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 MichaelFFaller@gmail.com

General Dynamics Mission Systems, Inc.

150 Rustcraft Road Dedham MA 02026 United States of America keith.maver@gd-ms.com

WIT avtcore SCIP FNBDT NATO Department of Defense DoD NSA STANAG ICWG IICWG SCIPWG 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 session establishment, payload encryption, packetization and de-packetization of media, and reliable transport. This document provides a globally available reference that can be used for the development of network equipment and procurement of services that support SCIP traffic. The intended audience is network security policymakers; network administrators, architects, and original equipment manufacturers (OEMs); procurement personnel; and government agency and commercial industry representatives. IESG Note This IETF specification depends upon a second technical specification that is not available publicly, namely . The IETF was therefore unable to conduct a security review of that specification, independently or when carried inside Audio/Video Transport (AVT). Implementers need to be aware that the IETF hence cannot verify any of the security claims contained in this document.

Status of This Memo This is an Internet Standards Track document. This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Further information on Internet Standards is available in Section 2 of RFC 7841. Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at .

Copyright Notice Copyright (c) 2024 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents () in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Revised BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Revised BSD License.

Table of Contents

• .  Key Points
• .  Introduction
  • .  Conventions
  • .  Abbreviations
• .  Background
• .  Payload Format
  • .  RTP Header Fields
  • .  Congestion Control Considerations
  • .  Use of Augmented RTPs with SCIP
• .  Payload Format Parameters
  • .  Media Subtype "audio/scip"
  • .  Media Subtype "video/scip"
  • .  Mapping to SDP
  • .  SDP Offer/Answer Considerations
• .  Security Considerations
• .  IANA Considerations
• .  SCIP Contact Information
• .  References
  • .  Normative References
  • .  Informative References
• Authors' Addresses

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 as defined by .
• SCIP transmits encrypted traffic and 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.
• SCIP includes built-in mechanisms that negotiate protocol message versions and capabilities. To avoid SCIP protocol ossification (as described in ), it is important for middleboxes to not attempt parsing of the SCIP payload. As described in this document, such parsing serves no useful purpose.
• 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 and 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 middleboxes 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.

Introduction This 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 how:

1. encrypted audio and video codec payloads are transported over RTP;
2. the IP network layer does not implement SCIP as a protocol since SCIP operates at the application layer in endpoints;
3. the IP network layer enables SCIP traffic to transparently pass through the network;
4. some network devices do not recognize SCIP and may remove the SCIP codecs from the SDP media payload declaration before forwarding to the next network node; and finally,
5. SCIP endpoint devices do not operate on networks if the SCIP media subtype is removed from the SDP media payload declaration.

The United States, along with its NATO Partners, have implemented SCIP in 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 and de-packetization, retransmission, capability exchange, version negotiation, and payload encryption. Since the traffic is encrypted, neither the RTP transport nor middleboxes 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 middleboxes, 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. The best current practices for writing an RTP payload format specification, as per and , 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-Visual Profile

AVPF:

Audio-Visual Profile with Feedback

FNBDT:

Future Narrowband Digital Terminal

ICWG:

Interoperability Control Working Group

IICWG:

International Interoperability Control Working Group

MELPe:

Mixed Excitation Linear Prediction Enhanced

MTU:

Maximum Transmission Unit

NATO:

North Atlantic Treaty Organization

OEM:

Original Equipment Manufacturer

SAVP:

Secure Audio-Visual Profile

SAVPF:

Secure Audio-Visual Profile with 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 as a secure communications protocol. 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 and 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 the "audio/scip" and "video/scip" media subtypes that enable 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 that 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 identifies and indicates support for 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 . 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. 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 and 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 , to synchronize "audio/scip" and "video/scip" streams. 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 | +--------------------------------------------------+

* Packetizer:

The SCIP application layer will ensure that all traffic sent to the RTP layer will not exceed the MTU size. The receiving SCIP RTP layer will handle packet identification, ordering, and reassembly. When required, the SCIP application layer handles error detection and retransmission.

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 . SCIP traffic may be continuous or discontinuous. The Timestamp field MUST increment based on the sampling clock for discontinuous transmission as described in . 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., Mixed Excitation Linear Prediction Enhanced (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 are OPTIONAL when transporting video data.

Use of Augmented RTPs with SCIP The SCIP application-layer protocol uses RTP as a basic transport for the "audio/scip" and "video/scip" payloads. Additional RTPs 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 (SDP)" a.k.a. 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 from . 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"

Type name:

audio

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 transcoding of the audio stream as it traverses the network.

Security considerations:

See .

Interoperability considerations:

N/A

Published specification:

Applications that use this media type:

N/A

Fragment identifier considerations:

none

Additional information:

Deprecated alias names for this type:

N/A

Magic number(s):

N/A

File extension(s):

N/A

Macintosh file type code(s):

N/A

Person & email address to contact for further information:

Michael Faller (michael.faller@gd-ms.com or MichaelFFaller@gmail.com) and Daniel Hanson (dan.hanson@gd-ms.com)

Intended usage:

COMMON

Restrictions on usage:

N/A

Authors:

Michael Faller (michael.faller@gd-ms.com or MichaelFFaller@gmail.com) and Daniel Hanson (dan.hanson@gd-ms.com)

Change controller:

SCIP Working Group (ncia.cis3@ncia.nato.int)

Media Subtype "video/scip"

Type name:

video

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 transcoding of the video stream as it traverses the network.

Security considerations:

See .

Interoperability considerations:

N/A

Published specification:

Applications that use this media type:

N/A

Fragment identifier considerations:

none

Additional information:

Deprecated alias names for this type:

N/A

Magic number(s):

N/A

File extension(s):

N/A

Macintosh file type code(s):

N/A

Person & email address to contact for further information:

Michael Faller (michael.faller@gd-ms.com or MichaelFFaller@gmail.com) and Daniel Hanson (dan.hanson@gd-ms.com)

Intended usage:

COMMON

Restrictions on usage:

N/A

Authors:

Michael Faller (michael.faller@gd-ms.com or MichaelFFaller@gmail.com) and 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 . 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: m=audio 50000 RTP/AVP 96 a=rtpmap:96 scip/8000 An example mapping for "video/scip" is: m=video 50002 RTP/AVP 97 a=rtpmap:97 scip/90000 An example mapping for both "audio/scip" and "video/scip" is: m=audio 50000 RTP/AVP 96 a=rtpmap:96 scip/8000 m=video 50002 RTP/AVP 97 a=rtpmap:97 scip/90000

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: m=audio 50000 RTP/AVP 96 0 8 a=rtpmap:96 scip/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000

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 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 headers 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 in the "Media Types" registry . IANA has updated these registrations to reference this document.

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 ncia.cis3@ncia.nato.int

An older public version of the SCIP-210 specification can be downloaded from . A U.S. Department of Defense Root Certificate should be installed to access this website.

References Normative References 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. 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. 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] This memorandum describes RTP, the real-time transport protocol. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of- service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers. Most of the text in this memorandum is identical to RFC 1889 which it obsoletes. There are no changes in the packet formats on the wire, only changes to the rules and algorithms governing how the protocol is used. The biggest change is an enhancement to the scalable timer algorithm for calculating when to send RTCP packets in order to minimize transmission in excess of the intended rate when many participants join a session simultaneously. [STANDARDS-TRACK] This document 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] 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] 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] 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] 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. 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. Informative References IANA 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] 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] 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. 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] 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] 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] 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. 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. 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. 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. 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. 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. 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. 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. 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. IETF SCIP Working Group Available by request via email to <ncia.cis3@ncia.nato.int>.

Authors' Addresses 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 MichaelFFaller@gmail.com

General Dynamics Mission Systems, Inc.

150 Rustcraft Road Dedham MA 02026 United States of America keith.maver@gd-ms.com