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RFC 8088 - How to Write an RTP Payload Format

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Internet Engineering Task Force (IETF)                     M. Westerlund
Request for Comments: 8088                                      Ericsson
Updates: 2736                                                   May 2017
Category: Informational
ISSN: 2070-1721

                   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.

Status of This Memo

   This document is not an Internet Standards Track specification; it is
   published for informational purposes.

   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).  Not all documents
   approved by the IESG are a candidate for any level of Internet
   Standard; see 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) 2017 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
   (http://trustee.ietf.org/license-info) 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 Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1. Introduction ....................................................4
      1.1. Structure ..................................................4
   2. Terminology .....................................................5
      2.1. Definitions ................................................5
      2.2. Abbreviations ..............................................5
      2.3. Use of Normative Requirements Language .....................6
   3. Preparations ....................................................6
      3.1. Read and Understand the Media Coding Specification .........6
      3.2. Recommended Reading ........................................7
           3.2.1. IETF Process and Publication ........................7
           3.2.2. RTP .................................................9
      3.3. Important RTP Details .....................................13
           3.3.1. The RTP Session ....................................13
           3.3.2. RTP Header .........................................14
           3.3.3. RTP Multiplexing ...................................16
           3.3.4. RTP Synchronization ................................16
      3.4. Signaling Aspects .........................................18
           3.4.1. Media Types ........................................19
           3.4.2. Mapping to SDP .....................................20
      3.5. Transport Characteristics .................................23
           3.5.1. Path MTU ...........................................23
           3.5.2. Different Queuing Algorithms .......................23
           3.5.3. Quality of Service .................................24
   4. Standardization Process for an RTP Payload Format ..............24
      4.1. IETF ......................................................25
           4.1.1. Steps from Idea to Publication .....................25
           4.1.2. WG Meetings ........................................27
           4.1.3. Draft Naming .......................................27
           4.1.4. Writing Style ......................................28
           4.1.5. How to Speed Up the Process ........................29
      4.2. Other Standards Bodies ....................................29
      4.3. Proprietary and Vendor Specific ...........................30
      4.4. Joint Development of Media Coding Specification
           and RTP Payload Format ....................................31
   5. Designing Payload Formats ......................................31
      5.1. Features of RTP Payload Formats ...........................32
           5.1.1. Aggregation ........................................32
           5.1.2. Fragmentation ......................................33
           5.1.3. Interleaving and Transmission Rescheduling .........33
           5.1.4. Media Back Channels ................................34
           5.1.5. Media Scalability ..................................34
           5.1.6. High Packet Rates ..................................37
      5.2. Selecting Timestamp Definition ............................37

   6. Noteworthy Aspects in Payload Format Design ....................39
      6.1. Audio Payloads ............................................39
      6.2. Video .....................................................40
      6.3. Text ......................................................41
      6.4. Application ...............................................41
   7. Important Specification Sections ...............................42
      7.1. Media Format Description ..................................42
      7.2. Security Considerations ...................................43
      7.3. Congestion Control ........................................44
      7.4. IANA Considerations .......................................45
   8. Authoring Tools ................................................45
      8.1. Editing Tools .............................................46
      8.2. Verification Tools ........................................46
   9. Security Considerations ........................................47
   10. Informative References ........................................47
   Appendix A. RTP Payload Format Template ...........................58
     A.1.  Title .....................................................58
     A.2.  Front-Page Boilerplate ....................................58
     A.3.  Abstract ..................................................58
     A.4.  Table of Contents .........................................58
     A.5.  Introduction ..............................................59
     A.6.  Conventions, Definitions, and Abbreviations ...............59
     A.7.  Media Format Description ..................................59
     A.8.  Payload Format ............................................59
       A.8.1.  RTP Header Usage ......................................59
       A.8.2.  Payload Header ........................................59
       A.8.3.  Payload Data ..........................................60
     A.9.  Payload Examples ..........................................60
     A.10. Congestion Control Considerations .........................60
     A.11. Payload Format Parameters .................................60
       A.11.1.  Media Type Definition ................................60
       A.11.2.  Mapping to SDP .......................................62
     A.12. IANA Considerations .......................................63
     A.13. Security Considerations ...................................63
     A.14. RFC Editor Considerations .................................64
     A.15. References ................................................64
       A.15.1.  Normative References .................................64
       A.15.2.  Informative References ...............................64
     A.16. Authors' Addresses ........................................64
   Acknowledgements ..................................................64
   Contributors ......................................................65
   Author's Address ..................................................65

1.  Introduction

   RTP [RFC3550] payload formats define how a specific real-time data
   format is structured in the payload of an RTP packet.  A real-time
   data format without a payload format specification cannot be
   transported using RTP.  This creates an interest in many individuals/
   organizations with media encoders or other types of real-time data to
   define RTP payload formats.  However, the specification of a well-
   designed RTP payload format is nontrivial and requires knowledge of
   both RTP and the real-time data format.

   This document is intended to help any author of an RTP payload format
   specification make important design decisions, consider important
   features of RTP and RTP security, etc.  The document is also intended
   to be a good starting point for any person with little experience in
   the IETF and/or RTP to learn the necessary steps.

   This document extends and updates the information that is available
   in "Guidelines for Writers of RTP Payload Format Specifications"
   [RFC2736].  Since that RFC was written, further experience has been
   gained on the design and specification of RTP payload formats.
   Several new RTP profiles and robustness tools have been defined, and
   these need to be considered.

   This document also discusses the possible venues for defining an RTP
   payload format: the IETF, other standards bodies, and proprietary

   Note, this document does discuss IETF, IANA, and RFC Editor processes
   and rules as they were when this document was published.  This to
   make clear how the work to specify an RTP payload formats depends,
   uses, and interacts with these rules and processes.  However, these
   rules and processes are subject to change and the formal rule and
   process specifications always takes precedence over what is written

1.1.  Structure

   This document has several different parts discussing different
   aspects of the creation of an RTP payload format specification.
   Section 3 discusses the preparations the author(s) should make before
   starting to write a specification.  Section 4 discusses the different
   processes used when specifying and completing a payload format, with
   focus on working inside the IETF.  Section 5 discusses the design of
   payload formats themselves in detail.  Section 6 discusses current
   design trends and provides good examples of practices that should be
   followed when applicable.  Following that, Section 7 provides a
   discussion on important sections in the RTP payload format

   specification itself such as Security Considerations and IANA
   Considerations.  This document ends with an appendix containing a
   template that can be used when writing RTP payload formats

2.  Terminology

2.1.  Definitions

   RTP Stream:  A sequence of RTP packets that together carry part or
      all of the content of a specific media (audio, video, text, or
      data whose form and meaning are defined by a specific real-time
      application) from a specific sender source within a given RTP

   RTP Session:  An association among a set of participants
      communicating with RTP.  The distinguishing feature of an RTP
      session is that each session maintains a full, separate space of
      synchronization source (SSRC) identifiers.  See also
      Section 3.3.1.

   RTP Payload Format:  The RTP payload format specifies how units of a
      specific encoded media are put into the RTP packet payloads and
      how the fields of the RTP packet header are used, thus enabling
      the format to be used in RTP applications.

   A Taxonomy of Semantics and Mechanisms for Real-Time Transport
   Protocol (RTP) Sources [RFC7656] defines many useful terms.

2.2.  Abbreviations

   ABNF:  Augmented Backus-Naur Form [RFC5234]

   ADU:  Application Data Unit

   ALF:  Application Level Framing

   ASM:  Any-Source Multicast

   BCP:  Best Current Practice

   I-D:  Internet-Draft

   IESG:  Internet Engineering Steering Group

   MTU:  Maximum Transmission Unit

   WG:  Working Group

   QoS:  Quality of Service

   RFC:  Request For Comments

   RTP:  Real-time Transport Protocol

   RTCP:  RTP Control Protocol

   RTT:  Round-Trip Time

   SSM:  Source-Specific Multicast

2.3.  Use of Normative Requirements Language

   As this document is both Informational and instructional rather than
   a specification, this document does not use any RFC 2119 language and
   the use of "may", "should", "recommended", and "must" carries no
   special connotation.

3.  Preparations

   RTP is a complex real-time media delivery framework, and it has a lot
   of details that need to be considered when writing an RTP payload
   format.  It is also important to have a good understanding of the
   media codec / format so that all of its important features and
   properties are considered.  Only when one has sufficient
   understanding of both parts can one produce an RTP payload format of
   high quality.  On top of this, one needs to understand the process
   within the IETF and especially the Working Group responsible for
   standardizing payload formats (currently the PAYLOAD WG) to go
   quickly from the initial idea stage to a finished RFC.  This and the
   next sections help an author prepare himself in those regards.

3.1.  Read and Understand the Media Coding Specification

   It may be obvious, but it is necessary for an author of an RTP
   payload specification to have a solid understanding of the media to
   be transported.  Important are not only the specifically spelled out
   transport aspects (if any) in the media coding specification, but
   also core concepts of the underlying technology.  For example, an RTP
   payload format for video coded with inter-picture prediction will
   perform poorly if the payload designer does not take the use of
   inter-picture prediction into account.  On the other hand, some
   (mostly older) media codecs offer error-resilience tools against bit
   errors, which, when misapplied over RTP, in almost all cases would
   only introduce overhead with no measurable return.

3.2.  Recommended Reading

   The following subsections list a number of documents.  Not all need
   to be read in full detail.  However, an author basically needs to be
   aware of everything listed below.

3.2.1.  IETF Process and Publication

   Newcomers to the IETF are strongly recommended to read the "Tao of
   the IETF" [TAO] that goes through most things that one needs to know
   about the IETF: the history, organizational structure, how the WGs
   and meetings work, etc.

   It is very important to note and understand the IETF Intellectual
   Property Rights (IPR) policy that requires early disclosures based on
   personal knowledge from anyone contributing in IETF.  The IETF
   policies associated with IPR are documented in BCP 78 [BCP78]
   (related to copyright, including software copyright, for example,
   code) and BCP 79 [BCP79] (related to patent rights).  These rules may
   be different from other standardization organizations.  For example,
   a person that has a patent or a patent application that he or she
   reasonably and personally believes to cover a mechanism that gets
   added to the Internet-Draft they are contributing to (e.g., by
   submitting the draft, posting comments or suggestions on a mailing
   list, or speaking at a meeting) will need to make a timely IPR
   disclosure.  Read the above documents for the authoritative rules.
   Failure to follow the IPR rules can have dire implications for the
   specification and the author(s) as discussed in [RFC6701].

      Note: These IPR rules apply on what is specified in the RTP
      payload format Internet-Draft (and later RFC); an IPR that relates
      to a codec specification from an external body does not require
      IETF IPR disclosure.  Informative text explaining the nature of
      the codec would not normally require an IETF IPR declaration.
      Appropriate IPR declarations for the codec itself would normally
      be found in files of the external body defining the codec, in
      accordance with that external body's own IPR rules.

   The main part of the IETF process is formally defined in BCP 9
   [BCP9].  BCP 25 [BCP25] describes the WG process, the relation
   between the IESG and the WG, and the responsibilities of WG Chairs
   and participants.

   It is important to note that the RFC Series contains documents of
   several different publication streams as defined by The RFC Series
   and RFC Editor [RFC4844].  The most important stream for RTP payload
   formats authors is the IETF Stream.  In this stream, the work of the
   IETF is published.  The stream contains documents of several

   different categories: Standards Track, Informational, Experimental,
   Best Current Practice, and Historic.  "Standards Track" contains two
   maturity levels: Proposed Standard and Internet Standard [RFC6410].
   A Standards Track document must start as a Proposed Standard; after
   successful deployment and operational experience with at least two
   implementations, it can be moved to an Internet Standard.  The
   Independent Submission Stream could appear to be of interest as it
   provides a way of publishing documents of certain categories such as
   Experimental and Informational with a different review process.
   However, as long as IETF has a WG that is chartered to work on RTP
   payload formats, this stream should not be used.

   As the content of a given RFC is not allowed to change once
   published, the only way to modify an RFC is to write and publish a
   new one that either updates or replaces the old one.  Therefore,
   whether reading or referencing an RFC, it is important to consider
   both the Category field in the document header and to check if the
   RFC is the latest on the subject and still valid.  One way of
   checking the current status of an RFC is to use the RFC Editor's RFC
   search page (https://www.rfc-editor.org/search), which displays the
   current status and which if any RFC has updated or obsoleted it.  The
   RFC Editor search engine will also indicate if there exist any errata
   reports for the RFC.  Any verified errata report contains issues of
   significant importance with the RFC; thus, they should be known prior
   to an update and replacement publication.

   Before starting to write a draft, one should also read the Internet-
   Draft writing guidelines (http://www.ietf.org/ietf/1id-
   guidelines.txt), the I-D checklist (http://www.ietf.org/ID-
   Checklist.html), and the RFC Style Guide [RFC7322].  Another document
   that can be useful is "Guide for Internet Standards Writers"

   There are also a number of documents to consider in the process of
   writing drafts intended to become RFCs.  These are important when
   writing certain types of text.

   RFC 2606:  When writing examples using DNS names in Internet-Drafts,
      those names shall be chosen from the example.com, example.net, and
      example.org domains.

   RFC 3849:  Defines the range of IPv6 unicast addresses
      (2001:DB8::/32) that should be used in any examples.

   RFC 5737:  Defines the ranges of IPv4 unicast addresses reserved for
      documentation and examples:,, and

   RFC 5234:  Augmented Backus-Naur Form (ABNF) is often used when
      writing text field specifications.  Not commonly used in RTP
      payload formats, but may be useful when defining media type
      parameters of some complexity.

3.2.2.  RTP

   The recommended reading for RTP consists of several different parts:
   design guidelines, the RTP protocol, profiles, robustness tools, and
   media-specific recommendations.

   Any author of RTP payload formats should start by reading "Guidelines
   for Writers of RTP Payload Format Specifications" [RFC2736], which
   contains an introduction to the Application Level Framing (ALF)
   principle, the channel characteristics of IP channels, and design
   guidelines for RTP payload formats.  The goal of ALF is to be able to
   transmit Application Data Units (ADUs) that are independently usable
   by the receiver in individual RTP packets, thus minimizing
   dependencies between RTP packets and the effects of packet loss.

   Then, it is advisable to learn more about the RTP protocol, by
   studying the RTP specification "RTP: A Transport Protocol for Real-
   Time Applications" [RFC3550] and the existing profiles.  As a
   complement to the Standards Track documents, there exists a book
   totally dedicated to RTP [CSP-RTP].  There exist several profiles for
   RTP today, but all are based on "RTP Profile for Audio and Video
   Conferences with Minimal Control" [RFC3551] (abbreviated as RTP/AVP).
   The other profiles that one should know about are "The Secure Real-
   time Transport Protocol (SRTP)" (RTP/SAVP) [RFC3711], "Extended RTP
   Profile for RTCP-based Feedback (RTP/AVPF)" [RFC4585], and "Extended
   Secure Real-time Transport Control Protocol (RTCP)-Based Feedback
   (RTP/SAVPF)" [RFC5124].  It is important to understand RTP and the
   RTP/AVP profile in detail.  For the other profiles, it is sufficient
   to have an understanding of what functionality they provide and the
   limitations they create.

   A number of robustness tools have been developed for RTP.  The tools
   are for different use cases and real-time requirements.

   RFC 2198:  "RTP Payload for Redundant Audio Data" [RFC2198] provides
      functionalities to transmit redundant copies of audio or text
      payloads.  These redundant copies are sent together with a primary
      format in the same RTP payload.  This format relies on the RTP
      timestamp to determine where data belongs in a sequence;
      therefore, it is usually most suitable to be used with audio.
      However, the RTP Payload format for T.140 [RFC4103] text format
      also uses this format.  The format's major property is that it
      only preserves the timestamp of the redundant payloads, not the

      original sequence number.  This makes it unusable for most video
      formats.  This format is also only suitable for media formats that
      produce relatively small RTP payloads.

   RFC 6354:  The "Forward-Shifted RTP Redundancy Payload Support"
      [RFC6354] is a variant of RFC 2198 that allows the redundant data
      to be transmitted prior to the original.

   RFC 5109:  The "RTP Payload Format for Generic Forward Error
      Correction" [RFC5109] provides an XOR-based Forward Error
      Correction (FEC) of the whole or parts of a number of RTP packets.
      This specification replaced the previous specification for XOR-
      based FEC [RFC2733].  These FEC packets are sent in a separate
      stream or as a redundant encoding using RFC 2198.  This FEC scheme
      has certain restrictions in the number of packets it can protect.
      It is suitable for applications with low-to-medium delay tolerance
      with a limited amount of RTP packets.

   RFC 6015:  "RTP Payload Format for 1-D Interleaved Parity Forward
      Error Correction (FEC)" [RFC6015] provides a variant of the XOR-
      based Generic protection defined in [RFC2733].  The main
      difference is to use interleaving scheme on which packets gets
      included as source packets for a particular protection packet.
      The interleaving is defined by using every L packets as source
      data and then producing protection data over D number of packets.
      Thus, each block of D x L source packets will result in L number
      of Repair packets, each capable of repairing one loss.  The goal
      is to provide better burst-error robustness when the packet rate
      is higher.

   FEC Framework:  "Forward Error Correction (FEC) Framework" [RFC6363]
      defines how to use FEC protection for arbitrary packet flows.
      This framework can be applied for RTP/RTCP packet flows, including
      using RTP for transmission of repair symbols, an example is in
      "RTP Payload Format for Raptor Forward Error Correction (FEC)"

   RTP Retransmission:  The RTP retransmission scheme [RFC4588] is used
      for semi-reliability of the most important RTP packets in a RTP
      stream.  The level of reliability between semi- and in-practice
      full reliability depends on the targeted properties and situation
      where parameters such as round-trip time (RTT) allowed additional
      overhead and allowable delay.  It often requires the application
      to be quite delay tolerant as a minimum of one round-trip time
      plus processing delay is required to perform a retransmission.
      Thus, it is mostly suitable for streaming applications but may
      also be usable in certain other cases when operating in networks
      with short round-trip times.

   RTP over TCP:  RFC 4571 [RFC4571] defines how one sends RTP and RTCP
      packets over connection-oriented transports like TCP.  If one uses
      TCP, one gets reliability for all packets but loses some of the
      real-time behavior that RTP was designed to provide.  Issues with
      TCP transport of real-time media include head-of-line blocking and
      wasting resources on retransmission of data that is already late.
      TCP is also limited to point-to-point connections, which further
      restricts its applicability.

   There have been both discussion and design of RTP payload formats,
   e.g., Adaptive Multi-Rate (AMR) and AMR Wideband (AMR-WB) [RFC4867],
   supporting the unequal error detection provided by UDP-Lite
   [RFC3828].  The idea is that by not having a checksum over part of
   the RTP payload one can allow bit errors from the lower layers.  By
   allowing bit errors one can increase the efficiency of some link
   layers and also avoid unnecessary discarding of data when the payload
   and media codec can get at least some benefit from the data.  The
   main issue is that one has no idea of the level of bit errors present
   in the unprotected part of the payload.  This makes it hard or
   impossible to determine whether or not one can design something
   usable.  Payload format designers are not recommended to consider
   features for unequal error detection using UDP-Lite unless very clear
   requirements exist.

   There also exist some management and monitoring extensions.

   RFC 2959:  The RTP protocol Management Information Database (MIB)
      [RFC2959] that is used with SNMP [RFC3410] to configure and
      retrieve information about RTP sessions.

   RFC 3611:  The RTCP Extended Reports (RTCP XR) [RFC3611] consists of
      a framework for reports sent within RTCP.  It can easily be
      extended by defining new report formats, which has and is
      occurring.  The XRBLOCK WG in the IETF is chartered (at the time
      of writing) with defining new report formats.  The list of
      specified formats is available in IANA's RTCP XR Block Type
      registry (http://www.iana.org/assignments/rtcp-xr-block-types/).
      The report formats that are defined in RFC 3611 provide report
      information on packet loss, packet duplication, packet reception
      times, RTCP statistics summary, and VoIP Quality.  [RFC3611] also
      defines a mechanism that allows receivers to calculate the RTT to
      other session participants when used.

   RMONMIB:  The Remote Network Monitoring WG has defined a mechanism
      [RFC3577] based on usage of the MIB that can be an alternative to
      RTCP XR.

   A number of transport optimizations have also been developed for use
   in certain environments.  They are all intended to be transparent and
   do not require special consideration by the RTP payload format
   writer.  Thus, they are primarily listed here for informational

   RFC 2508:  "Compressing IP/UDP/RTP Headers for Low-Speed Serial
      Links" (CRTP) [RFC2508] is the first IETF-developed RTP header
      compression mechanism.  It provides quite good compression;
      however, it has clear performance problems when subject to packet
      loss or reordering between compressor and decompressor.

   RFCs 3095 and 5795:  These are the base specifications of the robust
      header compression (ROHC) protocol version 1 [RFC3095] and version
      2 [RFC5795].  This solution was created as a result of CRTP's lack
      of performance when compressed packets are subject to loss.

   RFC 3545:  Enhanced compressed RTP (E-CRTP) [RFC3545] was developed
      to provide extensions to CRTP that allow for better performance
      over links with long RTTs, packet loss, and/or reordering.

   RFC 4170:  "Tunneling Multiplexed Compressed RTP (TCRTP)" [RFC4170]
      is a solution that allows header compression within a tunnel
      carrying multiple multiplexed RTP flows.  This is primarily used
      in voice trunking.

   There exist a couple of different security mechanisms that may be
   used with RTP.  By definition, generic mechanisms are transparent for
   the RTP payload format and do not need special consideration by the
   format designer.  The main reason that different solutions exist is
   that different applications have different requirements; thus,
   different solutions have been developed.  For more discussion on
   this, please see "Options for Securing RTP Sessions" [RFC7201] and
   "Securing the RTP Framework: Why RTP Does Not Mandate a Single Media
   Security Solution" [RFC7202].  The main properties for an RTP
   security mechanism are to provide confidentiality for the RTP
   payload, integrity protection to detect manipulation of payload and
   headers, and source authentication.  Not all mechanisms provide all
   of these features, a point that will need to be considered when a
   specific mechanisms is chosen.

   The profile for Secure RTP - SRTP (RTP/SAVP) [RFC3711] and the
   derived profile (RTP/SAVPF [RFC5124]) are a solution that enables
   confidentiality, integrity protection, replay protection, and partial
   source authentication.  It is the solution most commonly used with
   RTP at the time of writing this document.  There exist several key-
   management solutions for SRTP, as well other choices, affecting the

   security properties.  For a more in-depth review of the options and
   solutions other than SRTP consult "Options for Securing RTP Sessions"

3.3.  Important RTP Details

   This section reviews a number of RTP features and concepts that are
   available in RTP, independent of the payload format.  The RTP payload
   format can make use of these when appropriate, and even affect the
   behavior (RTP timestamp and marker bit), but it is important to note
   that not all features and concepts are relevant to every payload
   format.  This section does not remove the necessity to read up on
   RTP.  However, it does point out a few important details to remember
   when designing a payload format.

3.3.1.  The RTP Session

   The definition of the RTP session from RFC 3550 is:

      An association among a set of participants communicating with RTP.
      A participant may be involved in multiple RTP sessions at the same
      time.  In a multimedia session, each medium is typically carried
      in a separate RTP session with its own RTCP packets unless the
      encoding itself multiplexes multiple media into a single data
      stream.  A participant distinguishes multiple RTP sessions by
      reception of different sessions using different pairs of
      destination transport addresses, where a pair of transport
      addresses comprises one network address plus a pair of ports for
      RTP and RTCP.  All participants in an RTP session may share a
      common destination transport address pair, as in the case of IP
      multicast, or the pairs may be different for each participant, as
      in the case of individual unicast network addresses and port
      pairs.  In the unicast case, a participant may receive from all
      other participants in the session using the same pair of ports, or
      may use a distinct pair of ports for each.

      The distinguishing feature of an RTP session is that each session
      maintains a full, separate space of SSRC identifiers (defined
      next).  The set of participants included in one RTP session
      consists of those that can receive an SSRC identifier transmitted
      by any one of the participants either in RTP as the SSRC or a CSRC
      (also defined below) or in RTCP.  For example, consider a three-
      party conference implemented using unicast UDP with each
      participant receiving from the other two on separate port pairs.
      If each participant sends RTCP feedback about data received from
      one other participant only back to that participant, then the
      conference is composed of three separate point-to-point RTP
      sessions.  If each participant provides RTCP feedback about its

      reception of one other participant to both of the other
      participants, then the conference is composed of one multi-party
      RTP session.  The latter case simulates the behavior that would
      occur with IP multicast communication among the three

      The RTP framework allows the variations defined here, but a
      particular control protocol or application design will usually
      impose constraints on these variations.

3.3.2.  RTP Header

   The RTP header contains a number of fields.  Two fields always
   require additional specification by the RTP payload format, namely
   the RTP timestamp and the marker bit.  Certain RTP payload formats
   also use the RTP sequence number to realize certain functionalities,
   primarily related to the order of their application data units.  The
   payload type is used to indicate the used payload format.  The SSRC
   is used to distinguish RTP packets from multiple senders and media
   sources identifying the RTP stream.  Finally, [RFC5285] specifies how
   to transport payload format independent metadata relating to the RTP
   packet or stream.

   Marker Bit:  A single bit normally used to provide important
      indications.  In audio, it is normally used to indicate the start
      of a talk burst.  This enables jitter buffer adaptation prior to
      the beginning of the burst with minimal audio quality impact.  In
      video, the marker bit is normally used to indicate the last packet
      part of a frame.  This enables a decoder to finish decoding the
      picture, where it otherwise may need to wait for the next packet
      to explicitly know that the frame is finished.

   Timestamp:  The RTP timestamp indicates the time instance the media
      sample belongs to.  For discrete media like video, it normally
      indicates when the media (frame) was sampled.  For continuous
      media, it normally indicates the first time instance the media
      present in the payload represents.  For audio, this is the
      sampling time of the first sample.  All RTP payload formats must
      specify the meaning of the timestamp value and the clock rates
      allowed.  Selecting a timestamp rate is an active design choice
      and is further discussed in Section 5.2.

      Discontinuous Transmission (DTX) that is common among speech
      codecs, typically results in gaps or jumps in the timestamp values
      due to that there is no media payload to transmit and the next
      used timestamp value represent the actual sampling time of the
      data transmitted.

   Sequence Number:  The sequence number is monotonically increasing and
      is set as the packet is sent.  This property is used in many
      payload formats to recover the order of everything from the whole
      stream down to fragments of application data units (ADUs) and the
      order they need to be decoded.  Discontinuous transmissions do not
      result in gaps in the sequence number, as it is monotonically
      increasing for each sent RTP packet.

   Payload Type:  The payload type is used to indicate, on a per-packet
      basis, which format is used.  The binding between a payload type
      number and a payload format and its configuration are dynamically
      bound and RTP session specific.  The configuration information can
      be bound to a payload type value by out-of-band signaling
      (Section 3.4).  An example of this would be video decoder
      configuration information.  Commonly, the same payload type is
      used for a media stream for the whole duration of a session.
      However, in some cases it may be necessary to change the payload
      format or its configuration during the session.

   SSRC:  The synchronization source (SSRC) identifier is normally not
      used by a payload format other than to identify the RTP timestamp
      and sequence number space a packet belongs to, allowing
      simultaneously reception of multiple media sources.  However, some
      of the RTP mechanisms for improving resilience to packet loss uses
      multiple SSRCs to separate original data and repair or redundant
      data, as well as multi-stream transmission of scalable codecs.

   Header Extensions:  RTP payload formats often need to include
      metadata relating to the payload data being transported.  Such
      metadata is sent as a payload header, at the start of the payload
      section of the RTP packet.  The RTP packet also includes space for
      a header extension [RFC5285]; this can be used to transport
      payload format independent metadata, for example, an SMPTE time
      code for the packet [RFC5484].  The RTP header extensions are not
      intended to carry headers that relate to a particular payload
      format, and must not contain information needed in order to decode
      the payload.

   The remaining fields do not commonly influence the RTP payload
   format.  The padding bit is worth clarifying as it indicates that one
   or more bytes are appended after the RTP payload.  This padding must
   be removed by a receiver before payload format processing can occur.
   Thus, it is completely separate from any padding that may occur
   within the payload format itself.

3.3.3.  RTP Multiplexing

   RTP has three multiplexing points that are used for different
   purposes.  A proper understanding of this is important to correctly
   use them.

   The first one is separation of RTP streams of different types or
   usages, which is accomplished using different RTP sessions.  So, for
   example, in the common multimedia session with audio and video, RTP
   commonly multiplexes audio and video in different RTP sessions.  To
   achieve this separation, transport-level functionalities are used,
   normally UDP port numbers.  Different RTP sessions can also be used
   to realize layered scalability as it allows a receiver to select one
   or more layers for multicast RTP sessions simply by joining the
   multicast groups over which the desired layers are transported.  This
   separation also allows different Quality of Service (QoS) to be
   applied to different media types.  Use of multiple transport flows
   has potential issues due to NAT and firewall traversal.  The choices
   how one applies RTP sessions as well as transport flows can affect
   the transport properties an RTP media stream experiences.

   The next multiplexing point is separation of different RTP streams
   within an RTP session.  Here, RTP uses the SSRC to identify
   individual sources of RTP streams.  An example of individual media
   sources would be the capture of different microphones that are
   carried in an RTP session for audio, independently of whether they
   are connected to the same host or different hosts.  There also exist
   cases where a single media source, is transmitted using multiple RTP
   streams.  For each SSRC, a unique RTP sequence number and timestamp
   space is used.

   The third multiplexing point is the RTP header payload type field.
   The payload type identifies what format the content in the RTP
   payload has.  This includes different payload format configurations,
   different codecs, and also usage of robustness mechanisms like the
   one described in RFC 2198 [RFC2198].

3.3.4.  RTP Synchronization

   There are several types of synchronization, and we will here describe
   how RTP handles the different types:

   Intra media:  The synchronization within a media stream from a
      synchronization source (SSRC) is accomplished using the RTP
      timestamp field.  Each RTP packet carries the RTP timestamp, which
      specifies the position in time of the media payload contained in
      this packet relative to the content of other RTP packets in the
      same RTP stream (i.e., a given SSRC).  This is especially useful

      in cases of discontinuous transmissions.  Discontinuities can be
      caused by network conditions; when extensive losses occur the RTP
      timestamp tells the receiver how much later than previously
      received media the present media should be played out.

   Inter-media:  Applications commonly have a desire to use several
      media sources, possibly of different media types, at the same
      time.  Thus, there exists a need to synchronize different media
      from the same endpoint.  This puts two requirements on RTP: the
      possibility to determine which media are from the same endpoint
      and if they should be synchronized with each other and the
      functionality to facilitate the synchronization itself.

   The first step in inter-media synchronization is to determine which
   SSRCs in each session should be synchronized with each other.  This
   is accomplished by comparing the CNAME fields in the RTCP source
   description (SDES) packets.  SSRCs with the same CNAME sent in any of
   multiple RTP sessions can be synchronized.

   The actual RTCP mechanism for inter-media synchronization is based on
   the idea that each RTP stream provides a position on the media
   specific time line (measured in RTP timestamp ticks) and a common
   reference time line.  The common reference time line is expressed in
   RTCP as a wall-clock time in the Network Time Protocol (NTP) format.
   It is important to notice that the wall-clock time is not required to
   be synchronized between hosts, for example, by using NTP [RFC5905].
   It can even have nothing at all to do with the actual time; for
   example, the host system's up-time can be used for this purpose.  The
   important factor is that all media streams from a particular source
   that are being synchronized use the same reference clock to derive
   their relative RTP timestamp time scales.  The type of reference
   clock and its timebase can be signaled using RTP Clock Source
   Signaling [RFC7273].

   Figure 1 illustrates how if one receives RTCP Sender Report (SR)
   packet P1 for one RTP stream and RTCP SR packet P2 for the other RTP
   stream, then one can calculate the corresponding RTP timestamp values
   for any arbitrary point in time T.  However, to be able to do that,
   it is also required to know the RTP timestamp rates for each RTP
   stream currently used in the sessions.

   TS1   --+---------------+------->
           |               |
          P1               |
           |               |
   NTP  ---+-----+---------T------>
                 |         |
                P2         |
                 |         |
   TS2  ---------+---------+---X-->

   Figure 1: RTCP Synchronization

   Assume that medium 1 uses an RTP timestamp clock rate of 16 kHz, and
   medium 2 uses a clock rate of 90 kHz.  Then, TS1 and TS2 for point T
   can be calculated in the following way: TS1(T) = TS1(P1) + 16000 *
   (NTP(T)-NTP(P1)) and TS2(T) = TS2(P2) + 90000 * (NTP(T)-NTP(P2)).
   This calculation is useful as it allows the implementation to
   generate a common synchronization point for which all time values are
   provided (TS1(T), TS2(T) and T).  So, when one wishes to calculate
   the NTP time that the timestamp value present in packet X corresponds
   to, one can do that in the following way: NTP(X) = NTP(T) + (TS2(X) -

   Improved signaling for layered codecs and fast tune-in have been
   specified in "Rapid Synchronization for RTP Flows" [RFC6051].

   Leap seconds are extra seconds added or seconds removed to keep our
   clocks in sync with the earth's rotation.  Adding or removing seconds
   can impact the reference clock as discussed in "RTP and Leap Seconds"
   [RFC7164]; also, in cases where the RTP timestamp values are derived
   using the wall clock during the leap second event, errors can occur.
   Implementations need to consider leap seconds and should consider the
   recommendations in [RFC7164].

3.4.  Signaling Aspects

   RTP payload formats are used in the context of application signaling
   protocols such as SIP [RFC3261] using the Session Description
   Protocol (SDP) [RFC4566] with Offer/Answer [RFC3264], RTSP [RFC7826],
   or the Session Announcement Protocol [RFC2974].  These examples all
   use out-of-band signaling to indicate which type of RTP streams are
   desired to be used in the session and how they are configured.  To be
   able to declare or negotiate the media format and RTP payload
   packetization, the payload format must be given an identifier.  In
   addition to the identifier, many payload formats also have the need
   to signal further configuration information out-of-band for the RTP
   payloads prior to the media transport session.

   The above examples of session-establishing protocols all use SDP, but
   other session description formats may be used.  For example, there
   was discussion of a new XML-based session description format within
   the IETF (SDP-NG).  In the end, the proposal did not get beyond draft
   protocol specification because of the enormous installed base of SDP
   implementations.  However, to avoid locking the usage of RTP to SDP
   based out-of-band signaling, the payload formats are identified using
   a separate definition format for the identifier and associated
   parameters.  That format is the media type.

3.4.1.  Media Types

   Media types [RFC6838] are identifiers originally created for
   identifying media formats included in email.  In this usage, they
   were known as MIME types, where the expansion of the MIME acronym
   includes the word "mail".  The term "media type" was introduced to
   reflect a broader usage, which includes HTTP [RFC7231], Message
   Session Relay Protocol (MSRP) [RFC4975], and many other protocols to
   identify arbitrary content carried within the protocols.  Media types
   also provide a media hierarchy that fits RTP payload formats well.
   Media type names are of two parts and consist of content type and
   sub-type separated with a slash, e.g., 'audio/PCMA' or 'video/
   h263-2000'.  It is important to choose the correct content-type when
   creating the media type identifying an RTP payload format.  However,
   in most cases, there is little doubt what content type the format
   belongs to.  Guidelines for choosing the correct media type and
   registration rules for media type names are provided in "Media Type
   Specifications and Registration Procedures" [RFC6838].  The
   additional rules for media types for RTP payload formats are provided
   in "Media Type Registration of RTP Payload Formats" [RFC4855].

   Registration of the RTP payload name is something that is required to
   avoid name collision in the future.  Note that "x-" names are not
   suitable for any documented format as they have the same problem with
   name collision and can't be registered.  The list of already-
   registered media types can be found at

   Media types are allowed any number of parameters, which may be
   required or optional for that media type.  They are always specified
   on the form "name=value".  There exist no restrictions on how the
   value is defined from the media type's perspective, except that
   parameters must have a value.  However, the usage of media types in

   SDP, etc., has resulted in the following restrictions that need to be
   followed to make media types usable for RTP-identifying payload

   1.  Arbitrary binary content in the parameters is allowed, but it
       needs to be encoded so that it can be placed within text-based
       protocols.  Base64 [RFC4648] is recommended, but for shorter
       content Base16 [RFC4648] may be more appropriate as it is simpler
       to interpret for humans.  This needs to be explicitly stated when
       defining a media type parameter with binary values.

   2.  The end of the value needs to be easily found when parsing a
       message.  Thus, parameter values that are continuous and not
       interrupted by common text separators, such as space and
       semicolon characters, are recommended.  If that is not possible,
       some type of escaping should be used.  Usage of quote (") is
       recommended; do not forget to provide a method of encoding any
       character used for quoting inside the quoted element.

   3.  A common representation form for the media type and its
       parameters is on a single line.  In that case, the media type is
       followed by a semicolon-separated list of the parameter value
       pairs, e.g.:

       audio/amr octet-align=0; mode-set=0,2,5,7; mode-change-period=2

3.4.2.  Mapping to SDP

   Since SDP [RFC4566] is so commonly used as an out-of-band signaling
   protocol, a mapping of the media type into SDP exists.  The details
   on how to map the media type and its parameters into SDP are
   described in [RFC4855].  However, this is not sufficient to explain
   how certain parameters must be interpreted, for example, in the
   context of Offer/Answer negotiation [RFC3264].  The Offer/Answer Model

   The Offer/Answer (O/A) model allows SIP to negotiate which media
   formats and payload formats are to be used in a session and how they
   are to be configured.  However, O/A does not define a default
   behavior; instead, it points out the need to define how parameters
   behave.  To make things even more complex, the direction of media
   within a session has an impact on these rules, so that some cases may
   require separate descriptions for RTP streams that are send-only,
   receive-only, or both sent and received as identified by the SDP
   attributes a=sendonly, a=recvonly, and a=sendrecv.  In addition, the
   usage of multicast adds further limitations as the same RTP stream is

   delivered to all participants.  If those multicast-imposed
   restrictions are too limiting for unicast, then separate rules for
   unicast and multicast will be required.

   The simplest and most common O/A interpretation is that a parameter
   is defined to be declarative; i.e., the SDP Offer/Answer sending
   agent can declare a value and that has no direct impact on the other
   agent's values.  This declared value applies to all media that are
   going to be sent to the declaring entity.  For example, most video
   codecs have a level parameter that tells the other participants the
   highest complexity the video decoder supports.  The level parameter
   can be declared independently by two participants in a unicast
   session as it will be the media sender's responsibility to transmit a
   video stream that fulfills the limitation the other side has
   declared.  However, in multicast, it will be necessary to send a
   stream that follows the limitation of the weakest receiver, i.e., the
   one that supports the lowest level.  To simplify the negotiation in
   these cases, it is common to require any answerer to a multicast
   session to take a yes or no approach to parameters.

   A "negotiated" parameter is a different case, for which both sides
   need to agree on its value.  Such a parameter requires the answerer
   to either accept it as it is offered or remove the payload type the
   parameter belonged to from its answer.  The removal of the payload
   type from the answer indicates to the offerer the lack of support for
   the parameter values presented.  An unfortunate implication of the
   need to use complete payload types to indicate each possible
   configuration so as to maximize the chances of achieving
   interoperability, is that the number of necessary payload types can
   quickly grow large.  This is one reason to limit the total number of
   sets of capabilities that may be implemented.

   The most problematic type of parameters are those that relate to the
   media the entity sends.  They do not really fit the O/A model, but
   can be shoehorned in.  Examples of such parameters can be found in
   the H.264 video codec's payload format [RFC6184], where the name of
   all parameters with this property starts with "sprop-".  The issue
   with these parameters is that they declare properties for a RTP
   stream that the other party may not accept.  The best one can make of
   the situation is to explain the assumption that the other party will
   accept the same parameter value for the media it will receive as the
   offerer of the session has proposed.  If the answerer needs to change
   any declarative parameter relating to streams it will receive, then
   the offerer may be required to make a new offer to update the
   parameter values for its outgoing RTP stream.

   Another issue to consider is the send-only RTP streams in offers.
   Parameters that relate to what the answering entity accepts to
   receive have no meaning other than to provide a template for the
   answer.  It is worth pointing out in the specification that these
   really provide a set of parameter values that the sender recommends.
   Note that send-only streams in answers will need to indicate the
   offerer's parameters to ensure that the offerer can match the answer
   to the offer.

   A further issue with Offer/Answer that complicates things is that the
   answerer is allowed to renumber the payload types between offer and
   answer.  This is not recommended, but allowed for support of gateways
   to the ITU conferencing suite.  This means that it must be possible
   to bind answers for payload types to the payload types in the offer
   even when the payload type number has been changed, and some of the
   proposed payload types have been removed.  This binding must normally
   be done by matching the configurations originally offered against
   those in the answer.  This may require specification in the payload
   format of which parameters that constitute a configuration, for
   example, as done in Section 8.2.2 of the H.264 RTP Payload format
   [RFC6184], which states: "The parameters identifying a media format
   configuration for H.264 are profile-level-id and packetization-mode".  Declarative Usage in RTSP and SAP

   SAP (Session Announcement Protocol) [RFC2974] was experimentally used
   for announcing multicast sessions.  Similar but better protocols are
   using SDP in a declarative style to configure multicast-based
   applications.  Independently of the usage of Source-Specific
   Multicast (SSM) [RFC3569] or Any-Source Multicast (ASM), the SDP
   provided by these configuration delivery protocols applies to all
   participants.  All media that is sent to the session must follow the
   RTP stream definition as specified by the SDP.  This enables everyone
   to receive the session if they support the configuration.  Here, SDP
   provides a one-way channel with no possibility to affect the
   configuration that the session creator has decided upon.  Any RTP
   payload format that requires parameters for the send direction and
   that needs individual values per implementation or instance will fail
   in a SAP session for a multicast session allowing anyone to send.

   Real-Time Streaming Protocol (RTSP) [RFC7826] allows the negotiation
   of transport parameters for RTP streams that are part of a streaming
   session between a server and client.  RTSP has divided the transport
   parameters from the media configuration.  SDP is commonly used for
   media configuration in RTSP and is sent to the client prior to
   session establishment, either through use of the DESCRIBE method or

   by means of an out-of-band channel like HTTP, email, etc.  The SDP is
   used to determine which RTP streams and what formats are being used
   prior to session establishment.

   Thus, both SAP and RTSP use SDP to configure receivers and senders
   with a predetermined configuration for a RTP stream including the
   payload format and any of its parameters.  All parameters are used in
   a declarative fashion.  This can result in different treatment of
   parameters between Offer/Answer and declarative usage in RTSP and
   SAP.  Any such difference will need to be spelled out by the payload
   format specification.

3.5.  Transport Characteristics

   The general channel characteristics that RTP flows experience are
   documented in Section 3 of "Guidelines for Writers of RTP Payload
   Format Specifications" [RFC2736].  The discussion below provides
   additional information.

3.5.1.  Path MTU

   At the time of writing, the most common IP Maximum Transmission Unit
   (MTU) in commonly deployed link layers is 1500 bytes (Ethernet data
   payload).  However, there exist both links with smaller MTUs and
   links with much larger MTUs.  An example for links with small MTU
   size is older generation cellular links.  Certain parts of the
   Internet already support an IP MTU of 8000 bytes or more, but these
   are limited islands.  The most likely places to find MTUs larger than
   1500 bytes are within enterprise networks, university networks, data
   centers, storage networks, and over high capacity (10 Gbps or more)
   links.  There is a slow, ongoing evolution towards larger MTU sizes.
   However, at the same time, it has become common to use tunneling
   protocols, often multiple ones, whose overhead when added together
   can shrink the MTU significantly.  Thus, there exists a need both to
   consider limited MTUs as well as enable support of larger MTUs.  This
   should be considered in the design, especially in regard to features
   such as aggregation of independently decodable data units.

3.5.2.  Different Queuing Algorithms

   Routers and switches on the network path between an IP sender and a
   particular receiver can exhibit different behaviors affecting the
   end-to-end characteristics.  One of the more important aspects of
   this is queuing behavior.  Routers and switches have some amount of
   queuing to handle temporary bursts of data that designated to leave
   the switch or router on the same egress link.  A queue, when not
   empty, results in an increased path delay.

   The implementation of the queuing affects the delay and also how
   congestion signals (Explicit Congestion Notification (ECN) [RFC6679]
   or packet drops) are provided to the flow.  The other aspects are if
   the flow shares the queue with other flows and how the implementation
   affects the flow interaction.  This becomes important, for example,
   when real-time flows interact with long-lived TCP flows.  TCP has a
   built-in behavior in its congestion control that strives to fill the
   buffer; thus, all flows sharing the buffer experienced the delay
   build up.

   A common, but quite poor, queue-handling mechanism is tail-drop,
   i.e., only drop packets when the incoming packet doesn't fit in the
   queue.  If a bad queuing algorithm is combined with too much queue
   space, the queuing time can grow to be very significant and can even
   become multiple seconds.  This is called "bufferbloat" [BLOAT].
   Active Queue Management (AQM) is a term covering mechanisms that try
   to do something smarter by actively managing the queue, for example,
   sending congestion signals earlier by dropping packets earlier in the
   queue.  The behavior also affects the flow interactions.  For
   example, Random Early Detection (RED) [RED] selects which packet(s)
   to drop randomly.  This gives flows that have more packets in the
   queue a higher probability to experience the packet loss (congestion
   signal).  There is ongoing work in the IETF WG AQM to find suitable
   mechanisms to recommend for implementation and reduce the use of

3.5.3.  Quality of Service

   Using best-effort Internet has no guarantees for the path's
   properties.  QoS mechanisms are intended to provide the possibility
   to bound the path properties.  Where Diffserv [RFC2475] markings
   affect the queuing and forwarding behaviors of routers, the mechanism
   provides only statistical guarantees and care in how much marked
   packets of different types that are entering the network.  Flow-based
   QoS, like IntServ [RFC1633], has the potential for stricter
   guarantees as the properties are agreed on by each hop on the path,
   at the cost of per-flow state in the network.

4.  Standardization Process for an RTP Payload Format

   This section discusses the recommended process to produce an RTP
   payload format in the described venues.  This is to document the best
   current practice on how to get a well-designed and specified payload
   format as quickly as possible.  For specifications that are defined
   by standards bodies other than the IETF, the primary milestone is the
   registration of the media type for the RTP payload format.  For

   proprietary media formats, the primary goal depends on whether
   interoperability is desired at the RTP level.  However, there is also
   the issue of ensuring best possible quality of any specification.

4.1.  IETF

   For all standardized media formats, it is recommended that the
   payload format be specified in the IETF.  The main reason is to
   provide an openly available RTP payload format specification that has
   been reviewed by people experienced with RTP payload formats.  At the
   time of writing, this work is done in the PAYLOAD Working Group (WG),
   but that may change in the future.

4.1.1.  Steps from Idea to Publication

   There are a number of steps that an RTP payload format should go
   through from the initial idea until it is published.  This also
   documents the process that the PAYLOAD WG applies when working with
   RTP payload formats.

   Idea:   Determine the need for an RTP payload format as an IETF

   Initial effort:   Using this document as a guideline, one should be
      able to get started on the work.  If one's media codec doesn't fit
      any of the common design patterns or one has problems
      understanding what the most suitable way forward is, then one
      should contact the PAYLOAD WG and/or the WG Chairs.  The goal of
      this stage is to have an initial individual draft.  This draft
      needs to focus on the introductory parts that describe the real-
      time media format and the basic idea on how to packetize it.  Not
      all the details are required to be filled in.  However, the
      security chapter is not something that one should skip, even
      initially.  From the start, it is important to consider any
      serious security risks that need to be solved.  The first step is
      completed when one has a draft that is sufficiently detailed for a
      first review by the WG.  The less confident one is of the
      solution, the less work should be spent on details; instead,
      concentrate on the codec properties and what is required to make
      the packetization work.

   Submission of the first version:   When one has performed the above,
      one submits the draft as an individual draft
      (https://datatracker.ietf.org/submit/).  This can be done at any
      time, except for a period prior to an IETF meeting (see important
      dates related to the next IETF meeting for draft submission cutoff
      date).  When the Internet-Draft announcement has been sent out on

      the draft announcement list
      (https://www.ietf.org/mailman/listinfo/I-D-Announce), forward it
      to the PAYLOAD WG (https://www.ietf.org/mailman/listinfo/payload)
      and request that it be reviewed.  In the email, outline any issues
      the authors currently have with the design.

   Iterative improvements:   Taking the feedback received into account,
      one updates the draft and tries resolve issues.  New revisions of
      the draft can be submitted at any time (again except for a short
      period before meetings).  It is recommended to submit a new
      version whenever one has made major updates or has new issues that
      are easiest to discuss in the context of a new draft version.

   Becoming a WG document:   Given that the definition of RTP payload
      formats is part of the PAYLOAD WG's charter, RTP payload formats
      that are going to be published as Standards Track RFCs need to
      become WG documents.  Becoming a WG document means that the WG
      Chairs or an appointed document shepherd are responsible for
      administrative handling, for example, issuing publication
      requests.  However, be aware that making a document into a WG
      document changes the formal ownership and responsibility from the
      individual authors to the WG.  The initial authors normally
      continue being the document editors, unless unusual circumstances
      occur.  The PAYLOAD WG accepts new RTP payload formats based on
      their suitability and document maturity.  The document maturity is
      a requirement to ensure that there are dedicated document editors
      and that there exists a good solution.

   Iterative improvements:  The updates and review cycles continue until
      the draft has reached the level of maturity suitable for
      publication.  The authors are responsible for judging when the
      document is ready for the next step, most likely WG Last Call, but
      they can ask the WG chairs or Shepherd.

   WG Last Call:   A WG Last Call of at least two weeks is always
      performed for payload formats in the PAYLOAD WG (see Section 7.4
      of [RFC2418]).  The authors request WG Last Call for a draft when
      they think it is mature enough for publication.  The WG Chairs or
      shepherd perform a review to check if they agree with the authors'
      assessment.  If the WG Chairs or shepherd agree on the maturity,
      the WG Last Call is announced on the WG mailing list.  If there
      are issues raised, these need to be addressed with an updated
      draft version.  For any more substantial changes to the draft, a
      new WG Last Call is announced for the updated version.  Minor
      changes, like editorial fixes, can be progressed without an
      additional WG Last Call.

   Publication requested:   For WG documents, the WG Chairs or shepherd
      request publication of the draft after it has passed WG Last Call.
      After this, the approval and publication process described in BCP
      9 [BCP9] is performed.  The status after the publication has been
      requested can be tracked using the IETF Datatracker [TRACKER].
      Documents do not expire as they normally do after publication has
      been requested, so authors do not have to issue keep-alive
      updates.  In addition, any submission of document updates requires
      the approval of WG Chair(s).  The authors are commonly asked to
      address comments or issues raised by the IESG.  The authors also
      do one last review of the document immediately prior to its
      publication as an RFC to ensure that no errors or formatting
      problems have been introduced during the publication process.

4.1.2.  WG Meetings

   WG meetings are for discussing issues, not presentations.  This means
   that most RTP payload formats should never need to be discussed in a
   WG meeting.  RTP payload formats that would be discussed are either
   those with controversial issues that failed to be resolved on the
   mailing list or those including new design concepts worth a general

   There exists no requirement to present or discuss a draft at a WG
   meeting before it becomes published as an RFC.  Thus, even authors
   who lack the possibility to go to WG meetings should be able to
   successfully specify an RTP payload format in the IETF.  WG meetings
   may become necessary only if the draft gets stuck in a serious debate
   that cannot easily be resolved.

4.1.3.  Draft Naming

   To simplify the work of the PAYLOAD WG Chairs and WG members, a
   specific Internet-Draft file-naming convention shall be used for RTP
   payload formats.  Individual submissions shall be named using the
   template: draft-<lead author family name>-payload-rtp-<descriptive
   name>-<version>.  The WG documents shall be named according to this
   template: draft-ietf-payload-rtp-<descriptive name>-<version>.  The
   inclusion of "payload" in the draft file name ensures that the search
   for "payload-" will find all PAYLOAD-related drafts.  Inclusion of
   "rtp" tells us that it is an RTP payload format draft.  The
   descriptive name should be as short as possible while still
   describing what the payload format is for.  It is recommended to use
   the media format or codec abbreviation.  Please note that the version
   must start at 00 and is increased by one for each submission to the
   IETF secretary of the draft.  No version numbers may be skipped.  For
   more details on draft naming, please see Section 7 of [ID-GUIDE].

4.1.4.  Writing Style

   When writing an Internet-Draft for an RTP payload format, one should
   observe some few considerations (that may be somewhat divergent from
   the style of other IETF documents and/or the media coding spec's
   author group may use):

   Include Motivations:  In the IETF, it is common to include the
      motivation for why a particular design or technical path was
      chosen.  These are not long statements: a sentence here and there
      explaining why suffice.

   Use the Defined Terminology:  There exists defined terminology both
      in RTP and in the media codec specification for which the RTP
      payload format is designed.  A payload format specification needs
      to use both to make clear the relation of features and their
      functions.  It is unwise to introduce or, worse, use without
      introduction, terminology that appears to be more accessible to
      average readers but may miss certain nuances that the defined
      terms imply.  An RTP payload format author can assume the reader
      to be reasonably familiar with the terminology in the media coding

   Keeping It Simple:  The IETF has a history of specifications that are
      focused on their main usage.  Historically, some RTP payload
      formats have a lot of modes and features, while the actual
      deployments have only included the most basic features that had
      very clear requirements.  Time and effort can be saved by focusing
      on only the most important use cases and keeping the solution
      simple.  An extension mechanism should be provided to enable
      backward-compatible extensions, if that is an organic fit.

   Normative Requirements:  When writing specifications, there is
      commonly a need to make it clear when something is normative and
      at what level.  In the IETF, the most common method is to use "Key
      words for use in RFCs to Indicate Requirement Levels" [RFC2119],
      which defines the meaning of "MUST", "MUST NOT", "REQUIRED",

4.1.5.  How to Speed Up the Process

   There a number of ways to lose a lot of time in the above process.
   This section discusses what to do and what to avoid.

   o  Do not update the draft only for the meeting deadline.  An update
      to each meeting automatically limits the draft to three updates
      per year.  Instead, ignore the meeting schedule and publish new
      versions as soon as possible.

   o  Try to avoid requesting reviews when people are busy, like the few
      weeks before a meeting.  It is actually more likely that people
      have time for them directly after a meeting.

   o  Perform draft updates quickly.  A common mistake is that the
      authors let the draft slip.  By performing updates to the draft
      text directly after getting resolution on an issue, things speed
      up.  This minimizes the delay that the author has direct control
      over.  The time taken for reviews, responses from Area Directors
      and WG Chairs, etc., can be much harder to speed up.

   o  Do not fail to take human nature into account.  It happens that
      people forget or need to be reminded about tasks.  Send a kind
      reminder to the people you are waiting for if things take longer
      than expected.  Ask people to estimate when they expect to fulfill
      the requested task.

   o  Ensure there is enough review.  It is common that documents take a
      long time and many iterations because not enough review is
      performed in each iteration.  To improve the amount of review you
      get on your own document, trade review time with other document
      authors.  Make a deal with some other document author that you
      will review their draft if they review yours.  Even inexperienced
      reviewers can help with language, editorial, or clarity issues.
      Also, try approaching the more experienced people in the WG and
      getting them to commit to a review.  The WG Chairs cannot, even if
      desirable, be expected to review all versions.  Due to workload,
      the Chairs may need to concentrate on key points in a draft
      evolution like checking on initial submissions, a draft's
      readiness to become a WG document, or its readiness for WG Last

4.2.  Other Standards Bodies

   Other standards bodies may define RTP payloads in their own
   specifications.  When they do this, they are strongly recommended to
   contact the PAYLOAD WG Chairs and request review of the work.  It is
   recommended that at least two review steps are performed.  The first

   should be early in the process when more fundamental issues can be
   easily resolved without abandoning a lot of effort.  Then, when
   nearing completion, but while it is still possible to update the
   specification, a second review should be scheduled.  In that pass,
   the quality can be assessed; hopefully, no updates will be needed.
   Using this procedure can avoid both conflicting definitions and
   serious mistakes, like breaking certain aspects of the RTP model.

   RTP payload media types may be registered in the standards tree by
   other standards bodies.  The requirements on the organization are
   outlined in the media types registration documents [RFC4855] and
   [RFC6838]).  This registration requires a request to the IESG, which
   ensures that the filled-in registration template is acceptable.  To
   avoid last-minute problems with these registrations the registration
   template must be sent for review both to the PAYLOAD WG and the media
   types list (ietf-types@iana.org) and is something that should be
   included in the IETF reviews of the payload format specification.

4.3.  Proprietary and Vendor Specific

   Proprietary RTP payload formats are commonly specified when the real-
   time media format is proprietary and not intended to be part of any
   standardized system.  However, there are reasons why also proprietary
   formats should be correctly documented and registered:

   o  Usage in a standardized signaling environment, such as SIP/SDP.
      RTP needs to be configured with the RTP profiles, payload formats,
      and their payload types being used.  To accomplish this, it is
      desirable to have registered media type names to ensure that the
      names do not collide with those of other formats.

   o  Sharing with business partners.  As RTP payload formats are used
      for communication, situations often arise where business partners
      would like to support a proprietary format.  Having a well-written
      specification of the format will save time and money for both
      parties, as interoperability will be much easier to accomplish.

   o  To ensure interoperability between different implementations on
      different platforms.

   To avoid name collisions, there is a central registry keeping track
   of the registered media type names used by different RTP payload
   formats.  When it comes to proprietary formats, they should be
   registered in the vendor's own tree.  All vendor-specific
   registrations use sub-type names that start with "vnd.<vendor-name>".
   Names in the vendor's own tree are not required to be registered with
   IANA.  However, registration [RFC6838] is recommended if the media
   type is used at all in public environments.

   If interoperability at the RTP level is desired, a payload type
   specification should be standardized in the IETF following the
   process described above.  The IETF does not require full disclosure
   of the codec when defining an RTP payload format to carry that codec,
   but a description must be provided that is sufficient to allow the
   IETF to judge whether the payload format is well designed.  The media
   type identifier assigned to a standardized payload format of this
   sort will lie in the standards tree rather than the vendor tree.

4.4.  Joint Development of Media Coding Specification and RTP Payload

   In the last decade, there have been a few cases where the media codec
   and the associated RTP payload format have been developed
   concurrently and jointly.  Developing the two specs not only
   concurrently but also jointly, in close cooperation with the group
   developing the media codec, allows one to leverage the benefits joint
   source/channel coding can provide.  Doing so has historically
   resulted in well-performing payload formats and in success of both
   the media coding specification and associated RTP payload format.
   Insofar, whenever the opportunity presents it, it may be useful to
   closely keep the media coding group in the loop (through appropriate
   liaison means whatever those may be) and influence the media coding
   specification to be RTP friendly.  One example for such a media
   coding specification is H.264, where the RTP payload header co-serves
   as the H.264 NAL unit header and vice versa, and is documented in
   both specifications.

5.  Designing Payload Formats

   The best summary of payload format design is KISS (Keep It Simple,
   Stupid).  A simple payload format is easier to review for
   correctness, easier to implement, and has low complexity.
   Unfortunately, contradictory requirements sometimes make it hard to
   do things simply.  Complexity issues and problems that occur for RTP
   payload formats are:

   Too many configurations:  Contradictory requirements lead to the
      result that one configuration is created for each conceivable
      case.  Such contradictory requirements are often between
      functionality and bandwidth.  This outcome has two big
      disadvantages; First all configurations need to be implemented.
      Second, the user application must select the most suitable
      configuration.  Selecting the best configuration can be very
      difficult and, in negotiating applications, this can create
      interoperability problems.  The recommendation is to try to select

      a very limited set of configurations (preferably one) that perform
      well for the most common cases and are capable of handling the
      other cases, but maybe not that well.

   Hard to implement:  Certain payload formats may become difficult to
      implement both correctly and efficiently.  This needs to be
      considered in the design.

   Interaction with general mechanisms:  Special solutions may create
      issues with deployed tools for RTP, such as tools for more robust
      transport of RTP.  For example, a requirement for an unbroken
      sequence number space creates issues for mechanisms relying on
      payload type switching interleaving media-independent resilience
      within a stream.

5.1.  Features of RTP Payload Formats

   There are a number of common features in RTP payload formats.  There
   is no general requirement to support these features; instead, their
   applicability must be considered for each payload format.  In fact,
   it may be that certain features are not even applicable.

5.1.1.  Aggregation

   Aggregation allows for the inclusion of multiple Application Data
   Units (ADUs) within the same RTP payload.  This is commonly supported
   for codecs that produce ADUs of sizes smaller than the IP MTU.  One
   reason for the use of aggregation is the reduction of header overhead
   (IP/UDP/RTP headers).  When setting into relation the ADU size and
   the MTU size, do remember that the MTU may be significantly larger
   than 1500 bytes.  An MTU of 9000 bytes is available today and an MTU
   of 64k may be available in the future.  Many speech codecs have the
   property of ADUs of a few fixed sizes.  Video encoders may generally
   produce ADUs of quite flexible sizes.  Thus, the need for aggregation
   may be less.  But some codecs produce small ADUs mixed with large
   ones, for example, H.264 Supplemental Enhancement Information (SEI)
   messages.  Sending individual SEI message in separate packets are not
   efficient compared to combing the with other ADUs.  Also, some small
   ADUs are, within the media domain, semantically coupled to the larger
   ADUs (for example, in-band parameter sets in H.264 [RFC6184]).  In
   such cases, aggregation is sensible, even if not required from a
   payload/header overhead viewpoint.  There also exist cases when the
   ADUs are pre-produced and can't be adopted to a specific networks
   MTU.  Instead, their packetization needs to be adopted to the
   network.  All above factors should be taken into account when
   deciding on the inclusion of aggregation, and weighting its benefits

   against the complexity of defining them (which can be significant
   especially when aggregation is performed over ADUs with different
   playback times).

   The main disadvantage of aggregation, beyond implementation
   complexity, is the extra delay introduced (due to buffering until a
   sufficient number of ADUs have been collected at the sender) and
   reduced robustness against packet loss.  Aggregation also introduces
   buffering requirements at the receiver.

5.1.2.  Fragmentation

   If the real-time media format has the property that it may produce
   ADUs that are larger than common MTU sizes, then fragmentation
   support should be considered.  An RTP payload format may always fall
   back on IP fragmentation; however, as discussed in RFC 2736, this has
   some drawbacks.  Perhaps the most important reason to avoid IP
   fragmentation is that IP fragmented packets commonly are discarded in
   the network, especially by NATs or firewalls.  The usage of
   fragmentation at the RTP payload format level allows for more
   efficient usage of RTP packet loss recovery mechanisms.  It may also
   in some cases also allow better usage of partial ADUs by doing media
   specific fragmentation at media-specific boundaries.  In use cases
   where the ADUs are pre-produced and can't be adopted to the network's
   MTU size, support for fragmentation can be crucial.

5.1.3.  Interleaving and Transmission Rescheduling

   Interleaving has been implemented in a number of payload formats to
   allow for less quality reduction when packet loss occurs.  When
   losses are bursty and several consecutive packets are lost, the
   impact on quality can be quite severe.  Interleaving is used to
   convert that burst loss to several spread-out individual packet
   losses.  It can also be used when several ADUs are aggregated in the
   same packets.  A loss of an RTP packet with several ADUs in the
   payload has the same effect as a burst loss if the ADUs would have
   been transmitted in individual packets.  To reduce the burstiness of
   the loss, the data present in an aggregated payload may be
   interleaved, thus, spreading the loss over a longer time period.

   A requirement for doing interleaving within an RTP payload format is
   the aggregation of multiple ADUs.  For formats that do not use
   aggregation, there is still a possibility of implementing a
   transmission order rescheduling mechanism.  That has the effect that
   the packets transmitted consecutively originate from different points
   in the RTP stream.  This can be used to mitigate burst losses, which
   may be useful if one transmits packets at frequent intervals.
   However, it may also be used to transmit more significant data

   earlier in combination with RTP retransmission to allow for more
   graceful degradation and increased possibility to receive the most
   important data, e.g., intra frames of video.

   The drawback of interleaving is the significantly increased
   transmission buffering delay, making it less useful for low-delay
   applications.  It may also create significant buffering requirements
   on the receiver.  That buffering is also problematic, as it is
   usually difficult to indicate when a receiver may start consume data
   and still avoid buffer under run caused by the interleaving mechanism
   itself.  Transmission rescheduling is only useful in a few specific
   cases, as in streaming with retransmissions.  The potential gains
   must be weighed against the complexity of these schemes.

5.1.4.  Media Back Channels

   A few RTP payload formats have implemented back channels within the
   media format.  Those have been for specific features, like the AMR
   [RFC4867] codec mode request (CMR) field.  The CMR field is used in
   the operation of gateways to circuit-switched voice to allow an IP
   terminal to react to the circuit-switched network's need for a
   specific encoder mode.  A common motivation for media back channels
   is the need to have signaling in direct relation to the media or the
   media path.

   If back channels are considered for an RTP payload format they should
   be for a specific requirements which cannot be easily satisfied by
   more generic mechanisms within RTP or RTCP.

5.1.5.  Media Scalability

   Some codecs support various types of media scalability, i.e. some
   data of a RTP stream may be removed to adapt the media's properties,
   such as bitrate and quality.  The adaptation may be applied in the
   following dimensions of the media:

   Temporal:  For most video codecs it is possible to adapt the frame
      rate without any specific definition of a temporal scalability
      mode, e.g., for H.264 [RFC6184].  In these cases, the sender
      changes which frames it delivers and the RTP timestamp makes it
      clear the frame interval and each frames relative capture time.
      H.264 Scalable Video Coding (SVC) [RFC6190] has more explicit
      support for temporal scalability.

   Spatial:  Video codecs supporting scalability may adapt the
      resolution, e.g., in SVC [RFC6190].

   Quality:  The quality of the encoded stream may be scaled by adapting
      the accuracy of the coding process, as, e.g.  possible with Signal
      to Noise Ratio (SNR) fidelity scalability of SVC [RFC6190].

   At the time of writing this document, codecs that support scalability
   have a bit of a revival.  It has been realized that getting the
   required functionality for supporting the features of the media
   stream into the RTP framework is quite challenging.  One of the
   recent examples for layered and scalable codecs is SVC [RFC6190].

   SVC is a good example for a payload format supporting media
   scalability features, which have been in its basic form already
   included in RTP.  A layered codec supports the dropping of data parts
   of a RTP stream, i.e., RTP packets may not be transmitted or
   forwarded to a client in order to adapt the RTP streams bitrate as
   well as the received encoded stream's quality, while still providing
   a decodable subset of the encoded stream to a client.  One example
   for using the scalability feature may be an RTP Mixer (Multipoint
   Control Unit) [RFC7667], which controls the rate and quality sent out
   to participants in a communication based on dropping RTP packets or
   removing part of the payload.  Another example may be a transport
   channel, which allows for differentiation in Quality of Service (QoS)
   parameters based on RTP sessions in a multicast session.  In such a
   case, the more important packets of the scalable encoded stream (base
   layer) may get better QoS parameters than the less important packets
   (enhancement layer) in order to provide some kind of graceful
   degradation.  The scalability features required for allowing an
   adaptive transport, as described in the two examples above, are based
   on RTP multiplexing in order to identify the packets to be dropped or
   transmitted/forwarded.  The multiplexing features defined for
   Scalable Video Coding [RFC6190] are:

      Single Session Transmission (SST), where all media layers of the
      media are transported as a single synchronization source (SSRC) in
      a single RTP session; as well as

      Multi-Session Transmission (MST), which should more accurately be
      called multi-stream transmission, where different media layers or
      a set of media layers are transported in different RTP streams,
      i.e., using multiple sources (SSRCs).

   In the first case (SST), additional in-band as well as out-of-band
   signaling is required in order to allow identification of packets
   belonging to a specific media layer.  Furthermore, an adaptation of
   the encoded stream requires dropping of specific packets in order to
   provide the client with a compliant encoded stream.  In case of using
   encryption, it is typically required for an adapting network device

   to be in the security context to allow packet dropping and providing
   an intact RTP session to the client.  This typically requires the
   network device to be an RTP mixer.

   In general, having a media-unaware network device dropping excessive
   packets will be more problematic than having a Media-Aware Network
   Entity (MANE).  First is the need to understand the media format and
   know which ADUs or payloads belong to the layers, that no other layer
   will be dependent on after the dropping.  Second, if the MANE can
   work as an RTP mixer or translator, it can rewrite the RTP and RTCP
   in such a way that the receiver will not suspect unintentional RTP
   packet losses needing repair actions.  This as the receiver can't
   determine if a lost packet was an important base layer packet or one
   of the less important extension layers.

   In the second case (MST), the RTP packet streams can be sent using a
   single or multiple RTP session, and thus transport flows, e.g., on
   different multicast groups.  Transmitting the streams in different
   RTP sessions, then the out-of-band signaling typically provides
   enough information to identify the media layers and its properties.
   The decision on dropping packets is based on the Network Address that
   identifies the RTP session to be dropped.  In order to allow correct
   data provisioning to a decoder after reception from different
   sessions, data realignment mechanisms are required.  In some cases,
   existing generic tools, as described below, can be employed to enable
   such realignment; when those generic mechanisms are sufficient, they
   should be used.  For example, "Rapid Synchronisation for RTP Flows"
   [RFC6051], uses existing RTP mechanisms, i.e. the NTP timestamp, to
   ensure timely inter-session synchronization.  Another is the
   signaling feature for indicating dependencies of RTP sessions in SDP,
   as defined in the Media Decoding Dependency Grouping in SDP

   Using MST within a single RTP session is also possible and allows
   stream level handling instead of looking deeper into the packets by a
   MANE.  However, transport flow-level properties will be the same
   unless packet based mechanisms like Diffserv is used.

   When QoS settings, e.g., Diffserv markings, are used to ensure that
   the extension layers are dropped prior the base layer the receiving
   endpoint has the benefit in MST to know which layer or set of layers
   the missing packets belong to as it will be bound to different RTP
   sessions or RTP packet streams (SSRCs), thus, explicitly indicating
   the importance of the loss.

5.1.6.  High Packet Rates

   Some media codecs require high packet rates; in these cases, the RTP
   sequence number wraps too quickly.  As a rule of thumb, it must not
   be possible to wrap the sequence number space within at least three
   RTCP reporting intervals.  As the reporting interval can vary widely
   due to configuration and session properties, and also must take into
   account the randomization of the interval, one can use the TCP
   maximum segment lifetime (MSL), i.e., 2 minutes, in ones
   consideration.  If earlier wrapping may occur, then the payload
   format should specify an extended sequence number field to allow the
   receiver to determine where a specific payload belongs in the
   sequence, even in the face of extensive reordering.  The RTP payload
   format for uncompressed video [RFC4175] can be used as an example for
   such a field.

   RTCP is also affected by high packet rates.  For RTCP mechanisms that
   do not use extended counters, there is significant risk that they
   wrap multiple times between RTCP reporting or feedback; thus,
   producing uncertainty about which packet(s) are referenced.  The
   payload designer can't effect the RTCP packet formats used and their
   design, but can note this considerations when configuring RTCP
   bandwidth and reporting intervals to avoid to wrapping issues.

5.2.  Selecting Timestamp Definition

   The RTP timestamp is an important part and has two design choices
   associated with it.  The first is the definition that determines what
   the timestamp value in a particular RTP packet will be, the second is
   which timestamp rate should be used.

   The timestamp definition needs to explicitly define what the
   timestamp value in the RTP packet represent for a particular payload
   format.  Two common definitions are used; for discretely sampled
   media, like video frames, the sampling time of the earliest included
   video frame which the data represent (fully or partially) is used;
   for continuous media like audio, the sampling time of the earliest
   sample which the payload data represent.  There exist cases where
   more elaborate or other definitions are used.

   RTP payload formats with a timestamp definition that results in no or
   little correlation between the media time instance and its
   transmission time cause the RTCP jitter calculation to become
   unusable due to the errors introduced on the sender side.  A common
   example is a payload format for a video codec where the RTP timestamp
   represents the capture time of the video frame, but frames are large

   enough that multiple RTP packets need to be sent for each frame
   spread across the framing interval.  It should be noted whether or
   not the payload format has this property.

   An RTP payload format also needs to define what timestamp rates, or
   clock rates (as it is also called), may be used.  Depending on the
   RTP payload format, this may be a single rate or multiple ones or
   theoretically any rate.  So what needs to be considered when
   selecting a rate?

   The rate needs be selected so that one can determine where in the
   time line of the media a particular sample (e.g., individual audio
   sample, or video frame) or set of samples (e.g., audio frames)
   belong.  To enable correct synchronization of this data with previous
   frames, including over periods of discontinuous transmission or

   For audio, it is common to require audio sample accuracy.  Thus, one
   commonly selects the input sampling rate as the timestamp rate.  This
   can, however, be challenging for audio codecs that support multiple
   different sampling frequencies, either as codec input or being used
   internally but effecting output, for example, frame duration.
   Depending on how one expects to use these different sampling rates
   one can allow multiple timestamp rates, each matching a particular
   codec input or sampling rate.  However, due to the issues with using
   multiple different RTP timestamp rates for the same source (SSRC)
   [RFC7160], this should be avoided if one expects to need to switch
   between modes.

   Then, an alternative is to find a common denominator frequency
   between the different modes, e.g., OPUS [RFC7587] that uses 48 kHz.
   If the different modes uses or can use a common input/output
   frequency, then selecting this also needs to be considered.  However,
   it is important to consider all aspects as the case of AMR-WB+
   [RFC4352] illustrates.  AMR-WB+'s RTP timestamp rate has the very
   unusual value of 72 kHz, despite the fact that output normally is at
   a sample rate of 48kHz.  The design is motivated by the media codec's
   production of a large range of different frame lengths in time
   perspective.  The 72 kHz timestamp rate is the smallest found value
   that would make all of the frames the codec could produce result in
   an integer frame length in RTP timestamp ticks.  This way, a receiver
   can always correctly place the frames in relation to any other frame,
   even when the frame length changes.  The downside is that the decoder
   outputs for certain frame lengths are, in fact, partial samples.  The
   result is that the output in samples from the codec will vary from
   frame to frame, potentially making implementation more difficult.

   Video codecs have commonly been using 90 kHz; the reason is this is a
   common denominator between the usually used frame rates such as 24,
   25, 30, 50 and 60, and NTSC's odd 29.97 Hz.  There does, however,
   exist at least one exception in the payload format for SMPTE 292M
   video [RFC3497] that uses a clock rate of 148.5 MHz.  The reason here
   is that the timestamp then identify the exact start sample within a
   video frame.

   Timestamp rates below 1000 Hz are not appropriate, because this will
   cause a resolution too low in the RTCP measurements that are
   expressed in RTP timestamps.  This is the main reason that the text
   RTP payload formats, like T.140 [RFC4103], use 1000 Hz.

6.  Noteworthy Aspects in Payload Format Design

   This section provides a few examples of payload formats that are
   worth noting for good or bad design in general or in specific

6.1.  Audio Payloads

   The AMR [RFC4867], AMR-WB [RFC4867], EVRC [RFC3558], SMV [RFC3558]
   payload formats are all quite similar.  They are all for frame-based
   audio codecs and use a table of contents structure.  Each frame has a
   table of contents entry that indicates the type of the frame and if
   additional frames are present.  This is quite flexible, but produces
   unnecessary overhead if the ADU is of fixed size and if, when
   aggregating multiple ADUs, they are commonly of the same type.  In
   that case, a solution like the one in AMR-WB+ [RFC4352] may be more

   The RTP payload format for MIDI [RFC6295] contains some interesting
   features.  MIDI is an audio format sensitive to packet losses, as the
   loss of a "note off" command will result in a note being stuck in an
   "on" state.  To counter this, a recovery journal is defined that
   provides a summarized state that allows the receiver to recover from
   packet losses quickly.  It also uses RTCP and the reported highest
   sequence number to be able to prune the state the recovery journal
   needs to contain.  These features appear limited in applicability to
   media formats that are highly stateful and primarily use symbolic
   media representations.

   There exists a security concern with variable bitrate audio and
   speech codecs that changes their payload length based on the input
   data.  This can leak information, especially in structured
   communication like a speech recognition prompt service that asks
   people to enter information verbally.  This issue also exists to some
   degree for discontinuous transmission as that allows the length of

   phrases to be determined.  The issue is further discussed in
   "Guidelines for the Use of Variable Bit Rate Audio with Secure RTP"
   [RFC6562], which needs to be read by anyone writing an RTP payload
   format for an audio or speech codec with these properties.

6.2.  Video

   The definition of RTP payload formats for video has seen an evolution
   from the early ones such as H.261 [RFC4587] towards the latest for
   VP8 [RFC7741] and H.265/HEVC [RFC7798].

   The H.264 RTP payload format [RFC3984] can be seen as a smorgasbord
   of functionality: some of it, such as the interleaving, being pretty
   advanced.  The reason for this was to ensure that the majority of
   applications considered by the ITU-T and MPEG that can be supported
   by RTP are indeed supported.  This has created a payload format that
   rarely is fully implemented.  Despite that, no major issues with
   interoperability has been reported with one exception namely the
   Offer/Answer and parameter signaling, which resulted in a revised
   specification [RFC6184].  However, complaints about its complexity
   are common.

   The RTP payload format for uncompressed video [RFC4175] must be
   mentioned in this context as it contains a special feature not
   commonly seen in RTP payload formats.  Due to the high bitrate and
   thus packet rate of uncompressed video (gigabits rather than megabits
   per second) the payload format includes a field to extend the RTP
   sequence number since the normal 16-bit one can wrap in less than a
   second.  [RFC4175] also specifies a registry of different color sub-
   samplings that can be reused in other video RTP payload formats.

   Both the H.264 and the uncompressed video format enable the
   implementer to fulfill the goals of application-level framing, i.e.,
   each individual RTP Packet's payload can be independently decoded and
   its content used to create a video frame (or part of) and that
   irrespective of whether preceding packets has been lost (see
   Section 4) [RFC2736].  For uncompressed, this is straightforward as
   each pixel is independently represented from others and its location
   in the video frame known.  H.264 is more dependent on the actual
   implementation, configuration of the video encoder and usage of the
   RTP payload format.

   The common challenge with video is that, in most cases, a single
   compressed video frame doesn't fit into a single IP packet.  Thus,
   the compressed representation of a video frame needs to be split over
   multiple packets.  This can be done unintelligently with a basic
   payload level fragmentation method or more integrated by interfacing
   with the encoder's possibilities to create ADUs that are independent

   and fit the MTU for the RTP packet.  The latter is more robust and
   commonly recommended unless strong packet loss mechanisms are used
   and sufficient delay budget for the repair exist.  Commonly, both
   payload-level fragmentation as well as explaining how tailored ADUs
   can be created are needed in a video payload format.  Also, the
   handling of crucial metadata, like H.264 Parameter Sets, needs to be
   considered as decoding is not possible without receiving the used
   parameter sets.

6.3.  Text

   Only a single format text format has been standardized in the IETF,
   namely T.140 [RFC4103].  The 3GPP Timed Text format [RFC4396] should
   be considered to be text, even though in the end was registered as a
   video format.  It was registered in that part of the tree because it
   deals with decorated text, usable for subtitles and other
   embellishments of video.  However, it has many of the properties that
   text formats generally have.

   The RTP payload format for T.140 was designed with high reliability
   in mind as real-time text commonly is an extremely low bitrate
   application.  Thus, it recommends the use of RFC 2198 with many
   generations of redundancy.  However, the format failed to provide a
   text-block-specific sequence number and instead relies on the RTP one
   to detect loss.  This makes detection of missing text blocks
   unnecessarily difficult and hinders deployment with other robustness
   mechanisms that would involve switching the payload type, as that may
   result in erroneous error marking in the T.140 text stream.

6.4.  Application

   At the time of writing, the application content type contains two
   media types that aren't RTP transport robustness tools such as FEC
   [RFC3009] [RFC5109] [RFC6015] [RFC6682] and RTP retransmission

   The first one is H.224 [RFC4573], which enables far-end camera
   control over RTP.  This is not an IETF-defined RTP format, only an
   IETF-performed registration.

   The second one is "RTP Payload Format for Society of Motion Picture
   and Television Engineers (SMPTE) ST 336 Encoded Data" [RFC6597],
   which carries generic key length value (KLV) triplets.  These pairs
   may contain arbitrary binary metadata associated with video
   transmissions.  It has a very basic fragmentation mechanism requiring
   reception without packet loss, not only of the triplet itself but
   also one packet before and after the sequence of fragmented KLV
   triplet, to ensure correct reception.  Specific KLV triplets

   themselves may have recommendations on how to handle incomplete ones
   allowing the use and repair of them.  In general, the application
   using such a mechanism must be robust to errors and also use some
   combination of application-level repetition, RTP-level transport
   robustness tools, and network-level requirements to achieve low
   levels of packet loss rates and repair of KLV triplets.

   An author should consider applying for a media subtype under the
   application media type (application/<foo>) when the payload format is
   of a generic nature or does not clearly match any of the media types
   described above (audio, video, or text).  However, existing
   limitations in, for example, SDP, have resulted in generic mechanisms
   normally registered in all media types possibly having been
   associated with any existing media types in an RTP session.

7.  Important Specification Sections

   A number of sections in the payload format draft need special
   consideration.  These include the Security Considerations and IANA
   Considerations sections that are required in all drafts.  Payload
   formats are also strongly recommended to have the media format
   description and congestion control considerations.  The included RTP
   payload format template (Appendix A) contains sample text for some of
   these sections.

7.1.  Media Format Description

   The intention of this section is to enable reviewers and other
   readers to get an overview of the capabilities and major properties
   of the media format.  It should be kept short and concise and is not
   a complete replacement for reading the media format specification.

   The actual specification of the RTP payload format generally uses
   normative references to the codec format specification to define how
   codec data elements are included in the payload format.  This
   normative reference can be to anything that have sufficient stability
   for a normative reference.  There exist no formal requirement on the
   codec format specification being publicly available or free to
   access.  However, it significantly helps in the review process if
   that specification is made available to any reviewer.  There exist
   RTP payload format RFCs for open-source project specifications as
   well as an individual company's proprietary format, and a large
   variety of standards development organizations or industrial forums.

7.2.  Security Considerations

   All Internet-Drafts require a Security Considerations section.  The
   Security Considerations section in an RTP payload format needs to
   concentrate on the security properties this particular format has.
   Some payload formats have very few specific issues or properties and
   can fully fall back on the security considerations for RTP in general
   and those of the profile being used.  Because those documents are
   always applicable, a reference to these is normally placed first in
   the Security Considerations section.  There is suggested text in the
   template below.

   The security issues of confidentiality, integrity protection, replay
   protection and source authentication are common issue for all payload
   formats.  These should be solved by mechanisms external to the
   payload and do not need any special consideration in the payload
   format except for a reminder on these issues.  There exist
   exceptions, such as payload formats that includes security
   functionality, like ISMAcrypt [ISMACrypt2].  Reasons for this
   division is further documented in "Securing the RTP Protocol
   Framework: Why RTP Does Not Mandate a Single Media Security Solution"
   [RFC7202].  For a survey of available mechanisms to meet these goals,
   review "Options for Securing RTP Sessions" [RFC7201].  This also
   includes key-exchange mechanisms for the security mechanisms, which
   can be both integrated or separate.  The choice of key-management can
   have significant impact on the security properties of the RTP-based
   application.  Suitable stock text to inform people about this is
   included in the template.

   Potential security issues with an RTP payload format and the media
   encoding that need to be considered if they are applicable:

   1.  The decoding of the payload format or its media results in
       substantial non-uniformity, either in output or in complexity to
       perform the decoding operation.  For example, a generic non-
       destructive compression algorithm may provide an output of almost
       an infinite size for a very limited input, thus consuming memory
       or storage space out of proportion with what the receiving
       application expected.  Such inputs can cause some sort of
       disruption, i.e., a denial-of-service attack on the receiver side
       by preventing that host from performing usable work.  Certain
       decoding operations may also vary in the amount of processing
       needed to perform those operations depending on the input.  This
       may also be a security risk if it is possible to raise processing
       load significantly above nominal simply by designing a malicious
       input sequence.  If such potential attacks exist, this must be

       made clear in the Security Considerations section to make
       implementers aware of the need to take precautions against such

   2.  The inclusion of active content in the media format or its
       transport.  "Active content" means scripts, etc., that allow an
       attacker to perform potentially arbitrary operations on the
       receiver.  Most active contents has limited possibility to access
       the system or perform operations outside a protected sandbox.
       RFC 4855 [RFC4855] has a requirement that it be noted in the
       media types registration whether or not the payload format
       contains active content.  If the payload format has active
       content, it is strongly recommended that references to any
       security model applicable for such content are provided.  A
       boilerplate text for "no active content" is included in the
       template.  This must be changed if the format actually carries
       active content.

   3.  Some media formats allow for the carrying of "user data", or
       types of data which are not known at the time of the
       specification of the payload format.  Such data may be a security
       risk and should be mentioned.

   4.  Audio or Speech codecs supporting variable bitrate based on
       'audio/speech' input or having discontinuous transmission support
       must consider the issues discussed in "Guidelines for the Use of
       Variable Bit Rate Audio with Secure RTP" [RFC6562].

   Suitable stock text for the Security Considerations section is
   provided in the template in Appendix A.  However, authors do need to
   actively consider any security issues from the start.  Failure to
   address these issues may block approval and publication.

7.3.  Congestion Control

   RTP and its profiles do discuss congestion control.  There is ongoing
   work in the IETF with both a basic circuit-breaker mechanism
   [RFC8083] using basic RTCP messages intended to prevent persistent
   congestion and also work on more capable congestion avoidance /
   bitrate adaptation mechanism in the RMCAT WG.

   Congestion control is an important issue in any usage in networks
   that are not dedicated.  For that reason, it is recommended that all
   RTP payload format documents discuss the possibilities that exist to
   regulate the bitrate of the transmissions using the described RTP
   payload format.  Some formats may have limited or step-wise
   regulation of bitrate.  Such limiting factors should be discussed.

7.4.  IANA Considerations

   Since all RTP payload formats contain a media type specification,
   they also need an IANA Considerations section.  The media type name
   must be registered, and this is done by requesting that IANA register
   that media name.  When that registration request is written, it shall
   also be requested that the media type is included under the "RTP
   Payload Format media types" subregistry of the RTP registry

   Parameters for the payload format need to be included in this
   registration and can be specified as required or optional ones.  The
   format of these parameters should be such that they can be included
   in the SDP attribute "a=fmtp" string (see Section 6 [RFC4566]), which
   is the common mapping.  Some parameters, such as "Channel" are
   normally mapped to the rtpmap attribute instead; see Section 3 of

   In addition to the above request for media type registration, some
   payload formats may have parameters where, in the future, new
   parameter values need to be added.  In these cases, a registry for
   that parameter must be created.  This is done by defining the
   registry in the IANA Considerations section.  BCP 26 [BCP26] provides
   guidelines to specifying such registries.  Care should be taken when
   defining the policy for new registrations.

   Before specifying a new registry, it is worth checking the existing
   ones in the IANA "MIME Media Type Sub-Parameter Registries".  For
   example, video formats that need a media parameter expressing color
   sub-sampling may be able to reuse those defined for 'video/raw'

8.  Authoring Tools

   This section provides information about some tools that may be used.
   Don't feel pressured to follow these recommendations.  There exist a
   number of alternatives, including the ones listed at
   <http://tools.ietf.org>.  But these suggestions are worth checking
   out before deciding that the grass is greener somewhere else.

   Note that these options are related to the old text only RFC format,
   and do not cover tools for at the time of publication recently
   approved new RFC format, see [RFC7990].

8.1.  Editing Tools

   There are many choices when it comes to tools to choose for authoring
   Internet-Drafts.  However, in the end, they need to be able to
   produce a draft that conforms to the Internet-Draft requirements.  If
   you don't have any previous experience with authoring Internet-
   Drafts, xml2rfc does have some advantages.  It helps by creating a
   lot of the necessary boilerplate in accordance with the latest rules,
   thus reducing the effort.  It also speeds up publication after
   approval as the RFC Editor can use the source XML document to produce
   the RFC more quickly.

   Another common choice is to use Microsoft Word and a suitable
   template (see [RFC5385]) to produce the draft and print that to file
   using the generic text printer.  It has some advantages when it comes
   to spell checking and change bars.  However, Word may also produce
   some problems, like changing formatting, and inconsistent results
   between what one sees in the editor and in the generated text
   document, at least according to the author's personal experience.

8.2.  Verification Tools

   There are a few tools that are very good to know about when writing a
   draft.  These help check and verify parts of one's work.  These tools
   can be found at <http://tools.ietf.org>.

   o  I-D Nits checker (https://tools.ietf.org/tools/idnits/).  It
      checks that the boilerplate and some other things that are easily
      verifiable by machine are okay in your draft.  Always use it
      before submitting a draft to avoid direct refusal in the
      submission step.

   o  ABNF Parser and verification (https://tools.ietf.org/tools/bap/
      abnf.cgi).  Checks that your ABNF parses correctly and warns about
      loose ends, like undefined symbols.  However, the actual content
      can only be verified by humans knowing what it intends to

   o  RFC diff (https://tools.ietf.org/rfcdiff).  A diff tool that is
      optimized for drafts and RFCs.  For example, it does not point out
      that the footer and header have moved in relation to the text on
      every page.

9.  Security Considerations

   As this is an Informational RFC about writing drafts that are
   intended to become RFCs, there are no direct security considerations.
   However, the document does discuss the writing of Security
   Considerations sections and what should be particularly considered
   when specifying RTP payload formats.

10.  Informative References

   [BCP9]     Bradner, S., "The Internet Standards Process -- Revision
              3", BCP 9, RFC 2026, October 1996.

              Kolkman, O., Bradner, S., and S. Turner, "Characterization
              of Proposed Standards", BCP 9, RFC 7127, January 2014.

              Dusseault, L. and R. Sparks, "Guidance on Interoperation
              and Implementation Reports for Advancement to Draft
              Standard", BCP 9, RFC 5657, September 2009.

              Housley, R., Crocker, D., and E. Burger, "Reducing the
              Standards Track to Two Maturity Levels", BCP 9, RFC 6410,
              October 2011.

              Resnick, P., "Retirement of the "Internet Official
              Protocol Standards" Summary Document", BCP 9, RFC 7100,
              December 2013.

              Dawkins, S., "Increasing the Number of Area Directors in
              an IETF Area", BCP 9, RFC 7475, March 2015.


   [BCP25]    Wasserman, M., "Updates to RFC 2418 Regarding the
              Management of IETF Mailing Lists", BCP 25, RFC 3934,
              October 2004.

              Bradner, S., "IETF Working Group Guidelines and
              Procedures", BCP 25, RFC 2418, September 1998.

              Resnick, P. and A. Farrel, "IETF Anti-Harassment
              Procedures", BCP 25, RFC 7776, March 2016.


   [BCP26]    Narten, T. and H. Alvestrand, "Guidelines for Writing an
              IANA Considerations Section in RFCs", BCP 26, RFC 5226,
              May 2008, <http://www.rfc-editor.org/info/bcp26>.

   [BCP78]    Bradner, S., Ed. and J. Contreras, Ed., "Rights
              Contributors Provide to the IETF Trust", BCP 78, RFC 5378,
              November 2008, <http://www.rfc-editor.org/info/bcp78>.

   [BCP79]    Bradner, S., Ed., "Intellectual Property Rights in IETF
              Technology", BCP 79, RFC 3979, March 2005.

              Narten, T., "Clarification of the Third Party Disclosure
              Procedure in RFC 3979", BCP 79, RFC 4879, April 2007.


   [BLOAT]    Nichols, K. and V. Jacobson, "Controlling Queue Delay",
              ACM Networks, Vol. 10, No. 5, DOI 10.1145/2208917.2209336,
              May 2012, <http://queue.acm.org/detail.cfm?id=2209336>.

   [CSP-RTP]  Perkins, C., "RTP: Audio and Video for the Internet",
              Addison-Wesley Professional, ISBN 0-672-32249-8, June

   [ID-GUIDE] Housley, R., "Guidelines to Authors of Internet-Drafts",
              December 2010,

              Internet Streaming Media Alliance (ISMA), "ISMA Encryption
              and Authentication, Version 2.0 release version", November
              2007, <http://www.oipf.tv/docs/mpegif/isma_easpec2.0.pdf>.

   [RED]      Floyd, S. and V. Jacobson, "Random Early Detection (RED)
              gateways for Congestion Avoidance", IEEE/ACM Transactions
              on Networking 1(4) 397--413, August 1993,

   [RFC1633]  Braden, R., Clark, D., and S. Shenker, "Integrated
              Services in the Internet Architecture: an Overview",
              RFC 1633, DOI 10.17487/RFC1633, June 1994,

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

   [RFC2198]  Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
              Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
              Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
              DOI 10.17487/RFC2198, September 1997,

   [RFC2360]  Scott, G., "Guide for Internet Standards Writers", BCP 22,
              RFC 2360, DOI 10.17487/RFC2360, June 1998,

   [RFC2418]  Bradner, S., "IETF Working Group Guidelines and
              Procedures", BCP 25, RFC 2418, DOI 10.17487/RFC2418,
              September 1998, <http://www.rfc-editor.org/info/rfc2418>.

   [RFC2475]  Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z.,
              and W. Weiss, "An Architecture for Differentiated
              Services", RFC 2475, DOI 10.17487/RFC2475, December 1998,

   [RFC2508]  Casner, S. and V. Jacobson, "Compressing IP/UDP/RTP
              Headers for Low-Speed Serial Links", RFC 2508,
              DOI 10.17487/RFC2508, February 1999,

   [RFC2733]  Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format
              for Generic Forward Error Correction", RFC 2733,
              DOI 10.17487/RFC2733, December 1999,

   [RFC2736]  Handley, M. and C. Perkins, "Guidelines for Writers of RTP
              Payload Format Specifications", BCP 36, RFC 2736,
              DOI 10.17487/RFC2736, December 1999,

   [RFC2959]  Baugher, M., Strahm, B., and I. Suconick, "Real-Time
              Transport Protocol Management Information Base", RFC 2959,
              DOI 10.17487/RFC2959, October 2000,

   [RFC2974]  Handley, M., Perkins, C., and E. Whelan, "Session
              Announcement Protocol", RFC 2974, DOI 10.17487/RFC2974,
              October 2000, <http://www.rfc-editor.org/info/rfc2974>.

   [RFC3009]  Rosenberg, J. and H. Schulzrinne, "Registration of
              parityfec MIME types", RFC 3009, DOI 10.17487/RFC3009,
              November 2000, <http://www.rfc-editor.org/info/rfc3009>.

   [RFC3095]  Bormann, C., Burmeister, C., Degermark, M., Fukushima, H.,
              Hannu, H., Jonsson, L-E., Hakenberg, R., Koren, T., Le,
              K., Liu, Z., Martensson, A., Miyazaki, A., Svanbro, K.,
              Wiebke, T., Yoshimura, T., and H. Zheng, "RObust Header
              Compression (ROHC): Framework and four profiles: RTP, UDP,
              ESP, and uncompressed", RFC 3095, DOI 10.17487/RFC3095,
              July 2001, <http://www.rfc-editor.org/info/rfc3095>.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              DOI 10.17487/RFC3261, June 2002,

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              DOI 10.17487/RFC3264, June 2002,

   [RFC3410]  Case, J., Mundy, R., Partain, D., and B. Stewart,
              "Introduction and Applicability Statements for Internet-
              Standard Management Framework", RFC 3410,
              DOI 10.17487/RFC3410, December 2002,

   [RFC3497]  Gharai, L., Perkins, C., Goncher, G., and A. Mankin, "RTP
              Payload Format for Society of Motion Picture and
              Television Engineers (SMPTE) 292M Video", RFC 3497,
              DOI 10.17487/RFC3497, March 2003,

   [RFC3545]  Koren, T., Casner, S., Geevarghese, J., Thompson, B., and
              P. Ruddy, "Enhanced Compressed RTP (CRTP) for Links with
              High Delay, Packet Loss and Reordering", RFC 3545,
              DOI 10.17487/RFC3545, July 2003,

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
              July 2003, <http://www.rfc-editor.org/info/rfc3550>.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              DOI 10.17487/RFC3551, July 2003,

   [RFC3558]  Li, A., "RTP Payload Format for Enhanced Variable Rate
              Codecs (EVRC) and Selectable Mode Vocoders (SMV)",
              RFC 3558, DOI 10.17487/RFC3558, July 2003,

   [RFC3569]  Bhattacharyya, S., Ed., "An Overview of Source-Specific
              Multicast (SSM)", RFC 3569, DOI 10.17487/RFC3569, July
              2003, <http://www.rfc-editor.org/info/rfc3569>.

   [RFC3577]  Waldbusser, S., Cole, R., Kalbfleisch, C., and D.
              Romascanu, "Introduction to the Remote Monitoring (RMON)
              Family of MIB Modules", RFC 3577, DOI 10.17487/RFC3577,
              August 2003, <http://www.rfc-editor.org/info/rfc3577>.

   [RFC3611]  Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed.,
              "RTP Control Protocol Extended Reports (RTCP XR)",
              RFC 3611, DOI 10.17487/RFC3611, November 2003,

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, DOI 10.17487/RFC3711, March 2004,

   [RFC3828]  Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., Ed.,
              and G. Fairhurst, Ed., "The Lightweight User Datagram
              Protocol (UDP-Lite)", RFC 3828, DOI 10.17487/RFC3828, July
              2004, <http://www.rfc-editor.org/info/rfc3828>.

   [RFC3984]  Wenger, S., Hannuksela, M., Stockhammer, T., Westerlund,
              M., and D. Singer, "RTP Payload Format for H.264 Video",
              RFC 3984, DOI 10.17487/RFC3984, February 2005,

   [RFC4103]  Hellstrom, G. and P. Jones, "RTP Payload for Text
              Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,

   [RFC4170]  Thompson, B., Koren, T., and D. Wing, "Tunneling
              Multiplexed Compressed RTP (TCRTP)", BCP 110, RFC 4170,
              DOI 10.17487/RFC4170, November 2005,

   [RFC4175]  Gharai, L. and C. Perkins, "RTP Payload Format for
              Uncompressed Video", RFC 4175, DOI 10.17487/RFC4175,
              September 2005, <http://www.rfc-editor.org/info/rfc4175>.

   [RFC4352]  Sjoberg, J., Westerlund, M., Lakaniemi, A., and S. Wenger,
              "RTP Payload Format for the Extended Adaptive Multi-Rate
              Wideband (AMR-WB+) Audio Codec", RFC 4352,
              DOI 10.17487/RFC4352, January 2006,

   [RFC4396]  Rey, J. and Y. Matsui, "RTP Payload Format for 3rd
              Generation Partnership Project (3GPP) Timed Text",
              RFC 4396, DOI 10.17487/RFC4396, February 2006,

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
              July 2006, <http://www.rfc-editor.org/info/rfc4566>.

   [RFC4571]  Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
              and RTP Control Protocol (RTCP) Packets over Connection-
              Oriented Transport", RFC 4571, DOI 10.17487/RFC4571, July
              2006, <http://www.rfc-editor.org/info/rfc4571>.

   [RFC4573]  Even, R. and A. Lochbaum, "MIME Type Registration for RTP
              Payload Format for H.224", RFC 4573, DOI 10.17487/RFC4573,
              July 2006, <http://www.rfc-editor.org/info/rfc4573>.

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
              DOI 10.17487/RFC4585, July 2006,

   [RFC4587]  Even, R., "RTP Payload Format for H.261 Video Streams",
              RFC 4587, DOI 10.17487/RFC4587, August 2006,

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              DOI 10.17487/RFC4588, July 2006,

   [RFC4648]  Josefsson, S., "The Base16, Base32, and Base64 Data
              Encodings", RFC 4648, DOI 10.17487/RFC4648, October 2006,

   [RFC4844]  Daigle, L., Ed. and Internet Architecture Board, "The RFC
              Series and RFC Editor", RFC 4844, DOI 10.17487/RFC4844,
              July 2007, <http://www.rfc-editor.org/info/rfc4844>.

   [RFC4855]  Casner, S., "Media Type Registration of RTP Payload
              Formats", RFC 4855, DOI 10.17487/RFC4855, February 2007,

   [RFC4867]  Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie,
              "RTP Payload Format and File Storage Format for the
              Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband
              (AMR-WB) Audio Codecs", RFC 4867, DOI 10.17487/RFC4867,
              April 2007, <http://www.rfc-editor.org/info/rfc4867>.

   [RFC4975]  Campbell, B., Ed., Mahy, R., Ed., and C. Jennings, Ed.,
              "The Message Session Relay Protocol (MSRP)", RFC 4975,
              DOI 10.17487/RFC4975, September 2007,

   [RFC5109]  Li, A., Ed., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, DOI 10.17487/RFC5109, December
              2007, <http://www.rfc-editor.org/info/rfc5109>.

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
              2008, <http://www.rfc-editor.org/info/rfc5124>.

   [RFC5234]  Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax
              Specifications: ABNF", STD 68, RFC 5234,
              DOI 10.17487/RFC5234, January 2008,

   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
              2008, <http://www.rfc-editor.org/info/rfc5285>.

   [RFC5385]  Touch, J., "Version 2.0 Microsoft Word Template for
              Creating Internet Drafts and RFCs", RFC 5385,
              DOI 10.17487/RFC5385, February 2010,

   [RFC5484]  Singer, D., "Associating Time-Codes with RTP Streams",
              RFC 5484, DOI 10.17487/RFC5484, March 2009,

   [RFC5583]  Schierl, T. and S. Wenger, "Signaling Media Decoding
              Dependency in the Session Description Protocol (SDP)",
              RFC 5583, DOI 10.17487/RFC5583, July 2009,

   [RFC5795]  Sandlund, K., Pelletier, G., and L-E. Jonsson, "The RObust
              Header Compression (ROHC) Framework", RFC 5795,
              DOI 10.17487/RFC5795, March 2010,

   [RFC5905]  Mills, D., Martin, J., Ed., Burbank, J., and W. Kasch,
              "Network Time Protocol Version 4: Protocol and Algorithms
              Specification", RFC 5905, DOI 10.17487/RFC5905, June 2010,

   [RFC6015]  Begen, A., "RTP Payload Format for 1-D Interleaved Parity
              Forward Error Correction (FEC)", RFC 6015,
              DOI 10.17487/RFC6015, October 2010,

   [RFC6051]  Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
              Flows", RFC 6051, DOI 10.17487/RFC6051, November 2010,

   [RFC6184]  Wang, Y., Even, R., Kristensen, T., and R. Jesup, "RTP
              Payload Format for H.264 Video", RFC 6184,
              DOI 10.17487/RFC6184, May 2011,

   [RFC6190]  Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
              "RTP Payload Format for Scalable Video Coding", RFC 6190,
              DOI 10.17487/RFC6190, May 2011,

   [RFC6295]  Lazzaro, J. and J. Wawrzynek, "RTP Payload Format for
              MIDI", RFC 6295, DOI 10.17487/RFC6295, June 2011,

   [RFC6354]  Xie, Q., "Forward-Shifted RTP Redundancy Payload Support",
              RFC 6354, DOI 10.17487/RFC6354, August 2011,

   [RFC6363]  Watson, M., Begen, A., and V. Roca, "Forward Error
              Correction (FEC) Framework", RFC 6363,
              DOI 10.17487/RFC6363, October 2011,

   [RFC6410]  Housley, R., Crocker, D., and E. Burger, "Reducing the
              Standards Track to Two Maturity Levels", BCP 9, RFC 6410,
              DOI 10.17487/RFC6410, October 2011,

   [RFC6562]  Perkins, C. and JM. Valin, "Guidelines for the Use of
              Variable Bit Rate Audio with Secure RTP", RFC 6562,
              DOI 10.17487/RFC6562, March 2012,

   [RFC6597]  Downs, J., Ed. and J. Arbeiter, Ed., "RTP Payload Format
              for Society of Motion Picture and Television Engineers
              (SMPTE) ST 336 Encoded Data", RFC 6597,
              DOI 10.17487/RFC6597, April 2012,

   [RFC6679]  Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
              and K. Carlberg, "Explicit Congestion Notification (ECN)
              for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August
              2012, <http://www.rfc-editor.org/info/rfc6679>.

   [RFC6682]  Watson, M., Stockhammer, T., and M. Luby, "RTP Payload
              Format for Raptor Forward Error Correction (FEC)",
              RFC 6682, DOI 10.17487/RFC6682, August 2012,

   [RFC6701]  Farrel, A. and P. Resnick, "Sanctions Available for
              Application to Violators of IETF IPR Policy", RFC 6701,
              DOI 10.17487/RFC6701, August 2012,

   [RFC6838]  Freed, N., Klensin, J., and T. Hansen, "Media Type
              Specifications and Registration Procedures", BCP 13,
              RFC 6838, DOI 10.17487/RFC6838, January 2013,

   [RFC7160]  Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple
              Clock Rates in an RTP Session", RFC 7160,
              DOI 10.17487/RFC7160, April 2014,

   [RFC7164]  Gross, K. and R. Brandenburg, "RTP and Leap Seconds",
              RFC 7164, DOI 10.17487/RFC7164, March 2014,

   [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,

   [RFC7202]  Perkins, C. and M. Westerlund, "Securing the RTP
              Framework: Why RTP Does Not Mandate a Single Media
              Security Solution", RFC 7202, DOI 10.17487/RFC7202, April
              2014, <http://www.rfc-editor.org/info/rfc7202>.

   [RFC7231]  Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer
              Protocol (HTTP/1.1): Semantics and Content", RFC 7231,
              DOI 10.17487/RFC7231, June 2014,

   [RFC7273]  Williams, A., Gross, K., van Brandenburg, R., and H.
              Stokking, "RTP Clock Source Signalling", RFC 7273,
              DOI 10.17487/RFC7273, June 2014,

   [RFC7322]  Flanagan, H. and S. Ginoza, "RFC Style Guide", RFC 7322,
              DOI 10.17487/RFC7322, September 2014,

   [RFC7587]  Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format
              for the Opus Speech and Audio Codec", RFC 7587,
              DOI 10.17487/RFC7587, June 2015,

   [RFC7656]  Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
              B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
              for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
              DOI 10.17487/RFC7656, November 2015,

   [RFC7667]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
              DOI 10.17487/RFC7667, November 2015,

   [RFC7741]  Westin, P., Lundin, H., Glover, M., Uberti, J., and F.
              Galligan, "RTP Payload Format for VP8 Video", RFC 7741,
              DOI 10.17487/RFC7741, March 2016,

   [RFC7798]  Wang, Y., Sanchez, Y., Schierl, T., Wenger, S., and M.
              Hannuksela, "RTP Payload Format for High Efficiency Video
              Coding (HEVC)", RFC 7798, DOI 10.17487/RFC7798, March
              2016, <http://www.rfc-editor.org/info/rfc7798>.

   [RFC7826]  Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
              and M. Stiemerling, Ed., "Real-Time Streaming Protocol
              Version 2.0", RFC 7826, DOI 10.17487/RFC7826, December
              2016, <http://www.rfc-editor.org/info/rfc7826>.

   [RFC7990]  Flanagan, H., "RFC Format Framework", RFC 7990,
              DOI 10.17487/RFC7990, December 2016,

   [RFC8083]  Perkins, C. and V. Singh, "Multimedia Congestion Control:
              Circuit Breakers for Unicast RTP Sessions", RFC 8083,
              DOI 10.17487/RFC8083, March 2017,

   [TAO]      Hoffman, P., Ed., "The Tao of IETF: A Novice's Guide to
              the Internet Engineering Task Force", November 2012,

   [TRACKER]  "IETF Datatracker", <https://datatracker.ietf.org/>.

Appendix A.  RTP Payload Format Template

   This section contains a template for writing an RTP payload format in
   the form of an Internet-Draft.  Text within [...] are instructions
   and must be removed from the draft itself.  Some text proposals that
   are included are conditional. "..." is used to indicate where further
   text should be written.

A.1.  Title

   [The title shall be descriptive but as compact as possible.  RTP is
   allowed and recommended abbreviation in the title]

   RTP payload format for ...

A.2.  Front-Page Boilerplate

   Status of this Memo

   [Insert the IPR notice and copyright boilerplate from BCP 78 and 79
   that applies to this draft.]

   [Insert the current Internet-Draft document explanation.  At the time
   of publishing it was:]

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

A.3.  Abstract

   [A payload format abstract should mention the capabilities of the
   format, for which media format is used, and a little about that codec
   formats capabilities.  Any abbreviation used in the payload format
   must be spelled out here except the very well known like RTP.  No
   citations are allowed, and no use of language from RFC 2119 either.]

A.4.  Table of Contents

   [If your draft is approved for publication as an RFC, a Table of
   Contents is required, per [RFC7322].]

A.5.  Introduction

   [The Introduction should provide a background and overview of the
   payload format's capabilities.  No normative language in this
   section, i.e., no MUST, SHOULDs etc.]

A.6.  Conventions, Definitions, and Abbreviations

   [Define conventions, definitions, and abbreviations used in the
   document in this section.  The most common definition used in RTP
   payload formats are the RFC 2119 definitions of the uppercase
   normative words, e.g., MUST and SHOULD.]

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119.

A.7.  Media Format Description

   [The intention of this section is to enable reviewers and persons to
   get an overview of the capabilities and major properties of the media
   format.  It should be kept short and concise and is not a complete
   replacement for reading the media format specification.]

A.8.  Payload Format

   [Overview of payload structure]

A.8.1.  RTP Header Usage

   [RTP header usage needs to be defined.  The fields that absolutely
   need to be defined are timestamp and marker bit.  Further fields may
   be specified if used.  All the rest should be left to their RTP
   specification definition.]

   The remaining RTP header fields are used as specified in RTP

A.8.2.  Payload Header

   [Define how the payload header, if it exists, is structured and

A.8.3.  Payload Data

   [The payload data, i.e., what the media codec has produced.  Commonly
   done through reference to the media codec specification, which
   defines how the data is structured.  Rules for padding may need to be
   defined to bring data to octet alignment.]

A.9.  Payload Examples

   [One or more examples are good to help ease the understanding of the
   RTP payload format.]

A.10.  Congestion Control Considerations

   [This section is to describe the possibility to vary the bitrate as a
   response to congestion.  Below is also a proposal for an initial text
   that reference RTP and profiles definition of congestion control.]

   Congestion control for RTP SHALL be used in accordance with RFC 3550
   [RFC3550], and with any applicable RTP profile: e.g., RFC 3551
   [RFC3551].  An additional requirement if best-effort service is being
   used is users of this payload format MUST monitor packet loss to
   ensure that the packet loss rate is within acceptable parameters.
   Circuit Breakers [RFC8083] is an update to RTP [RFC3550] that defines
   criteria for when one is required to stop sending RTP Packet Streams.
   The circuit breakers is to be implemented and followed.

A.11.  Payload Format Parameters

   This RTP payload format is identified using the ... media type, which
   is registered in accordance with RFC 4855 [RFC4855] and using the
   template of RFC 6838 [RFC6838].

A.11.1.  Media Type Definition

   [Here the media type registration template from RFC 6838 is placed
   and filled out.  This template is provided with some common RTP

   Type name:

   Subtype name:

   Required parameters:

   Optional parameters:

   Encoding considerations:

      This media type is framed and binary; see Section 4.8 in RFC 6838

   Security considerations:

      Please see the Security Considerations section in RFC XXXX

   Interoperability considerations:

   Published specification:

   Applications that use this media type:

   Additional information:

      Deprecated alias names for this type:

         [Only applicable if there exists widely deployed alias for this
         media type; see Section 4.2.9 of [RFC6838].  Remove or use N/A

      Magic number(s):

         [Only applicable for media types that has file format
         specification.  Remove or use N/A otherwise.]

      File extension(s):

         [Only applicable for media types that has file format
         specification.  Remove or use N/A otherwise.]

      Macintosh file type code(s):

         [Only applicable for media types that has file format
         specification.  Even for file formats they can be skipped as
         they are not relied on after Mac OS 9.X.  Remove or use N/A

   Person & email address to contact for further information:

   Intended usage:


   Restrictions on usage:

      [The below text is for media types that is only defined for RTP
      payload formats.  There exist certain media types that are defined
      both as RTP payload formats and file transfer.  The rules for such
      types are documented in RFC 4855 [RFC4855].]

      This media type depends on RTP framing and, hence, is only defined
      for transfer via RTP [RFC3550].  Transport within other framing
      protocols is not defined at this time.


   Change controller:

   IETF Payload working group delegated from the IESG.

   Provisional registration? (standards tree only):


   (Any other information that the author deems interesting may be added
   below this line.)

   [From RFC 6838:

      "N/A", written exactly that way, can be used in any field if
      desired to emphasize the fact that it does not apply or that the
      question was not omitted by accident.  Do not use 'none' or other
      words that could be mistaken for a response.

      Limited-use media types should also note in the applications list
      whether or not that list is exhaustive.]

A.11.2.  Mapping to SDP

   The mapping of the above defined payload format media type and its
   parameters SHALL be done according to Section 3 of RFC 4855

   [More specific rules only need to be included if some parameter does
   not match these rules.]

A.11.2.1.  Offer/Answer Considerations

   [Here write your Offer/Answer considerations section; please see
   Section for help.]

A.11.2.2.  Declarative SDP Considerations

   [Here write your considerations for declarative SDP, please see
   Section for help.]

A.12.  IANA Considerations

   This memo requests that IANA registers [insert media type name here]
   as specified in Appendix A.11.1.  The media type is also requested to
   be added to the IANA registry for "RTP Payload Format MIME types"

   [See Section 7.4 and consider if any of the parameter needs a
   registered name space.]

A.13.  Security Considerations

   [See Section 7.2.]

   RTP packets using the payload format defined in this specification
   are subject to the security considerations discussed in the RTP
   specification [RFC3550] , and in any applicable RTP profile such as
   RTP/AVP [RFC3551], RTP/AVPF [RFC4585], RTP/SAVP [RFC3711], or RTP/
   SAVPF [RFC5124].  However, as "Securing the RTP Protocol Framework:
   Why RTP Does Not Mandate a Single Media Security Solution" [RFC7202]
   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 lays on anyone using RTP in an
   application.  They can find guidance on available security mechanisms
   and important considerations in "Options for Securing RTP Sessions"
   [RFC7201].  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 are unlikely to pose a
   denial-of-service threat due to the receipt of pathological data.
   Nor does the RTP payload format contain any active content.

   [The previous paragraph may need editing due to the format breaking
   either of the statements.  Fill in here any further potential
   security threats created by the payload format itself.]

A.14.  RFC Editor Considerations

   Note to RFC Editor: This section may be removed after carrying out
   all the instructions of this section.

   RFC XXXX is to be replaced by the RFC number this specification
   receives when published.

A.15.  References

   [References must be classified as either normative or informative and
   added to the relevant section.  References should use descriptive
   reference tags.]

A.15.1.  Normative References

   [Normative references are those that are required to be used to
   correctly implement the payload format.  Also, when requirements
   language is used, as in the sample text for "Congestion Control
   Considerations" above, there should be a normative reference to

A.15.2.  Informative References

   [All other references.]

A.16.  Authors' Addresses

   [All authors need to include their name and email address as a
   minimum: postal mail and possibly phone numbers are included

   [The Template Ends Here!]


   The author would like to thank the individuals who have provided
   input to this document.  These individuals include Richard Barnes,
   Ali C. Begen, Bo Burman, Ross Finlayson, Russ Housley, John Lazzaro,
   Jonathan Lennox, Colin Perkins, Tom Taylor, Stephan Wenger, and Qin


   The author would like to thank Tom Taylor for the editing pass of the
   whole document and contributing text regarding proprietary RTP
   payload formats.  Thanks also goes to Thomas Schierl who contributed
   text regarding Media Scalability features in payload formats
   (Section 5.1.5).  Stephan Wenger has contributed text on the need to
   understand the media coding (Section 3.1) as well as joint
   development of payload format with the media coding (Section 4.4).

Author's Address

   Magnus Westerlund
   Farogatan 2
   SE-164 80 Kista

   Phone: +46 10 714 82 87
   Email: magnus.westerlund@ericsson.com


User Contributions:

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