]> nil, "sortrefs"=>nil, "symrefs"=>nil}="yes"?> RTP Payload Format for the Secure Communication Interoperability Protocol (SCIP) Codec General Dynamics Mission Systems, Inc.
150 Rustcraft Road Dedham MA 02026 United States of America dan.hanson@gd-ms.com
General Dynamics Mission Systems, Inc.
150 Rustcraft Road Dedham MA 02026 United States of America michael.faller@gd-ms.com
General Dynamics Mission Systems, Inc.
150 Rustcraft Road Dedham MA 02026 United States of America keith.maver@gd-ms.com
Payload Working Group This document describes the RTP payload format of the Secure Communication Interoperability Protocol (SCIP). SCIP is an application layer protocol that provides end-to-end capability exchange, packetization/de-packetization of media, reliable transport, and security services such as confidentiality and integrity protection. SCIP handles packetization/de-packetization of the encrypted media and acts as a tunneling protocol, treating SCIP payloads as opaque octets to be encapsulated within RTP payloads prior to transmission or decapsulated on reception. SCIP payloads are sized to fit within the maximum transmission unit (MTU) when transported over RTP thereby avoiding fragmentation. SCIP provides confidentiality and integrity of the tunneled media, therefore the SCIP payload does not require the use of Secure RTP (SRTP) for payload protection. SCIP also provides for reliable transport at the application layer, so it is not necessary to negotiate RTCP retransmission capabilities. To establish reliable communications using SCIP over RTP, it is important that middle boxes avoid parsing or modifying SCIP payloads. Because SCIP payloads are confidentiality and integrity protected and are only decipherable by the originating and receiving SCIP devices, modification of the payload by middle boxes would be detected as an integrity failure in SCIP devices, resulting in retransmission and/or communication failure. Middle boxes do not need to parse the SCIP payloads to correctly transport them. Not only is parsing unnecessary to tunnel/detunnel SCIP within RTP, but the parsing and filtering of SCIP payloads by middle boxes would likely lead to ossification of the evolving SCIP protocol.
Key Points
  • SCIP is an application layer protocol that uses RTP as a transport. This document defines the SCIP media subtypes to be listed in the Session Description Protocol (SDP) and only requires a basic RTP transport channel for SCIP payloads. This basic transport channel is comparable to Clearmode.
  • SCIP is designed to be network agnostic. It can operate over any digital link, including non-IP modem-based PSTN and ISDN. The SCIP media subtypes listed in this document were developed for SCIP to operate over RTP.
  • SCIP handles packetization/de-packetization of payloads by producing encrypted media packets that are not greater than the MTU size. The SCIP payload is opaque to the network, therefore, SCIP functions as a tunneling protocol for the encrypted media, without the need for middle boxes to parse SCIP payloads. Since SCIP payloads are integrity protected, modification of the SCIP payload is detected as an integrity violation by SCIP endpoints leading to communication failure.
  • SCIP includes built-in mechanisms that negotiate protocol message versions and capabilities. To avoid SCIP protocol ossification (as described in ), it is important for middle boxes to not attempt parsing of the SCIP payload. As described in this document, such parsing serves no useful purpose.
Introduction The purpose of this document is to provide enough information to enable SCIP payloads to be transported through the network without modification or filtering. The document provides a reference for network security policymakers; network equipment OEMs, administrators, and architects; procurement personnel; and government agency and commercial industry representatives. The document details usage of the "audio/scip" and "video/scip" pseudo-codecs , as a secure session establishment protocol and media transport protocol over RTP. It discusses (1) how encrypted audio and video codec payloads are transported over RTP; (2) the IP network layer not implementing SCIP as a protocol since SCIP operates at the application layer in endpoints; (3) the IP network layer enabling SCIP traffic to transparently pass through the network; (4) network devices not recognizing SCIP, and thus removing the scip codecs from the SDP media payload declaration before forwarding to the next network node; and finally, (5) SCIP endpoint devices not operating on networks due to the scip media subtype removal from the SDP media payload declaration. SCIP is presently implemented in the United States and NATO secure voice, video, and data products operating on commercial, private, and tactical IP networks worldwide using the scip media subtype. The SCIP data traversing the network is encrypted, and network equipment in-line with the session cannot interpret the traffic stream in any way. SCIP-based RTP traffic is opaque and can vary significantly in structure and frequency making traffic profiling not possible. Also, as the SCIP protocol continues to evolve independently of this document, any network device that attempts to filter traffic (e.g., deep packet inspection) may cause unintended consequences in the future when changes to the SCIP traffic may not be recognized by the network device. The SCIP protocol defined in SCIP-210 includes built-in support for packetization/de-packetization, retransmission, capability exchange, version negotiation and secure key establishment as well as security services such as end-to-end confidentiality and integrity protection. As a result, neither the RTP transport nor middle boxes can usefully parse or modify SCIP payloads; modifications are detected as integrity violations resulting in retransmission, and eventually, communication failure. Because knowledge of the SCIP payload format is not needed to transport SCIP signaling or media through middle boxes, SCIP-210 represents an informative reference. While older versions of the SCIP-210 specification are publicly available, the authors strongly encourage network implementers to treat SCIP payloads as opaque octets. When handled correctly, such treatment does not require referring to SCIP-210, and any assumptions about the format of SCIP messages defined in SCIP-210 are likely to lead to protocol ossification and communication failures as the protocol evolves.
Conventions The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14 when, and only when, they appear in all capitals, as shown here. Best current practices for writing an RTP payload format specification were followed . When referring to the Secure Communication Interoperability Protocol, the uppercase acronym "SCIP" is used. When referring to the media subtype scip, lowercase "scip" is used.
Abbreviations The following abbreviations are used in this document.
AVP:
Audio/Video Profile
AVPF:
Audio/Video Profile Feedback
ICWG:
Interoperability Control Working Group
IICWG:
International Interoperability Control Working Group
NATO:
North Atlantic Treaty Organization
OEM:
Original Equipment Manufacturer
SAVP:
Secure Audio/Video Profile
SAVPF:
Secure Audio/Video Profile Feedback
SCIP:
Secure Communication Interoperability Protocol
SDP:
Session Description Protocol
SRTP:
Secure Real-Time Transport Protocol
STANAG:
Standardization Agreement
Background The Secure Communication Interoperability Protocol (SCIP) allows the negotiation of several voice, data, and video applications using various cryptographic suites. SCIP also provides several important characteristics that have led to its broad acceptance in the United States and within NATO. These capabilities include end-to-end security at the application layer, authentication of user identity, the ability to apply different security levels for each secure session, and secure communication over any end-to-end data connection. SCIP began in the United States as the Future Narrowband Digital Terminal (FNBDT) Protocol in the late 1990s. A combined U.S. Department of Defense and vendor consortium formed a governing organization named the Interoperability Control Working Group (ICWG) to manage the protocol. In time, the group expanded to include NATO, NATO partners and European vendors under the name International Interoperability Control Working Group (IICWG), which was later renamed the SCIP Working Group. First generation SCIP devices operated on circuit-switched networks. SCIP was then expanded to radio and IP networks. The scip media subtype transports SCIP secure session establishment signaling and secure application traffic. The built-in negotiation and flexibility provided by the SCIP protocols make it a natural choice for many scenarios that require various secure applications and associated encryption suites. SCIP has been adopted by NATO in STANAG 5068. SCIP standards are currently available to participating government/military communities and select OEMs of equipment that support SCIP. However, SCIP must operate over global networks (including private and commercial networks). Without access to necessary information to support SCIP, some networks may not support the SCIP media subtypes. Issues may occur simply because information is not as readily available to OEMs, network administrators, and network architects. This document provides essential information about audio/scip and video/scip media subtypes that enables network equipment manufacturers to include settings for "scip" as a known audio and video media subtype in their equipment. This enables network administrators to define and implement a compatible security policy which includes audio and video media subtypes "audio/scip" and "video/scip", respectively, as permitted codecs on the network. All current IP-based SCIP endpoints implement "scip" as a media subtype. Registration of scip as a media subtype provides a common reference for network equipment manufacturers to recognize SCIP in an SDP payload declaration.
Payload Format The "scip" media subtype indicates support for and identifies SCIP traffic that is being transported over RTP. Transcoding, lossy compression, or other data modifications MUST NOT be performed by the network on the SCIP RTP payload. The audio/scip and video/scip media subtype data streams within the network, including the VoIP network, MUST be a transparent relay and be treated as "clear-channel data", similar to the Clearmode media subtype defined by . RFC 4040 is referenced because Clearmode does not define specific RTP payload content, packet size, or packet intervals, but rather enables Clearmode devices to signal that they support a compatible mode of operation and defines a transparent channel on which devices may communicate. This document takes a similar approach. Network devices that implement support for SCIP need to enable SCIP endpoints to signal that they support SCIP and provide a transparent channel on which SCIP endpoints may communicate. SCIP is an application layer protocol that is defined in SCIP-210. The SCIP traffic consists of encrypted SCIP control messages and codec data. The payload size and interval will vary considerably depending on the state of the SCIP protocol within the SCIP device. Figure 1 below illustrates the RTP payload format for SCIP.
SCIP RTP Payload Format 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | RTP Header | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | | | SCIP payload | | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The SCIP codec produces an encrypted bitstream that is transported over RTP. Unlike other codecs, SCIP does not have its own upper layer syntax (e.g., no Network Adaptation Layer (NAL) units), but rather encrypts the output of the audio/video codecs that it uses (e.g., G.729D, H.264 , etc.). SCIP achieves this by encapsulating the encrypted codec output that has been previously formatted according to the relevant RTP payload specification for that codec. SCIP endpoints MAY employ mechanisms, such as Inter-media RTP Synchronization as described in Section 3.3.4, to synchronize audio/scip and video/scip streams. Figure 2 below illustrates notionally how codec packets and SCIP control messages are packetized for transmission over RTP.
SCIP RTP Architecture +-----------+ +-----------------------+ | Codec | | SCIP control messages | +-----------+ +-----------------------+ | | | | V V +--------------------------------------------------+ | Packetizer* (<= MTU size) | +--------------------------------------------------+ | | | | V | +--------------+ | | Encryption | | +--------------+ | | | | | V V +--------------------------------------------------+ | RTP | +--------------------------------------------------+
As described above, the SCIP RTP payload format is variable and cannot be described in specificity in this document. Details can be found in SCIP-210. SCIP will continue to evolve and as such the SCIP RTP traffic MUST NOT be filtered by network devices based upon what currently is observed or documented. The focus of this document is for network devices to consider the SCIP RTP payload as opaque and allow it to traverse the network. Network devices MUST NOT modify SCIP RTP packets.
RTP Header Fields The SCIP RTP header fields SHALL conform to RFC 3550. SCIP traffic may be continuous or discontinuous. The Timestamp field MUST increment based on the sampling clock for discontinuous transmission as described in , Section 5.1. The Timestamp field for continuous transmission applications is dependent on the sampling rate of the media as specified in the media subtype's specification (e.g., MELPe). Note that during a SCIP session, both discontinuous and continuous traffic are highly probable. The Marker bit SHALL be set to zero for discontinuous traffic. The Marker bit for continuous traffic is based on the underlying media subtype specification. The underlying media is opaque within SCIP RTP packets.
Congestion Control Considerations The bitrate of SCIP may be adjusted depending on the capability of the underlying codec (such as MELPe , G.729D , etc.). The number of encoded audio frames per packet may also be adjusted to control congestion. Discontinuous transmission may also be used if supported by the underlying codec. Since UDP does not provide congestion control, applications that use RTP over UDP SHOULD implement their own congestion control above the UDP layer and MAY also implement a transport circuit breaker . Work in the RTP Media Congestion Avoidance Techniques (RMCAT) working group describes the interactions and conceptual interfaces necessary between the application components that relate to congestion control, including the RTP layer, the higher-level media codec control layer, and the lower-level transport interface, as well as components dedicated to congestion control functions. Use of the packet loss feedback mechanisms in AVPF and SAVPF are OPTIONAL because SCIP itself manages retransmissions of some errored or lost packets. Specifically, the Payload-Specific Feedback Messages defined in RFC 4585 section 6.3 are OPTIONAL when transporting video data.
Use of Augmented RTP Transport Protocols with SCIP The SCIP application layer protocol uses RTP as a basic transport for the audio/scip and video/scip payloads. Additional RTP transport protocols that do not modify the SCIP payload are considered OPTIONAL in this document and are discretionary for a SCIP device vendor to implement. Some examples include but are not limited to:
  • RTP Payload Format for Generic Forward Error Correction
  • Multiplexing RTP Data and Control Packets on a Single Port
  • Symmetric RTP/RTP Control Protocol (RTCP)
  • Negotiating Media Multiplexing Using the Session Description Protocol (BUNDLE)
Payload Format Parameters The SCIP RTP payload format is identified using the scip media subtype, which is registered in accordance with and per the media type registration template form . A clock rate of 8000 Hz SHALL be used for "audio/scip". A clock rate of 90000 Hz SHALL be used for "video/scip".
Media Subtype "audio/scip" Media type name: audio Media subtype name: scip Required parameters: N/A Optional parameters: N/A Encoding considerations: Binary. This media subtype is only defined for transfer via RTP. There SHALL be no encoding/decoding (transcoding) of the audio stream as it traverses the network. Security considerations: See Section 7. Interoperability considerations: N/A Published specifications: Applications which use this media: N/A Fragment Identifier considerations: none Restrictions on usage: N/A Additional information: 1. Deprecated alias names for this type: N/A 2. Magic number(s): N/A 3. File extension(s): N/A 4. Macintosh file type code: N/A 5. Object Identifiers: N/A Person to contact for further information: 1. Name: Michael Faller and Daniel Hanson 2. Email: michael.faller@gd-ms.com and dan.hanson@gd-ms.com Intended usage: Common Authors: Michael Faller - michael.faller@gd-ms.com Daniel Hanson - dan.hanson@gd-ms.com Change controller: SCIP Working Group - ncia.cis3@ncia.nato.int
Media Subtype "video/scip" Media type name: video Media subtype name: scip Required parameters: N/A Optional parameters: N/A Encoding considerations: Binary. This media subtype is only defined for transfer via RTP. There SHALL be no encoding/decoding (transcoding) of the video stream as it traverses the network. Security considerations: See Section 7. Interoperability considerations: N/A Published specifications: Applications which use this media: N/A Fragment Identifier considerations: none Restrictions on usage: N/A Additional information: 1. Deprecated alias names for this type: N/A 2. Magic number(s): N/A 3. File extension(s): N/A 4. Macintosh file type code: N/A 5. Object Identifiers: N/A Person to contact for further information: 1. Name: Michael Faller and Daniel Hanson 2. Email: michael.faller@gd-ms.com and dan.hanson@gd-ms.com Intended usage: Common Authors: Michael Faller - michael.faller@gd-ms.com Daniel Hanson - dan.hanson@gd-ms.com Change controller: SCIP Working Group - ncia.cis3@ncia.nato.int
Mapping to SDP The mapping of the above defined payload format media subtype and its parameters SHALL be implemented according to Section 3 of . Since SCIP includes its own facilities for capabilities exchange, it is only necessary to negotiate the use of SCIP within SDP Offer/Answer; the specific codecs to be encapsulated within SCIP are then negotiated via the exchange of SCIP control messages. The information carried in the media type specification has a specific mapping to fields in the Session Description Protocol (SDP) , which is commonly used to describe RTP sessions. When SDP is used to specify sessions employing the SCIP codec, the mapping is as follows:
  • The media type ("audio") goes in SDP "m=" as the media name for audio/scip, and the media type ("video") goes in SDP "m=" as the media name for video/scip.
  • The media subtype ("scip") goes in SDP "a=rtpmap" as the encoding name. The required parameter "rate" also goes in "a=rtpmap" as the clock rate.
  • The optional parameters "ptime" and "maxptime" go in the SDP "a=ptime" and "a=maxptime" attributes, respectively.
An example mapping for audio/scip is:
An example mapping for video/scip is:
An example mapping for both audio/scip and video/scip is:
SDP Offer/Answer Considerations In accordance with the SDP Offer/Answer model , the SCIP device SHALL list the SCIP payload type number in order of preference in the "m" media line. For example, an SDP Offer with scip as the preferred audio media subtype:
Security Considerations RTP packets using the payload format defined in this specification are subject to the security considerations discussed in the RTP specification , and in any applicable RTP profile such as RTP/AVP , RTP/AVPF , RTP/SAVP , or RTP/SAVPF . However, as "Securing the RTP Protocol Framework: Why RTP Does Not Mandate a Single Media Security Solution" discusses, it is not an RTP payload format's responsibility to discuss or mandate what solutions are used to meet the basic security goals like confidentiality, integrity, and source authenticity for RTP in general. This responsibility lies on anyone using RTP in an application. They can find guidance on available security mechanisms and important considerations in "Options for Securing RTP Sessions" . Applications SHOULD use one or more appropriate strong security mechanisms. The rest of this Security Considerations section discusses the security impacting properties of the payload format itself. This RTP payload format and its media decoder do not exhibit any significant non-uniformity in the receiver-side computational complexity for packet processing, and thus do not inherently pose a denial-of-service threat due to the receipt of pathological data. Nor does the RTP payload format contain any active content. SCIP only encrypts the contents transported in the RTP payload; it does not protect the RTP header or RTCP packets. Applications requiring additional RTP header and/or RTCP security might consider mechanisms such as SRTP , however these additional mechanisms are considered OPTIONAL in this document.
IANA Considerations The audio/scip and video/scip media subtypes have previously been registered with IANA . IANA should update and to reference this document upon publication.
SCIP Contact Information The SCIP protocol is maintained by the SCIP Working Group. The current SCIP-210 specification may be requested from the email address below. SCIP Working Group, CIS3 Partnership
NATO Communications and Information Agency
Oude Waalsdorperweg 61
2597 AK The Hague, Netherlands
Email: ncia.cis3@ncia.nato.int An older public version of the SCIP-210 specification can be downloaded from .
Key words for use in RFCs to Indicate Requirement Levels In many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. 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. An Offer/Answer Model with Session Description Protocol (SDP) This document defines a mechanism by which two entities can make use of the Session Description Protocol (SDP) to arrive at a common view of a multimedia session between them. In the model, one participant offers the other a description of the desired session from their perspective, and the other participant answers with the desired session from their perspective. This offer/answer model is most useful in unicast sessions where information from both participants is needed for the complete view of the session. The offer/answer model is used by protocols like the Session Initiation Protocol (SIP). [STANDARDS-TRACK] 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] 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] 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] 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] 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] Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words RFC 2119 specifies common key words that may be used in protocol specifications. This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the defined special meanings. SDP: Session Description Protocol This memo defines the Session Description Protocol (SDP). SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. This document obsoletes RFC 4566. audio/scip: Internet Assigned Numbers Authority (IANA) 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] 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] Symmetric RTP / RTP Control Protocol (RTCP) 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 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] 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] 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] Media Type Specifications and Registration Procedures This document defines procedures for the specification and registration of media types for use in HTTP, MIME, and other Internet protocols. This memo documents an Internet Best Current Practice. Options for Securing RTP Sessions The Real-time Transport Protocol (RTP) is used in a large number of different application domains and environments. This heterogeneity implies that different security mechanisms are needed to provide services such as confidentiality, integrity, and source authentication of RTP and RTP Control Protocol (RTCP) packets suitable for the various environments. The range of solutions makes it difficult for RTP-based application developers to pick the most suitable mechanism. This document provides an overview of a number of security solutions for RTP and gives guidance for developers on how to choose the appropriate security mechanism. Securing the RTP Framework: Why RTP Does Not Mandate a Single Media Security Solution This memo discusses the problem of securing real-time multimedia sessions. It also explains why the Real-time Transport Protocol (RTP) and the associated RTP Control Protocol (RTCP) do not mandate a single media security mechanism. This is relevant for designers and reviewers of future RTP extensions to ensure that appropriate security mechanisms are mandated and that any such mechanisms are specified in a manner that conforms with the RTP architecture. 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. UDP Usage Guidelines The User Datagram Protocol (UDP) provides a minimal message-passing transport that has no inherent congestion control mechanisms. This document provides guidelines on the use of UDP for the designers of applications, tunnels, and other protocols that use UDP. Congestion control guidelines are a primary focus, but the document also provides guidance on other topics, including message sizes, reliability, checksums, middlebox traversal, the use of Explicit Congestion Notification (ECN), Differentiated Services Code Points (DSCPs), and ports. Because congestion control is critical to the stable operation of the Internet, applications and other protocols that choose to use UDP as an Internet transport must employ mechanisms to prevent congestion collapse and to establish some degree of fairness with concurrent traffic. They may also need to implement additional mechanisms, depending on how they use UDP. Some guidance is also applicable to the design of other protocols (e.g., protocols layered directly on IP or via IP-based tunnels), especially when these protocols do not themselves provide congestion control. This document obsoletes RFC 5405 and adds guidelines for multicast UDP usage. How to Write an RTP Payload Format This document contains information on how best to write an RTP payload format specification. It provides reading tips, design practices, and practical tips on how to produce an RTP payload format specification quickly and with good results. A template is also included with instructions. RTP Payload Format for the Mixed Excitation Linear Prediction Enhanced (MELPe) Codec This document describes the RTP payload format for the Mixed Excitation Linear Prediction Enhanced (MELPe) speech coder. MELPe's three different speech encoding rates and sample frame sizes are supported. Comfort noise procedures and packet loss concealment are described in detail. Negotiating Media Multiplexing Using the Session Description Protocol (SDP) This specification defines a new Session Description Protocol (SDP) Grouping Framework extension called 'BUNDLE'. The extension can be used with the SDP offer/answer mechanism to negotiate the usage of a single transport (5-tuple) for sending and receiving media described by multiple SDP media descriptions ("m=" sections). Such transport is referred to as a "BUNDLE transport", and the media is referred to as "bundled media". The "m=" sections that use the BUNDLE transport form a BUNDLE group. This specification defines a new RTP Control Protocol (RTCP) Source Description (SDES) item and a new RTP header extension. This specification updates RFCs 3264, 5888, and 7941. This specification obsoletes RFC 8843. Long-Term Viability of Protocol Extension Mechanisms The ability to change protocols depends on exercising the extension and version-negotiation mechanisms that support change. This document explores how regular use of new protocol features can ensure that it remains possible to deploy changes to a protocol. Examples are given where lack of use caused changes to be more difficult or costly. RTP Media Congestion Avoidance Techniques (rmcat) Working Group IETF SCIP Signaling Plan SCIP Working Group SCIP-210, r3.11 video/scip: Internet Assigned Numbers Authority (IANA)