| Rfc | 7657 |
|---|
| Title | Differentiated Services (Diffserv) and Real-Time Communication |
|---|
| Author | D.
Black, Ed., P. Jones |
|---|
| Date | November 2015 |
|---|
| Format: | TXT, HTML |
|---|
| Status: | INFORMATIONAL |
|---|
|
Internet Engineering Task Force (IETF) D. Black, Ed.
Request for Comments: 7657 EMC
Category: Informational P. Jones
ISSN: 2070-1721 Cisco
November 2015
Differentiated Services (Diffserv) and Real-Time Communication
Abstract
This memo describes the interaction between Differentiated Services
(Diffserv) network quality-of-service (QoS) functionality and real-
time network communication, including communication based on the
Real-time Transport Protocol (RTP). Diffserv is based on network
nodes applying different forwarding treatments to packets whose IP
headers are marked with different Diffserv Codepoints (DSCPs).
WebRTC applications, as well as some conferencing applications, have
begun using the Session Description Protocol (SDP) bundle negotiation
mechanism to send multiple traffic streams with different QoS
requirements using the same network 5-tuple. The results of using
multiple DSCPs to obtain different QoS treatments within a single
network 5-tuple have transport protocol interactions, particularly
with congestion control functionality (e.g., reordering). In
addition, DSCP markings may be changed or removed between the traffic
source and destination. This memo covers the implications of these
Diffserv aspects for real-time network communication, including
WebRTC.
Status of This Memo
This document is not an Internet Standards Track specification; it is
published for informational purposes.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Not all documents
approved by the IESG are a candidate for any level of Internet
Standard; see Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc7657.
Copyright Notice
Copyright (c) 2015 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Real-Time Communications . . . . . . . . . . . . . . . . . . 3
2.1. RTP Background . . . . . . . . . . . . . . . . . . . . . 4
2.2. RTP Multiplexing . . . . . . . . . . . . . . . . . . . . 6
3. Differentiated Services (Diffserv) . . . . . . . . . . . . . 7
3.1. Diffserv Per-Hop Behaviors (PHBs) . . . . . . . . . . . . 10
3.2. Traffic Classifiers and DSCP Remarking . . . . . . . . . 10
4. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 12
5. Diffserv Interactions . . . . . . . . . . . . . . . . . . . . 13
5.1. Diffserv, Reordering, and Transport Protocols . . . . . . 13
5.2. Diffserv, Reordering, and Real-Time Communication . . . . 15
5.3. Drop Precedence and Transport Protocols . . . . . . . . . 16
5.4. Diffserv and RTCP . . . . . . . . . . . . . . . . . . . . 17
6. Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . 18
7. Security Considerations . . . . . . . . . . . . . . . . . . . 19
8. References . . . . . . . . . . . . . . . . . . . . . . . . . 20
8.1. Normative References . . . . . . . . . . . . . . . . . . 20
8.2. Informative References . . . . . . . . . . . . . . . . . 22
Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . . 26
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 26
1. Introduction
This memo describes the interactions between Differentiated Services
(Diffserv) network quality-of-service (QoS) functionality [RFC2475]
and real-time network communication, including communication based on
the Real-time Transport Protocol (RTP) [RFC3550]. Diffserv is based
on network nodes applying different forwarding treatments to packets
whose IP headers are marked with different Diffserv Codepoints
(DSCPs) [RFC2474]. In the past, distinct RTP streams have been sent
over different transport-level flows, sometimes multiplexed with the
RTP Control Protocol (RTCP). WebRTC applications, as well as some
conferencing applications, are now using the Session Description
Protocol (SDP) [RFC4566] bundle negotiation mechanism [SDP-BUNDLE] to
send multiple traffic streams with different QoS requirements using
the same network 5-tuple. The results of using multiple DSCPs to
obtain different QoS treatments within a single network 5-tuple have
transport protocol interactions, particularly with congestion control
functionality (e.g., reordering). In addition, DSCP markings may be
changed or removed between the traffic source and destination. This
memo covers the implications of these Diffserv aspects for real-time
network communication, including WebRTC traffic [WEBRTC-OVERVIEW].
The memo is organized as follows. Background is provided in
Section 2 on real-time communications and Section 3 on Differentiated
Services. Section 4 describes some examples of Diffserv usage with
real-time communications. Section 5 explains how use of Diffserv
features interacts with both transport and real-time communications
protocols and Section 6 provides guidance on Diffserv feature usage
to control undesired interactions. Security considerations are
discussed in Section 7.
2. Real-Time Communications
Real-time communications enables communication in real time over an
IP network using voice, video, text, content sharing, etc. It is
possible to use more than one of these modes concurrently to provide
a rich communication experience.
A simple example of real-time communications is a voice call placed
over the Internet where an audio stream is transmitted in each
direction between two users. A more complex example is an immersive
videoconferencing system that has multiple video screens, multiple
cameras, multiple microphones, and some means of sharing content.
For such complex systems, there may be multiple media and non-media
streams transmitted via a single IP address and port or via multiple
IP addresses and ports.
2.1. RTP Background
The most common protocol used for real-time media is RTP [RFC3550].
RTP defines a common encapsulation format and handling rules for
real-time data transmitted over the Internet. Unfortunately, RTP
terminology usage has been inconsistent. For example, RFC 7656
[RFC7656] on RTP terminology observes that:
RTP [RFC3550] uses media stream, audio stream, video stream, and a
stream of (RTP) packets interchangeably, which are all RTP
streams.
Terminology in this memo is based on that RTP terminology document
with the following terms being of particular importance (see that
terminology document for full definitions):
Source Stream: A reference clock synchronized, time progressing,
digital media stream.
RTP Stream: A stream of RTP packets containing media data, which may
be source data or redundant data. The RTP stream is identified by
an RTP synchronization source (SSRC) belonging to a particular RTP
session. An RTP stream may be a secured RTP stream when RTP-based
security is used.
In addition, this memo follows [RFC3550] in using the term "SSRC" to
designate both the identifier of an RTP stream and the entity that
sends that RTP stream.
Media encoding and packetization of a source stream results in a
source RTP stream plus zero or more redundancy RTP streams that
provide resilience against loss of packets from the source RTP stream
[RFC7656]. Redundancy information may also be carried in the same
RTP stream as the encoded source stream, e.g., see Section 7.2 of
[RFC5109]. With most applications, a single media type (e.g., audio)
is transmitted within a single RTP session. However, it is possible
to transmit multiple, distinct source streams over the same RTP
session as one or more individual RTP streams. This is referred to
as RTP multiplexing. In addition, an RTP stream may contain multiple
source streams, e.g., components or programs in an MPEG Transport
Stream [H.221].
The number of source streams and RTP streams in an overall real-time
interaction can be surprisingly large. In addition to a voice source
stream and a video source stream, there could be separate source
streams for each of the cameras or microphones on a videoconferencing
system. As noted above, there might also be separate redundancy RTP
streams that provide protection to a source RTP stream, using
techniques such as forward error correction. Another example is
simulcast transmission, where a video source stream can be
transmitted as high resolution and low resolution RTP streams at the
same time. In this case, a media processing function might choose to
send one or both RTP streams onward to a receiver based on bandwidth
availability or who the active speaker is in a multipoint conference.
Lastly, a transmitter might send the same media content concurrently
as two RTP streams using different encodings (e.g., video encoded as
VP8 [RFC6386] in parallel with H.264 [H.264]) to allow a media
processing function to select a media encoding that best matches the
capabilities of the receiver.
For the WebRTC protocol suite [WEBRTC-TRANSPORTS], an individual
source stream is a MediaStreamTrack, and a MediaStream contains one
or more MediaStreamTracks [W3C.WD-mediacapture-streams-20130903]. A
MediaStreamTrack is transmitted as a source RTP stream plus zero or
more redundant RTP streams, so a MediaStream that consists of one
MediaStreamTrack is transmitted as a single source RTP stream plus
zero or more redundant RTP streams. For more information on use of
RTP in WebRTC, see [RTP-USAGE].
RTP is usually carried over a datagram protocol, such as UDP
[RFC768], UDP-Lite [RFC3828], or the Datagram Congestion Control
Protocol (DCCP) [RFC4340]; UDP is most commonly used, but a non-
datagram protocol (e.g., TCP [RFC793]) may also be used. Transport
protocols other than UDP or UDP-Lite may also be used to transmit
real-time data or near-real-time data. For example, the Stream
Control Transmission Protocol (SCTP) [RFC4960] can be utilized to
carry application-sharing or whiteboarding information as part of an
overall interaction that includes real-time media. These additional
transport protocols can be multiplexed with an RTP session via UDP
encapsulation, thereby using a single pair of UDP ports.
The WebRTC protocol suite encompasses a number of forms of
multiplexing:
1. Individual source streams are carried in one or more individual
RTP streams. These RTP streams can be multiplexed onto a single
transport-layer flow or sent as separate transport-layer flows.
This memo only considers the case where the RTP streams are to be
multiplexed onto a single transport-layer flow, forming a single
RTP session as described in [RFC3550];
2. RTCP (see [RFC3550]) may be multiplexed onto the same transport-
layer flow as the RTP streams with which it is associated, as
described in [RFC5761], or it may be sent on a separate
transport-layer flow;
3. An RTP session could be multiplexed with a single SCTP
association over Datagram Transport Layer Security (DTLS) and
with both Session Traversal Utilities for NAT (STUN) [RFC5389]
and TURN [RFC5766] traffic into a single transport-layer flow as
described in [RFC5764] with the updates in [SRTP-DTLS]. The STUN
[RFC5389] and Traversal Using Relays around NAT (TURN) [RFC5766]
protocols provide NAT/FW (Network Address Translator / Firewall)
traversal and port mapping.
The resulting transport-layer flow is identified by a network
5-tuple, i.e., a combination of two IP addresses (source and
destination), two ports (source and destination), and the transport
protocol used (e.g., UDP). SDP bundle negotiation restrictions
[SDP-BUNDLE] limit WebRTC to using at most a single DTLS session per
network 5-tuple. In contrast to WebRTC use of a single SCTP
association with DTLS, multiple SCTP associations can be directly
multiplexed over a single UDP 5-tuple as specified in [RFC6951].
The STUN and TURN protocols were originally designed to use UDP as a
transport; however, TURN has been extended to use TCP as a transport
for situations in which UDP does not work [RFC6062]. When TURN
selects use of TCP, the entire real-time communications session is
carried over a single TCP connection (i.e., 5-tuple).
For IPv6, addition of the flow label [RFC6437] to network 5-tuples
results in network 6-tuples (or 7-tuples for bidirectional flows),
but in practice, use of a flow label is unlikely to result in a
finer-grain traffic subset than the corresponding network 5-tuple
(e.g., the flow label is likely to represent the combination of two
ports with use of the UDP protocol). For that reason, discussion in
this document focuses on UDP 5-tuples.
2.2. RTP Multiplexing
Section 2.1 explains how source streams can be multiplexed in a
single RTP session, which can in turn be multiplexed over UDP with
packets generated by other transport protocols. This section
provides background on why this level of multiplexing is desirable.
The rationale in this section applies both to multiplexing of source
streams in a single RTP session and multiplexing of an RTP session
with traffic from other transport protocols via UDP encapsulation.
Multiplexing reduces the number of ports utilized for real-time and
related communication in an overall interaction. While a single
endpoint might have plenty of ports available for communication, this
traffic often traverses points in the network that are constrained on
the number of available ports or whose performance degrades as the
number of ports in use increases. A good example is a NAT/FW device
sitting at the network edge. As the number of simultaneous protocol
sessions increases, so does the burden placed on these devices to
provide port mapping.
Another reason for multiplexing is to help reduce the time required
to establish bidirectional communication. Since any two
communicating users might be situated behind different NAT/FW
devices, it is necessary to employ techniques like STUN and TURN
along with Interactive Connectivity Establishment (ICE) [RFC5245] to
get traffic to flow between the two devices [WEBRTC-TRANSPORTS].
Performing the tasks required by these protocols takes time,
especially when multiple protocol sessions are involved. While tasks
for different sessions can be performed in parallel, it is
nonetheless necessary for applications to wait for all sessions to be
opened before communication between two users can begin. Reducing
the number of STUN/ICE/TURN steps reduces the likelihood of loss of a
packet for one of these protocols; any such loss adds delay to
setting up a communication session. Further, reducing the number of
STUN/ICE/TURN tasks places a lower burden on the STUN and TURN
servers.
Multiplexing may reduce the complexity and resulting load on an
endpoint. A single instance of STUN/ICE/TURN is simpler to execute
and manage than multiple instances STUN/ICE/TURN operations happening
in parallel, as the latter require synchronization and create more
complex failure situations that have to be cleaned up by additional
code.
3. Differentiated Services (Diffserv)
The Diffserv architecture [RFC2475][RFC4594] is intended to enable
scalable service discrimination in the Internet without requiring
each node in the network to store per-flow state and participate in
per-flow signaling. The services may be end to end or within a
network; they include both those that can satisfy quantitative
performance requirements (e.g., peak bandwidth) and those based on
relative performance (e.g., "class" differentiation). Services can
be constructed by a combination of well-defined building blocks
deployed in network nodes that:
o classify traffic and set bits in an IP header field at network
boundaries or hosts,
o use those bits to determine how packets are forwarded by the nodes
inside the network, and
o condition the marked packets at network boundaries in accordance
with the requirements or rules of each service.
Traffic conditioning may include changing the DSCP in a packet
(remarking it), delaying the packet (as a consequence of traffic
shaping), or dropping the packet (as a consequence of traffic
policing).
A network node that supports Diffserv includes a classifier that
selects packets based on the value of the DS field in IP headers (the
Diffserv codepoint or DSCP), along with buffer management and packet
scheduling mechanisms capable of delivering the specific packet
forwarding treatment indicated by the DS field value. Setting of the
DS field and fine-grain conditioning of marked packets need only be
performed at network boundaries; internal network nodes operate on
traffic aggregates that share a DS field value, or in some cases, a
small set of related values.
The Diffserv architecture [RFC2475] maintains distinctions among:
o the QoS service provided to a traffic aggregate,
o the conditioning functions and per-hop behaviors (PHBs) used to
realize services,
o the DSCP in the IP header used to mark packets to select a per-hop
behavior, and
o the particular implementation mechanisms that realize a per-hop
behavior.
This memo focuses on PHBs and the usage of DSCPs to obtain those
behaviors. In a network node's forwarding path, the DSCP is used to
map a packet to a particular forwarding treatment, or to a per-hop
behavior (PHB) that specifies the forwarding treatment.
The specification of a PHB describes the externally observable
forwarding behavior of a network node for network traffic marked with
a DSCP that selects that PHB. In this context, "forwarding behavior"
is a general concept - for example, if only one DSCP is used for all
traffic on a link, the observable forwarding behavior (e.g., loss,
delay, jitter) will often depend only on the loading of the link. To
obtain useful behavioral differentiation, multiple traffic subsets
are marked with different DSCPs for different PHBs for which node
resources such as buffer space and bandwidth are allocated. PHBs
provide the framework for a Diffserv network node to allocate
resources to traffic subsets, with network-scope Differentiated
Services constructed on top of this basic hop-by-hop resource
allocation mechanism.
The codepoints (DSCPs) may be chosen from a small set of fixed values
(the class selector codepoints), from a set of recommended values
defined in PHB specifications, or from values that have purely local
meanings to a specific network that supports Diffserv; in general,
packets may be forwarded across multiple such networks between source
and destination.
The mandatory DSCPs are the class selector codepoints as specified in
[RFC2474]. The class selector codepoints (CS0-CS7) extend the
deprecated concept of IP Precedence in the IPv4 header; three bits
are added, so that the class selector DSCPs are of the form 'xxx000'.
The all-zero DSCP ('000000' or CS0) is always assigned to a Default
PHB that provides best-effort forwarding behavior, and the remaining
class selector codepoints are intended to provide relatively better
per-hop-forwarding behavior in increasing numerical order, but:
o A network endpoint cannot rely upon different class selector
codepoints providing Differentiated Services via assignment to
different PHBs, as adjacent class selector codepoints may use the
same pool of resources on each network node in some networks.
This generalizes to ranges of class selector codepoints, but with
limits -- for example, CS6 and CS7 are often used for network
control (e.g., routing) traffic [RFC4594] and hence are likely to
provide better forwarding behavior under network load to
prioritize network recovery from disruptions. There is no
effective way for a network endpoint to determine which PHBs are
selected by the class selector codepoints on a specific network,
let alone end to end.
o CS1 ('001000') was subsequently designated as the recommended
codepoint for the Lower Effort (LE) PHB [RFC3662]. An LE service
forwards traffic with "lower" priority than best effort and can be
"starved" by best-effort and other "higher" priority traffic. Not
all networks offer an LE service, hence traffic marked with the
CS1 DSCP may not receive lower effort forwarding; such traffic may
be forwarded with a different PHB (e.g., the Default PHB),
remarked to another DSCP (e.g., CS0) and forwarded accordingly, or
dropped. A network endpoint cannot rely upon the presence of an
LE service that is selected by the CS1 DSCP on a specific network,
let alone end to end. Packets marked with the CS1 DSCP may be
forwarded with best-effort service or another "higher" priority
service; see [RFC2474]. See [RFC3662] for further discussion of
the LE PHB and service.
3.1. Diffserv Per-Hop Behaviors (PHBs)
Although Differentiated Services is a general architecture that may
be used to implement a variety of services, three fundamental
forwarding behaviors (PHBs) have been defined and characterized for
general use. These are:
1. Default Forwarding (DF) for elastic traffic [RFC2474]. The
Default PHB is always selected by the all-zero DSCP and provides
best-effort forwarding.
2. Assured Forwarding (AF) [RFC2597] to provide Differentiated
Service to elastic traffic. Each instance of the AF behavior
consists of three PHBs that differ only in drop precedence, e.g.,
AF11, AF12, and AF13; such a set of three AF PHBs is referred to
as an AF class, e.g., AF1x. There are four defined AF classes,
AF1x through AF4x, with higher numbered classes intended to
receive better forwarding treatment than lower numbered classes.
Use of multiple PHBs from a single AF class (e.g., AF1x) does not
enable network traffic reordering within a single network
5-tuple, although such reordering may occur for other transient
reasons (e.g., routing changes or ECMP rebalancing).
3. Expedited Forwarding (EF) [RFC3246] intended for inelastic
traffic. Beyond the basic EF PHB, the VOICE-ADMIT PHB [RFC5865]
is an admission-controlled variant of the EF PHB. Both of these
PHBs are based on preconfigured limited forwarding capacity;
traffic in excess of that capacity is expected to be dropped.
3.2. Traffic Classifiers and DSCP Remarking
DSCP markings are not end to end in general. Each network can make
its own decisions about what PHBs to use and which DSCP maps to each
PHB. While every PHB specification includes a recommended DSCP, and
RFC 4594 [RFC4594] recommends their end-to-end usage, there is no
requirement that every network support any PHBs (aside from the
Default PHB for best-effort forwarding) or use any specific DSCPs,
with the exception of the support requirements for the class selector
codepoints (see RFC 2474 [RFC2474]). When Diffserv is used, the edge
or boundary nodes of a network are responsible for ensuring that all
traffic entering that network conforms to that network's policies for
DSCP and PHB usage, and such nodes may change DSCP markings on
traffic to achieve that result. As a result, DSCP remarking is
possible at any network boundary, including the first network node
that traffic sent by a host encounters. Remarking is also possible
within a network, e.g., for traffic shaping.
DSCP remarking is part of traffic conditioning; the traffic
conditioning functionality applied to packets at a network node is
determined by a traffic classifier [RFC2475]. Edge nodes of a
Diffserv network classify traffic based on selected packet header
fields; typical implementations do not look beyond the traffic's
network 5-tuple in the IP and transport protocol headers (e.g., for
SCTP or RTP encapsulated in UDP, header-based classification is
unlikely to look beyond the outer UDP header). As a result, when
multiple DSCPs are used for traffic that shares a network 5-tuple,
remarking at a network boundary may result in all of the traffic
being forwarded with a single DSCP, thereby removing any
differentiation within the network 5-tuple downstream of the
remarking location. Network nodes within a Diffserv network
generally classify traffic based solely on DSCPs, but may perform
finer-grain traffic conditioning similar to that performed by edge
nodes.
So, for two arbitrary network endpoints, there can be no assurance
that the DSCP set at the source endpoint will be preserved and
presented at the destination endpoint. Rather, it is quite likely
that the DSCP will be set to zero (e.g., at the boundary of a network
operator that distrusts or does not use the DSCP field) or to a value
deemed suitable by an ingress classifier for whatever network 5-tuple
it carries.
In addition, remarking may remove application-level distinctions in
forwarding behavior - e.g., if multiple PHBs within an AF class are
used to distinguish different types of frames within a video RTP
stream, token-bucket-based remarkers operating in color-blind mode
(see [RFC2697] and [RFC2698] for examples) may remark solely based on
flow rate and burst behavior, removing the drop precedence
distinctions specified by the source.
Backbone and other carrier networks may employ a small number of
DSCPs (e.g., less than half a dozen) to manage a small number of
traffic aggregates; hosts that use a larger number of DSCPs can
expect to find that much of their intended differentiation is removed
by such networks. Better results may be achieved when DSCPs are used
to spread traffic among a smaller number of Diffserv-based traffic
subsets or aggregates; see [DIFFSERV-INTERCON] for one proposal.
This is of particular importance for MPLS-based networks due to the
limited size of the Traffic Class (TC) field in an MPLS label
[RFC5462] that is used to carry Diffserv information and the use of
that TC field for other purposes, e.g., Explicit Congestion
Notification (ECN) [RFC5129]. For further discussion on use of
Diffserv with MPLS, see [RFC3270] and [RFC5127].
4. Examples
For real-time communications, one might want to mark the audio
packets using EF and the video packets as AF41. However, a video
conference receiving the audio packets significantly ahead of the
video is not useful because lip sync is necessary between audio and
video. It may still be desirable to send audio with a PHB that
provides better service, because more reliable arrival of audio helps
assure smooth audio rendering, which is often more important than
fully faithful video rendering. There are also limits, as some
devices have difficulties in synchronizing voice and video when
packets that need to be rendered together arrive at significantly
different times. It makes more sense to use different PHBs when the
audio and video source streams do not share a strict timing
relationship. For example, video content may be shared within a
video conference via playback, perhaps of an unedited video clip that
is intended to become part of a television advertisement. Such
content sharing video does not need precise synchronization with
video conference audio, and could use a different PHB, as content
sharing video is more tolerant to jitter, loss, and delay.
Within a layered video RTP stream, ordering of frame communication is
preferred, but importance of frame types varies, making use of PHBs
with different drop precedences appropriate. For example, I-frames
that contain an entire image are usually more important than P-frames
that contain only changes from the previous image because loss of a
P-frame (or part thereof) can be recovered (at the latest) via the
next I-frame, whereas loss of an I-frame (or part thereof) may cause
rendering problems for all of the P-frames that depend on the missing
I-frame. For this reason, it is appropriate to mark I-frame packets
with a PHB that has lower drop precedence than the PHB used for
P-frames, as long as the PHBs preserve ordering among frames (e.g.,
are in a single AF class) - AF41 for I-frames and AF43 for P-frames
is one possibility. Additional spatial and temporal layers beyond
the base video layer could also be marked with higher drop precedence
than the base video layer, as their loss reduces video quality, but
does not disrupt video rendering.
Additional RTP streams in a real-time communication interaction could
be marked with CS0 and carried as best-effort traffic. One example
is real-time text transmitted as specified in RFC 4103 [RFC4103].
Best-effort forwarding suffices because such real-time text has loose
timing requirements; RFC 4103 recommends sending text in chunks every
300 ms. Such text is technically real-time, but does not need a PHB
promising better service than best effort, in contrast to audio or
video.
A WebRTC application may use one or more RTP streams, as discussed
above. In addition, it may use an SCTP-based data channel
[DATA-CHAN] whose QoS treatment depends on the nature of the
application. For example, best-effort treatment of data channels is
likely to suffice for messaging, shared white board, and guided
browsing applications, whereas latency-sensitive games might desire
better QoS for their data channels.
5. Diffserv Interactions
5.1. Diffserv, Reordering, and Transport Protocols
Transport protocols provide data communication behaviors beyond those
possible at the IP layer. An important example is that TCP [RFC793]
provides reliable in-order delivery of data with congestion control.
SCTP [RFC4960] provides additional properties such as preservation of
message boundaries, and the ability to avoid head-of-line blocking
that may occur with TCP.
In contrast, UDP [RFC768] is a basic unreliable datagram protocol
that provides port-based multiplexing and demultiplexing on top of
IP. Two other unreliable datagram protocols are UDP-Lite [RFC3828],
a variant of UDP that may deliver partially corrupt payloads when
errors occur, and DCCP [RFC4340], which provides a range of
congestion control modes for its unreliable datagram service.
Transport protocols that provide reliable delivery (e.g., TCP, SCTP)
are sensitive to network reordering of traffic. When a protocol that
provides reliable delivery receives a packet other than the next
expected packet, the protocol usually assumes that the expected
packet has been lost and updates the peer, which often causes a
retransmission. In addition, congestion control functionality in
transport protocols (including DCCP) usually infers congestion when
packets are lost. This creates additional sensitivity to significant
network packet reordering, as such reordering may be (mis)interpreted
as loss of the out-of-order packets, causing a congestion control
response.
This sensitivity to reordering remains even when ECN [RFC3168] is in
use, as ECN receivers are required to treat missing packets as
potential indications of congestion, because:
o Severe congestion may cause ECN-capable network nodes to drop
packets, and
o ECN traffic may be forwarded by network nodes that do not support
ECN and hence drop packets to indicate congestion.
Congestion control is an important aspect of the Internet
architecture; see [RFC2914] for further discussion.
In general, marking packets with different DSCPs results in different
PHBs being applied at nodes in the network, making reordering very
likely due to use of different pools of forwarding resources for each
PHB. This should not be done within a single network 5-tuple for
current transport protocols, with the important exceptions of UDP and
UDP-Lite.
When PHBs that enable reordering are mixed within a single network
5-tuple, the effect is to mix QoS-based traffic classes within the
scope of a single transport protocol connection or association. As
these QoS-based traffic classes receive different network QoS
treatments, they use different pools of network resources and hence
may exhibit different levels of congestion. The result for
congestion-controlled protocols is that a separate instance of
congestion control functionality is needed per QoS-based traffic
class. Current transport protocols support only a single instance of
congestion control functionality for an entire connection or
association; extending that support to multiple instances would add
significant protocol complexity. Traffic in different QoS-based
classes may use different paths through the network; this complicates
path integrity checking in connection- or association-based
protocols, as those paths may fail independently.
The primary example where usage of multiple PHBs does not enable
reordering within a single network 5-tuple is use of PHBs from a
single AF class (e.g., AF1x). Traffic reordering within the scope of
a network 5-tuple that uses a single PHB or AF class may occur for
other transient reasons (e.g., routing changes or ECMP rebalancing).
Reordering also affects other forms of congestion control, such as
techniques for RTP congestion control that were under development
when this memo was published; see [RMCAT-CC] for requirements. These
techniques prefer use of a common (coupled) congestion controller for
RTP streams between the same endpoints to reduce packet loss and
delay by reducing competition for resources at any shared bottleneck.
Shared bottlenecks can be detected via techniques such as correlation
of one-way delay measurements across RTP streams. An alternate
approach is to assume that the set of packets on a single network
5-tuple marked with DSCPs that do not enable reordering will utilize
a common network path and common forwarding resources at each network
node. Under that assumption, any bottleneck encountered by such
packets is shared among all of them, making it safe to use a common
(coupled) congestion controller (see [COUPLED-CC]). This is not a
safe assumption when the packets involved are marked with DSCP values
that enable reordering because a bottleneck may not be shared among
all such packets (e.g., when the DSCP values result in use of
different queues at a network node, but only one queue is a
bottleneck).
UDP and UDP-Lite are not sensitive to reordering in the network,
because they do not provide reliable delivery or congestion control.
On the other hand, when used to encapsulate other protocols (e.g., as
UDP is used by WebRTC; see Section 2.1), the reordering
considerations for the encapsulated protocols apply. For the
specific usage of UDP by WebRTC, every encapsulated protocol (i.e.,
RTP, SCTP, and TCP) is sensitive to reordering as further discussed
in this memo. In addition, [RFC5405] provides general guidelines for
use of UDP (and UDP-Lite); the congestion control guidelines in that
document apply to protocols encapsulated in UDP (or UDP-Lite).
5.2. Diffserv, Reordering, and Real-Time Communication
Real-time communications are also sensitive to network reordering of
packets. Such reordering may lead to unneeded retransmission and
spurious retransmission control signals (such as NACK) in reliable
delivery protocols (see Section 5.1). The degree of sensitivity
depends on protocol or stream timers, in contrast to reliable
delivery protocols that usually react to all reordering.
Receiver jitter buffers have important roles in the effect of
reordering on real-time communications:
o Minor packet reordering that is contained within a jitter buffer
usually has no effect on rendering of the received RTP stream
because packets that arrive out of order are retrieved in order
from the jitter buffer for rendering.
o Packet reordering that exceeds the capacity of a jitter buffer can
cause user-perceptible quality problems (e.g., glitches, noise)
for delay-sensitive communication, such as interactive
conversations for which small jitter buffers are necessary to
preserve human perceptions of real-time interaction. Interactive
real-time communication implementations often discard data that is
sufficiently late so that it cannot be rendered in source stream
order, making retransmission counterproductive. For this reason,
implementations of interactive real-time communication often do
not use retransmission.
o In contrast, replay of recorded media can tolerate significantly
longer delays than interactive conversations, so replay is likely
to use larger jitter buffers than interactive conversations.
These larger jitter buffers increase the tolerance of replay to
reordering by comparison to interactive conversations. The size
of the jitter buffer imposes an upper bound on replay tolerance to
reordering but does enable retransmission to be used when the
jitter buffer is significantly larger than the amount of data that
can be expected to arrive during the round-trip latency for
retransmission.
Network packet reordering has no effective upper bound and can exceed
the size of any reasonable jitter buffer. In practice, the size of
jitter buffers for replay is limited by external factors such as the
amount of time that a human is willing to wait for replay to start.
5.3. Drop Precedence and Transport Protocols
Packets within the same network 5-tuple that use PHBs within a single
AF class can be expected to draw upon the same forwarding resources
on network nodes (e.g., use the same router queue), and hence use of
multiple drop precedences within an AF class is not expected to cause
latency variation. When PHBs within a single AF class are mixed
within a flow, the resulting overall likelihood that packets will be
dropped from that flow is a mix of the drop likelihoods of the PHBs
involved.
There are situations in which drop precedences should not be mixed.
A simple example is that there is little value in mixing drop
precedences within a TCP connection, because TCP's ordered delivery
behavior results in any drop requiring the receiver to wait for the
dropped packet to be retransmitted. Any resulting delay depends on
the RTT and not the packet that was dropped. Hence a single DSCP
should be used for all packets in a TCP connection.
As a consequence, when TCP is selected for NAT/FW traversal (e.g., by
TURN), a single DSCP should be used for all traffic on that TCP
connection. An additional reason for this recommendation is that
packetization for STUN/ICE/TURN occurs before passing the resulting
packets to TCP; TCP resegmentation may result in a different
packetization on the wire, breaking any association between DSCPs and
specific data to which they are intended to apply.
SCTP [RFC4960] differs from TCP in a number of ways, including the
ability to deliver messages in an order that differs from the order
in which they were sent and support for unreliable streams. However,
SCTP performs congestion control and retransmission across the entire
association, and not on a per-stream basis. Although there may be
advantages to using multiple drop precedence across SCTP streams or
within an SCTP stream that does not use reliable ordered delivery,
there is no practical operational experience in doing so (e.g., the
SCTP sockets API [RFC6458] does not support use of more than one DSCP
for an SCTP association). As a consequence, the impacts on SCTP
protocol and implementation behavior are unknown and difficult to
predict. Hence a single DSCP should be used for all packets in an
SCTP association, independent of the number or nature of streams in
that association. Similar reasoning applies to a DCCP connection; a
single DSCP should be used because the scope of congestion control is
the connection and there is no operational experience with using more
than one DSCP. This recommendation may be revised in the future if
experiments, analysis, and operational experience provide compelling
reasons to change it.
Guidance on transport protocol design and implementation to provide
support for use of multiple PHBs and DSCPs in a transport protocol
connection (e.g., DCCP) or transport protocol association (e.g.,
SCTP) is out of scope for this memo.
5.4. Diffserv and RTCP
RTCP [RFC3550] is used with RTP to monitor quality of service and
convey information about RTP session participants. A sender of RTCP
packets that also sends RTP packets (i.e., originates an RTP stream)
should use the same DSCP marking for both types of packets. If an
RTCP sender doesn't send any RTP packets, it should mark its RTCP
packets with the DSCP that it would use if it did send RTP packets
with media similar to the RTP traffic that it receives. If the RTCP
sender uses or would use multiple DSCPs that differ only in drop
precedence for RTP, then it should use the DSCP with the least
likelihood of drop for RTCP to increase the likelihood of RTCP packet
delivery.
If the SDP bundle extension [SDP-BUNDLE] is used to negotiate sending
multiple types of media in a single RTP session, then receivers will
send separate RTCP reports for each type of media, using a separate
SSRC for each media type; each RTCP report should be marked with the
DSCP corresponding to the type of media handled by the reporting
SSRC.
This guidance may result in different DSCP markings for RTP streams
and RTCP receiver reports about those RTP streams. The resulting
variation in network QoS treatment by traffic direction is necessary
to obtain representative round-trip time (RTT) estimates that
correspond to the media path RTT, which may differ from the transport
protocol RTT. RTCP receiver reports may be relatively infrequent,
and hence the resulting RTT estimates are of limited utility for
transport protocol congestion control (although those RTT estimates
have other important uses; see [RFC3550]). For this reason, it is
important that RTCP receiver reports sent by an SSRC receive the same
network QoS treatment as the RTP stream being sent by that SSRC.
6. Guidelines
The only use of multiple standardized PHBs and DSCPs that does not
enable network reordering among packets marked with different DSCPs
is use of PHBs within a single AF class. All other uses of multiple
PHBs and/or the class selector DSCPs enable network reordering of
packets that are marked with different DSCPs. Based on this and the
foregoing discussion, the guidelines in this section apply to use of
Diffserv with real-time communications.
Applications and other traffic sources (including RTP SSRCs):
o Should limit use of DSCPs within a single RTP stream to those
whose corresponding PHBs do not enable packet reordering. If this
is not done, significant network reordering may overwhelm
implementation assumptions about reordering limits, e.g., jitter
buffer size, causing poor user experiences (see Section 5.2).
This guideline applies to all of the RTP streams that are within
the scope of a common (coupled) congestion controller when that
controller does not use per-RTP-stream measurements for bottleneck
detection.
o Should use a single DSCP for RTCP packets, which should be a DSCP
used for RTP packets that are or would be sent by that SSRC (see
Section 5.4).
o Should use a single DSCP for all packets within a reliable
transport protocol session (e.g., TCP connection, SCTP
association) or DCCP connection (see Sections 5.1 and 5.3). For
SCTP, this requirement applies across the entire SCTP association,
and not just to individual streams within an association. When
TURN selects TCP for NAT/FW traversal, this guideline applies to
all traffic multiplexed onto that TCP connection, in contrast to
use of UDP for NAT/FW traversal.
o May use different DSCPs whose corresponding PHBs enable reordering
within a single UDP or UDP-Lite 5-tuple, subject to the above
constraints. The service differentiation provided by such usage
is unreliable, as it may be removed or changed by DSCP remarking
at network boundaries as described in Section 3.2 above.
o Cannot rely on end-to-end preservation of DSCPs as network node
remarking can change DSCPs and remove drop precedence distinctions
(see Section 3.2). For example, if a source uses drop precedence
distinctions within an AF class to identify different types of
video frames, using those DSCP values at the receiver to identify
frame type is inherently unreliable.
o Should limit use of the CS1 codepoint to traffic for which best
effort forwarding is acceptable, as network support for use of CS1
to select a "less than best-effort" PHB is inconsistent. Further,
some networks may treat CS1 as providing "better than best-effort"
forwarding behavior.
There is no guidance in this memo on how network operators should
differentiate traffic. Networks may support all of the PHBs
discussed herein, classify EF and AFxx traffic identically, or even
remark all traffic to best effort at some ingress points.
Nonetheless, it is useful for applications and other traffic sources
to provide finer granularity DSCP marking on packets for the benefit
of networks that offer QoS service differentiation. A specific
example is that traffic originating from a browser may benefit from
QoS service differentiation in within-building and residential access
networks, even if the DSCP marking is subsequently removed or
simplified. This is because such networks and the boundaries between
them are likely traffic bottleneck locations (e.g., due to customer
aggregation onto common links and/or speed differences among links
used by the same traffic).
7. Security Considerations
The security considerations for all of the technologies discussed in
this memo apply; in particular, see the security considerations for
RTP in [RFC3550] and Diffserv in [RFC2474] and [RFC2475].
Multiplexing of multiple protocols onto a single UDP 5-tuple via
encapsulation has implications for network functionality that
monitors or inspects individual protocol flows, e.g., firewalls and
traffic monitoring systems. When implementations of such
functionality lack visibility into encapsulated traffic (likely for
many current implementations), it may be difficult or impossible to
apply network security policy and associated controls at a finer
granularity than the overall UDP 5-tuple.
Use of multiple DSCPs that enable reordering within an overall real-
time communication interaction enlarges the set of network forwarding
resources used by that interaction, thereby increasing exposure to
resource depletion or failure, independent of whether the underlying
cause is benign or malicious. This represents an increase in the
effective attack surface of the interaction and is a consideration in
selecting an appropriate degree of QoS differentiation among the
components of the real-time communication interaction. See
Section 3.3.2.1 of [RFC6274] for related discussion of DSCP security
considerations.
Use of multiple DSCPs to provide differentiated QoS service may
reveal information about the encrypted traffic to which different
service levels are provided. For example, DSCP-based identification
of RTP streams combined with packet frequency and packet size could
reveal the type or nature of the encrypted source streams. The IP
header used for forwarding has to be unencrypted for obvious reasons,
and the DSCP likewise has to be unencrypted to enable different IP
forwarding behaviors to be applied to different packets. The nature
of encrypted traffic components can be disguised via encrypted dummy
data padding and encrypted dummy packets, e.g., see the discussion of
traffic flow confidentiality in [RFC4303]. Encrypted dummy packets
could even be added in a fashion that an observer of the overall
encrypted traffic might mistake for another encrypted RTP stream.
8. References
8.1. Normative References
[RFC768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
DOI 10.17487/RFC0768, August 1980,
<http://www.rfc-editor.org/info/rfc768>.
[RFC793] Postel, J., "Transmission Control Protocol", STD 7,
RFC 793, DOI 10.17487/RFC0793, September 1981,
<http://www.rfc-editor.org/info/rfc793>.
[RFC2474] Nichols, K., Blake, S., Baker, F., and D. Black,
"Definition of the Differentiated Services Field (DS
Field) in the IPv4 and IPv6 Headers", RFC 2474,
DOI 10.17487/RFC2474, December 1998,
<http://www.rfc-editor.org/info/rfc2474>.
[RFC2475] Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z.,
and W. Weiss, "An Architecture for Differentiated
Services", RFC 2475, DOI 10.17487/RFC2475, December 1998,
<http://www.rfc-editor.org/info/rfc2475>.
[RFC2597] Heinanen, J., Baker, F., Weiss, W., and J. Wroclawski,
"Assured Forwarding PHB Group", RFC 2597,
DOI 10.17487/RFC2597, June 1999,
<http://www.rfc-editor.org/info/rfc2597>.
[RFC3246] Davie, B., Charny, A., Bennet, J., Benson, K., Le Boudec,
J., Courtney, W., Davari, S., Firoiu, V., and D.
Stiliadis, "An Expedited Forwarding PHB (Per-Hop
Behavior)", RFC 3246, DOI 10.17487/RFC3246, March 2002,
<http://www.rfc-editor.org/info/rfc3246>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3662] Bless, R., Nichols, K., and K. Wehrle, "A Lower Effort
Per-Domain Behavior (PDB) for Differentiated Services",
RFC 3662, DOI 10.17487/RFC3662, December 2003,
<http://www.rfc-editor.org/info/rfc3662>.
[RFC3828] Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., Ed.,
and G. Fairhurst, Ed., "The Lightweight User Datagram
Protocol (UDP-Lite)", RFC 3828, DOI 10.17487/RFC3828, July
2004, <http://www.rfc-editor.org/info/rfc3828>.
[RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram
Congestion Control Protocol (DCCP)", RFC 4340,
DOI 10.17487/RFC4340, March 2006,
<http://www.rfc-editor.org/info/rfc4340>.
[RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol",
RFC 4960, DOI 10.17487/RFC4960, September 2007,
<http://www.rfc-editor.org/info/rfc4960>.
[RFC5405] Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
for Application Designers", BCP 145, RFC 5405,
DOI 10.17487/RFC5405, November 2008,
<http://www.rfc-editor.org/info/rfc5405>.
[RFC5865] Baker, F., Polk, J., and M. Dolly, "A Differentiated
Services Code Point (DSCP) for Capacity-Admitted Traffic",
RFC 5865, DOI 10.17487/RFC5865, May 2010,
<http://www.rfc-editor.org/info/rfc5865>.
[RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream
Control Transmission Protocol (SCTP) Packets for End-Host
to End-Host Communication", RFC 6951,
DOI 10.17487/RFC6951, May 2013,
<http://www.rfc-editor.org/info/rfc6951>.
[RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
for the Real-Time Transport Protocol (RTP) Sources",
RFC 7656, DOI 10.17487/RFC7656, November 2015,
<http://www.rfc-editor.org/info/rfc7656>.
8.2. Informative References
[COUPLED-CC]
Welzl, M., Islam, S., and S. Gjessing, "Coupled congestion
control for RTP media", Work in Progress,
draft-welzl-rmcat-coupled-cc-05, June 2015.
[DATA-CHAN]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", Work in Progress, draft-ietf-rtcweb-data-
channel-13, January 2015.
[DIFFSERV-INTERCON]
Geib, R., Ed. and D. Black, "Diffserv interconnection
classes and practice", Work in Progress, draft-ietf-tsvwg-
diffserv-intercon-03, October 2015.
[H.221] ITU-T, "Frame structure for a 64 to 1920 kbit/s channel in
audiovisual teleservices", Recommendation H.221, March
2009.
[H.264] ITU-T, "Advanced video coding for generic audiovisual
services", Recommendation H.264, February 2014.
[RFC2697] Heinanen, J. and R. Guerin, "A Single Rate Three Color
Marker", RFC 2697, DOI 10.17487/RFC2697, September 1999,
<http://www.rfc-editor.org/info/rfc2697>.
[RFC2698] Heinanen, J. and R. Guerin, "A Two Rate Three Color
Marker", RFC 2698, DOI 10.17487/RFC2698, September 1999,
<http://www.rfc-editor.org/info/rfc2698>.
[RFC2914] Floyd, S., "Congestion Control Principles", BCP 41,
RFC 2914, DOI 10.17487/RFC2914, September 2000,
<http://www.rfc-editor.org/info/rfc2914>.
[RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP",
RFC 3168, DOI 10.17487/RFC3168, September 2001,
<http://www.rfc-editor.org/info/rfc3168>.
[RFC3270] Le Faucheur, F., Wu, L., Davie, B., Davari, S., Vaananen,
P., Krishnan, R., Cheval, P., and J. Heinanen, "Multi-
Protocol Label Switching (MPLS) Support of Differentiated
Services", RFC 3270, DOI 10.17487/RFC3270, May 2002,
<http://www.rfc-editor.org/info/rfc3270>.
[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text
Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005,
<http://www.rfc-editor.org/info/rfc4103>.
[RFC4303] Kent, S., "IP Encapsulating Security Payload (ESP)",
RFC 4303, DOI 10.17487/RFC4303, December 2005,
<http://www.rfc-editor.org/info/rfc4303>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <http://www.rfc-editor.org/info/rfc4566>.
[RFC4594] Babiarz, J., Chan, K., and F. Baker, "Configuration
Guidelines for DiffServ Service Classes", RFC 4594,
DOI 10.17487/RFC4594, August 2006,
<http://www.rfc-editor.org/info/rfc4594>.
[RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, DOI 10.17487/RFC5109, December
2007, <http://www.rfc-editor.org/info/rfc5109>.
[RFC5127] Chan, K., Babiarz, J., and F. Baker, "Aggregation of
Diffserv Service Classes", RFC 5127, DOI 10.17487/RFC5127,
February 2008, <http://www.rfc-editor.org/info/rfc5127>.
[RFC5129] Davie, B., Briscoe, B., and J. Tay, "Explicit Congestion
Marking in MPLS", RFC 5129, DOI 10.17487/RFC5129, January
2008, <http://www.rfc-editor.org/info/rfc5129>.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
DOI 10.17487/RFC5245, April 2010,
<http://www.rfc-editor.org/info/rfc5245>.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
DOI 10.17487/RFC5389, October 2008,
<http://www.rfc-editor.org/info/rfc5389>.
[RFC5462] Andersson, L. and R. Asati, "Multiprotocol Label Switching
(MPLS) Label Stack Entry: "EXP" Field Renamed to "Traffic
Class" Field", RFC 5462, DOI 10.17487/RFC5462, February
2009, <http://www.rfc-editor.org/info/rfc5462>.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761,
DOI 10.17487/RFC5761, April 2010,
<http://www.rfc-editor.org/info/rfc5761>.
[RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for the Secure
Real-time Transport Protocol (SRTP)", RFC 5764,
DOI 10.17487/RFC5764, May 2010,
<http://www.rfc-editor.org/info/rfc5764>.
[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766,
DOI 10.17487/RFC5766, April 2010,
<http://www.rfc-editor.org/info/rfc5766>.
[RFC6062] Perreault, S., Ed. and J. Rosenberg, "Traversal Using
Relays around NAT (TURN) Extensions for TCP Allocations",
RFC 6062, DOI 10.17487/RFC6062, November 2010,
<http://www.rfc-editor.org/info/rfc6062>.
[RFC6274] Gont, F., "Security Assessment of the Internet Protocol
Version 4", RFC 6274, DOI 10.17487/RFC6274, July 2011,
<http://www.rfc-editor.org/info/rfc6274>.
[RFC6386] Bankoski, J., Koleszar, J., Quillio, L., Salonen, J.,
Wilkins, P., and Y. Xu, "VP8 Data Format and Decoding
Guide", RFC 6386, DOI 10.17487/RFC6386, November 2011,
<http://www.rfc-editor.org/info/rfc6386>.
[RFC6437] Amante, S., Carpenter, B., Jiang, S., and J. Rajahalme,
"IPv6 Flow Label Specification", RFC 6437,
DOI 10.17487/RFC6437, November 2011,
<http://www.rfc-editor.org/info/rfc6437>.
[RFC6458] Stewart, R., Tuexen, M., Poon, K., Lei, P., and V.
Yasevich, "Sockets API Extensions for the Stream Control
Transmission Protocol (SCTP)", RFC 6458,
DOI 10.17487/RFC6458, December 2011,
<http://www.rfc-editor.org/info/rfc6458>.
[RMCAT-CC] Jesup, R. and Z. Sarker, "Congestion Control Requirements
for Interactive Real-Time Media", Work in Progress,
draft-ietf-rmcat-cc-requirements-09, December 2014.
[RTP-USAGE]
Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
Communication (WebRTC): Media Transport and Use of RTP",
Work in Progress, draft-ietf-rtcweb-rtp-usage-25, June
2015.
[SDP-BUNDLE]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", Work in Progress, draft-ietf-
mmusic-sdp-bundle-negotiation-23, July 2015.
[SRTP-DTLS]
Petit-Huguenin, M. and G. Salgueiro, "Multiplexing Scheme
Updates for Secure Real-time Transport Protocol (SRTP)
Extension for Datagram Transport Layer Security (DTLS)",
Work in Progress, draft-petithuguenin-avtcore-rfc5764-mux-
fixes-02, March 2015.
[W3C.WD-mediacapture-streams-20130903]
Burnett, D., Bergkvist, A., Jennings, C., and A.
Narayanan, "Media Capture and Streams", World Wide Web
Consortium Recommendation WD-mediacapture-streams-
20130903, September 2013, <http://www.w3.org/TR/2013/
WD-mediacapture-streams-20130903>.
[WEBRTC-OVERVIEW]
Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", Work in Progress,
draft-ietf-rtcweb-overview-14, June 2015.
[WEBRTC-TRANSPORTS]
Alvestrand, H., "Transports for WebRTC", Work in
Progress, draft-ietf-rtcweb-transports-10, October 2015.
Acknowledgements
This memo is the result of many conversations that have occurred
within the DART working group and other working groups in the RAI and
Transport areas. Many thanks to Aamer Akhter, Harald Alvestrand,
Fred Baker, Richard Barnes, Erin Bournival, Ben Campbell, Brian
Carpenter, Spencer Dawkins, Keith Drage, Gorry Fairhurst, Ruediger
Geib, Cullen Jennings, Jonathan Lennox, Karen Nielsen, Colin Perkins,
James Polk, Robert Sparks, Tina Tsou, Michael Welzl, Dan York, and
the DART WG participants for their reviews and comments.
Authors' Addresses
David Black (editor)
EMC
176 South Street
Hopkinton, MA 01748
United States
Phone: +1 508 293-7953
Email: david.black@emc.com
Paul Jones
Cisco
7025 Kit Creek Road
Research Triangle Park, NC 27502
United States
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