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Versions: (draft-ietf-avt-rtp-howto) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 RFC 8088

Payload Working Group                                      M. Westerlund
Internet-Draft                                                  Ericsson
Intended status: Informational                        September 21, 2013
Expires: March 25, 2014


                   How to Write an RTP Payload Format
                    draft-ietf-payload-rtp-howto-06

Abstract

   This document contains information on how to best write an RTP
   payload format.  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 that can be used when writing an RTP payload format.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   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."

   This Internet-Draft will expire on March 25, 2014.

Copyright Notice

   Copyright (c) 2013 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.



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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
     1.1.  Structure . . . . . . . . . . . . . . . . . . . . . . . .   4
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   4
     2.1.  Definitions . . . . . . . . . . . . . . . . . . . . . . .   4
     2.2.  Acronyms  . . . . . . . . . . . . . . . . . . . . . . . .   5
   3.  Preparations  . . . . . . . . . . . . . . . . . . . . . . . .   5
     3.1.  Recommend Reading . . . . . . . . . . . . . . . . . . . .   6
       3.1.1.  IETF Process and Publication  . . . . . . . . . . . .   6
       3.1.2.  RTP . . . . . . . . . . . . . . . . . . . . . . . . .   7
     3.2.  Important RTP details . . . . . . . . . . . . . . . . . .  11
       3.2.1.  The RTP Session . . . . . . . . . . . . . . . . . . .  12
       3.2.2.  RTP Header  . . . . . . . . . . . . . . . . . . . . .  12
       3.2.3.  RTP Multiplexing  . . . . . . . . . . . . . . . . . .  14
       3.2.4.  RTP Synchronization . . . . . . . . . . . . . . . . .  15
     3.3.  Signalling Aspects  . . . . . . . . . . . . . . . . . . .  17
       3.3.1.  Media Types . . . . . . . . . . . . . . . . . . . . .  17
       3.3.2.  Mapping to SDP  . . . . . . . . . . . . . . . . . . .  19
     3.4.  Transport Characteristics . . . . . . . . . . . . . . . .  21
       3.4.1.  Path MTU  . . . . . . . . . . . . . . . . . . . . . .  21
       3.4.2.  Different Queuing Algorithms  . . . . . . . . . . . .  21
       3.4.3.  Quality of Service  . . . . . . . . . . . . . . . . .  22
   4.  Standardisation Process for an RTP Payload Format . . . . . .  22
     4.1.  IETF  . . . . . . . . . . . . . . . . . . . . . . . . . .  23
       4.1.1.  Steps from Idea to Publication  . . . . . . . . . . .  23
       4.1.2.  WG meetings . . . . . . . . . . . . . . . . . . . . .  25
       4.1.3.  Draft Naming  . . . . . . . . . . . . . . . . . . . .  25
       4.1.4.  How to speed up the process . . . . . . . . . . . . .  25
     4.2.  Other Standards bodies  . . . . . . . . . . . . . . . . .  26
     4.3.  Proprietary and Vendor Specific . . . . . . . . . . . . .  27
   5.  Designing Payload Formats . . . . . . . . . . . . . . . . . .  28
     5.1.  Features of RTP Payload Formats . . . . . . . . . . . . .  28
       5.1.1.  Aggregation . . . . . . . . . . . . . . . . . . . . .  29
       5.1.2.  Fragmentation . . . . . . . . . . . . . . . . . . . .  29
       5.1.3.  Interleaving and Transmission Re-Scheduling . . . . .  29
       5.1.4.  Media Back Channels . . . . . . . . . . . . . . . . .  30
       5.1.5.  Media Scalability . . . . . . . . . . . . . . . . . .  30
       5.1.6.  High Packet Rates . . . . . . . . . . . . . . . . . .  32
     5.2.  Selecting Timestamp Definition  . . . . . . . . . . . . .  33
   6.  Noteworthy Aspects in Payload Format Design . . . . . . . . .  35
     6.1.  Audio Payloads  . . . . . . . . . . . . . . . . . . . . .  35
     6.2.  Video . . . . . . . . . . . . . . . . . . . . . . . . . .  36
     6.3.  Text  . . . . . . . . . . . . . . . . . . . . . . . . . .  37
     6.4.  Application . . . . . . . . . . . . . . . . . . . . . . .  37
   7.  Important Specification Sections  . . . . . . . . . . . . . .  38
     7.1.  Media Format Description  . . . . . . . . . . . . . . . .  38
     7.2.  Security Considerations . . . . . . . . . . . . . . . . .  38



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     7.3.  Congestion Control  . . . . . . . . . . . . . . . . . . .  40
     7.4.  IANA Considerations . . . . . . . . . . . . . . . . . . .  40
   8.  Authoring Tools . . . . . . . . . . . . . . . . . . . . . . .  41
     8.1.  Editing Tools . . . . . . . . . . . . . . . . . . . . . .  41
     8.2.  Verification Tools  . . . . . . . . . . . . . . . . . . .  42
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  42
   10. Security Considerations . . . . . . . . . . . . . . . . . . .  42
   11. Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  42
   12. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  42
   13. Informative References  . . . . . . . . . . . . . . . . . . .  43
   Appendix A.  RTP Payload Format Template  . . . . . . . . . . . .  49
     A.1.  Title . . . . . . . . . . . . . . . . . . . . . . . . . .  49
     A.2.  Front page boilerplate  . . . . . . . . . . . . . . . . .  49
     A.3.  Abstract  . . . . . . . . . . . . . . . . . . . . . . . .  50
     A.4.  Table of Content  . . . . . . . . . . . . . . . . . . . .  50
     A.5.  Introduction  . . . . . . . . . . . . . . . . . . . . . .  50
     A.6.  Conventions, Definitions and Acronyms . . . . . . . . . .  50
     A.7.  Media Format Description  . . . . . . . . . . . . . . . .  50
     A.8.  Payload format  . . . . . . . . . . . . . . . . . . . . .  51
       A.8.1.  RTP Header Usage  . . . . . . . . . . . . . . . . . .  51
       A.8.2.  Payload Header  . . . . . . . . . . . . . . . . . . .  51
       A.8.3.  Payload Data  . . . . . . . . . . . . . . . . . . . .  51
     A.9.  Payload Examples  . . . . . . . . . . . . . . . . . . . .  51
     A.10. Congestion Control Considerations . . . . . . . . . . . .  51
     A.11. Payload Format Parameters . . . . . . . . . . . . . . . .  52
       A.11.1.  Media Type Definition  . . . . . . . . . . . . . . .  52
       A.11.2.  Mapping to SDP . . . . . . . . . . . . . . . . . . .  54
     A.12. IANA Considerations . . . . . . . . . . . . . . . . . . .  54
     A.13. Security Considerations . . . . . . . . . . . . . . . . .  54
     A.14. RFC Editor Considerations . . . . . . . . . . . . . . . .  55
     A.15. References  . . . . . . . . . . . . . . . . . . . . . . .  55
       A.15.1.  Normative References . . . . . . . . . . . . . . . .  55
       A.15.2.  Informative References . . . . . . . . . . . . . . .  55
     A.16. Author Addresses  . . . . . . . . . . . . . . . . . . . .  55
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . .  56

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 can't 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 non-trivial and requires knowledge of
   both RTP and the real-time data format.





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   This document is intended to help any author of an RTP payload format
   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 have been defined, and robustness tools have
   also been defined, and these need to be considered.

   We also discuss the possible venues for defining an RTP payload
   format, in IETF, by other standard bodies and proprietary ones.

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 do 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 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

   Media Stream:  A sequence of RTP packets that together carry part or
      all of the content of a specific medium (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
      session.

   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
      SSRC identifiers.  See also Section 3.2.1.





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   RTP Payload Format:  The RTP payload format specifies how units of a
      specific encoded medium 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 sessions.

2.2.  Acronyms

   ABNF:  Augmented Backus-Naur Form [RFC5234]

   ADU:  Application Data Unit

   ALF:  Application Level Framing

   ASM:  Any-Source Multicast

   BCP:  Best Current Practice

   ID:  Internet Draft

   IESG:  Internet Engineering Steering Group

   MTU:  Maximum Transmission Unit

   WG:  Working Group

   QoS:  Quality of Service

   RFC:  Request For Comment

   RTP:  Real-time Transport Protocol

   RTCP:  RTP Control Protocol

   RTT:  Round Trip Time

   SSM:  Source Specific Multicast

3.  Preparations

   RTP is a complex real-time media delivery framework and it has a lot
   of details that needs 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 one can produce an RTP payload format of
   high quality.  On top of this, one needs to understand the process
   within IETF and especially the Working Group responsible for
   standardizing payload formats (currently PAYLOAD) to go quickly from



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   initial idea stage to a finished RFC.  This and the next section help
   an author prepare himself in those regards.

3.1.  Recommend Reading

   The following sub-sections 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.

3.1.1.  IETF Process and Publication

   Newcomers to the IETF are strongly recommended to read the "Tao of
   the IETF" [RFC6722] that goes through most things that one needs to
   know about the IETF.  This contains information about history,
   organizational structure, how the WG and meetings work and many more
   details.

   The main part of the IETF process is formally defined in RFC 2026
   [RFC2026].  In addition an author needs to understands the IETF rules
   and rights associated with copyright and IPR documented in BCP 78
   [RFC5378] and BCP 79 [RFC3979].  RFC 2418 [RFC2418] 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 The RFC
   Series and RFC Editor [RFC4844].  The most important stream for RTP
   payload formats authors are the IETF Stream.  In this streams the
   work of IETF is published.  The stream contains documents of several
   different categories: standards track, informational, experimental,
   best current practice (BCP), and historic.  The standard track
   previously allowed for documents of three different maturity
   classifications, proposed, draft and Internet Standard.  Since
   October 2011 this has been reduced to only two levels, Proposed
   Standard and Internet Standard [RFC6410].  A standards track document
   must start as proposed; after successful deployment and operational
   experience with at least two implementations it can be moved to
   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 which 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



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   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 engine, which displays the current status and which if any RFC
   update or obsolete it.  The RFC-editor search engine will also
   indicate if there exist any RFC-errata.  Any approved Errata is
   issues of significant importance with the RFC and thus should be
   known also 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 ID checklist (http://www.ietf.org/ID-
   Checklist.html) and the RFC editorial guidelines and procedures
   [RFC-ED].  Another document that can be useful is the "Guide for
   Internet Standards Writers" [RFC2360].

   There are also a number of documents to consider in process of
   writing of drafts intended to become RFCs.  These are important when
   writing certain type 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: 192.0.2.0/24, 198.51.100.0/24, and
      203.0.113.0/24.

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

3.1.2.  RTP

   The recommended reading for RTP consist 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 layer 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



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   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 RFC 3550 [RFC3550] and the existing
   profiles.  As a complement to the standards document there exists a
   book totally dedicated to RTP [CSP-RTP].  There exist several
   profiles for RTP today, but all are based on the "RTP Profile for
   Audio and Video Conferences with Minimal Control" (RFC 3551)
   [RFC3551] (abbreviated RTP/AVP).  The other profiles that one should
   know about are Secure RTP (RTP/SAVP) [RFC3711], "Extended RTP Profile
   for RTCP-based Feedback (RTP/AVPF)" [RFC4585] and "Extended Secure
   RTP Profile for 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:  The "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 and
      therefore 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 which allows the redundant data
      to transmitted prior to the original.

   RFC 5109:  The "RTP Payload Format for Generic Forward Error
      Correction (FEC)" [RFC5109] provides an XOR based 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 low-to-
      medium delay tolerant applications with limited amount of RTP
      packets.





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   RFC 6015:  The "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 produce 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:  The Forward Error Correction (FEC) Framework
      [RFC6363] defines how to FEC protect arbitrary packet flows.  This
      can be applied for RTP and also can also use RTP for transmission
      of repair symbols for the Raptor FEC scheme [RFC6682].

   RTP Retransmission:  The RTP retransmission scheme [RFC4588] is used
      for semi-reliability of the most important RTP packets in a media
      stream.  The scheme is not intended, to provide full reliability.
      It requires the application to be quite delay tolerant as a
      minimum of one round-trip time plus processing delay is required
      to perform an 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
      (RTT).

   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 already late data.  TCP is
      also limited to point-to-point connections which further restricts
      its applicability.
















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   There has also been both discussion and design of RTP payload
   formats, e.g AMR and 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
   if one can design something usable or not.  Payload format designers
   are recommended against considering features for unequal error
   detection 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 occurred.  The
      list of specified formats are available in IANA's RTCP XR Block
      Type registry (http://www.iana.org/assignments/rtcp-xr-block-types
      /rtcp-xr-block-types.xhtml).  The report formats that are defined
      in RFC3611 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 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
   reasons and do not require deeper studies.

   RFC 2508:  Compressing IP/UDP/RTP headers for slow 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.





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   RFC 3095 & RFC 5795:  This is 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.  Generic mechanisms by definition 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
   [I-D.ietf-avtcore-rtp-security-options] and Why RTP Does Not Mandate
   a Single Security Mechanism [I-D.ietf-avt-srtp-not-mandatory].  The
   main properties for a 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 which will need
   to be considered when one of these mechanisms is used.

   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 time of writing.  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 also other solutions
   than SRTP please consult "Options for Securing RTP Sessions"
   [I-D.ietf-avtcore-rtp-security-options].

3.2.  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 in some cases even
   effect the behavior (RTP Timestamp and marker bit).  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.



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3.2.1.  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 participants."

   "The RTP framework allows the variations defined here (RFC3550), but
   a particular control protocol or application design will usually
   impose constraints on these variations."

3.2.2.  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.
   The payload type is used to indicate the used payload format.  The
   Sender Source Identifier (SSRC) is used to distinguish RTP packets



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   from multiple senders and media streams.  Finally, [RFC5285]
   specifies how to extend the RTP header to carry metadata relating to
   the payload when this is desirable.

   Marker bit:  A single bit normally used to provide important
      indications.  In audio it is normally used to indicate the start
      of an 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 an 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 timestamp rate is an active design choice and is further
      discussed in Section 5.2.

      Discontinuous transmissions (DTX) that is common among speech
      codecs, typically results in gaps or jumps in the timestamp values
      due that there are 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 does
      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 signalling
      (Section 3.3).  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.




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   SSRC:  The Synchronisation Source Identifier (SSRC) 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.

   Header extensions:  Payload formats normally specifies the carrying
      of meta data related to the media payload in the payload header.
      In some cases the meta data may be more generic, e.g. relevant
      more to the media type (e.g. audio or video), or RTP packets in
      general.  In those cases the carrying of this additional meta data
      can be specified in a header extension.  One example is the
      transport of SMPTE time-codes in the RTP header [RFC5484].  As
      [RFC5285] specifies, header extensions must not contain
      information required in order to decode the payload successfully.

   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.

3.2.3.  RTP Multiplexing

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

   The first one is separation of media streams of different types or
   usages, which is accomplished using different RTP sessions.  So for
   example in the common multi-media 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 are also 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.









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   The next multiplexing point is separation of different sources within
   an RTP session.  Here RTP uses the SSRC to identify individual
   sources.  An example of individual sources in an audio RTP session
   would be different microphones, independently of whether they are
   connected to the same host or different hosts.  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].

   For more discussion and consideration of how and when to use the
   different RTP multiplexing points see
   [I-D.ietf-avtcore-multiplex-guidelines].

3.2.4.  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 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 media 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 also different
      media from the same end-point.  This puts two requirements on RTP:
      the possibility to determine which media are from the same end-
      point 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 SDES
   packets.  SSRCs with the same CNAME sent in any of multiple RTP
   sessions can be synchronized.





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   The actual RTCP mechanism for inter-media synchronization is based on
   the idea that each media 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 signalled using RTP Clock Source
   Signalling [I-D.ietf-avtcore-clksrc].

   Figure 1 illustrates how if one receives RTCP Sender Report (SR)
   packet P1 in one media stream and RTCP SR packet P2 in the other
   session, 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
   medium 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) -
   TS2(T))/90000.

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




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   Leap Seconds are extra seconds added or seconds removed to keep our
   clocks in sync with earths rotation.  Adding or removing seconds can
   impact first of all the reference clock as discussed in "RTP and Leap
   Seconds" [I-D.ietf-avtcore-leap-second].  But also in cases where the
   RTP timestamp value are derived using the wall clock during the leap
   second event errors can occur.  Implementations needs to consider
   leap seconds and should consider the recommendations in
   [I-D.ietf-avtcore-leap-second].

3.3.  Signalling Aspects

   RTP payload formats are used in the context of application signalling
   protocols such as SIP [RFC3261] using the Session Description
   Protocol (SDP) [RFC4566] with Offer/Answer [RFC3264], RTSP [RFC2326]
   or SAP [RFC2326].  These examples all use out-of-band signalling to
   indicate which type of RTP media streams that 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 have also 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 IETF
   (SDP-NG).  In the event, the proposal did not get beyond the initial
   protocol specification because of the enormous embedded base of SDP
   implementations.  However, to avoid locking the usage of RTP to SDP
   based out-of-band signalling, the payload formats are identified
   using a separate definition format for the identifier and associated
   parameters.  That format is the Media Type.

3.3.1.  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 [RFC2616], 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 two-part 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



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   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 IANA Web site (http://
   www.iana.org).

   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 exists no restrictions on how the
   value is defined from 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
   formats:

   1.  Arbitrary binary content in the parameters is allowed but 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 semi-
       colon, are recommended.  If that is not possible some type of
       escaping should be used.  Usage of quote (") is recommended, and
       don't 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









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3.3.2.  Mapping to SDP

   Since SDP [RFC4566] is so commonly used as an out-of-band signalling
   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 RFC 4855 [RFC4855].  However this is not sufficient to
   explain how certain parameters must be interpreted for example in the
   context of Offer/Answer negotiation [RFC3264].

3.3.2.1.  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
   and instead 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 media 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 media 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 which 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 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 that the
   answerer 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



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   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 shoe-horned 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 media 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 an new offer to update the
   parameter values for its outgoing media stream.

   Another issue to consider is the sendonly media 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 sendonly 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 which 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.

3.3.2.2.  Declarative usage in RTSP and SAP

   SAP (Session Announcement Protocol) [RFC2974] is used for announcing
   multicast sessions.  Independently of the usage of Source Specific
   Multicast (SSM) [RFC3569] or Any-Source Multicast (ASM), the SDP
   provided by SAP applies to all participants.  All media that is sent
   to the session must follow the media 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



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   has decided upon.  Any RTP Payload format that requires parameters
   for the send direction and which 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) [RFC2326] allows the negotiation
   of transport parameters for media streams which 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 media 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 media 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.

3.4.  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.

3.4.1.  Path MTU

   At the time of writing this document the most common IP Maximum
   Transmission Unit (MTU) in used link layers is 1500 bytes (Ethernet
   data payload).  However there exist both links with smaller MTUs and
   links with much larger MTUs.  Certain parts of the Internet already
   today support an IP MTU of 9000 bytes or more.  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 who's overhead when added together can shrink the MTU
   significant.  This should be considered in the design, especially in
   regards to features such as aggregation of independently decodable
   data units.

3.4.2.  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



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   to end characteristics.  One of the more important aspects of this is
   queuing behavior.  Routers and Switches has 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 effects the delay and also how
   congestion signals (Explicint Congestion Marking (ECN) 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 interacts with long lived TCP flows.  TCP have a built in
   behavior in its congestion control that strive 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.
   Active Queue Management (AQM) is a term covering mechanism that tries
   to do something smarter buy actively managing the queue, for example
   by sending congestion signals earlier by droping packets earlier in
   the queue.  The behavior also affects the flow interactions.  For
   example Random Early Drop (RED) selects which packet to drop
   randomly.  This to give flows that have more packets in the queue a
   higher probability to experience the packet loss (congestion signal).
   There is ongoing work to find suitable mechanisms to recommend for
   implementation and reduce the use of tail-drop.

3.4.3.  Quality of Service

   Using best effort Internet one have no guaratees for the paths
   properties.  Quality of Service (QoS) mechanism are intended to
   provide the possility to bound the path properties.  Where Diffserv
   [RFC2475] markings effects the queing 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 [RFC1663] has the potential for
   stricter guarantees as the properties are agreed on by each hop on
   the path.

4.  Standardisation 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
   registration of the Media Type for the RTP payload format.  For
   proprietary media formats the primary goal depends on whether



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   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 Working Group applies when
   working with RTP payload formats.

   Idea:   Determined the need for an RTP payload format as an IETF
      specification.

   Initial effort:   Using this document as 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 Working Group 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.  It
      is important to consider from the start 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 first version:   When one has performed the above one
      submits the draft as an individual draft.  This can be done at any
      time except the 3 weeks (current deadline at the time of writing,
      consult current announcements) prior to an IETF meeting.  When the
      IETF draft announcement has been sent out on the draft
      announcement list, forward it to the PAYLOAD WG and request that
      it be reviewed.  In the email outline any issues the authors
      currently have with the design.




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   Iterative improvements:   Taking the feedback 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 buffer
      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 chairs
      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 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.

   WG Last Call:   A WG Last Call of at least 2 weeks is always
      performed for payload formats in the PAYLOAD WG.  The authors
      request WG last call for a draft when they think it is mature
      enough for publication.  The chairs perform a review to check if
      they agree with the authors' assessment.  If the chairs 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 updates of 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.















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   Publication Requested:   For WG documents the chairs request
      publication of the draft, after it has passed WG Last Call.  After
      this the approval and publication process described in RFC 2026
      [RFC2026] are performed.  The status after the publication has
      been requested can be tracked using the IETF data 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 its correctness.

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
   discussion.

   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 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 its WG members a
   specific draft file naming convention shall be used for RTP payload
   formats.  Individual submissions shall be named 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 filename 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 acronym.
   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.

4.1.4.  How to speed up the process




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   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
      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 are
      speed up.  This minimizes the delay that the author has direct
      control over.  The time taken for reviews, responses from area
      directors and 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.
      Try also 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 initial submissions, checking if a draft is ready
      to become a WG document or ready for WG last call.

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



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   specification, a second review should be scheduled.  In that pass the
   quality can be assessed and 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 standard bodies.  The requirements on the organization are
   outlined in the media types registration document [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 signalling 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
      sides, 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 register keeping tracks
   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 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



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   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.

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 a non-broken
      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 requirements to support these features; instead, their
   applicability must be considered for each payload format.  It may in
   fact be that certain features are not even applicable.





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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.  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.
   However in certain use cases the possibility to aggregate multiple
   ADUs especially for different playback times is useful.

   The main disadvantage of aggregation 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.  Mainly that IP fragmented packets commonly are discarded
   in the network, especially by Network Address Translators or
   Firewalls.  The usage of RTP payload format-level fragmentation
   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.

5.1.3.  Interleaving and Transmission Re-Scheduling

   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 affect 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 spread 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



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   aggregation there is still a possibility of implementing a
   transmission order re-scheduling mechanism.  That has the effect that
   the packets transmitted consecutively originate from different points
   in the media 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 re-scheduling is only useful in a few specific
   cases, as in streaming with retransmissions.  The potential gains
   must be weighted 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 signalling 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 media 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 video codecs it is possible to adapt the frame rate,
      e.g. for H.264 [RFC6184].

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




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   Quality:  The quality of the media 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 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 Scalable Video
   Coding [RFC6190] (SVC).

   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 media stream, i.e. RTP packets may be not transmitted or
   forwarded to a client in order to adapt the media stream rate as well
   as the media stream quality, while still providing a decodable subset
   of the media stream to a client.  One example for using the
   scalability feature may be an RTP Mixer (Multipoint Control Unit)
   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 an 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 media stream (base layer) may get
   better QoS parameters, then 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 single 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 media stream requires dropping of specific packets in order to
   provide the client with a compliant media stream.  In case of using
   encryption, it is typically required for an adapting network device



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   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 that belongs to the layers, that no other
   layer will be dependent on after the dropping.  Secondly, if the MANE
   can work as RTP mixer or translator it can rewrite the RTP and RTCP
   in such a way that the receiver will not suspect non-intentional 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 sessions, and thus transport flows, e.g. on
   differnt 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 for dropping packets is based on the Network Address which
   identifies the RTP session to be dropped.  In order to allow correct
   data provision to a decoder after reception from different sessions,
   data re-alignment mechanisms are described for Scalable Video Coding
   [RFC6190].  A more generic one is also described in Rapid Sync for
   RTP flows [RFC6051], which is purely based on existing RTP
   mechanisms, i.e. the NTP timestamp, for inter-session
   synchronization.  Another signaling feature is the generic indication
   of dependencies of RTP sessions in SDP, as defined in the Media
   Decoding Dependency Grouping in SDP [RFC5583].

   Using MST within a single RTP session is also possible and allow
   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
   end-point has the benefit in MST to know which layer or set of layers
   the missing packets belong as it will be bound to different RTP
   sessions.  Thus explicitly indicating the importance of the loss.

5.1.6.  High Packet Rates

   Some media codecs require high packet rates, and in these cases the
   RTP sequence number wraps too quickly.  As rule of thumb, it must not
   be possible to wrap the sequence number space in less than 2 minutes
   (TCP maximum segment lifetime).  If earlier wrapping may occur then



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   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 mechanism that
   do not use extended counters there is significant risk that they wrap
   multiple times between RTCP reporting or feedback, thus producing
   uncertainty which packet(s) that 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 have 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 which 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.  Such
   examples are for example payload formats for video that requires
   multiple packets, where a set of packets will have the same
   timestamp, another are payloads that includes alternating set of
   redundant information, if the definition used are for the earliest
   data included, the the timestamp may jump back and forth.  It should
   be noted if the payload format has this property or not.

   A RTP payload format also needs to define what timestamp rates, or
   clock rates (as it is also called), that 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 rate?





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   The rate needs be selected so that one can determine where in the
   timeline 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
   irregularities.

   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 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)
   [I-D.ietf-avtext-multiple-clock-rates] this should be avoided if one
   expects to need to switch between modes.

   An alternatives then is to find a common denominator frequency
   between the different modes, e.g. OPUS [I-D.ietf-payload-rtp-opus]
   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 is 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 has 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 it will
   cause a too low resolution in the RTCP measurements that are



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   expressed in RTP timestamps.  This is the main reason that the text
   RTP payload formats, like T.140 [RFC4103] uses 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 specific details of
   their design.

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 content 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 that in AMR-WB+ [RFC4352] may be more
   suitable.

   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 exist a security concern with variable bit-rate audio and
   speech codecs that changes their payload length based on the input
   data.  This can leak information, especially in structured
   communication like speech recognition prompt service that asks people
   to enter information verbally.  This issue also exist 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 read by anyone writing an RTP payload format for an audio or
   speech codec with these properties.









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6.2.  Video

   The definition of RTP payload formats for video has seen an evolution
   from the early ones such as H.261 towards the latest for VC-1 and
   H.264.

   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 signalling, 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 bit-rate and
   thus packet rate of uncompressed video (gigabits rather than
   megabits) 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 re-used in other video RTP payload formats.

   Both, the H.264 and the uncompressed video format enables the
   implementor 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 don'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 unintelligent with a basic
   payload level fragmentation method or more integrated by interfacing
   with the encoder's possibilities to create ADUs that are
   independently and fits the MTU for the RTP packet.  The later 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 an video payload format.  Also the



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   handling of crucial meta data, 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 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 bit-rate
   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 relies instead of 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

   The application content type contains at the time of writing two
   media types that aren't RTP transport robustness tools such as FEC
   [RFC3009][RFC5109][RFC6015][RFC6682] and RTP retransmission
   [RFC4588].

   The first one is H224 [RFC4573] which enables far end camera control
   over RTP.  This is not an IETF defined RTP format, only an IETF
   performed registration.

















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   The second one is the 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 meta data associated with
   video transmissions.  It has a very basic fragmentation mechanism
   requiring packet loss free reception 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 recommendation on how to handle non-complete 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.

   The application top media type (application/<foo>) should be
   considered to be used when the payload format defined is not clearly
   matching any of the existing media types (audio, video or text) or
   are of a generic nature.  However, existing limitations in for
   example SDP, has resulted in that generic mechanisms normally are
   registered in all media types to be possible to have 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 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 draft 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.

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



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   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 an 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"
   [I-D.ietf-avt-srtp-not-mandatory].  For a survey of available
   mechanisms to meet these goals, please review "Options for Securing
   RTP Sessions" [I-D.ietf-avtcore-rtp-security-options].  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 needs 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
       behavior.

   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



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       media types registration if the payload format contains active
       content or not.  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 bit-rate 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 the appendix.  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 are
   ongoing work in IETF with both a basic circuit breaker mechanism
   [I-D.ietf-avtcore-rtp-circuit-breakers] using basic RTCP messages
   intended to prevent persistent congestion, but also work on more
   capable congestion avoidance / bit-rate adaptation mechanism in the
   RMCAT WG.

   Congestion control is an important issue in any usage in non-
   dedicated networks.  For that reason it is recommended that all RTP
   payload format documents discuss the possibilities that exist to
   regulate the bit-rate of the transmissions using the described RTP
   payload format.  Some formats may have limited or step wise
   regulation of bit-rate.  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" list part of the RTP registry (http://
   www.iana.org/assignments/rtp-parameters).




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   Parameters for the payload format needs to be included in this
   registration and can be specified as required or optional ones.  The
   format of these parameter 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 [RFC4855].

   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 [RFC5226] 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" list.
   For example video formats needing a media parameter expressing color
   sub-sampling may be able to reuse those defined for video/raw
   [RFC4175].

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.  But these suggestions are worth checking out
   before deciding that the field is greener somewhere else.

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 create a lot of the
   necessary boiler plate 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 authors' personal experience.




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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  ID Nits checker.  It checks that the boiler plate 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.  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 describe.

   o  RFC diff.  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.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an
   RFC.

10.  Security Considerations

   As this is an informational document 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.

11.  Contributors

   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.

12.  Acknowledgements

   The author would like to thank the individuals who have provided
   input to this document.  These individuals include Ali C. Begen, John
   Lazzaro, Jonathan Lennox, Colin Perkins, Tom Taylor and Qin Wu.




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13.  Informative References

   [CSP-RTP]  Perkins, C., "RTP: Audio and Video for the Internet", June
              2003.

   [I-D.ietf-avt-srtp-not-mandatory]
              Perkins, C. and M. Westerlund, "Securing the RTP Protocol
              Framework: Why RTP Does Not Mandate a Single Media
              Security Solution", draft-ietf-avt-srtp-not-mandatory-13
              (work in progress), May 2013.

   [I-D.ietf-avtcore-clksrc]
              Williams, A., Gross, K., Brandenburg, R., and H. Stokking,
              "RTP Clock Source Signalling", draft-ietf-avtcore-
              clksrc-06 (work in progress), September 2013.

   [I-D.ietf-avtcore-leap-second]
              Gross, K. and R. Brandenburg, "RTP and Leap Seconds",
              draft-ietf-avtcore-leap-second-04 (work in progress),
              August 2013.

   [I-D.ietf-avtcore-multiplex-guidelines]
              Westerlund, M., Perkins, C., and H. Alvestrand,
              "Guidelines for using the Multiplexing Features of RTP to
              Support Multiple Media Streams", draft-ietf-avtcore-
              multiplex-guidelines-01 (work in progress), July 2013.

   [I-D.ietf-avtcore-rtp-circuit-breakers]
              Perkins, C. and V. Singh, "Multimedia Congestion Control:
              Circuit Breakers for Unicast RTP Sessions", draft-ietf-
              avtcore-rtp-circuit-breakers-03 (work in progress), July
              2013.

   [I-D.ietf-avtcore-rtp-security-options]
              Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", draft-ietf-avtcore-rtp-security-options-06
              (work in progress), August 2013.

   [I-D.ietf-avtext-multiple-clock-rates]
              Petit-Huguenin, M. and G. Zorn, "Support for Multiple
              Clock Rates in an RTP Session", draft-ietf-avtext-
              multiple-clock-rates-10 (work in progress), September
              2013.

   [I-D.ietf-payload-rtp-opus]
              Spittka, J., Vos, K., and J. Valin, "RTP Payload Format
              for Opus Speech and Audio Codec", draft-ietf-payload-rtp-
              opus-01 (work in progress), August 2013.



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   [ISMACrypt2]
              , "ISMA Encryption and Authentication, Version 2.0 release
              version", November 2007.

   [MACOSFILETYPES]
              Apple Knowledge Base Article
              55381<http://www.info.apple.com/kbnum/n55381>, "Mac OS:
              File Type and Creator Codes, and File Formats", 1993.

   [RFC-ED]   http://www.rfc-editor.org/policy.html, "RFC Editorial
              Guidelines and Procedures", July 2008.

   [RFC1663]  Rand, D., "PPP Reliable Transmission", RFC 1663, July
              1994.

   [RFC2026]  Bradner, S., "The Internet Standards Process -- Revision
              3", BCP 9, RFC 2026, October 1996.

   [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,
              September 1997.

   [RFC2326]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
              Streaming Protocol (RTSP)", RFC 2326, April 1998.

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

   [RFC2418]  Bradner, S., "IETF Working Group Guidelines and
              Procedures", BCP 25, RFC 2418, September 1998.

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

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

   [RFC2616]  Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
              Masinter, L., Leach, P., and T. Berners-Lee, "Hypertext
              Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999.

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




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

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

   [RFC2974]  Handley, M., Perkins, C., and E. Whelan, "Session
              Announcement Protocol", RFC 2974, October 2000.

   [RFC3009]  Rosenberg, J. and H. Schulzrinne, "Registration of
              parityfec MIME types", RFC 3009, November 2000.

   [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, July 2001.

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

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

   [RFC3410]  Case, J., Mundy, R., Partain, D., and B. Stewart,
              "Introduction and Applicability Statements for Internet-
              Standard Management Framework", RFC 3410, 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, 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, July
              2003.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.




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

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

   [RFC3569]  Bhattacharyya, S., "An Overview of Source-Specific
              Multicast (SSM)", RFC 3569, July 2003.

   [RFC3577]  Waldbusser, S., Cole, R., Kalbfleisch, C., and D.
              Romascanu, "Introduction to the Remote Monitoring (RMON)
              Family of MIB Modules", RFC 3577, August 2003.

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

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

   [RFC3828]  Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., and
              G. Fairhurst, "The Lightweight User Datagram Protocol
              (UDP-Lite)", RFC 3828, July 2004.

   [RFC3979]  Bradner, S., "Intellectual Property Rights in IETF
              Technology", BCP 79, RFC 3979, March 2005.

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

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

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

   [RFC4175]  Gharai, L. and C. Perkins, "RTP Payload Format for
              Uncompressed Video", RFC 4175, September 2005.

   [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, January 2006.




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

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [RFC4571]  Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
              and RTP Control Protocol (RTCP) Packets over Connection-
              Oriented Transport", RFC 4571, July 2006.

   [RFC4573]  Even, R. and A. Lochbaum, "MIME Type Registration for RTP
              Payload Format for H.224", RFC 4573, July 2006.

   [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, July
              2006.

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

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

   [RFC4844]  Daigle, L. Internet Architecture Board, "The RFC Series
              and RFC Editor", RFC 4844, July 2007.

   [RFC4855]  Casner, S., "Media Type Registration of RTP Payload
              Formats", RFC 4855, 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, April 2007.

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

   [RFC5109]  Li, A., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, December 2007.

   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, February 2008.





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   [RFC5226]  Narten, T. and H. Alvestrand, "Guidelines for Writing an
              IANA Considerations Section in RFCs", BCP 26, RFC 5226,
              May 2008.

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

   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", RFC 5285, July 2008.

   [RFC5378]  Bradner, S. and J. Contreras, "Rights Contributors Provide
              to the IETF Trust", BCP 78, RFC 5378, November 2008.

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

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

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

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

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

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

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

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

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

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




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

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

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

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

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

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

   [RFC6722]  Hoffman, P., "Publishing the "Tao of the IETF" as a Web
              Page", RFC 6722, August 2012.

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

Appendix A.  RTP Payload Format Template

   This section contains a template for writing an RTP payload format in
   form as a Internet draft.  Text within [...] are instructions and
   must be removed.  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




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   [Insert the IPR notice and copyright boiler plate 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
   references are allowed, no use of RFC 2119 language either.]

A.4.  Table of Content

   [All drafts over 15 pages in length must have an Table of Content.]

A.5.  Introduction

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

A.6.  Conventions, Definitions and Acronyms

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

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119.

A.7.  Media Format Description





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   [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 field 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 [RFC
   3550].

A.8.2.  Payload Header

   [Define how the payload header, if it exist, is structured and used.]

A.8.3.  Payload Data

   [The payload data, i.e. what the media codec has produced.  Commonly
   done through reference to 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 bit-rate 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 [I-D.ietf-avtcore-rtp-circuit-breakers] is an update



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   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
   boilerplate.]

   Type name:

   Subtype name:

   Required parameters:

   Optional parameters:

   Encoding considerations:

      This media type is framed and binary, see section 4.8 in RFC6838
      [RFC6838].

   Security considerations:

   Please see security consideration in RFCXXXX

   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
         otherwise.]

      Magic number(s):



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         [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.  Remove or use N/A otherwise.]

   Person & email address to contact for further information:

   Intended usage:

      [One of COMMON, LIMITED USE or OBSOLETE.]

   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.

   Author:

   Change controller:

   IETF Payload working group delegated from the IESG.

   Provisional registration? (standards tree only):

      No

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

   [From RFC 6838:

      Some discussion of Macintosh file type codes and their purpose can
      be found in [MACOSFILETYPES].




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      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
   [RFC4855].

   [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 consideration section, please see
   Section 3.3.2.1 for help.]

A.11.2.2.  Declarative SDP Considerations

   [Here write your considerations for declarative SDP, please see
   Section 3.3.2.2 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"
   (http://www.iana.org/assignments/rtp-parameters).

   [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"
   [I-D.ietf-avt-srtp-not-mandatory] discusses it is not an RTP payload



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   formats 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 [I-D.ietf-avtcore-rtp-security-options].
   The rest of the this security consideration 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.

   RFCXXXX 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.]

A.15.2.  Informative References

   [All other references.]

A.16.  Author Addresses

   [All Authors need to include their Name and email addresses as a
   minimal.  Commonly also surface mail and possibly phone numbers are
   included.]




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   [The Template Ends Here!]

Author's Address

   Magnus Westerlund
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

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







































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