STANDARDS-TRACK]

The RTP Control Protocol (RTCP) is used along with the Real-time Transport Protocol (RTP) to provide a control channel between media senders and receivers. This allows constructing a feedback loop to enable application adaptation and monitoring, among other uses. The basic reporting mechanisms offered by RTCP are generic, yet quite powerful and suffice to cover a range of uses. This document provides guidelines on extending RTCP if those basic mechanisms prove insufficient. This document is not an Internet Standards Track specification; it is published for informational purposes.

This document defines a new report block type within the framework of RTP Control Protocol (RTCP) Extended Reports (XRs). One of the initial XR report block types is the Loss Run Length Encoding (RLE) Report Block. This report conveys information regarding the individual Real-time Transport Protocol (RTP) packet receipt and loss events experienced during the RTCP interval preceding the transmission of the report. The new report, which is referred to as the Post-repair Loss RLE report, carries information regarding the packets that remain lost after all loss-repair methods are applied. By comparing the RTP packet receipts/losses before and after the loss repair is completed, one can determine the effectiveness of the loss- repair methods in an aggregated fashion. This document also defines the signaling of the Post-repair Loss RLE report in the Session Description Protocol (SDP). [STANDARDS-TRACK]

This document specifies an extension to the Real-time Transport Control Protocol (RTCP) to use unicast feedback to a multicast sender. The proposed extension is useful for single-source multicast sessions such as Source-Specific Multicast (SSM) communication where the traditional model of many-to-many group communication is either not available or not desired. In addition, it can be applied to any group that might benefit from a sender-controlled summarized reporting mechanism. [STANDARDS-TRACK]

This memo discusses benefits and issues that arise when allowing Real-time Transport Protocol (RTCP) packets to be transmitted with reduced size. The size can be reduced if the rules on how to create compound packets outlined in RFC 3550 are removed or changed. Based on that analysis, this memo defines certain changes to the rules to allow feedback messages to be sent as Reduced-Size RTCP packets under certain conditions when using the RTP/AVPF (Real-time Transport Protocol / Audio-Visual Profile with Feedback) profile (RFC 4585). This document updates RFC 3550, RFC 3711, and RFC 4585. [STANDARDS-TRACK]

An RTP profile (SAVP) for secure real-time communications and another profile (AVPF) to provide timely feedback from the receivers to a sender are defined in RFC 3711 and RFC 4585, respectively. This memo specifies the combination of both profiles to enable secure RTP communications with feedback. [STANDARDS-TRACK]

This document describes an RTCP XR report block, which reports packet transport parameters. The report block was developed by BT for pre-standards use in BT's next-generation network. This document has been produced to describe the report block in sufficient detail to register the block type with IANA in accordance with the Specification Required policy of RFC 3611. This specification does not standardise the new report block for use outside BT's network. This memo provides information for the Internet community.

This document recommends using one UDP port pair for both communication directions of bidirectional RTP and RTP Control Protocol (RTCP) sessions, commonly called "symmetric RTP" and "symmetric RTCP". This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.

RTP retransmission is an effective packet loss recovery technique for real-time applications with relaxed delay bounds. This document describes an RTP payload format for performing retransmissions. Retransmitted RTP packets are sent in a separate stream from the original RTP stream. It is assumed that feedback from receivers to senders is available. In particular, it is assumed that Real-time Transport Control Protocol (RTCP) feedback as defined in the extended RTP profile for RTCP-based feedback (denoted RTP/AVPF) is available in this memo. [STANDARDS-TRACK]

This memo defines a method for framing Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) packets onto connection-oriented transport (such as TCP). The memo also defines how session descriptions may specify RTP streams that use the framing method. [STANDARDS-TRACK]

Real-time media streams that use RTP are, to some degree, resilient against packet losses. Receivers may use the base mechanisms of the Real-time Transport Control Protocol (RTCP) to report packet reception statistics and thus allow a sender to adapt its transmission behavior in the mid-term. This is the sole means for feedback and feedback-based error repair (besides a few codec-specific mechanisms). This document defines an extension to the Audio-visual Profile (AVP) that enables receivers to provide, statistically, more immediate feedback to the senders and thus allows for short-term adaptation and efficient feedback-based repair mechanisms to be implemented. This early feedback profile (AVPF) maintains the AVP bandwidth constraints for RTCP and preserves scalability to large groups. [STANDARDS-TRACK]

This document describes the results achieved when simulating the timing rules of the Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback, denoted AVPF. Unicast and multicast topologies are considered as well as several protocol and environment configurations. The results show that the timing rules result in better performance regarding feedback delay and still preserve the well-accepted RTP rules regarding allowed bit rates for control traffic. This memo provides information for the Internet community.

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 memorandum describes RTP, the real-time transport protocol. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of- service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers. Most of the text in this memorandum is identical to RFC 1889 which it obsoletes. There are no changes in the packet formats on the wire, only changes to the rules and algorithms governing how the protocol is used. The biggest change is an enhancement to the scalable timer algorithm for calculating when to send RTCP packets in order to minimize transmission in excess of the intended rate when many participants join a session simultaneously. [STANDARDS-TRACK]

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]