Avt Workgroup RFCs

Browse Avt Workgroup RFCs by Number

RFC1889 - RTP: A Transport Protocol for Real-Time Applications: 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. [STANDARDS-TRACK]
RFC1890 - RTP Profile for Audio and Video Conferences with Minimal Control: This memo describes a profile for the use of the real-time transport protocol (RTP), version 2, and the associated control protocol, RTCP, within audio and video multiparticipant conferences with minimal control. [STANDARDS-TRACK]
RFC2029 - RTP Payload Format of Sun's CellB Video Encoding: This memo describes a packetization scheme for the CellB video encoding. The scheme proposed allows applications to transport CellB video flows over protocols used by RTP. This document is meant for implementors of video applications that want to use RTP and CellB. [STANDARDS-TRACK]
RFC2032 - RTP Payload Format for H.261 Video Streams: This memo describes a scheme to packetize an H.261 video stream for transport using the Real-time Transport Protocol, RTP, with any of the underlying protocols that carry RTP. [STANDARDS-TRACK]
RFC2035 - RTP Payload Format for JPEG-compressed Video: This memo describes the RTP payload format for JPEG video streams. The packet format is optimized for real-time video streams where codec parameters change rarely from frame to frame. [STANDARDS-TRACK]
RFC2038 - RTP Payload Format for MPEG1/MPEG2 Video: This memo describes a packetization scheme for MPEG video and audio streams. The scheme proposed can be used to transport such a video or audio flow over the transport protocols supported by RTP. [STANDARDS-TRACK]
RFC2190 - RTP Payload Format for H.263 Video Streams: This document specifies the payload format for encapsulating an H.263 bitstream in the Real-Time Transport Protocol (RTP). [STANDARDS-TRACK]
RFC2198 - RTP Payload for Redundant Audio Data: This document describes a payload format for use with the real-time transport protocol (RTP), version 2, for encoding redundant audio data. [STANDARDS-TRACK]
RFC2250 - RTP Payload Format for MPEG1/MPEG2 Video: This memo describes a packetization scheme for MPEG video and audio streams. [STANDARDS-TRACK] The purpose of this document is to express the general Internet community's expectations of Computer Security Incident Response Teams (CSIRTs). It is not possible to define a set of requirements that would be appropriate for all teams, but it is possible and helpful to list and describe the general set of topics and issues which are of concern and interest to constituent communities. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.
RFC2343 - RTP Payload Format for Bundled MPEG: This document describes a payload type for bundled, MPEG-2 encoded video and audio data that may be used with RTP, version 2. This memo defines an Experimental Protocol for the Internet community. This memo does not specify an Internet standard of any kind. Discussion and suggestions for improvement are requested.
RFC2354 - Options for Repair of Streaming Media: This document summarizes a range of possible techniques for the repair of continuous media streams subject to packet loss. This memo provides information for the Internet community. This memo does not specify an Internet standard of any kind.
RFC2429 - RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video (H.263+): This document specifies an RTP payload header format applicable to the transmission of video streams generated based on the 1998 version of ITU-T Recommendation H.263. [STANDARDS-TRACK]
RFC2431 - RTP Payload Format for BT.656 Video Encoding: This document specifies the RTP payload format for encapsulating ITU Recommendation BT.656-3 video streams in the Real-Time Transport Protocol (RTP). [STANDARDS-TRACK]
RFC2435 - RTP Payload Format for JPEG-compressed Video: This memo describes the RTP payload format for JPEG video streams. [STANDARDS-TRACK]
RFC2508 - Compressing IP/UDP/RTP Headers for Low-Speed Serial Links: This document describes a method for compressing the headers of IP/UDP/RTP datagrams to reduce overhead on low-speed serial links. [STANDARDS-TRACK]
RFC2733 - An RTP Payload Format for Generic Forward Error Correction: This document specifies a payload format for generic forward error correction of media encapsulated in RTP. [STANDARDS-TRACK]
RFC2736 - Guidelines for Writers of RTP Payload Format Specifications: This document provides general guidelines aimed at assisting the authors of RTP Payload Format specifications in deciding on good formats. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.
RFC2762 - Sampling of the Group Membership in RTP: This document discusses mechanisms for sampling of this group membership table in order to reduce the memory requirements. This memo defines an Experimental Protocol for the Internet community.
RFC2793 - RTP Payload for Text Conversation: This memo describes how to carry text conversation session contents in RTP packets. Text conversation session contents are specified in ITU-T Recommendation T.140. [STANDARDS-TRACK]
RFC2833 - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals: This memo describes how to carry dual-tone multifrequency (DTMF) signaling, other tone signals and telephony events in RTP packets. [STANDARDS-TRACK]
RFC2862 - RTP Payload Format for Real-Time Pointers: This document describes an RTP payload format for transporting the coordinates of a dynamic pointer that may be used during a presentation. [STANDARDS-TRACK]
RFC2959 - Real-Time Transport Protocol Management Information Base: This memo defines a portion of the Management Information Base (MIB) for use with network management protocols in the Internet community. [STANDARDS-TRACK]
RFC3009 - Registration of parityfec MIME types: The RTP (Real-time Transport Protocol) payload format for generic forward error correction allows RTP participants to improve loss resiliency through the use of traditional parity-based channel codes. This payload format requires four new MIME types, audio/parityfec, video/parityfec, text/parityfec and application/parityfec. This document serves as the MIME type registration for those formats. [STANDARDS-TRACK]
RFC3016 - RTP Payload Format for MPEG-4 Audio/Visual Streams: This document describes Real-Time Transport Protocol (RTP) payload formats for carrying each of MPEG-4 Audio and MPEG-4 Visual bitstreams without using MPEG-4 Systems. [STANDARDS-TRACK]
RFC3047 - RTP Payload Format for ITU-T Recommendation G.722.1: This document describes the payload format for including G.722.1 generated bit streams within an RTP packet. Also included here are the necessary details for the use of G.722.1 with MIME and SDP. [STANDARDS-TRACK]
RFC3119 - A More Loss-Tolerant RTP Payload Format for MP3 Audio: This document describes a RTP (Real-Time Protocol) payload format for transporting MPEG (Moving Picture Experts Group) 1 or 2, layer III audio (commonly known as "MP3"). This format is an alternative to that described in RFC 2250, and performs better if there is packet loss. [STANDARDS-TRACK]
RFC3158 - RTP Testing Strategies: This memo describes a possible testing strategy for RTP (real-time transport protocol) implementations. This memo provides information for the Internet community.
RFC3189 - RTP Payload Format for DV (IEC 61834) Video: This document specifies the packetization scheme for encapsulating the compressed digital video data streams commonly known as "DV" into a payload format for the Real-Time Transport Protocol (RTP). [STANDARDS-TRACK]
RFC3190 - RTP Payload Format for 12-bit DAT Audio and 20- and 24-bit Linear Sampled Audio: This document specifies a packetization scheme for encapsulating 12-bit nonlinear, 20-bit linear, and 24-bit linear audio data streams using the Real-time Transport Protocol (RTP). This document also specifies the format of a Session Description Protocol (SDP) parameter to indicate when audio data is preemphasized before sampling. The parameter may be used with other audio payload formats, in particular L16 (16-bit linear). [STANDARDS-TRACK]
RFC3267 - Real-Time Transport Protocol (RTP) Payload Format and File Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs: This document specifies a real-time transport protocol (RTP) payload format to be used for Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) encoded speech signals. The payload format is designed to be able to interoperate with existing AMR and AMR-WB transport formats on non-IP networks. In addition, a file format is specified for transport of AMR and AMR-WB speech data in storage mode applications such as email. Two separate MIME type registrations are included, one for AMR and one for AMR-WB, specifying use of both the RTP payload format and the storage format. [STANDARDS-TRACK]
RFC3389 - Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)
RFC3497 - RTP Payload Format for Society of Motion Picture and Television Engineers (SMPTE) 292M Video: This memo specifies an RTP payload format for encapsulating uncompressed High Definition Television (HDTV) as defined by the Society of Motion Picture and Television Engineers (SMPTE) standard, SMPTE 292M. SMPTE is the main standardizing body in the motion imaging industry and the SMPTE 292M standard defines a bit-serial digital interface for local area HDTV transport. [STANDARDS-TRACK]
RFC3545 - Enhanced Compressed RTP (CRTP) for Links with High Delay, Packet Loss and Reordering: This document describes a header compression scheme for point to point links with packet loss and long delays. It is based on Compressed Real-time Transport Protocol (CRTP), the IP/UDP/RTP header compression described in RFC 2508. CRTP does not perform well on such links: packet loss results in context corruption and due to the long delay, many more packets are discarded before the context is repaired. To correct the behavior of CRTP over such links, a few extensions to the protocol are specified here. The extensions aim to reduce context corruption by changing the way the compressor updates the context at the decompressor: updates are repeated and include updates to full and differential context parameters. With these extensions, CRTP performs well over links with packet loss, packet reordering and long delays. [STANDARDS-TRACK]
RFC3550 - RTP: A Transport Protocol for Real-Time Applications: 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]
RFC3551 - RTP Profile for Audio and Video Conferences with Minimal Control: This document describes a profile called "RTP/AVP" for the use of the real-time transport protocol (RTP), version 2, and the associated control protocol, RTCP, within audio and video multiparticipant conferences with minimal control. It provides interpretations of generic fields within the RTP specification suitable for audio and video conferences. In particular, this document defines a set of default mappings from payload type numbers to encodings. This document also describes how audio and video data may be carried within RTP. It defines a set of standard encodings and their names when used within RTP. The descriptions provide pointers to reference implementations and the detailed standards. This document is meant as an aid for implementors of audio, video and other real-time multimedia applications. This memorandum obsoletes RFC 1890. It is mostly backwards-compatible except for functions removed because two interoperable implementations were not found. The additions to RFC 1890 codify existing practice in the use of payload formats under this profile and include new payload formats defined since RFC 1890 was published. [STANDARDS-TRACK]
RFC3555 - MIME Type Registration of RTP Payload Formats: This document defines the procedure to register RTP Payload Formats as audio, video or other MIME subtype names. This is useful in a text- based format or control protocol to identify the type of an RTP transmission. This document also registers all the RTP payload formats defined in the RTP Profile for Audio and Video Conferences as MIME subtypes. Some of these may also be used for transfer modes other than RTP. [STANDARDS-TRACK]
RFC3556 - Session Description Protocol (SDP) Bandwidth Modifiers for RTP Control Protocol (RTCP) Bandwidth: 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]
RFC3557 - RTP Payload Format for European Telecommunications Standards Institute (ETSI) European Standard ES 201 108 Distributed Speech Recognition Encoding: This document specifies an RTP payload format for encapsulating European Telecommunications Standards Institute (ETSI) European Standard (ES) 201 108 front-end signal processing feature streams for distributed speech recognition (DSR) systems. [STANDARDS-TRACK]
RFC3558 - RTP Payload Format for Enhanced Variable Rate Codecs (EVRC) and Selectable Mode Vocoders (SMV): This document describes the RTP payload format for Enhanced Variable Rate Codec (EVRC) Speech and Selectable Mode Vocoder (SMV) Speech. Two sub-formats are specified for different application scenarios. A bundled/interleaved format is included to reduce the effect of packet loss on speech quality and amortize the overhead of the RTP header over more than one speech frame. A non-bundled format is also supported for conversational applications. [STANDARDS-TRACK]
RFC3611 - RTP Control Protocol Extended Reports (RTCP XR): 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.
RFC3640 - RTP Payload Format for Transport of MPEG-4 Elementary Streams: The Motion Picture Experts Group (MPEG) Committee (ISO/IEC JTC1/SC29 WG11) is a working group in ISO that produced the MPEG-4 standard. MPEG defines tools to compress content such as audio-visual information into elementary streams. This specification defines a simple, but generic RTP payload format for transport of any non-multiplexed MPEG-4 elementary stream.
RFC3711 - The Secure Real-time Transport Protocol (SRTP): This document describes the Secure Real-time Transport Protocol (SRTP), a profile of the Real-time Transport Protocol (RTP), which can provide confidentiality, message authentication, and replay protection to the RTP traffic and to the control traffic for RTP, the Real-time Transport Control Protocol (RTCP). [STANDARDS-TRACK]
RFC3951 - Internet Low Bit Rate Codec (iLBC): This document specifies a speech codec suitable for robust voice communication over IP. The codec is developed by Global IP Sound (GIPS). It is designed for narrow band speech and results in a payload bit rate of 13.33 kbit/s for 30 ms frames and 15.20 kbit/s for 20 ms frames. The codec enables graceful speech quality degradation in the case of lost frames, which occurs in connection with lost or delayed IP packets. This memo defines an Experimental Protocol for the Internet community.
RFC3952 - Real-time Transport Protocol (RTP) Payload Format for internet Low Bit Rate Codec (iLBC) Speech: This document describes the Real-time Transport Protocol (RTP) payload format for the internet Low Bit Rate Codec (iLBC) Speech developed by Global IP Sound (GIPS). Also, within the document there are included necessary details for the use of iLBC with MIME and Session Description Protocol (SDP). This memo defines an Experimental Protocol for the Internet community.
RFC3984 - RTP Payload Format for H.264 Video: This memo describes an RTP Payload format for the ITU-T Recommendation H.264 video codec and the technically identical ISO/IEC International Standard 14496-10 video codec. The RTP payload format allows for packetization of one or more Network Abstraction Layer Units (NALUs), produced by an H.264 video encoder, in each RTP payload. The payload format has wide applicability, as it supports applications from simple low bit-rate conversational usage, to Internet video streaming with interleaved transmission, to high bit-rate video-on-demand. [STANDARDS-TRACK]
RFC4040 - RTP Payload Format for a 64 kbit/s Transparent Call: This document describes how to carry 64 kbit/s channel data transparently in RTP packets, using a pseudo-codec called "Clearmode". It also serves as registration for a related MIME type called "audio/clearmode".; "Clearmode" is a basic feature of VoIP Media Gateways. [STANDARDS-TRACK]
RFC4060 - RTP Payload Formats for European Telecommunications Standards Institute (ETSI) European Standard ES 202 050, ES 202 211, and ES 202 212 Distributed Speech Recognition Encoding: This document specifies RTP payload formats for encapsulating European Telecommunications Standards Institute (ETSI) European Standard ES 202 050 DSR Advanced Front-end (AFE), ES 202 211 DSR Extended Front-end (XFE), and ES 202 212 DSR Extended Advanced Front-end (XAFE) signal processing feature streams for distributed speech recognition (DSR) systems. [STANDARDS-TRACK]
RFC4102 - Registration of the text/red MIME Sub-Type: This document defines the text/red MIME sub-type. "Red" is short for redundant. The actual RTP packetization for this MIME type is specified in RFC 2198. [STANDARDS-TRACK]
RFC4103 - RTP Payload for Text Conversation: This memo obsoletes RFC 2793; it describes how to carry real-time text conversation session contents in RTP packets. Text conversation session contents are specified in ITU-T Recommendation T.140.; One payload format is described for transmitting text on a separate RTP session dedicated for the transmission of text.; This RTP payload description recommends a method to include redundant text from already transmitted packets in order to reduce the risk of text loss caused by packet loss. [STANDARDS-TRACK]
RFC4170 - Tunneling Multiplexed Compressed RTP (TCRTP): This document describes a method to improve the bandwidth utilization of RTP streams over network paths that carry multiple Real-time Transport Protocol (RTP) streams in parallel between two endpoints, as in voice trunking. The method combines standard protocols that provide compression, multiplexing, and tunneling over a network path for the purpose of reducing the bandwidth used when multiple RTP streams are carried over that path. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements.
RFC4175 - RTP Payload Format for Uncompressed Video: This memo specifies a packetization scheme for encapsulating uncompressed video into a payload format for the Real-time Transport Protocol, RTP. It supports a range of standard- and high-definition video formats, including common television formats such as ITU BT.601, and standards from the Society of Motion Picture and Television Engineers (SMPTE), such as SMPTE 274M and SMPTE 296M. The format is designed to be applicable and extensible to new video formats as they are developed. [STANDARDS-TRACK]
RFC4184 - RTP Payload Format for AC-3 Audio: This document describes an RTP payload format for transporting audio data using the AC-3 audio compression standard. AC-3 is a high quality, multichannel audio coding system that is used for United States HDTV, DVD, cable television, satellite television and other media. The RTP payload format presented in this document includes support for data fragmentation. [STANDARDS-TRACK]
RFC4247 - Requirements for Header Compression over MPLS: Voice over IP (VoIP) typically uses the encapsulation voice/RTP/UDP/IP. When MPLS labels are added, this becomes voice/RTP/UDP/IP/MPLS-labels. For an MPLS VPN, the packet header is typically 48 bytes, while the voice payload is often no more than 30 bytes, for example. Header compression can significantly reduce the overhead through various compression mechanisms, such as enhanced compressed RTP (ECRTP) and robust header compression (ROHC). We consider using MPLS to route compressed packets over an MPLS Label Switched Path (LSP) without compression/decompression cycles at each router. This approach can increase the bandwidth efficiency as well as processing scalability of the maximum number of simultaneous flows that use header compression at each router. In this document, we give a problem statement, goals and requirements, and an example scenario. This memo provides information for the Internet community.
RFC4298 - RTP Payload Format for BroadVoice Speech Codecs: This document describes the RTP payload format for the BroadVoice(R) narrowband and wideband speech codecs. The narrowband codec, called BroadVoice16, or BV16, has been selected by CableLabs as a mandatory codec in PacketCable 1.5 and has a CableLabs specification. The document also provides specifications for the use of BroadVoice with MIME and the Session Description Protocol (SDP). [STANDARDS-TRACK]
RFC4348 - Real-Time Transport Protocol (RTP) Payload Format for the Variable-Rate Multimode Wideband (VMR-WB) Audio Codec: This document specifies a real-time transport protocol (RTP) payload format to be used for the Variable-Rate Multimode Wideband (VMR-WB) speech codec. The payload format is designed to be able to interoperate with existing VMR-WB transport formats on non-IP networks. A media type registration is included for VMR-WB RTP payload format.; VMR-WB is a variable-rate multimode wideband speech codec that has a number of operating modes, one of which is interoperable with AMR-WB (i.e., RFC 3267) audio codec at certain rates. Therefore, provisions have been made in this document to facilitate and simplify data packet exchange between VMR-WB and AMR-WB in the interoperable mode with no transcoding function involved. [STANDARDS-TRACK]
RFC4351 - Real-Time Transport Protocol (RTP) Payload for Text Conversation Interleaved in an Audio Stream: This memo describes how to carry real-time text conversation session contents in RTP packets. Text conversation session contents are specified in ITU-T Recommendation T.140.; One payload format is described for transmitting audio and text data within a single RTP session.; This RTP payload description recommends a method to include redundant text from already transmitted packets in order to reduce the risk of text loss caused by packet loss. This memo defines a Historic Document for the Internet community.
RFC4352 - RTP Payload Format for the Extended Adaptive Multi-Rate Wideband (AMR-WB+) Audio Codec: This document specifies a Real-time Transport Protocol (RTP) payload format for Extended Adaptive Multi-Rate Wideband (AMR-WB+) encoded audio signals. The AMR-WB+ codec is an audio extension of the AMR-WB speech codec. It encompasses the AMR-WB frame types and a number of new frame types designed to support high-quality music and speech. A media type registration for AMR-WB+ is included in this specification. [STANDARDS-TRACK]
RFC4396 - RTP Payload Format for 3rd Generation Partnership Project (3GPP) Timed Text: This document specifies an RTP payload format for the transmission of 3GPP (3rd Generation Partnership Project) timed text. 3GPP timed text is a time-lined, decorated text media format with defined storage in a 3GP file. Timed Text can be synchronized with audio/video contents and used in applications such as captioning, titling, and multimedia presentations. In the following sections, the problems of streaming timed text are addressed, and a payload format for streaming 3GPP timed text over RTP is specified. [STANDARDS-TRACK]
RFC4421 - RTP Payload Format for Uncompressed Video: Additional Colour Sampling Modes: The RFC Payload Format for Uncompressed Video, RFC 4175, defines a scheme to packetise uncompressed, studio-quality, video streams for transport using RTP. This memo extends the format to support additional colour sampling modes. [STANDARDS-TRACK]
RFC4424 - Real-Time Transport Protocol (RTP) Payload Format for the Variable-Rate Multimode Wideband (VMR-WB) Extension Audio Codec: This document is an addendum to RFC 4348, which specifies the RTP payload format for the Variable-Rate Multimode Wideband (VMR-WB) speech codec. This document specifies some updates in RFC 4348 to enable support for the new operating mode of VMR-WB standard (i.e., VMR-WB mode 4). These updates do not affect the existing modes of VMR-WB already specified in RFC 4348.; The payload formats and their associated parameters, as well as all provisions, restrictions, use cases, features, etc., that are specified in RFC 4348 are applicable to the new operating mode with no exception. [STANDARDS-TRACK]
RFC4425 - RTP Payload Format for Video Codec 1 (VC-1): This memo specifies an RTP payload format for encapsulating Video Codec 1 (VC-1) compressed bit streams, as defined by the Society of Motion Picture and Television Engineers (SMPTE) standard, SMPTE 421M. SMPTE is the main standardizing body in the motion imaging industry, and the SMPTE 421M standard defines a compressed video bit stream format and decoding process for television. [STANDARDS-TRACK]
RFC4571 - Framing Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) Packets over Connection-Oriented Transport: 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]
RFC4573 - MIME Type Registration for RTP Payload Format for H.224: In conversational video applications, far-end camera control protocol is used by participants to control the remote camera. The protocol that is commonly used is ITU H.281 over H.224. The document registers the H224 media type. It defines the syntax and the semantics of the Session Description Protocol (SDP) parameters needed to support far-end camera control protocol using H.224. [STANDARDS-TRACK]
RFC4585 - Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF): 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]
RFC4586 - Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback: Results of the Timing Rule Simulations: 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.
RFC4587 - RTP Payload Format for H.261 Video Streams: This memo describes a scheme to packetize an H.261 video stream for transport using the Real-time Transport Protocol, RTP, with any of the underlying protocols that carry RTP.; The memo also describes the syntax and semantics of the Session Description Protocol (SDP) parameters needed to support the H.261 video codec. A media type registration is included for this payload format.; This specification obsoletes RFC 2032. [STANDARDS-TRACK]
RFC4588 - RTP Retransmission Payload Format: 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]
RFC4598 - Real-time Transport Protocol (RTP) Payload Format for Enhanced AC-3 (E-AC-3) Audio: This document describes a Real-time Transport Protocol (RTP) payload format for transporting Enhanced AC-3 (E-AC-3) encoded audio data. E-AC-3 is a high-quality, multichannel audio coding format and is an extension of the AC-3 audio coding format, which is used in US High-Definition Television (HDTV), DVD, cable and satellite television, and other media. E-AC-3 is an optional audio format in US and world wide digital television and high-definition DVD formats. The RTP payload format as presented in this document includes support for data fragmentation. [STANDARDS-TRACK]
RFC4628 - RTP Payload Format for H.263 Moving RFC 2190 to Historic Status: The first RFC that describes RTP payload format for ITU Telecommunication Standardization Sector (ITU-T) recommendation H.263 is RFC 2190. This specification discusses why to move RFC 2190 to historic status. This memo provides information for the Internet community.
RFC4629 - RTP Payload Format for ITU-T Rec. H.263 Video: This document describes a scheme to packetize an H.263 video stream for transport using the Real-time Transport Protocol (RTP) with any of the underlying protocols that carry RTP.; The document also describes the syntax and semantics of the Session Description Protocol (SDP) parameters needed to support the H.263 video codec.; The document obsoletes RFC 2429 and updates the H263-1998 and H263-2000 MIME media type in RFC 3555. [STANDARDS-TRACK]
RFC4695 - RTP Payload Format for MIDI: This memo describes a Real-time Transport Protocol (RTP) payload format for the MIDI (Musical Instrument Digital Interface) command language. The format encodes all commands that may legally appear on a MIDI 1.0 DIN cable. The format is suitable for interactive applications (such as network musical performance) and content-delivery applications (such as file streaming). The format may be used over unicast and multicast UDP and TCP, and it defines tools for graceful recovery from packet loss. Stream behavior, including the MIDI rendering method, may be customized during session setup. The format also serves as a mode for the mpeg4-generic format, to support the MPEG 4 Audio Object Types for General MIDI, Downloadable Sounds Level 2, and Structured Audio. [STANDARDS-TRACK]
RFC4696 - An Implementation Guide for RTP MIDI: This memo offers non-normative implementation guidance for the Real-time Protocol (RTP) MIDI (Musical Instrument Digital Interface) payload format. The memo presents its advice in the context of a network musical performance application. In this application two musicians, located in different physical locations, interact over a network to perform as they would if located in the same room. Underlying the performances are RTP MIDI sessions over unicast UDP. Algorithms for sending and receiving recovery journals (the resiliency structure for the payload format) are described in detail. Although the memo focuses on network musical performance, the presented implementation advice is relevant to other RTP MIDI applications. [STANDARDS-TRACK]
RFC4733 - RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals: This memo describes how to carry dual-tone multifrequency (DTMF) signalling, other tone signals, and telephony events in RTP packets. It obsoletes RFC 2833.; This memo captures and expands upon the basic framework defined in RFC 2833, but retains only the most basic event codes. It sets up an IANA registry to which other event code assignments may be added. Companion documents add event codes to this registry relating to modem, fax, text telephony, and channel-associated signalling events. The remainder of the event codes defined in RFC 2833 are conditionally reserved in case other documents revive their use.; This document provides a number of clarifications to the original document. However, it specifically differs from RFC 2833 by removing the requirement that all compliant implementations support the DTMF events. Instead, compliant implementations taking part in out-of-band negotiations of media stream content indicate what events they support. This memo adds three new procedures to the RFC 2833 framework: subdivision of long events into segments, reporting of multiple events in a single packet, and the concept and reporting of state events. [STANDARDS-TRACK]
RFC4734 - Definition of Events for Modem, Fax, and Text Telephony Signals: This memo updates RFC 4733 to add event codes for modem, fax, and text telephony signals when carried in the telephony event RTP payload. It supersedes the assignment of event codes for this purpose in RFC 2833, and therefore obsoletes that part of RFC 2833. [STANDARDS-TRACK]
RFC4749 - RTP Payload Format for the G.729.1 Audio Codec: This document specifies a Real-time Transport Protocol (RTP) payload format to be used for the International Telecommunication Union (ITU-T) G.729.1 audio codec. A media type registration is included for this payload format. [STANDARDS-TRACK]
RFC4788 - Enhancements to RTP Payload Formats for EVRC Family Codecs: This document updates the Enhanced Variable Rate Codec (EVRC) RTP payload formats defined in RFC 3558 with several enhancements and extensions. In particular, it defines support for the header-free and interleaved/bundled packet formats for the EVRC-B codec, a new compact bundled format for the EVRC and EVRC-B codecs, as well as discontinuous transmission (DTX) support for EVRC and EVRC-B-encoded speech transported via RTP. Voice over IP (VoIP) applications operating over low bandwidth dial-up and wireless networks require such enhancements for efficient use of the bandwidth. [STANDARDS-TRACK]
RFC4855 - Media Type Registration of RTP Payload Formats: This document specifies the procedure to register RTP payload formats as audio, video, or other media subtype names. This is useful in a text-based format description or control protocol to identify the type of an RTP transmission. [STANDARDS-TRACK]
RFC4856 - Media Type Registration of Payload Formats in the RTP Profile for Audio and Video Conferences: This document specifies media type registrations for the RTP payload formats defined in the RTP Profile for Audio and Video Conferences. Some of these may also be used for transfer modes other than RTP. [STANDARDS-TRACK]
RFC4867 - RTP Payload Format and File Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs: This document specifies a Real-time Transport Protocol (RTP) payload format to be used for Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) encoded speech signals. The payload format is designed to be able to interoperate with existing AMR and AMR-WB transport formats on non-IP networks. In addition, a file format is specified for transport of AMR and AMR-WB speech data in storage mode applications such as email. Two separate media type registrations are included, one for AMR and one for AMR-WB, specifying use of both the RTP payload format and the storage format. This document obsoletes RFC 3267. [STANDARDS-TRACK]
RFC4901 - Protocol Extensions for Header Compression over MPLS: This specification defines how to use Multi-Protocol Label Switching (MPLS) to route Header-Compressed (HC) packets over an MPLS label switched path. HC can significantly reduce packet-header overhead and, in combination with MPLS, can also increases bandwidth efficiency and processing scalability in terms of the maximum number of simultaneous compressed flows that use HC at each router). Here we define how MPLS pseudowires are used to transport the HC context and control messages between the ingress and egress MPLS label switching routers. This is defined for a specific set of existing HC mechanisms that might be used, for example, to support voice over IP. This specification also describes extension mechanisms to allow support for future, as yet to be defined, HC protocols. In this specification, each HC protocol operates independently over a single pseudowire instance, very much as it would over a single point-to-point link. [STANDARDS-TRACK]
RFC5104 - Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF): This document specifies a few extensions to the messages defined in the Audio-Visual Profile with Feedback (AVPF). They are helpful primarily in conversational multimedia scenarios where centralized multipoint functionalities are in use. However, some are also usable in smaller multicast environments and point-to-point calls.; The extensions discussed are messages related to the ITU-T Rec. H.271 Video Back Channel, Full Intra Request, Temporary Maximum Media Stream Bit Rate, and Temporal-Spatial Trade-off. [STANDARDS-TRACK]
RFC5109 - RTP Payload Format for Generic Forward Error Correction: This document specifies a payload format for generic Forward Error Correction (FEC) for media data encapsulated in RTP. It is based on the exclusive-or (parity) operation. The payload format described in this document allows end systems to apply protection using various protection lengths and levels, in addition to using various protection group sizes to adapt to different media and channel characteristics. It enables complete recovery of the protected packets or partial recovery of the critical parts of the payload depending on the packet loss situation. This scheme is completely compatible with non-FEC-capable hosts, so the receivers in a multicast group that do not implement FEC can still work by simply ignoring the protection data. This specification obsoletes RFC 2733 and RFC 3009. The FEC specified in this document is not backward compatible with RFC 2733 and RFC 3009. [STANDARDS-TRACK]
RFC5117 - RTP Topologies: This document discusses multi-endpoint topologies used in Real-time Transport Protocol (RTP)-based environments. In particular, centralized topologies commonly employed in the video conferencing industry are mapped to the RTP terminology. This memo provides information for the Internet community.
RFC5124 - Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF): 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]
RFC5188 - RTP Payload Format for the Enhanced Variable Rate Wideband Codec (EVRC-WB) and the Media Subtype Updates for EVRC-B Codec: This document specifies Real-time Transport Protocol (RTP) payload formats to be used for the Enhanced Variable Rate Wideband Codec (EVRC-WB) and updates the media type registrations for EVRC-B codec. Several media type registrations are included for EVRC-WB RTP payload formats. In addition, a file format is specified for transport of EVRC-WB speech data in storage mode applications such as email. [STANDARDS-TRACK]
RFC5215 - RTP Payload Format for Vorbis Encoded Audio: This document describes an RTP payload format for transporting Vorbis encoded audio. It details the RTP encapsulation mechanism for raw Vorbis data and the delivery mechanisms for the decoder probability model (referred to as a codebook), as well as other setup information.; Also included within this memo are media type registrations and the details necessary for the use of Vorbis with the Session Description Protocol (SDP). [STANDARDS-TRACK]
RFC5219 - A More Loss-Tolerant RTP Payload Format for MP3 Audio: This document describes an RTP (Real-Time Protocol) payload format for transporting MPEG (Moving Picture Experts Group) 1 or 2, layer III audio (commonly known as "MP3"). This format is an alternative to that described in RFC 2250, and performs better if there is packet loss. This document obsoletes RFC 3119, correcting typographical errors in the "SDP usage" section and pseudo-code appendices. [STANDARDS-TRACK]
RFC5244 - Definition of Events for Channel-Oriented Telephony Signalling: This memo updates RFC 4733 to add event codes for telephony signals used for channel-associated signalling when carried in the telephony event RTP payload. It supersedes and adds to the original assignment of event codes for this purpose in Section 3.14 of RFC 2833. As documented in Appendix A of RFC 4733, some of the RFC 2833 events have been deprecated because their specification was ambiguous, erroneous, or redundant. In fact, the degree of change from Section 3.14 of RFC 2833 is such that implementations of the present document will be fully backward compatible with RFC 2833 implementations only in the case of full ABCD-bit signalling. This document expands and improves the coverage of signalling systems compared to RFC 2833. [STANDARDS-TRACK]
RFC5285 - 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 de-centralized. The actual extensions in use in a session are signaled in the setup information for that session. [STANDARDS-TRACK]
RFC5371 - RTP Payload Format for JPEG 2000 Video Streams: This memo describes an RTP payload format for the ISO/IEC International Standard 15444-1 | ITU-T Rec. T.800, better known as JPEG 2000. JPEG 2000 features are considered in the design of this payload format. JPEG 2000 is a truly scalable compression technology allowing applications to encode once and decode many different ways. The JPEG 2000 video stream is formed by extending from a single image to a series of JPEG 2000 images. [STANDARDS-TRACK]
RFC5372 - Payload Format for JPEG 2000 Video: Extensions for Scalability and Main Header Recovery: This memo describes extended uses for the payload header in "RTP Payload Format for JPEG 2000 Video Streams" as specified in RFC 5371, for better support of JPEG 2000 features such as scalability and main header recovery.; This memo must be accompanied with a complete implementation of "RTP Payload Format for JPEG 2000 Video Streams". That document is a complete description of the payload header and signaling, this document only describes additional processing for the payload header. There is an additional media type and Session Description Protocol (SDP) marker signaling for implementations of this document. [STANDARDS-TRACK]
RFC5391 - RTP Payload Format for ITU-T Recommendation G.711.1: This document specifies a Real-time Transport Protocol (RTP) payload format to be used for the ITU Telecommunication Standardization Sector (ITU-T) G.711.1 audio codec. Two media type registrations are also included. [STANDARDS-TRACK]
RFC5404 - RTP Payload Format for G.719: This document specifies the payload format for packetization of the G.719 full-band codec encoded audio signals into the Real-time Transport Protocol (RTP). The payload format supports transmission of multiple channels, multiple frames per payload, and interleaving. [STANDARDS-TRACK]
RFC5450 - Transmission Time Offsets in RTP Streams: This document describes a method to inform Real-time Transport Protocol (RTP) clients when RTP packets are transmitted at a time other than their 'nominal' transmission time. It also provides a mechanism to provide improved inter-arrival jitter reports from the clients, that take into account the reported transmission times. [STANDARDS-TRACK]
RFC5459 - G.729.1 RTP Payload Format Update: Discontinuous Transmission (DTX) Support: This document updates the Real-time Transport Protocol (RTP) payload format to be used for the International Telecommunication Union (ITU-T) Recommendation G.729.1 audio codec. It adds Discontinuous Transmission (DTX) support to the RFC 4749 specification, in a backward-compatible way. An updated media type registration is included for this payload format. [STANDARDS-TRACK]
RFC5484 - Associating Time-Codes with RTP Streams: This document describes a mechanism for associating \%time-codes, as defined by the Society of Motion Picture and Television Engineers (SMPTE), with media streams in a way that is independent of the RTP payload format of the media stream itself. [STANDARDS-TRACK]
RFC5506 - Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences: 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]
RFC5574 - RTP Payload Format for the Speex Codec: Speex is an open-source voice codec suitable for use in VoIP (Voice over IP) type applications. This document describes the payload format for Speex-generated bit streams within an RTP packet. Also included here are the necessary details for the use of Speex with the Session Description Protocol (SDP). [STANDARDS-TRACK]
RFC5577 - RTP Payload Format for ITU-T Recommendation G.722.1: International Telecommunication Union (ITU-T) Recommendation G.722.1 is a wide-band audio codec. This document describes the payload format for including G.722.1-generated bit streams within an RTP packet. The document also describes the syntax and semantics of the Session Description Protocol (SDP) parameters needed to support G.722.1 audio codec. [STANDARDS-TRACK]
RFC5584 - RTP Payload Format for the Adaptive TRansform Acoustic Coding (ATRAC) Family: This document describes an RTP payload format for efficient and flexible transporting of audio data encoded with the Adaptive TRansform Audio Coding (ATRAC) family of codecs. Recent enhancements to the ATRAC family of codecs support high-quality audio coding with multiple channels. The RTP payload format as presented in this document also includes support for data fragmentation, elementary redundancy measures, and a variation on scalable streaming. [STANDARDS-TRACK]
RFC5669 - The SEED Cipher Algorithm and Its Use with the Secure Real-Time Transport Protocol (SRTP): This document describes the use of the SEED block cipher algorithm in the Secure Real-time Transport Protocol (SRTP) for providing confidentiality for Real-time Transport Protocol (RTP) traffic and for the control traffic for RTP, the Real-time Transport Control Protocol (RTCP). [STANDARDS-TRACK]
RFC5686 - RTP Payload Format for mU-law EMbedded Codec for Low-delay IP Communication (UEMCLIP) Speech Codec: This document describes the RTP payload format of a mU-law EMbedded Coder for Low-delay IP communication (UEMCLIP), an enhanced speech codec of ITU-T G.711. The bitstream has a scalable structure with an embedded u-law bitstream, also known as PCMU, thus providing a handy transcoding operation between narrowband and wideband speech. [STANDARDS-TRACK]
RFC5691 - RTP Payload Format for Elementary Streams with MPEG Surround Multi-Channel Audio: This memo describes extensions for the RTP payload format defined in RFC 3640 for the transport of MPEG Surround multi-channel audio. Additional Media Type parameters are defined to signal backwards- compatible transmission inside an MPEG-4 Audio elementary stream. In addition, a layered transmission scheme that doesn't use the MPEG-4 systems framework is presented to transport an MPEG Surround elementary stream via RTP in parallel with an RTP stream containing the downmixed audio data. [STANDARDS-TRACK]
RFC5725 - Post-Repair Loss RLE Report Block Type for RTP Control Protocol (RTCP) Extended Reports (XRs): 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]
RFC5760 - RTP Control Protocol (RTCP) Extensions for Single-Source Multicast Sessions with Unicast Feedback: 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]
RFC5761 - Multiplexing RTP Data and Control Packets on a Single Port: This memo discusses issues that arise when multiplexing RTP data packets and RTP Control Protocol (RTCP) packets on a single UDP port. It updates RFC 3550 and RFC 3551 to describe when such multiplexing is and is not appropriate, and it explains how the Session Description Protocol (SDP) can be used to signal multiplexed sessions. [STANDARDS-TRACK]
RFC5764 - Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP): This document describes a Datagram Transport Layer Security (DTLS) extension to establish keys for Secure RTP (SRTP) and Secure RTP Control Protocol (SRTCP) flows. DTLS keying happens on the media path, independent of any out-of-band signalling channel present. [STANDARDS-TRACK]
RFC5968 - Guidelines for Extending the RTP Control Protocol (RTCP): 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.
RFC5993 - RTP Payload Format for Global System for Mobile Communications Half Rate (GSM-HR): This document specifies the payload format for packetization of Global System for Mobile Communications Half Rate (GSM-HR) speech codec data into the Real-time Transport Protocol (RTP). The payload format supports transmission of multiple frames per payload and packet loss robustness methods using redundancy. [STANDARDS-TRACK]
RFC6051 - Rapid Synchronisation of RTP Flows: 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]
RFC6128 - RTP Control Protocol (RTCP) Port for Source-Specific Multicast (SSM) Sessions: 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]
RFC6184 - RTP Payload Format for H.264 Video: This memo describes an RTP Payload format for the ITU-T Recommendation H.264 video codec and the technically identical ISO/IEC International Standard 14496-10 video codec, excluding the Scalable Video Coding (SVC) extension and the Multiview Video Coding extension, for which the RTP payload formats are defined elsewhere. The RTP payload format allows for packetization of one or more Network Abstraction Layer Units (NALUs), produced by an H.264 video encoder, in each RTP payload. The payload format has wide applicability, as it supports applications from simple low bitrate conversational usage, to Internet video streaming with interleaved transmission, to high bitrate video-on-demand.; This memo obsoletes RFC 3984. Changes from RFC 3984 are summarized in Section 14. Issues on backward compatibility to RFC 3984 are discussed in Section 15. [STANDARDS-TRACK]
RFC6185 - RTP Payload Format for H.264 Reduced-Complexity Decoding Operation (RCDO) Video: This document describes an RTP payload format for the Reduced- Complexity Decoding Operation (RCDO) for H.264 Baseline profile bitstreams, as specified in ITU-T Recommendation H.241. RCDO reduces the decoding cost and resource consumption of the video processing. The RCDO RTP payload format is based on the H.264 RTP payload format. [STANDARDS-TRACK]
RFC6188 - The Use of AES-192 and AES-256 in Secure RTP: This memo describes the use of the Advanced Encryption Standard (AES) with 192- and 256-bit keys within the Secure RTP (SRTP) protocol. It details counter mode encryption for SRTP and Secure Realtime Transport Control Protocol (SRTCP) and a new SRTP Key Derivation Function (KDF) for AES-192 and AES-256. [STANDARDS-TRACK]
RFC6222 - 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 are insufficient to achieve this uniqueness. This memo updates those guidelines to allow endpoints to choose unique RTCP CNAMEs. [STANDARDS-TRACK]
RFC6285 - Unicast-Based Rapid Acquisition of Multicast RTP Sessions: When an RTP receiver joins a multicast session, it may need to acquire and parse certain Reference Information before it can process any data sent in the multicast session. Depending on the join time, length of the Reference Information repetition (or appearance) interval, size of the Reference Information, and the application and transport properties, the time lag before an RTP receiver can usefully consume the multicast data, which we refer to as the Acquisition Delay, varies and can be large. This is an undesirable phenomenon for receivers that frequently switch among different multicast sessions, such as video broadcasts.; In this document, we describe a method using the existing RTP and RTP Control Protocol (RTCP) machinery that reduces the acquisition delay. In this method, an auxiliary unicast RTP session carrying the Reference Information to the receiver precedes or accompanies the multicast stream. This unicast RTP flow can be transmitted at a faster than natural bitrate to further accelerate the acquisition. The motivating use case for this capability is multicast applications that carry real-time compressed audio and video. However, this method can also be used in other types of multicast applications where the acquisition delay is long enough to be a problem. [STANDARDS-TRACK]