RFC 8088
How to Write an RTP Payload Format
3.3. Important RTP Details
This section reviews a number of RTP features and concepts that are available in RTP, independent of the payload format. The RTP payload format can make use of these when appropriate, and even affect the behavior (RTP timestamp and marker bit), but it is important to note that not all features and concepts are relevant to every payload format. This section does not remove the necessity to read up on RTP. However, it does point out a few important details to remember when designing a payload format.3.3.1. The RTP Session
The definition of the RTP session from RFC 3550 is: An association among a set of participants communicating with RTP. A participant may be involved in multiple RTP sessions at the same time. In a multimedia session, each medium is typically carried in a separate RTP session with its own RTCP packets unless the encoding itself multiplexes multiple media into a single data stream. A participant distinguishes multiple RTP sessions by reception of different sessions using different pairs of destination transport addresses, where a pair of transport addresses comprises one network address plus a pair of ports for RTP and RTCP. All participants in an RTP session may share a common destination transport address pair, as in the case of IP multicast, or the pairs may be different for each participant, as in the case of individual unicast network addresses and port pairs. In the unicast case, a participant may receive from all other participants in the session using the same pair of ports, or may use a distinct pair of ports for each. The distinguishing feature of an RTP session is that each session maintains a full, separate space of SSRC identifiers (defined next). The set of participants included in one RTP session consists of those that can receive an SSRC identifier transmitted by any one of the participants either in RTP as the SSRC or a CSRC (also defined below) or in RTCP. For example, consider a three- party conference implemented using unicast UDP with each participant receiving from the other two on separate port pairs. If each participant sends RTCP feedback about data received from one other participant only back to that participant, then the conference is composed of three separate point-to-point RTP sessions. If each participant provides RTCP feedback about its
reception of one other participant to both of the other
participants, then the conference is composed of one multi-party
RTP session. The latter case simulates the behavior that would
occur with IP multicast communication among the three
participants.
The RTP framework allows the variations defined here, but a
particular control protocol or application design will usually
impose constraints on these variations.
3.3.2. RTP Header
The RTP header contains a number of fields. Two fields always
require additional specification by the RTP payload format, namely
the RTP timestamp and the marker bit. Certain RTP payload formats
also use the RTP sequence number to realize certain functionalities,
primarily related to the order of their application data units. The
payload type is used to indicate the used payload format. The SSRC
is used to distinguish RTP packets from multiple senders and media
sources identifying the RTP stream. Finally, [RFC5285] specifies how
to transport payload format independent metadata relating to the RTP
packet or stream.
Marker Bit: A single bit normally used to provide important
indications. In audio, it is normally used to indicate the start
of a talk burst. This enables jitter buffer adaptation prior to
the beginning of the burst with minimal audio quality impact. In
video, the marker bit is normally used to indicate the last packet
part of a frame. This enables a decoder to finish decoding the
picture, where it otherwise may need to wait for the next packet
to explicitly know that the frame is finished.
Timestamp: The RTP timestamp indicates the time instance the media
sample belongs to. For discrete media like video, it normally
indicates when the media (frame) was sampled. For continuous
media, it normally indicates the first time instance the media
present in the payload represents. For audio, this is the
sampling time of the first sample. All RTP payload formats must
specify the meaning of the timestamp value and the clock rates
allowed. Selecting a timestamp rate is an active design choice
and is further discussed in Section 5.2.
Discontinuous Transmission (DTX) that is common among speech
codecs, typically results in gaps or jumps in the timestamp values
due to that there is no media payload to transmit and the next
used timestamp value represent the actual sampling time of the
data transmitted.
Sequence Number: The sequence number is monotonically increasing and
is set as the packet is sent. This property is used in many
payload formats to recover the order of everything from the whole
stream down to fragments of application data units (ADUs) and the
order they need to be decoded. Discontinuous transmissions do not
result in gaps in the sequence number, as it is monotonically
increasing for each sent RTP packet.
Payload Type: The payload type is used to indicate, on a per-packet
basis, which format is used. The binding between a payload type
number and a payload format and its configuration are dynamically
bound and RTP session specific. The configuration information can
be bound to a payload type value by out-of-band signaling
(Section 3.4). An example of this would be video decoder
configuration information. Commonly, the same payload type is
used for a media stream for the whole duration of a session.
However, in some cases it may be necessary to change the payload
format or its configuration during the session.
SSRC: The synchronization source (SSRC) identifier is normally not
used by a payload format other than to identify the RTP timestamp
and sequence number space a packet belongs to, allowing
simultaneously reception of multiple media sources. However, some
of the RTP mechanisms for improving resilience to packet loss uses
multiple SSRCs to separate original data and repair or redundant
data, as well as multi-stream transmission of scalable codecs.
Header Extensions: RTP payload formats often need to include
metadata relating to the payload data being transported. Such
metadata is sent as a payload header, at the start of the payload
section of the RTP packet. The RTP packet also includes space for
a header extension [RFC5285]; this can be used to transport
payload format independent metadata, for example, an SMPTE time
code for the packet [RFC5484]. The RTP header extensions are not
intended to carry headers that relate to a particular payload
format, and must not contain information needed in order to decode
the payload.
The remaining fields do not commonly influence the RTP payload
format. The padding bit is worth clarifying as it indicates that one
or more bytes are appended after the RTP payload. This padding must
be removed by a receiver before payload format processing can occur.
Thus, it is completely separate from any padding that may occur
within the payload format itself.
3.3.3. RTP Multiplexing
RTP has three multiplexing points that are used for different purposes. A proper understanding of this is important to correctly use them. The first one is separation of RTP streams of different types or usages, which is accomplished using different RTP sessions. So, for example, in the common multimedia session with audio and video, RTP commonly multiplexes audio and video in different RTP sessions. To achieve this separation, transport-level functionalities are used, normally UDP port numbers. Different RTP sessions can also be used to realize layered scalability as it allows a receiver to select one or more layers for multicast RTP sessions simply by joining the multicast groups over which the desired layers are transported. This separation also allows different Quality of Service (QoS) to be applied to different media types. Use of multiple transport flows has potential issues due to NAT and firewall traversal. The choices how one applies RTP sessions as well as transport flows can affect the transport properties an RTP media stream experiences. The next multiplexing point is separation of different RTP streams within an RTP session. Here, RTP uses the SSRC to identify individual sources of RTP streams. An example of individual media sources would be the capture of different microphones that are carried in an RTP session for audio, independently of whether they are connected to the same host or different hosts. There also exist cases where a single media source, is transmitted using multiple RTP streams. For each SSRC, a unique RTP sequence number and timestamp space is used. The third multiplexing point is the RTP header payload type field. The payload type identifies what format the content in the RTP payload has. This includes different payload format configurations, different codecs, and also usage of robustness mechanisms like the one described in RFC 2198 [RFC2198].3.3.4. RTP Synchronization
There are several types of synchronization, and we will here describe how RTP handles the different types: Intra media: The synchronization within a media stream from a synchronization source (SSRC) is accomplished using the RTP timestamp field. Each RTP packet carries the RTP timestamp, which specifies the position in time of the media payload contained in this packet relative to the content of other RTP packets in the same RTP stream (i.e., a given SSRC). This is especially useful
in cases of discontinuous transmissions. Discontinuities can be
caused by network conditions; when extensive losses occur the RTP
timestamp tells the receiver how much later than previously
received media the present media should be played out.
Inter-media: Applications commonly have a desire to use several
media sources, possibly of different media types, at the same
time. Thus, there exists a need to synchronize different media
from the same endpoint. This puts two requirements on RTP: the
possibility to determine which media are from the same endpoint
and if they should be synchronized with each other and the
functionality to facilitate the synchronization itself.
The first step in inter-media synchronization is to determine which
SSRCs in each session should be synchronized with each other. This
is accomplished by comparing the CNAME fields in the RTCP source
description (SDES) packets. SSRCs with the same CNAME sent in any of
multiple RTP sessions can be synchronized.
The actual RTCP mechanism for inter-media synchronization is based on
the idea that each RTP stream provides a position on the media
specific time line (measured in RTP timestamp ticks) and a common
reference time line. The common reference time line is expressed in
RTCP as a wall-clock time in the Network Time Protocol (NTP) format.
It is important to notice that the wall-clock time is not required to
be synchronized between hosts, for example, by using NTP [RFC5905].
It can even have nothing at all to do with the actual time; for
example, the host system's up-time can be used for this purpose. The
important factor is that all media streams from a particular source
that are being synchronized use the same reference clock to derive
their relative RTP timestamp time scales. The type of reference
clock and its timebase can be signaled using RTP Clock Source
Signaling [RFC7273].
Figure 1 illustrates how if one receives RTCP Sender Report (SR)
packet P1 for one RTP stream and RTCP SR packet P2 for the other RTP
stream, then one can calculate the corresponding RTP timestamp values
for any arbitrary point in time T. However, to be able to do that,
it is also required to know the RTP timestamp rates for each RTP
stream currently used in the sessions.
TS1 --+---------------+-------> | | P1 | | | NTP ---+-----+---------T------> | | P2 | | | TS2 ---------+---------+---X--> Figure 1: RTCP Synchronization Assume that medium 1 uses an RTP timestamp clock rate of 16 kHz, and medium 2 uses a clock rate of 90 kHz. Then, TS1 and TS2 for point T can be calculated in the following way: TS1(T) = TS1(P1) + 16000 * (NTP(T)-NTP(P1)) and TS2(T) = TS2(P2) + 90000 * (NTP(T)-NTP(P2)). This calculation is useful as it allows the implementation to generate a common synchronization point for which all time values are provided (TS1(T), TS2(T) and T). So, when one wishes to calculate the NTP time that the timestamp value present in packet X corresponds to, one can do that in the following way: NTP(X) = NTP(T) + (TS2(X) - TS2(T))/90000. Improved signaling for layered codecs and fast tune-in have been specified in "Rapid Synchronization for RTP Flows" [RFC6051]. Leap seconds are extra seconds added or seconds removed to keep our clocks in sync with the earth's rotation. Adding or removing seconds can impact the reference clock as discussed in "RTP and Leap Seconds" [RFC7164]; also, in cases where the RTP timestamp values are derived using the wall clock during the leap second event, errors can occur. Implementations need to consider leap seconds and should consider the recommendations in [RFC7164].3.4. Signaling Aspects
RTP payload formats are used in the context of application signaling protocols such as SIP [RFC3261] using the Session Description Protocol (SDP) [RFC4566] with Offer/Answer [RFC3264], RTSP [RFC7826], or the Session Announcement Protocol [RFC2974]. These examples all use out-of-band signaling to indicate which type of RTP streams are desired to be used in the session and how they are configured. To be able to declare or negotiate the media format and RTP payload packetization, the payload format must be given an identifier. In addition to the identifier, many payload formats also have the need to signal further configuration information out-of-band for the RTP payloads prior to the media transport session.
The above examples of session-establishing protocols all use SDP, but other session description formats may be used. For example, there was discussion of a new XML-based session description format within the IETF (SDP-NG). In the end, the proposal did not get beyond draft protocol specification because of the enormous installed base of SDP implementations. However, to avoid locking the usage of RTP to SDP based out-of-band signaling, the payload formats are identified using a separate definition format for the identifier and associated parameters. That format is the media type.3.4.1. Media Types
Media types [RFC6838] are identifiers originally created for identifying media formats included in email. In this usage, they were known as MIME types, where the expansion of the MIME acronym includes the word "mail". The term "media type" was introduced to reflect a broader usage, which includes HTTP [RFC7231], Message Session Relay Protocol (MSRP) [RFC4975], and many other protocols to identify arbitrary content carried within the protocols. Media types also provide a media hierarchy that fits RTP payload formats well. Media type names are of two parts and consist of content type and sub-type separated with a slash, e.g., 'audio/PCMA' or 'video/ h263-2000'. It is important to choose the correct content-type when creating the media type identifying an RTP payload format. However, in most cases, there is little doubt what content type the format belongs to. Guidelines for choosing the correct media type and registration rules for media type names are provided in "Media Type Specifications and Registration Procedures" [RFC6838]. The additional rules for media types for RTP payload formats are provided in "Media Type Registration of RTP Payload Formats" [RFC4855]. Registration of the RTP payload name is something that is required to avoid name collision in the future. Note that "x-" names are not suitable for any documented format as they have the same problem with name collision and can't be registered. The list of already- registered media types can be found at <https://www.iana.org/assignments/media-types/media-types.xhtml>. Media types are allowed any number of parameters, which may be required or optional for that media type. They are always specified on the form "name=value". There exist no restrictions on how the value is defined from the media type's perspective, except that parameters must have a value. However, the usage of media types in
SDP, etc., has resulted in the following restrictions that need to be
followed to make media types usable for RTP-identifying payload
formats:
1. Arbitrary binary content in the parameters is allowed, but it
needs to be encoded so that it can be placed within text-based
protocols. Base64 [RFC4648] is recommended, but for shorter
content Base16 [RFC4648] may be more appropriate as it is simpler
to interpret for humans. This needs to be explicitly stated when
defining a media type parameter with binary values.
2. The end of the value needs to be easily found when parsing a
message. Thus, parameter values that are continuous and not
interrupted by common text separators, such as space and
semicolon characters, are recommended. If that is not possible,
some type of escaping should be used. Usage of quote (") is
recommended; do not forget to provide a method of encoding any
character used for quoting inside the quoted element.
3. A common representation form for the media type and its
parameters is on a single line. In that case, the media type is
followed by a semicolon-separated list of the parameter value
pairs, e.g.:
audio/amr octet-align=0; mode-set=0,2,5,7; mode-change-period=2
3.4.2. Mapping to SDP
Since SDP [RFC4566] is so commonly used as an out-of-band signaling
protocol, a mapping of the media type into SDP exists. The details
on how to map the media type and its parameters into SDP are
described in [RFC4855]. However, this is not sufficient to explain
how certain parameters must be interpreted, for example, in the
context of Offer/Answer negotiation [RFC3264].
3.4.2.1. 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; instead, it points out the need to define how parameters
behave. To make things even more complex, the direction of media
within a session has an impact on these rules, so that some cases may
require separate descriptions for RTP streams that are send-only,
receive-only, or both sent and received as identified by the SDP
attributes a=sendonly, a=recvonly, and a=sendrecv. In addition, the
usage of multicast adds further limitations as the same RTP stream is
delivered to all participants. If those multicast-imposed restrictions are too limiting for unicast, then separate rules for unicast and multicast will be required. The simplest and most common O/A interpretation is that a parameter is defined to be declarative; i.e., the SDP Offer/Answer sending agent can declare a value and that has no direct impact on the other agent's values. This declared value applies to all media that are going to be sent to the declaring entity. For example, most video codecs have a level parameter that tells the other participants the highest complexity the video decoder supports. The level parameter can be declared independently by two participants in a unicast session as it will be the media sender's responsibility to transmit a video stream that fulfills the limitation the other side has declared. However, in multicast, it will be necessary to send a stream that follows the limitation of the weakest receiver, i.e., the one that supports the lowest level. To simplify the negotiation in these cases, it is common to require any answerer to a multicast session to take a yes or no approach to parameters. A "negotiated" parameter is a different case, for which both sides need to agree on its value. Such a parameter requires the answerer to either accept it as it is offered or remove the payload type the parameter belonged to from its answer. The removal of the payload type from the answer indicates to the offerer the lack of support for the parameter values presented. An unfortunate implication of the need to use complete payload types to indicate each possible configuration so as to maximize the chances of achieving interoperability, is that the number of necessary payload types can quickly grow large. This is one reason to limit the total number of sets of capabilities that may be implemented. The most problematic type of parameters are those that relate to the media the entity sends. They do not really fit the O/A model, but can be shoehorned in. Examples of such parameters can be found in the H.264 video codec's payload format [RFC6184], where the name of all parameters with this property starts with "sprop-". The issue with these parameters is that they declare properties for a RTP stream that the other party may not accept. The best one can make of the situation is to explain the assumption that the other party will accept the same parameter value for the media it will receive as the offerer of the session has proposed. If the answerer needs to change any declarative parameter relating to streams it will receive, then the offerer may be required to make a new offer to update the parameter values for its outgoing RTP stream.
Another issue to consider is the send-only RTP streams in offers. Parameters that relate to what the answering entity accepts to receive have no meaning other than to provide a template for the answer. It is worth pointing out in the specification that these really provide a set of parameter values that the sender recommends. Note that send-only streams in answers will need to indicate the offerer's parameters to ensure that the offerer can match the answer to the offer. A further issue with Offer/Answer that complicates things is that the answerer is allowed to renumber the payload types between offer and answer. This is not recommended, but allowed for support of gateways to the ITU conferencing suite. This means that it must be possible to bind answers for payload types to the payload types in the offer even when the payload type number has been changed, and some of the proposed payload types have been removed. This binding must normally be done by matching the configurations originally offered against those in the answer. This may require specification in the payload format of which parameters that constitute a configuration, for example, as done in Section 8.2.2 of the H.264 RTP Payload format [RFC6184], which states: "The parameters identifying a media format configuration for H.264 are profile-level-id and packetization-mode".3.4.2.2. Declarative Usage in RTSP and SAP
SAP (Session Announcement Protocol) [RFC2974] was experimentally used for announcing multicast sessions. Similar but better protocols are using SDP in a declarative style to configure multicast-based applications. Independently of the usage of Source-Specific Multicast (SSM) [RFC3569] or Any-Source Multicast (ASM), the SDP provided by these configuration delivery protocols applies to all participants. All media that is sent to the session must follow the RTP stream definition as specified by the SDP. This enables everyone to receive the session if they support the configuration. Here, SDP provides a one-way channel with no possibility to affect the configuration that the session creator has decided upon. Any RTP payload format that requires parameters for the send direction and that needs individual values per implementation or instance will fail in a SAP session for a multicast session allowing anyone to send. Real-Time Streaming Protocol (RTSP) [RFC7826] allows the negotiation of transport parameters for RTP streams that are part of a streaming session between a server and client. RTSP has divided the transport parameters from the media configuration. SDP is commonly used for media configuration in RTSP and is sent to the client prior to session establishment, either through use of the DESCRIBE method or
by means of an out-of-band channel like HTTP, email, etc. The SDP is used to determine which RTP streams and what formats are being used prior to session establishment. Thus, both SAP and RTSP use SDP to configure receivers and senders with a predetermined configuration for a RTP stream including the payload format and any of its parameters. All parameters are used in a declarative fashion. This can result in different treatment of parameters between Offer/Answer and declarative usage in RTSP and SAP. Any such difference will need to be spelled out by the payload format specification.3.5. Transport Characteristics
The general channel characteristics that RTP flows experience are documented in Section 3 of "Guidelines for Writers of RTP Payload Format Specifications" [RFC2736]. The discussion below provides additional information.3.5.1. Path MTU
At the time of writing, the most common IP Maximum Transmission Unit (MTU) in commonly deployed link layers is 1500 bytes (Ethernet data payload). However, there exist both links with smaller MTUs and links with much larger MTUs. An example for links with small MTU size is older generation cellular links. Certain parts of the Internet already support an IP MTU of 8000 bytes or more, but these are limited islands. The most likely places to find MTUs larger than 1500 bytes are within enterprise networks, university networks, data centers, storage networks, and over high capacity (10 Gbps or more) links. There is a slow, ongoing evolution towards larger MTU sizes. However, at the same time, it has become common to use tunneling protocols, often multiple ones, whose overhead when added together can shrink the MTU significantly. Thus, there exists a need both to consider limited MTUs as well as enable support of larger MTUs. This should be considered in the design, especially in regard to features such as aggregation of independently decodable data units.3.5.2. Different Queuing Algorithms
Routers and switches on the network path between an IP sender and a particular receiver can exhibit different behaviors affecting the end-to-end characteristics. One of the more important aspects of this is queuing behavior. Routers and switches have some amount of queuing to handle temporary bursts of data that designated to leave the switch or router on the same egress link. A queue, when not empty, results in an increased path delay.
The implementation of the queuing affects the delay and also how congestion signals (Explicit Congestion Notification (ECN) [RFC6679] or packet drops) are provided to the flow. The other aspects are if the flow shares the queue with other flows and how the implementation affects the flow interaction. This becomes important, for example, when real-time flows interact with long-lived TCP flows. TCP has a built-in behavior in its congestion control that strives to fill the buffer; thus, all flows sharing the buffer experienced the delay build up. A common, but quite poor, queue-handling mechanism is tail-drop, i.e., only drop packets when the incoming packet doesn't fit in the queue. If a bad queuing algorithm is combined with too much queue space, the queuing time can grow to be very significant and can even become multiple seconds. This is called "bufferbloat" [BLOAT]. Active Queue Management (AQM) is a term covering mechanisms that try to do something smarter by actively managing the queue, for example, sending congestion signals earlier by dropping packets earlier in the queue. The behavior also affects the flow interactions. For example, Random Early Detection (RED) [RED] selects which packet(s) to drop randomly. This gives flows that have more packets in the queue a higher probability to experience the packet loss (congestion signal). There is ongoing work in the IETF WG AQM to find suitable mechanisms to recommend for implementation and reduce the use of tail-drop.3.5.3. Quality of Service
Using best-effort Internet has no guarantees for the path's properties. QoS mechanisms are intended to provide the possibility to bound the path properties. Where Diffserv [RFC2475] markings affect the queuing and forwarding behaviors of routers, the mechanism provides only statistical guarantees and care in how much marked packets of different types that are entering the network. Flow-based QoS, like IntServ [RFC1633], has the potential for stricter guarantees as the properties are agreed on by each hop on the path, at the cost of per-flow state in the network.4. Standardization Process for an RTP Payload Format
This section discusses the recommended process to produce an RTP payload format in the described venues. This is to document the best current practice on how to get a well-designed and specified payload format as quickly as possible. For specifications that are defined by standards bodies other than the IETF, the primary milestone is the registration of the media type for the RTP payload format. For
proprietary media formats, the primary goal depends on whether interoperability is desired at the RTP level. However, there is also the issue of ensuring best possible quality of any specification.4.1. IETF
For all standardized media formats, it is recommended that the payload format be specified in the IETF. The main reason is to provide an openly available RTP payload format specification that has been reviewed by people experienced with RTP payload formats. At the time of writing, this work is done in the PAYLOAD Working Group (WG), but that may change in the future.4.1.1. Steps from Idea to Publication
There are a number of steps that an RTP payload format should go through from the initial idea until it is published. This also documents the process that the PAYLOAD WG applies when working with RTP payload formats. Idea: Determine the need for an RTP payload format as an IETF specification. Initial effort: Using this document as a guideline, one should be able to get started on the work. If one's media codec doesn't fit any of the common design patterns or one has problems understanding what the most suitable way forward is, then one should contact the PAYLOAD WG and/or the WG Chairs. The goal of this stage is to have an initial individual draft. This draft needs to focus on the introductory parts that describe the real- time media format and the basic idea on how to packetize it. Not all the details are required to be filled in. However, the security chapter is not something that one should skip, even initially. From the start, it is important to consider any serious security risks that need to be solved. The first step is completed when one has a draft that is sufficiently detailed for a first review by the WG. The less confident one is of the solution, the less work should be spent on details; instead, concentrate on the codec properties and what is required to make the packetization work. Submission of the first version: When one has performed the above, one submits the draft as an individual draft (https://datatracker.ietf.org/submit/). This can be done at any time, except for a period prior to an IETF meeting (see important dates related to the next IETF meeting for draft submission cutoff date). When the Internet-Draft announcement has been sent out on
the draft announcement list
(https://www.ietf.org/mailman/listinfo/I-D-Announce), forward it
to the PAYLOAD WG (https://www.ietf.org/mailman/listinfo/payload)
and request that it be reviewed. In the email, outline any issues
the authors currently have with the design.
Iterative improvements: Taking the feedback received into account,
one updates the draft and tries resolve issues. New revisions of
the draft can be submitted at any time (again except for a short
period before meetings). It is recommended to submit a new
version whenever one has made major updates or has new issues that
are easiest to discuss in the context of a new draft version.
Becoming a WG document: Given that the definition of RTP payload
formats is part of the PAYLOAD WG's charter, RTP payload formats
that are going to be published as Standards Track RFCs need to
become WG documents. Becoming a WG document means that the WG
Chairs or an appointed document shepherd are responsible for
administrative handling, for example, issuing publication
requests. However, be aware that making a document into a WG
document changes the formal ownership and responsibility from the
individual authors to the WG. The initial authors normally
continue being the document editors, unless unusual circumstances
occur. The PAYLOAD WG accepts new RTP payload formats based on
their suitability and document maturity. The document maturity is
a requirement to ensure that there are dedicated document editors
and that there exists a good solution.
Iterative improvements: The updates and review cycles continue until
the draft has reached the level of maturity suitable for
publication. The authors are responsible for judging when the
document is ready for the next step, most likely WG Last Call, but
they can ask the WG chairs or Shepherd.
WG Last Call: A WG Last Call of at least two weeks is always
performed for payload formats in the PAYLOAD WG (see Section 7.4
of [RFC2418]). The authors request WG Last Call for a draft when
they think it is mature enough for publication. The WG Chairs or
shepherd perform a review to check if they agree with the authors'
assessment. If the WG Chairs or shepherd agree on the maturity,
the WG Last Call is announced on the WG mailing list. If there
are issues raised, these need to be addressed with an updated
draft version. For any more substantial changes to the draft, a
new WG Last Call is announced for the updated version. Minor
changes, like editorial fixes, can be progressed without an
additional WG Last Call.
Publication requested: For WG documents, the WG Chairs or shepherd
request publication of the draft after it has passed WG Last Call.
After this, the approval and publication process described in BCP
9 [BCP9] is performed. The status after the publication has been
requested can be tracked using the IETF Datatracker [TRACKER].
Documents do not expire as they normally do after publication has
been requested, so authors do not have to issue keep-alive
updates. In addition, any submission of document updates requires
the approval of WG Chair(s). The authors are commonly asked to
address comments or issues raised by the IESG. The authors also
do one last review of the document immediately prior to its
publication as an RFC to ensure that no errors or formatting
problems have been introduced during the publication process.
4.1.2. WG Meetings
WG meetings are for discussing issues, not presentations. This means
that most RTP payload formats should never need to be discussed in a
WG meeting. RTP payload formats that would be discussed are either
those with controversial issues that failed to be resolved on the
mailing list or those including new design concepts worth a general
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 the IETF. WG meetings
may become necessary only if the draft gets stuck in a serious debate
that cannot easily be resolved.
4.1.3. Draft Naming
To simplify the work of the PAYLOAD WG Chairs and WG members, a
specific Internet-Draft file-naming convention shall be used for RTP
payload formats. Individual submissions shall be named using the
template: draft-<lead author family name>-payload-rtp-<descriptive
name>-<version>. The WG documents shall be named according to this
template: draft-ietf-payload-rtp-<descriptive name>-<version>. The
inclusion of "payload" in the draft file name ensures that the search
for "payload-" will find all PAYLOAD-related drafts. Inclusion of
"rtp" tells us that it is an RTP payload format draft. The
descriptive name should be as short as possible while still
describing what the payload format is for. It is recommended to use
the media format or codec abbreviation. Please note that the version
must start at 00 and is increased by one for each submission to the
IETF secretary of the draft. No version numbers may be skipped. For
more details on draft naming, please see Section 7 of [ID-GUIDE].
4.1.4. Writing Style
When writing an Internet-Draft for an RTP payload format, one should observe some few considerations (that may be somewhat divergent from the style of other IETF documents and/or the media coding spec's author group may use): Include Motivations: In the IETF, it is common to include the motivation for why a particular design or technical path was chosen. These are not long statements: a sentence here and there explaining why suffice. Use the Defined Terminology: There exists defined terminology both in RTP and in the media codec specification for which the RTP payload format is designed. A payload format specification needs to use both to make clear the relation of features and their functions. It is unwise to introduce or, worse, use without introduction, terminology that appears to be more accessible to average readers but may miss certain nuances that the defined terms imply. An RTP payload format author can assume the reader to be reasonably familiar with the terminology in the media coding specification. Keeping It Simple: The IETF has a history of specifications that are focused on their main usage. Historically, some RTP payload formats have a lot of modes and features, while the actual deployments have only included the most basic features that had very clear requirements. Time and effort can be saved by focusing on only the most important use cases and keeping the solution simple. An extension mechanism should be provided to enable backward-compatible extensions, if that is an organic fit. Normative Requirements: When writing specifications, there is commonly a need to make it clear when something is normative and at what level. In the IETF, the most common method is to use "Key words for use in RFCs to Indicate Requirement Levels" [RFC2119], which defines the meaning of "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL".
4.1.5. How to Speed Up the Process
There a number of ways to lose a lot of time in the above process. This section discusses what to do and what to avoid. o Do not update the draft only for the meeting deadline. An update to each meeting automatically limits the draft to three updates per year. Instead, ignore the meeting schedule and publish new versions as soon as possible. o Try to avoid requesting reviews when people are busy, like the few weeks before a meeting. It is actually more likely that people have time for them directly after a meeting. o Perform draft updates quickly. A common mistake is that the authors let the draft slip. By performing updates to the draft text directly after getting resolution on an issue, things speed up. This minimizes the delay that the author has direct control over. The time taken for reviews, responses from Area Directors and WG Chairs, etc., can be much harder to speed up. o Do not fail to take human nature into account. It happens that people forget or need to be reminded about tasks. Send a kind reminder to the people you are waiting for if things take longer than expected. Ask people to estimate when they expect to fulfill the requested task. o Ensure there is enough review. It is common that documents take a long time and many iterations because not enough review is performed in each iteration. To improve the amount of review you get on your own document, trade review time with other document authors. Make a deal with some other document author that you will review their draft if they review yours. Even inexperienced reviewers can help with language, editorial, or clarity issues. Also, try approaching the more experienced people in the WG and getting them to commit to a review. The WG Chairs cannot, even if desirable, be expected to review all versions. Due to workload, the Chairs may need to concentrate on key points in a draft evolution like checking on initial submissions, a draft's readiness to become a WG document, or its readiness for WG Last 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 specification, a second review should be scheduled. In that pass, the quality can be assessed; hopefully, no updates will be needed. Using this procedure can avoid both conflicting definitions and serious mistakes, like breaking certain aspects of the RTP model. RTP payload media types may be registered in the standards tree by other standards bodies. The requirements on the organization are outlined in the media types registration documents [RFC4855] and [RFC6838]). This registration requires a request to the IESG, which ensures that the filled-in registration template is acceptable. To avoid last-minute problems with these registrations the registration template must be sent for review both to the PAYLOAD WG and the media types list (ietf-types@iana.org) and is something that should be included in the IETF reviews of the payload format specification.4.3. Proprietary and Vendor Specific
Proprietary RTP payload formats are commonly specified when the real- time media format is proprietary and not intended to be part of any standardized system. However, there are reasons why also proprietary formats should be correctly documented and registered: o Usage in a standardized signaling environment, such as SIP/SDP. RTP needs to be configured with the RTP profiles, payload formats, and their payload types being used. To accomplish this, it is desirable to have registered media type names to ensure that the names do not collide with those of other formats. o Sharing with business partners. As RTP payload formats are used for communication, situations often arise where business partners would like to support a proprietary format. Having a well-written specification of the format will save time and money for both parties, as interoperability will be much easier to accomplish. o To ensure interoperability between different implementations on different platforms. To avoid name collisions, there is a central registry keeping track of the registered media type names used by different RTP payload formats. When it comes to proprietary formats, they should be registered in the vendor's own tree. All vendor-specific registrations use sub-type names that start with "vnd.<vendor-name>". Names in the vendor's own tree are not required to be registered with IANA. However, registration [RFC6838] is recommended if the media type is used at all in public environments.
If interoperability at the RTP level is desired, a payload type specification should be standardized in the IETF following the process described above. The IETF does not require full disclosure of the codec when defining an RTP payload format to carry that codec, but a description must be provided that is sufficient to allow the IETF to judge whether the payload format is well designed. The media type identifier assigned to a standardized payload format of this sort will lie in the standards tree rather than the vendor tree.4.4. Joint Development of Media Coding Specification and RTP Payload Format
In the last decade, there have been a few cases where the media codec and the associated RTP payload format have been developed concurrently and jointly. Developing the two specs not only concurrently but also jointly, in close cooperation with the group developing the media codec, allows one to leverage the benefits joint source/channel coding can provide. Doing so has historically resulted in well-performing payload formats and in success of both the media coding specification and associated RTP payload format. Insofar, whenever the opportunity presents it, it may be useful to closely keep the media coding group in the loop (through appropriate liaison means whatever those may be) and influence the media coding specification to be RTP friendly. One example for such a media coding specification is H.264, where the RTP payload header co-serves as the H.264 NAL unit header and vice versa, and is documented in both specifications.
(page 31 continued on part 3)
