Operational Considerations for Streaming Media
draft-ietf-mops-streaming-opcons-11
The information below is for an old version of the document.
| Document | Type |
This is an older version of an Internet-Draft that was ultimately published as RFC 9317.
|
|
|---|---|---|---|
| Authors | Jake Holland , Ali C. Begen , Spencer Dawkins | ||
| Last updated | 2022-07-11 | ||
| Replaces | draft-jholland-mops-taxonomy | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
| Reviews |
TSVART Last Call review
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by Michael Scharf
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| Additional resources | Mailing list discussion | ||
| Stream | WG state | Submitted to IESG for Publication | |
| Document shepherd | Sanjay Mishra | ||
| Shepherd write-up | Show Last changed 2022-03-01 | ||
| IESG | IESG state | Became RFC 9317 (Informational) | |
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| Responsible AD | Éric Vyncke | ||
| Send notices to | sanjay.mishra@verizon.com | ||
| IANA | IANA review state | Version Changed - Review Needed |
draft-ietf-mops-streaming-opcons-11
MOPS J. Holland
Internet-Draft Akamai Technologies, Inc.
Intended status: Informational A. Begen
Expires: 12 January 2023 Networked Media
S. Dawkins
Tencent America LLC
11 July 2022
Operational Considerations for Streaming Media
draft-ietf-mops-streaming-opcons-11
Abstract
This document provides an overview of operational networking and
transport protocol issues that pertain to the quality of experience
when streaming video and other high-bitrate media over the Internet.
This document is intended to explain characteristics of streaming
media delivery that have surprised network designers or transport
experts who lack specific media expertise, since streaming media
highlights key differences between common assumptions in existing
networking practices and observations of media delivery issues
encountered when streaming media over those existing networks.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on 12 January 2023.
Copyright Notice
Copyright (c) 2022 IETF Trust and the persons identified as the
document authors. All rights reserved.
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This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents (https://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 Revised BSD License text as
described in Section 4.e of the Trust Legal Provisions and are
provided without warranty as described in the Revised BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Document Scope . . . . . . . . . . . . . . . . . . . . . 4
1.2. Notes for Contributors and Reviewers . . . . . . . . . . 6
1.2.1. Venues for Contribution and Discussion . . . . . . . 6
2. Our Focus on Streaming Video . . . . . . . . . . . . . . . . 6
3. Bandwidth Provisioning . . . . . . . . . . . . . . . . . . . 7
3.1. Scaling Requirements for Media Delivery . . . . . . . . . 7
3.1.1. Video Bitrates . . . . . . . . . . . . . . . . . . . 8
3.1.2. Virtual Reality Bitrates . . . . . . . . . . . . . . 9
3.2. Path Bandwidth Constraints . . . . . . . . . . . . . . . 9
3.2.1. Recognizing Changes from a Baseline . . . . . . . . . 10
3.3. Path Requirements . . . . . . . . . . . . . . . . . . . . 12
3.4. Caching Systems . . . . . . . . . . . . . . . . . . . . . 12
3.5. Predictable Usage Profiles . . . . . . . . . . . . . . . 13
3.6. Unpredictable Usage Profiles . . . . . . . . . . . . . . 14
3.6.1. Peer-to-peer Applications . . . . . . . . . . . . . . 14
3.6.2. Impact of Global Pandemic . . . . . . . . . . . . . . 15
4. Latency Considerations . . . . . . . . . . . . . . . . . . . 16
4.1. Ultra-Low-Latency . . . . . . . . . . . . . . . . . . . . 16
4.1.1. Near-Realtime Latency . . . . . . . . . . . . . . . . 17
4.2. Low-Latency Live . . . . . . . . . . . . . . . . . . . . 17
4.3. Non-Low-Latency Live . . . . . . . . . . . . . . . . . . 18
4.4. On-Demand . . . . . . . . . . . . . . . . . . . . . . . . 19
5. Adaptive Encoding, Adaptive Delivery, and Measurement
Collection . . . . . . . . . . . . . . . . . . . . . . . 19
5.1. Overview . . . . . . . . . . . . . . . . . . . . . . . . 19
5.2. Adaptive Encoding . . . . . . . . . . . . . . . . . . . . 20
5.3. Adaptive Segmented Delivery . . . . . . . . . . . . . . . 21
5.4. Advertising . . . . . . . . . . . . . . . . . . . . . . . 21
5.5. Bitrate Detection Challenges . . . . . . . . . . . . . . 23
5.5.1. Idle Time between Segments . . . . . . . . . . . . . 23
5.5.2. Noisy Measurements . . . . . . . . . . . . . . . . . 24
5.5.3. Wide and Rapid Variation in Path Capacity . . . . . . 25
5.6. Measurement Collection . . . . . . . . . . . . . . . . . 25
6. Transport Protocol Behaviors and Their Implications for Media
Transport Protocols . . . . . . . . . . . . . . . . . . . 26
6.1. Media Transport Over Reliable Transport Protocols . . . . 27
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6.2. Media Transport Over Unreliable Transport Protocols . . . 28
6.3. QUIC and Changing Transport Protocol Behavior . . . . . . 30
7. Streaming Encrypted Media . . . . . . . . . . . . . . . . . . 32
7.1. General Considerations for Media Encryption . . . . . . . 33
7.2. Considerations for Hop-by-Hop Media Encryption . . . . . 34
7.3. Considerations for End-to-End Media Encryption . . . . . 36
8. Further Reading and References . . . . . . . . . . . . . . . 36
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 37
10. Security Considerations . . . . . . . . . . . . . . . . . . . 37
11. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 37
12. Informative References . . . . . . . . . . . . . . . . . . . 37
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 46
1. Introduction
This document provides an overview of operational networking and
transport protocol issues that pertain to the quality of experience
(QoE) when streaming video and other high-bitrate media over the
Internet.
This document is intended to explain characteristics of streaming
media delivery that have surprised network designers or transport
experts who lack specific media expertise, since streaming media
highlights key differences between common assumptions in existing
networking practices and observations of media delivery issues
encountered when streaming media over those existing networks.
This document defines "high-bitrate streaming media" as follows:
* "High-bitrate" is a context-sensitive term broadly intended to
capture rates that can be sustained over some but not all of the
target audience's network connections. A snapshot of values
commonly qualifying as high-bitrate on today's internet is given
by the higher-value entries in Section 3.1.1.
* "Streaming" means the continuous transmission of media segments
from a server to a client and its simultaneous consumption by the
client.
- The term "simultaneous" is critical, as media segment
transmission is not considered "streaming" if one downloads a
media file and plays it after the download is completed.
Instead, this would be called "download and play".
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- This has two implications. First, the sending rate for media
segments must match the client's consumption rate (whether
loosely or tightly) to provide uninterrupted playback. That
is, the client must not run out of media segments (buffer
underrun), and must not accept more media segments than it can
buffer before playback (buffer overrun).
- Second, the client's media segment consumption rate is limited
not only by the path's available bandwidth, but also by media
segment availability. The client cannot fetch media segments
that a media server cannot provide (yet).
* "Media" refers to any type of media and associated streams such as
video, audio, metadata, etc.
1.1. Document Scope
A full review of all streaming media considerations for all types of
media over all types of network paths is too broad a topic to cover
comprehensively in a single document.
This document focuses chiefly on large-scale delivery of streaming
high-bitrate media to end users. It is primarily intended for those
controlling endpoints involved in delivering streaming media traffic.
This can include origin servers publishing content, intermediaries
like content delivery networks (CDNs), and providers for client
devices and media players.
Most of the considerations covered in this document apply both to
"live media" (created and streamed as an event is in progress) and
"media on demand" (previously recorded media that is streamed from
storage), except where noted.
Most of the considerations covered in this document apply both to
media that is consumed by a media player, for viewing by a human, and
media that is consumed by a machine, such as a media recorder that is
executing an ABR algorithm, except where noted.
This document contains
* A short description of streaming video characteristics in
Section 2, to set the stage for the rest of the document,
* General guidance on bandwidth provisioning (Section 3) and latency
considerations (Section 4) for streaming video delivery,
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* A description of adaptive encoding and adaptive delivery
techniques in common use for streaming video, along with a
description of the challenges media senders face in detecting the
bitrate available between the media sender and media receiver, and
collection of measurements by a third party for use in analytics
(Section 5),
* A description of existing transport protocols used for video
streaming and the issues encountered when using those protocols,
along with a description of the QUIC transport protocol [RFC9000]
more recently used for streaming media (Section 6),
* A description of implications when streaming encrypted media
(Section 7), and
* Several pointers for further reading on this rapidly changing
subject (Section 8).
Topics outside this scope include:
* in-depth examination of real-time two-way interactive media, such
as video conferencing; although this document touches lightly on
topics related to this space, the intent is to let readers know
that for more in-depth coverage they should look to other
documents, since the techniques and issues for interactive real-
time two-way media differ so dramatically from those in large-
scale one-way delivery of streaming media.
* specific recommendations on operational practices to mitigate
issues described in this document; although some known mitigations
are mentioned in passing, the primary intent is to provide a point
of reference for future solution proposals to describe how new
technologies address or avoid existing problems.
* generalized network performance techniques; while things like
datacenter design and transit network design can be crucial
dependencies for a performant streaming media service, these are
considered independent topics better addressed by other documents.
* transparent tunnels; while tunnels can have an impact on streaming
media via issues like the round trip time and the maximum
transmission unit (MTU) of packets carried over tunnels, for the
purposes of this document these issues are considered as part of
the set of network path properties.
It is worth pointing out explicitly, because questions about "Web
Real-Time Communication" or "WebRTC" have come up often, that some
WebRTC protocols ([RFC8834], [RFC8835]) are mentioned in this
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document, including RTP, WebRTC's principal media transport protocol.
However, (as noted in Section 2) it is difficult to give general
guidance for unreliable media transport protocols used to carry
interactive real-time media.
1.2. Notes for Contributors and Reviewers
Note to RFC Editor: Please remove this section and its subsections
before publication.
This section is to provide references to make it easier to review the
development and discussion on the draft so far.
1.2.1. Venues for Contribution and Discussion
This document is in the GitHub repository at:
https://github.com/ietf-wg-mops/draft-ietf-mops-streaming-opcons
(https://github.com/ietf-wg-mops/draft-ietf-mops-streaming-opcons)
Readers are welcome to open issues and send pull requests for this
document.
Substantial discussion of this document should take place on the MOPS
working group mailing list (mops@ietf.org).
* Join: https://www.ietf.org/mailman/listinfo/mops
(https://www.ietf.org/mailman/listinfo/mops)
* Search: https://mailarchive.ietf.org/arch/browse/mops/
(https://mailarchive.ietf.org/arch/browse/mops/)
2. Our Focus on Streaming Video
As the Internet has grown, an increasingly large share of the traffic
delivered to end users has become video. The most recent available
estimates found that 75% of the total traffic to end users was video
in 2019 (as described in [RFC8404], such traffic surveys have since
become impossible to conduct due to ubiquitous encryption). At that
time, the share of video traffic had been growing for years and was
projected to continue growing (Appendix D of [CVNI]).
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A substantial part of this growth is due to the increased use of
streaming video. However, video traffic in real-time communications
(for example, online videoconferencing) has also grown significantly.
While both streaming video and videoconferencing have real-time
delivery and latency requirements, these requirements vary from one
application to another. For additional discussion of latency
requirements, see Section 4.
In many contexts, video traffic can be handled transparently as
generic application-level traffic. However, as the volume of video
traffic continues to grow, it is becoming increasingly important to
consider the effects of network design decisions on application-level
performance, with considerations for the impact on video delivery.
Much of the focus of this document is on media streaming over HTTP.
HTTP is widely used for media streaming because
* support for HTTP is widely available in a wide range of operating
systems,
* HTTP is also used in a wide variety of other applications,
* HTTP has been demonstrated to provide acceptable performance over
the open Internet,
* HTTP includes state of the art standardized security mechanisms,
and
* HTTP can use already-deployed caching infrastructure such as
content delivery networks (CDN), local proxies, and browser
caches.
Various HTTP versions have been used for media delivery. HTTP/1.0,
HTTP/1.1 and HTTP/2 are carried over TCP [I-D.ietf-tcpm-rfc793bis],
and TCP's transport behavior is described in Section 6.1. HTTP/3 is
carried over QUIC, and QUIC's transport behavior is described in
Section 6.3.
Unreliable media delivery using RTP and other UDP-based protocols is
also discussed in Section 4.1, Section 6.2, and Section 7.2, but it
is difficult to give general guidance for these applications. For
instance, when packet loss occurs, the most appropriate response may
depend on the type of codec being used.
3. Bandwidth Provisioning
3.1. Scaling Requirements for Media Delivery
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3.1.1. Video Bitrates
Video bitrate selection depends on many variables including the
resolution (height and width), frame rate, color depth, codec,
encoding parameters, scene complexity and amount of motion.
Generally speaking, as the resolution, frame rate, color depth, scene
complexity and amount of motion increase, the encoding bitrate
increases. As newer codecs with better compression tools are used,
the encoding bitrate decreases. Similarly, a multi-pass encoding
generally produces better quality output compared to single-pass
encoding at the same bitrate or delivers the same quality at a lower
bitrate.
Here are a few common resolutions used for video content, with
typical ranges of bitrates for the two most popular video codecs
[Encodings].
+============+================+============+============+
| Name | Width x Height | H.264 | H.265 |
+============+================+============+============+
| DVD | 720 x 480 | 1.0 Mbps | 0.5 Mbps |
+------------+----------------+------------+------------+
| 720p (1K) | 1280 x 720 | 3-4.5 Mbps | 2-4 Mbps |
+------------+----------------+------------+------------+
| 1080p (2K) | 1920 x 1080 | 6-8 Mbps | 4.5-7 Mbps |
+------------+----------------+------------+------------+
| 2160p (4k) | 3840 x 2160 | N/A | 10-20 Mbps |
+------------+----------------+------------+------------+
Table 1
* Note that these codecs do not take the actual "available
bandwidth" between streaming video servers and streaming video
receivers into account when encoding, because the codec does not
have any idea what network paths and network path conditions will
carry the encoded video, at some point in the future.
* Note that video receivers attempting to receive encoded video
across a network path with insufficient available path bandwidth
might request the video server to provide video encoded for lower
bitrates, as described in Section 5.3.
* In order to provide multiple encodings for video resources, the
codec must produce multiple versions of the video resource encoded
at various bitrates, as described in Section 5.2.
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3.1.2. Virtual Reality Bitrates
The bitrates given in Section 3.1.1 describe video streams that
provide the user with a single, fixed, point of view - so, the user
has no "degrees of freedom," and the user sees all of the video image
that is available.
Even basic virtual reality (360-degree) videos that allow users to
look around freely (referred to as "three degrees of freedom" or
3DoF) require substantially larger bitrates when they are captured
and encoded as such videos require multiple fields of view of the
scene. Yet, due to smart delivery methods such as viewport-based or
tile-based streaming, there is no need to send the whole scene to the
user. Instead, the user needs only the portion corresponding to its
viewpoint at any given time ([Survey360o]).
In more immersive applications, where limited user movement ("three
degrees of freedom plus" or 3DoF+) or full user movement ("six
degrees of freedom" or 6DoF) is allowed, the required bitrate grows
even further. In this case, immersive content is typically referred
to as volumetric media. One way to represent the volumetric media is
to use point clouds, where streaming a single object may easily
require a bitrate of 30 Mbps or higher. Refer to [MPEGI] and [PCC]
for more details.
3.2. Path Bandwidth Constraints
Even when the bandwidth requirements for video streams along a path
are well understood, additional analysis is required to understand
the constraints on bandwidth at various points in the network. This
analysis is necessary because media servers may react to bandwidth
constraints using two independent feedback loops:
* Media servers often respond to application-level feedback from the
media player that indicates a bottleneck somewhere along the path
by adjusting the number of media segments that the media server
will send to the media player in a given timeframe. This is
described in greater detail in Section 5.
* Media servers also typically rely on transport protocols with
capacity-seeking congestion controllers that probe for available
path bandwidth and adjust the media segment sending rate based on
transport mechanisms. This is described in greater detail in
Section 6.
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The result is that these two (potentially competing) "helpful"
mechanisms each respond to the same bottleneck with no coordination
between themselves, so that each is unaware of actions taken by the
other, and this can result in QoE for users that is significantly
lower than what could have been achieved.
One might wonder why media servers and transport protocols are each
unaware of what the other is doing, and there are multiple reasons
for that. One reason is that media servers are often implemented as
applications executing in user space, relying on a general-purpose
operating system that typically has its transport protocols
implemented in the operating system kernel, making decisions that the
media server never knows about.
In one example, if a media server overestimates the available
bandwidth to the media player,
* the transport protocol detects loss due to congestion and reduces
its sending window size per round trip,
* the media server adapts to application-level feedback from the
media player and reduces its own sending rate,
* the transport protocol sends media segments at the new, lower rate
and confirms that this new, lower rate is "safe" because no
transport-level loss is occurring, but
* because the media server continues to send at the new, lower rate,
the transport protocol's maximum sending rate is now limited by
the amount of information the media server queues for
transmission, so
* the transport protocol cannot probe for available path bandwidth
by sending at a higher rate, until the media receiver signals the
media server that the media server can increase its media segment
sending rate.
To avoid these types of situations, which can potentially affect all
the users whose streaming media segments traverse a bottleneck, there
are several possible mitigations that streaming operators can use.
However, the first step toward mitigating a problem is knowing that a
problem is occurring.
3.2.1. Recognizing Changes from a Baseline
There are many reasons why path characteristics might change in
normal operation, for example:
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* If the path topology changes. For example, routing changes, which
can happen in normal operation, may result in traffic being
carried over a new path topology that that is partially or
entirely disjoint from the previous path, especially if the new
path topology includes one or more path segments that are more
heavily loaded, offer lower total bandwidth, change the overall
Path MTU size, or simply cover more distance between the path
endpoints.
* If cross traffic that also traverses part or all of the same path
topology increases or decreases, especially if this new cross
traffic is "inelastic," and does not respond to indications of
path congestion.
* Wireless links (Wi-Fi, 5G, LTE, etc.) may see rapid changes to
capacity from changes in radio interference and signal strength as
endpoints move.
To recognize that a path carrying streaming media segments has
experienced a change, maintaining a baseline that captures its prior
properties is fundamental. Analytics that aid in that recognition
can be more or less sophisticated and can usefully operate on several
different time scales, from milliseconds to hours or days.
Useful properties to monitor for changes can include:
* round-trip times
* loss rate (and explicit congestion notification (ECN) ([RFC3168]
when in use)
* out of order packet rate
* packet and byte receive rate
* application level goodput
* properties of other connections carrying competing traffic, in
addition to the connections carrying the streaming media segments
* externally provided measurements, for example from network cards
or metrics collected by the operating system
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3.3. Path Requirements
The bitrate requirements in Section 3.1 are per end user actively
consuming a media feed, so in the worst case, the bitrate demands can
be multiplied by the number of simultaneous users to find the
bandwidth requirements for a delivery path with that number of users
downstream. For example, at a node with 10,000 downstream users
simultaneously consuming video streams, approximately 80 Gbps might
be necessary for all of them to get typical content at 1080p
resolution.
However, when there is some overlap in the feeds being consumed by
end users, it is sometimes possible to reduce the bandwidth
provisioning requirements for the network by performing some kind of
replication within the network. This can be achieved via object
caching with the delivery of replicated objects over individual
connections and/or by packet-level replication using multicast.
To the extent that replication of popular content can be performed,
bandwidth requirements at peering or ingest points can be reduced to
as low as a per-feed requirement instead of a per-user requirement.
3.4. Caching Systems
When demand for content is relatively predictable, and especially
when that content is relatively static, caching content close to
requesters and pre-loading caches to respond quickly to initial
requests is often useful (for example, HTTP/1.1 caching is described
in [I-D.ietf-httpbis-cache]). This is subject to the usual
considerations for caching - for example, how much data must be
cached to make a significant difference to the requester and how the
benefits of caching and pre-loading caches balances against the costs
of tracking stale content in caches and refreshing that content.
It is worth noting that not all high-demand content is "live"
content. One relevant example is when popular streaming content can
be staged close to a significant number of requesters, as can happen
when a new episode of a popular show is released. This content may
be largely stable, so low-cost to maintain in multiple places
throughout the Internet. This can reduce demands for high end-to-end
bandwidth without having to use mechanisms like multicast.
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Caching and pre-loading can also reduce exposure to peering point
congestion, since less traffic crosses the peering point exchanges if
the caches are placed in peer networks, especially when the content
can be pre-loaded during off-peak hours, and especially if the
transfer can make use of "Lower-Effort Per-Hop Behavior (LE PHB) for
Differentiated Services" [RFC8622], "Low Extra Delay Background
Transport (LEDBAT)" [RFC6817], or similar mechanisms.
All of this depends, of course, on the ability of a media provider to
predict usage and provision bandwidth, caching, and other mechanisms
to meet the needs of users. In some cases (Section 3.5), this is
relatively routine, but in other cases, it is more difficult
(Section 3.6).
And as with other parts of the ecosystem, new technology brings new
challenges. For example, with the emergence of ultra-low-latency
streaming, responses have to start streaming to the end user while
still being transmitted to the cache, and while the cache does not
yet know the size of the object. Some of the popular caching systems
were designed around cache footprint and had deeply ingrained
assumptions about knowing the size of objects that are being stored,
so the change in design requirements in long-established systems
caused some errors in production. Incidents occurred where a
transmission error in the connection from the upstream source to the
cache could result in the cache holding a truncated segment and
transmitting it to the end user's device. In this case, players
rendering the stream often had the video freeze until the player was
reset. In some cases the truncated object was even cached that way
and served later to other players as well, causing continued stalls
at the same spot in the video for all players playing the segment
delivered from that cache node.
3.5. Predictable Usage Profiles
Historical data shows that users consume more videos and these videos
are encoded at a higher bitrate than they were in the past.
Improvements in the codecs that help reduce the encoding bitrates
with better compression algorithms could not have offset the increase
in the demand for the higher quality video (higher resolution, higher
frame rate, better color gamut, better dynamic range, etc.). In
particular, mobile data usage has shown a large jump over the years
due to increased consumption of entertainment and conversational
video.
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3.6. Unpredictable Usage Profiles
It is also possible for usage profiles to change significantly and
suddenly. These changes are more difficult to plan for, but at a
minimum, recognizing that sudden changes are happening is critical.
Two examples are instructive.
3.6.1. Peer-to-peer Applications
In the first example, described in "Report from the IETF Workshop on
Peer-to-Peer (P2P) Infrastructure, May 28, 2008" ([RFC5594]), when
the BitTorrent filesharing application came into widespread use in
2005, sudden and unexpected growth in peer-to-peer traffic led to
complaints from ISP customers about the performance of delay-
sensitive traffic (VoIP and gaming). These performance issues
resulted from at least two causes:
* Many access networks for end users used underlying technologies
that are inherently asymmetric, favoring downstream bandwidth
(e.g. ADSL, cellular technologies, most IEEE 802.11 variants),
assuming that most users will need more downstream bandwidth than
upstream bandwidth. This is a good assumption for client-server
applications such as streaming video or software downloads, but
BitTorrent rewarded peers that uploaded as much as they
downloaded, so BitTorrent users had much more symmetric usage
profiles which interacted badly with these assymetric access
network technologies.
* BitTorrent also used distributed hash tables to organize peers
into a ring topology, where each peer knew its "next peer" and
"previous peer". There was no connection between the application-
level ring topology and the lower-level network topology, so a
peer's "next peer" might be anywhere on the reachable Internet.
Traffic models that expected most communication to take place with
a relatively small number of servers were unable to cope with
peer-to-peer traffic that was much less predictable.
Especially as end users increase use of video-based social networking
applications, it will be helpful for access network providers to
watch for increasing numbers of end users uploading significant
amounts of content.
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3.6.2. Impact of Global Pandemic
Early in 2020, the CoViD-19 pandemic and resulting quarantines and
shutdowns led to significant changes in traffic patterns, due to a
large number of people who suddenly started working and attending
school remotely and using more interactive applications (video
conferencing, in addition to streaming media). Subsequently, the
Internet Architecture Board (IAB) held a COVID-19 Network Impacts
Workshop [IABcovid] in November 2020. The following observations
from the workshop report are worth considering.
* Participants describing different types of networks reported
different kinds of impacts, but all types of networks saw impacts.
* Mobile networks saw traffic reductions and residential networks
saw significant increases.
* Reported traffic increases from ISPs and Internet Exchange Points
(IXP) over just a few weeks were as big as the traffic growth over
the course of a typical year, representing a 15-20% surge in
growth to land at a new normal that was much higher than
anticipated.
* At DE-CIX Frankfurt, the world's largest Internet Exchange Point
in terms of data throughput, the year 2020 has seen the largest
increase in peak traffic within a single year since the IXP was
founded in 1995.
* The usage pattern changed significantly as work-from-home and
videoconferencing usage peaked during normal work hours, which
would have typically been off-peak hours with adults at work and
children at school. One might expect that the peak would have had
more impact on networks if it had happened during typical evening
peak hours for video streaming applications.
* The increase in daytime bandwidth consumption reflected both
significant increases in essential applications such as
videoconferencing and virtual private networks (VPN), and
entertainment applications as people watched videos or played
games.
* At the IXP level, it was observed that physical link utilization
increased. This phenomenon could probably be explained by a
higher level of uncacheable traffic such as videoconferencing and
VPNs from residential users as they stopped commuting and switched
to work-at-home.
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Again, it will be helpful for streaming operators to monitor traffic
as described in Section 5.6, watching for sudden changes in
performance.
4. Latency Considerations
Streaming media latency refers to the "glass-to-glass" time duration,
which is the delay between the real-life occurrence of an event and
the streamed media being appropriately displayed on an end user's
device. Note that this is different from the network latency
(defined as the time for a packet to cross a network from one end to
another end) because it includes video encoding/decoding and
buffering time, and for most cases also the ingest to an intermediate
service such as a CDN or other video distribution service, rather
than a direct connection to an end user.
The team working on this document found these rough categories to be
useful when considering a streaming media application's latency
requirements:
* ultra-low-latency (less than 1 second)
* low-latency live (less than 10 seconds)
* non-low-latency live (10 seconds to a few minutes)
* on-demand (hours or more)
4.1. Ultra-Low-Latency
Ultra-low-latency delivery of media is defined here as having a
glass-to-glass delay target under one second.
Some media content providers aim to achieve this level of latency for
live media events. This introduces new challenges relative to less-
restricted levels of latency requirements because this latency is the
same scale as commonly observed end-to-end network latency variation
(for example, due to effects such as bufferbloat ([CoDel]), Wi-Fi
error correction, or packet reordering). These effects can make it
difficult to achieve this level of latency for the general case, and
may require tradeoffs in relatively frequent user-visible media
artifacts. However, for controlled environments or targeted networks
that provide mitigations against such effects, this level of latency
is potentially achievable with the right provisioning.
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Applications requiring ultra-low latency for media delivery are
usually tightly constrained on the available choices for media
transport technologies and sometimes may need to operate in
controlled environments to reliably achieve their latency and quality
goals.
Most applications operating over IP networks and requiring latency
this low use the Real-time Transport Protocol (RTP) [RFC3550] or
WebRTC [RFC8825], which uses RTP as its Media Transport Protocol,
along with several other protocols necessary for safe operation in
browsers.
Worth noting is that many applications for ultra-low-latency delivery
do not need to scale to more than a few users at a time, which
simplifies many delivery considerations relative to other use cases.
Recommended reading for applications adopting an RTP-based approach
also includes [RFC7656]. For increasing the robustness of the
playback by implementing adaptive playout methods, refer to [RFC4733]
and [RFC6843].
4.1.1. Near-Realtime Latency
Some internet applications that incorporate media streaming have
specific interactivity or control-feedback requirements that drive
much lower glass-to-glass media latency targets than one second.
These include videoconferencing or voice calls, remote video
gameplay, remote control of hardware platforms like drones, vehicles,
or surgical robots, and many other envisioned or deployed interactive
applications.
Applications with latency targets in these regimes are out of scope
for this document.
4.2. Low-Latency Live
Low-latency live delivery of media is defined here as having a glass-
to-glass delay target under 10 seconds.
This level of latency is targeted to have a user experience similar
to traditional broadcast TV delivery. A frequently cited problem
with failing to achieve this level of latency for live sporting
events is the user experience failure from having crowds within
earshot of one another who react audibly to an important play, or
from users who learn of an event in the match via some other channel,
for example, social media, before it has happened on the screen
showing the sporting event.
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Applications requiring low-latency live media delivery are generally
feasible at scale with some restrictions. This typically requires
the use of a premium service dedicated to the delivery of live video,
and some tradeoffs may be necessary relative to what is feasible in a
higher latency service. The tradeoffs may include higher costs,
delivering a lower quality video, reduced flexibility for adaptive
bitrates or reduced flexibility for available resolutions so that
fewer devices can receive an encoding tuned for their display. Low-
latency live delivery is also more susceptible to user-visible
disruptions due to transient network conditions than higher latency
services.
Implementation of a low-latency live video service can be achieved
with the use of low-latency extensions of HLS (called LL-HLS)
[I-D.draft-pantos-hls-rfc8216bis] and DASH (called LL-DASH)
[LL-DASH]. These extensions use the Common Media Application Format
(CMAF) standard [MPEG-CMAF] that allows the media to be packaged into
and transmitted in units smaller than segments, which are called
chunks in CMAF language. This way, the latency can be decoupled from
the duration of the media segments. Without a CMAF-like packaging,
lower latencies can only be achieved by using very short segment
durations. However, using shorter segments means using more frequent
intra-coded frames and that is detrimental to video encoding quality.
CMAF allows us to still use longer segments (improving encoding
quality) without penalizing latency.
While an LL-HLS client retrieves each chunk with a separate HTTP GET
request, an LL-DASH client uses the chunked transfer encoding feature
of the HTTP [CMAF-CTE] which allows the LL-DASH client to fetch all
the chunks belonging to a segment with a single GET request. An HTTP
server can transmit the CMAF chunks to the LL-DASH client as they
arrive from the encoder/packager. A detailed comparison of LL-HLS
and LL-DASH is given in [MMSP20].
4.3. Non-Low-Latency Live
Non-low-latency live delivery of media is defined here as a live
stream that does not have a latency target shorter than 10 seconds.
This level of latency is the historically common case for segmented
video delivery using HLS [RFC8216] and DASH [MPEG-DASH]. This level
of latency is often considered adequate for content like news or pre-
recorded content. This level of latency is also sometimes achieved
as a fallback state when some part of the delivery system or the
client-side players do not have the necessary support for the
features necessary to support low-latency live streaming.
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This level of latency can typically be achieved at scale with
commodity CDN services for HTTP(s) delivery, and in some cases, the
increased time window can allow for the production of a wider range
of encoding options relative to the requirements for a lower latency
service without the need for increasing the hardware footprint, which
can allow for wider device interoperability.
4.4. On-Demand
On-demand media streaming refers to the playback of pre-recorded
media based on a user's action. In some cases, on-demand media is
produced as a by-product of a live media production, using the same
segments as the live event, but freezing the manifest that describes
the media available from the media server after the live event has
finished. In other cases, on-demand media is constructed out of pre-
recorded assets with no streaming necessarily involved during the
production of the on-demand content.
On-demand media generally is not subject to latency concerns, but
other timing-related considerations can still be as important or even
more important to the user experience than the same considerations
with live events. These considerations include the startup time, the
stability of the media stream's playback quality, and avoidance of
stalls and video artifacts during the playback under all but the most
severe network conditions.
In some applications, optimizations are available to on-demand video
that are not always available to live events, such as pre-loading the
first segment for a startup time that doesn't have to wait for a
network download to begin.
5. Adaptive Encoding, Adaptive Delivery, and Measurement Collection
This section describes one of the best known ways to provide a good
user experience over a given network path, but one thing to keep in
mind is that application-level mechanisms cannot provide a better
experience than the underlying network path can support.
5.1. Overview
A simple model of video playback can be described as a video stream
consumer, a buffer, and a transport mechanism that fills the buffer.
The consumption rate is fairly static and is represented by the
content bitrate. The size of the buffer is also commonly a fixed
size. The fill process needs to be at least fast enough to ensure
that the buffer is never empty, however, it also can have significant
complexity when things like personalization or advertising insertion
workflows are introduced.
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The challenges in filling the buffer in a timely way fall into two
broad categories:
* Content selection comprises all of the steps needed to determine
which content variation to offer the client.
* Content variation is the number of content options that exist at
any given selection point.
The mechanism used to select the bitrate is part of the content
selection, and the content variation are all of the different bitrate
renditions.
Adaptive bitrate streaming ("ABR streaming", or simply "ABR") is a
commonly used technique for dynamically adjusting the compression
level and video quality of a stream to match bandwidth availability.
When this goal is achieved, the media server will tend to send enough
media that the media player does not "stall", without sending so much
media that the media player cannot accept it without exhausting all
available receive buffers.
ABR uses an application-level response strategy in which the
streaming client attempts to detect the available bandwidth of the
network path by first observing the successful application-layer
download speed, and then, given the available bandwidth, the client
chooses a bitrate for each of the video, audio, subtitles and
metadata (among a limited number of available options for each type
of media) that fits within that bandwidth, typically adjusting as
changes in available bandwidth occur in the network or changes in
capabilities occur during the playback (such as available memory,
CPU, display size, etc.).
5.2. Adaptive Encoding
Media servers can provide media streams at various bitrates because
the media has been encoded at various bitrates. This is a so-called
"ladder" of bitrates that can be offered to media players as part of
the manifest, so that the media player can select among the available
bitrate choices.
The media server may also choose to alter which bitrates are made
available to players by adding or removing bitrate options from the
ladder delivered to the player in subsequent manifests built and sent
to the player. This way, both the player, through its selection of
bitrate to request from the manifest, and the server, through its
construction of the bitrates offered in the manifest, are able to
affect network utilization.
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5.3. Adaptive Segmented Delivery
ABR playback is commonly implemented by streaming clients using HLS
[RFC8216] or DASH [MPEG-DASH] to perform a reliable segmented
delivery of media over HTTP. Different implementations use different
strategies [ABRSurvey], often relying on proprietary algorithms
(called rate adaptation or bitrate selection algorithms) to perform
available bandwidth estimation/prediction and the bitrate selection.
Many systems will do an initial probe or a very simple throughput
speed test at the start of video playback. This is done to get a
rough sense of the highest video bitrate in the ABR ladder that the
network between the server and player will likely be able to provide
under initial network conditions. After the initial testing, clients
tend to rely upon passive network observations and will make use of
player side statistics such as buffer fill rates to monitor and
respond to changing network conditions.
The choice of bitrate occurs within the context of optimizing for one
or more metrics monitored by the client, such as the highest
achievable video quality or lowest chances for a rebuffering event
(playback stall).
5.4. Advertising
The inclusion of advertising alongside or interspersed with streaming
media content is common in today's media landscape.
Some commonly used forms of advertising can introduce potential user
experience issues for a media stream. This section provides a very
brief overview of a complex and rapidly evolving space.
The same techniques used to allow a media player to switch between
renditions of different bitrates at segment or chunk boundaries can
also be used to enable the dynamic insertion of advertisements
(hereafter referred to as "ads"), but this does not mean that the
insertion of ads has no effect on the user's quality of experience.
Ads may be inserted either with Client-side Ad Insertion (CSAI) or
Server-side Ad Insertion (SSAI). - In CSAI, the ABR manifest will
generally include links to an external ad server for some segments of
the media stream, while in SSAI the server will remain the same
during advertisements, but will include media segments that contain
the advertising. - In SSAI, the media segments may or may not be
sourced from an external ad server like with CSAI.
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In general, the more targeted the ad request is, the more requests
the ad service needs to be able to handle concurrently. If
connectivity is poor to the ad service, this can cause rebuffering
even if the underlying video assets (both content and ads) can be
accessed quickly. The less targeted the ad request is, the more
likely that ad requests can be consolidated, and that ads can be
cached similarly to the video content.
In some cases, especially with SSAI, advertising space in a stream is
reserved for a specific advertiser and can be integrated with the
video so that the segments share the same encoding properties such as
bitrate, dynamic range, and resolution. However, in many cases, ad
servers integrate with a Supply Side Platform (SSP) that offers
advertising space in real-time auctions via an Ad Exchange, with bids
for the advertising space coming from Demand Side Platforms (DSPs)
that collect money from advertisers for delivering the
advertisements. Most such Ad Exchanges use application-level
protocol specifications published by the Interactive Advertising
Bureau [IAB-ADS], an industry trade organization.
This ecosystem balances several competing objectives, and integrating
with it naively can produce surprising user experience results. For
example, ad server provisioning and/or the bitrate of the ad segments
might be different from that of the main video, and either of these
differences can result in video stalls. For another example, since
the inserted ads are often produced independently, they might have a
different base volume level than the main video, which can make for a
jarring user experience.
Another major source of competing objectives comes from user privacy
considerations vs. the advertiser's incentives to target ads to user
segments based on behavioral data. Multiple studies, for example
[BEHAVE] and [BEHAVE2], have reported large improvements in ad
effectiveness when using behaviorally targeted ads, relative to
untargeted ads. This provides a strong incentive for advertisers to
gain access to the data necessary to perform behavioral targeting,
leading some to engage in what is indistinguishable from a pervasive
monitoring attack ([RFC7258]) based on user tracking in order to
collect the relevant data, A more complete review of issues in this
space is available in [BALANCING].
On top of these competing objectives, this market historically has
had incidents of misreporting of ad delivery to end users for
financial gain [ADFRAUD]. As a mitigation for concerns driven by
those incidents, some SSPs have required the use of specific media
players that include features like reporting of ad delivery, or
providing additional user information that can be used for tracking.
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In general, this is a rapidly developing space with many
considerations, and media streaming operators engaged in advertising
may need to research these and other concerns to find solutions that
meet their user experience, user privacy, and financial goals. For
further reading on mitigations, [BAP] has published some standards
and best practices based on user experience research.
5.5. Bitrate Detection Challenges
This kind of bandwidth-measurement system can experience trouble in
several ways that are affected by networking and transport protocol
issues. Because adaptive application-level response strategies are
often using rates as observed by the application layer, there are
sometimes inscrutable transport-level protocol behaviors that can
produce surprising measurement values when the application-level
feedback loop is interacting with a transport-level feedback loop.
A few specific examples of surprising phenomena that affect bitrate
detection measurements are described in the following subsections.
As these examples will demonstrate, it is common to encounter cases
that can deliver application-level measurements that are too low, too
high, and (possibly) correct but varying more quickly than a lab-
tested selection algorithm might expect.
These effects and others that cause transport behavior to diverge
from lab modeling can sometimes have a significant impact on bitrate
selection and on user QoE, especially where players use naive
measurement strategies and selection algorithms that don't account
for the likelihood of bandwidth measurements that diverge from the
true path capacity.
5.5.1. Idle Time between Segments
When the bitrate selection is chosen substantially below the
available capacity of the network path, the response to a segment
request will typically complete in much less absolute time than the
duration of the requested segment, leaving significant idle time
between segment downloads. This can have a few surprising
consequences:
* TCP slow-start when restarting after idle requires multiple RTTs
to re-establish a throughput at the network's available capacity.
When the active transmission time for segments is substantially
shorter than the time between segments, leaving an idle gap
between segments that triggers a restart of TCP slow-start, the
estimate of the successful download speed coming from the
application-visible receive rate on the socket can thus end up
much lower than the actual available network capacity. This, in
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turn, can prevent a shift to the most appropriate bitrate.
[RFC7661] provides some mitigations for this effect at the TCP
transport layer for senders who anticipate a high incidence of
this problem.
* Mobile flow-bandwidth spectrum and timing mapping can be impacted
by idle time in some networks. The carrier capacity assigned to a
physical or virtual link can vary with activity. Depending on the
idle time characteristics, this can result in a lower available
bitrate than would be achievable with a steadier transmission in
the same network.
Some receiver-side ABR algorithms such as [ELASTIC] are designed to
try to avoid this effect.
Another way to mitigate this effect is by the help of two
simultaneous TCP connections, as explained in [MMSys11] for Microsoft
Smooth Streaming. In some cases, the system-level TCP slow-start
restart can also be disabled, for example, as described in
[OReilly-HPBN].
5.5.2. Noisy Measurements
In addition to smoothing over an appropriate time scale to handle
network jitter (see [RFC5481]), ABR systems relying on measurements
at the application layer also have to account for noise from the in-
order data transmission at the transport layer.
For instance, in the event of a lost packet on a TCP connection with
SACK support (a common case for segmented delivery in practice), loss
of a packet can provide a confusing bandwidth signal to the receiving
application. Because of the sliding window in TCP, many packets may
be accepted by the receiver without being available to the
application until the missing packet arrives. Upon the arrival of
the one missing packet after retransmit, the receiver will suddenly
get access to a lot of data at the same time.
To a receiver measuring bytes received per unit time at the
application layer and interpreting it as an estimate of the available
network bandwidth, this appears as a high jitter in the goodput
measurement, presenting as a stall followed by a sudden leap that can
far exceed the actual capacity of the transport path from the server
when the hole in the received data is filled by a later
retransmission.
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5.5.3. Wide and Rapid Variation in Path Capacity
As many end devices have moved to wireless connections for the final
hop (such as Wi-Fi, 5G, LTE, etc.), new problems in bandwidth
detection have emerged.
In most real-world operating environments, wireless links can often
experience sudden changes in capacity as the end user device moves
from place to place or encounters new sources of interference.
Microwave ovens, for example, can cause a throughput degradation in
Wi-Fi of more than a factor of 2 while active [Micro].
These swings in actual transport capacity can result in user
experience issues when interacting with ABR algorithms that aren't
tuned to handle the capacity variation gracefully.
5.6. Measurement Collection
Media players use measurements to guide their segment-by-segment
adaptive streaming requests, but may also provide measurements to
streaming media providers.
In turn, media providers may base analytics on these measurements, to
guide decisions such as whether adaptive encoding bitrates in use are
the best ones to provide to media players, or whether current media
content caching is providing the best experience for viewers.
To that effect, the Consumer Technology Association (CTA), who owns
the Web Application Video Ecosystem (WAVE) project, has published two
important specifications.
* CTA-2066: Streaming Quality of Experience Events, Properties and
Metrics
[CTA-2066] specifies a set of media player events, properties, QoE
metrics and associated terminology for representing streaming media
QoE across systems, media players and analytics vendors. While all
these events, properties, metrics and associated terminology are used
across a number of proprietary analytics and measurement solutions,
they were used in slightly (or vastly) different ways that led to
interoperability issues. CTA-2066 attempts to address this issue by
defining a common terminology as well as how each metric should be
computed for consistent reporting.
* CTA-5004: Common Media Client Data (CMCD)
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Many assume that the CDNs have a holistic view of the health and
performance of the streaming clients. However, this is not the case.
The CDNs produce millions of log lines per second across hundreds of
thousands of clients and they have no concept of a "session" as a
client would have, so CDNs are decoupled from the metrics the clients
generate and report. A CDN cannot tell which request belongs to
which playback session, the duration of any media object, the
bitrate, or whether any of the clients have stalled and are
rebuffering or are about to stall and will rebuffer. The consequence
of this decoupling is that a CDN cannot prioritize delivery for when
the client needs it most, prefetch content, or trigger alerts when
the network itself may be underperforming. One approach to couple
the CDN to the playback sessions is for the clients to communicate
standardized media-relevant information to the CDNs while they are
fetching data. [CTA-5004] was developed exactly for this purpose.
6. Transport Protocol Behaviors and Their Implications for Media
Transport Protocols
Within this document, the term "Media Transport Protocol" is used to
describe any protocol that carries media metadata and media segments
in its payload, and the term "Transport Protocol" describes any
protocol that carries a Media Transport Protocol, or another
Transport Protocol, in its payload. This is easier to understand if
the reader assumes a protocol stack that looks something like this:
Media Segments
---------------------------
Media Format
---------------------------
Media Transport Protocol
---------------------------
Transport Protocol(s)
where
* "Media Segments" would be something like the output of a codec, or
some other source of media segments, such as closed-captioning,
* "Media Format" would be something like an RTP payload format
[RFC2736] or an ISOBMFF [ISOBMFF] profile,
* "Media Transport Protocol" would be something like RTP [RFC3550]
or DASH [MPEG-DASH], and
* "Transport Protocol" would be a protocol that provides appropriate
transport services, as described in Section 5 of [RFC8095].
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Not all possible streaming media applications follow this model, but
for the ones that do, it seems useful to distinguish between the
protocol layer that is aware it is transporting media segments, and
underlying protocol layers that are not aware.
As described in the [RFC8095] Abstract, the IETF has standardized a
number of protocols that provide transport services. Although these
protocols, taken in total, provide a wide variety of transport
services, Section 6 will distinguish between two extremes:
* Transport protocols used to provide reliable, in-order media
delivery to an endpoint, typically providing flow control and
congestion control (Section 6.1) and
* Transport protocols used to provide unreliable, unordered media
delivery to an endpoint, without flow control or congestion
control (Section 6.2).
Because newly standardized transport protocols such as QUIC [RFC9000]
that are typically implemented in userspace can evolve their
transport behavior more rapidly than currently-used transport
protocols that are typically implemented in operating system kernel
space, this document includes a description of how the path
characteristics that streaming media providers may see are likely to
evolve in Section 6.3.
It is worth noting explicitly that the Transport Protocol layer might
include more than one protocol. For example, a specific Media
Transport Protocol might run over HTTP, or over WebTransport, which
in turn runs over HTTP.
It is worth noting explicitly that more complex network protocol
stacks are certainly possible - for instance, when packets with this
protocol stack are carried in a tunnel or in a VPN, the entire packet
would likely appear in the payload of other protocols. If these
environments are present, streaming media operators may need to
analyze their effects on applications as well.
6.1. Media Transport Over Reliable Transport Protocols
The HLS [RFC8216] and DASH [MPEG-DASH] media transport protocols are
typically carried over HTTP, and HTTP has used TCP as its only
standardized transport protocol until HTTP/3 [RFC9114]. These media
transport protocols use ABR response strategies as described in
Section 5 to respond to changing path characteristics, and underlying
transport protocols are also attempting to respond to changing path
characteristics.
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The past success of the largely TCP-based Internet is evidence that
the various flow control and congestion control mechanisms TCP has
used to achieve equilibrium quickly, at a point where TCP senders do
not interfere with other TCP senders for sustained periods of time
([RFC5681]), have been largely successful. The Internet has
continued to work even when the specific TCP mechanisms used to reach
equilibrium changed over time ([RFC7414]). Because TCP provided a
common tool to avoid contention, even when significant TCP-based
applications like FTP were largely replaced by other significant TCP-
based applications like HTTP, the transport behavior remained safe
for the Internet.
Modern TCP implementations ([I-D.ietf-tcpm-rfc793bis]) continue to
probe for available bandwidth, and "back off" when a network path is
saturated, but may also work to avoid growing queues along network
paths, which can prevent older TCP senders from detecting quickly
when a network path is becoming saturated. Congestion control
mechanisms such as COPA [COPA18] and BBR
[I-D.cardwell-iccrg-bbr-congestion-control] make these decisions
based on measured path delays, assuming that if the measured path
delay is increasing, the sender is injecting packets onto the network
path faster than the network can forward them (or the receiver can
accept them) so the sender should adjust its sending rate
accordingly.
Although common TCP behavior has changed significantly since the days
of [Jacobson-Karels] and [RFC2001], even adding new congestion
controllers such as CUBIC [RFC8312], the common practice of
implementing TCP as part of an operating system kernel has acted to
limit how quickly TCP behavior can change. Even with the widespread
use of automated operating system update installation on many end-
user systems, streaming media providers could have a reasonable
expectation that they could understand TCP transport protocol
behaviors, and that those behaviors would remain relatively stable in
the short term.
6.2. Media Transport Over Unreliable Transport Protocols
Because UDP does not provide any feedback mechanism to senders to
help limit impacts on other users, UDP-based application-level
protocols have been responsible for the decisions that TCP-based
applications have delegated to TCP - what to send, how much to send,
and when to send it. Because UDP itself has no transport-layer
feedback mechanisms, UDP-based applications that send and receive
substantial amounts of information are expected to provide their own
feedback mechanisms, and to respond to the feedback the application
receives. This expectation is most recently codified as a Best
Current Practice [RFC8085].
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In contrast to adaptive segmented delivery over a reliable transport
as described in Section 5.3, some applications deliver streaming
media segments using an unreliable transport, and rely on a variety
of approaches, including:
* raw MPEG Transport Stream ("MPEG-TS")-formatted video [MPEG-TS]
over UDP, which makes no attempt to account for reordering or loss
in the transport,
* RTP [RFC3550], which can notice packetloss and repair some limited
reordering,
* SCTP [RFC9260], which can use partial reliability [RFC3758] to
recover from some loss, but can abandon recovery to limit head-of-
line blocking, and
* SRT [SRT], which can use forward error correction and time-bound
retransmission to recover from loss within certain limits, but can
abandon recovery to limit head-of-line blocking.
Under congestion and loss, approaches like the above generally
experience transient video artifacts more often and delay of playback
effects less often, as compared with reliable segment transport.
Often one of the key goals of using a UDP-based transport that allows
some unreliability is to reduce latency and better support
applications like videoconferencing, or for other live-action video
with interactive components, such as some sporting events.
Congestion avoidance strategies for deployments using unreliable
transport protocols vary widely in practice, ranging from being
entirely unresponsive to congestion, to using feedback signaling to
change encoder settings (as in [RFC5762]), to using fewer enhancement
layers (as in [RFC6190]), to using proprietary methods to detect QoE
issues and turn off video to allow less bandwidth-intensive media
such as audio to be delivered.
RTP relies on RTCP Sender and Receiver Reports [RFC3550] as its own
feedback mechanism, and even includes Circuit Breakers for Unicast
RTP Sessions [RFC8083] for situations when normal RTP congestion
control has not been able to react sufficiently to RTP flows sending
at rates that result in sustained packet loss.
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The notion of "Circuit Breakers" has also been applied to other UDP
applications in [RFC8084], such as tunneling packets over UDP that
are potentially not congestion-controlled (for example,
"Encapsulating MPLS in UDP," as described in [RFC7510]). If
streaming media segments are carried in tunnels encapsulated in UDP,
these media streams may encounter "tripped circuit breakers," with
resulting user-visible impacts.
6.3. QUIC and Changing Transport Protocol Behavior
The QUIC protocol, developed from a proprietary protocol into an IETF
standards-track protocol [RFC9000], turns many of the statements made
in Section 6.1 and Section 6.2 on their heads.
Although QUIC provides an alternative to the TCP and UDP transport
protocols, QUIC is itself encapsulated in UDP. As noted elsewhere in
Section 7.1, the QUIC protocol encrypts almost all of its transport
parameters, and all of its payload, so any intermediaries that
network operators may be using to troubleshoot HTTP streaming media
performance issues, perform analytics, or even intercept exchanges in
current applications will not work for QUIC-based applications
without making changes to their networks. Section 7 describes the
implications of media encryption in more detail.
While QUIC is designed as a general-purpose transport protocol, and
can carry different application-layer protocols, the current
standardized mapping is for HTTP/3 [RFC9114], which describes how
QUIC transport services are used for HTTP. The convention is for
HTTP/3 to run over UDP port 443 [Port443] but this is not a strict
requirement.
When HTTP/3 is encapsulated in QUIC, which is then encapsulated in
UDP, streaming operators (and network operators) might see UDP
traffic patterns that are similar to HTTP(S) over TCP. UDP ports may
be blocked for any port numbers that are not commonly used, such as
UDP 53 for DNS. Even when UDP ports are not blocked and QUIC packets
can flow, streaming operators (and network operators) may severely
rate-limit this traffic because they do not expect to see legitimate
high-bandwidth traffic such as streaming media over the UDP ports
that HTTP/3 is using.
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As noted in Section 5.5.2, because TCP provides a reliable, in-order
delivery service for applications, any packet loss for a TCP
connection causes head-of-line blocking, so that no TCP segments
arriving after a packet is lost will be delivered to the receiving
application until retransmission of the lost packet has been
received, allowing in-order delivery to the application to continue.
As described in [RFC9000], QUIC connections can carry multiple
streams, and when packet losses do occur, only the streams carried in
the lost packet are delayed.
A QUIC extension currently being specified ([I-D.ietf-quic-datagram])
adds the capability for "unreliable" delivery, similar to the service
provided by UDP, but these datagrams are still subject to the QUIC
connection's congestion controller, providing some transport-level
congestion avoidance measures, which UDP does not.
As noted in Section 6.1, there is an increasing interest in
congestion control algorithms that respond to delay measurements,
instead of responding to packet loss. These algorithms may deliver
an improved user experience, but in some cases, have not responded to
sustained packet loss, which exhausts available buffers along the
end-to-end path that may affect other users sharing that path. The
QUIC protocol provides a set of congestion control hooks that can be
used for algorithm agility, and [RFC9002] defines a basic congestion
control algorithm that is roughly similar to TCP NewReno [RFC6582].
However, QUIC senders can and do unilaterally choose to use different
algorithms such as loss-based CUBIC [RFC8312], delay-based COPA or
BBR, or even something completely different.
The Internet community does have experience with deploying new
congestion controllers without melting the Internet. As noted in
[RFC8312], both the CUBIC congestion controller and its predecessor
BIC have significantly different behavior from Reno-style congestion
controllers such as TCP NewReno [RFC6582], but both CUBIC and BIC
were added to the Linux kernel to allow experimentation and analysis,
and both were then selected as the default TCP congestion controllers
in Linux, and both were deployed globally.
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The point mentioned in Section 6.1 about TCP congestion controllers
being implemented in operating system kernels is different with QUIC.
Although QUIC can be implemented in operating system kernels, one of
the design goals when this work was chartered was "QUIC is expected
to support rapid, distributed development and testing of features,"
and to meet this expectation, many implementers have chosen to
implement QUIC in user space, outside the operating system kernel,
and to even distribute QUIC libraries with their own applications.
It is worth noting that streaming operators using HTTP/3, carried
over QUIC, can expect more frequent deployment of new congestion
controller behavior than has been the case with HTTP/1 and HTTP/2,
carried over TCP.
It is worth considering that if TCP-based HTTP traffic and UDP-based
HTTP/3 traffic are allowed to enter operator networks on roughly
equal terms, questions of fairness and contention will be heavily
dependent on interactions between the congestion controllers in use
for TCP-based HTTP traffic and UDP-based HTTP/3 traffic.
7. Streaming Encrypted Media
"Encrypted Media" has at least three meanings:
* Media encrypted at the application layer, typically using some
sort of Digital Rights Management (DRM) system, and typically
remaining encrypted at rest, when senders and receivers store it.
* Media encrypted by the sender at the transport layer, and
remaining encrypted until it reaches the ultimate media consumer
(in this document, referred to as end-to-end media encryption).
* Media encrypted by the sender at the transport layer, and
remaining encrypted until it reaches some intermediary that is
_not_ the ultimate media consumer, but has credentials allowing
decryption of the media content. This intermediary may examine
and even transform the media content in some way, before
forwarding re-encrypted media content (in this document referred
to as hop-by-hop media encryption).
This document focuses on media encrypted at the transport layer,
whether encryption is performed hop-by-hop or end-to-end. Because
media encrypted at the application layer will only be processed by
application-level entities, this encryption does not have transport-
layer implications. Of course, both hop-by-hop and end-to-end
encrypted transport may carry media that is, in addition, encrypted
at the application layer.
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Each of these encryption strategies is intended to achieve a
different goal. For instance, application-level encryption may be
used for business purposes, such as avoiding piracy or enforcing
geographic restrictions on playback, while transport-layer encryption
may be used to prevent media stream manipulation or to protect
manifests.
This document does not take a position on whether those goals are
"valid" (whatever that might mean).
Both end-to-end and hop-by-hop media encryption have specific
implications for streaming operators. These are described in
Section 7.2 and Section 7.3.
7.1. General Considerations for Media Encryption
The use of strong encryption does provide confidentiality for
encrypted streaming media, from the sender to either an intermediary
or the ultimate media consumer, and this does prevent Deep Packet
Inspection by any intermediary that does not possess credentials
allowing decryption. However, even encrypted content streams may be
vulnerable to traffic analysis. An intermediary that can identify an
encrypted media stream without decrypting it, may be able to
"fingerprint" the encrypted media stream of known content, and then
match the targeted media stream against the fingerprints of known
content. This protection can be lessened if a media provider is
repeatedly encrypting the same content. [CODASPY17] is an example of
what is possible when identifying HTTPS-protected videos over TCP
transport, based either on the length of entire resources being
transferred, or on characteristic packet patterns at the beginning of
a resource being transferred.
If traffic analysis is successful at identifying encrypted content
and associating it with specific users, this breaks privacy as
certainly as examining decrypted traffic.
Because HTTPS has historically layered HTTP on top of TLS, which is
in turn layered on top of TCP, intermediaries have historically had
access to unencrypted TCP-level transport information, such as
retransmissions, and some carriers exploited this information in
attempts to improve transport-layer performance [RFC3135]. The most
recent standardized version of HTTPS, HTTP/3 [RFC9114], uses the QUIC
protocol [RFC9000] as its transport layer. QUIC relies on the TLS
1.3 initial handshake [RFC8446] only for key exchange [RFC9001], and
encrypts almost all transport parameters itself except for a few
invariant header fields. In the QUIC short header, the only
transport-level parameter which is sent "in the clear" is the
Destination Connection ID [RFC8999], and even in the QUIC long
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header, the only transport-level parameters sent "in the clear" are
the Version, Destination Connection ID, and Source Connection ID.
For these reasons, HTTP/3 is significantly more "opaque" than HTTPS
with HTTP/1 or HTTP/2.
[I-D.ietf-quic-manageability] discusses the manageability of the QUIC
transport protocol that is used to encapsulate HTTP/3, focusing on
the implications of QUIC's design and wire image on network
operations involving QUIC traffic. It discusses what network
operators can consider in some detail.
More broadly, RFC 9065 [RFC9065], "Considerations around Transport
Header Confidentiality, Network Operations, and the Evolution of
Internet Transport Protocols" describes the impact of increased
encryption of transport headers in general terms.
It is also worth noting that considerations for heavily-encrypted
transport protocols also come into play when streaming media is
carried over IP-level VPNs and tunnels, with the additional
consideration that an intermediary that does not possess credentials
allowing decryption will not have visibility to the source and
destination IP addresses of the packets being carried inside the
tunnel.
7.2. Considerations for Hop-by-Hop Media Encryption
Although the IETF has put considerable emphasis on end-to-end
streaming media encryption, there are still important use cases that
require the insertion of intermediaries.
There are a variety of ways to involve intermediaries, and some are
much more intrusive than others.
From a media provider's perspective, a number of considerations are
in play. The first question is likely whether the media provider
intends that intermediaries are explicitly addressed from endpoints,
or whether the media provider is willing to allow intermediaries to
"intercept" streaming content transparently, with no awareness or
permission from either endpoint.
If a media provider does not actively work to avoid interception by
intermediaries, the effect will be indistinguishable from
"impersonation attacks," and endpoints cannot be assumed of any level
of privacy.
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Assuming that a media provider does intend to allow intermediaries to
participate in content streaming and does intend to provide some
level of privacy for endpoints, there are a number of possible tools,
either already available or still being specified. These include
* Server And Network assisted DASH [MPEG-DASH-SAND] - this
specification introduces explicit messaging between DASH clients
and network elements or between various network elements for the
purpose of improving the efficiency of streaming sessions by
providing information about real-time operational characteristics
of networks, servers, proxies, caches, CDNs, as well as DASH
client's performance and status.
* "Double Encryption Procedures for the Secure Real-Time Transport
Protocol (SRTP)" [RFC8723] - this specification provides a
cryptographic transform for the Secure Real-time Transport
Protocol that provides both hop-by-hop and end-to-end security
guarantees.
* Secure Media Frames [SFRAME] - [RFC8723] is closely tied to SRTP,
and this close association impeded widespread deployment, because
it could not be used for the most common media content delivery
mechanisms. A more recent proposal, Secure Media Frames [SFRAME],
also provides both hop-by-hop and end-to-end security guarantees,
but can be used with other transport protocols beyond SRTP.
The choice of whether to involve intermediaries sometimes requires
careful consideration. As an example, when ABR manifests were
commonly sent unencrypted, some networks would modify manifests
during peak hours by removing high-bitrate renditions to prevent
players from choosing those renditions, thus reducing the overall
bandwidth consumed for delivering these media streams and thereby
improving the network load and the user experience for their
customers. Now that ubiquitous encryption typically prevents this
kind of modification, a streaming media provider who used
intermediaries in the past, and who now wishes to maintain the same
level of network health and user experience, must choose between
adding intermediaries who are authorized to change the manifests or
adding some other form of complexity to their service.
Some resources that might inform other similar considerations are
further discussed in [RFC8824] (for WebRTC) and
[I-D.ietf-quic-manageability] (for HTTP/3 and QUIC).
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7.3. Considerations for End-to-End Media Encryption
End-to-end media encryption offers the potential of providing privacy
for streaming media consumers, with the idea being that if an
unauthorized intermediary cannot decrypt streaming media, the
intermediary cannot use Deep Packet Inspection to examine HTTP
request and response headers and identify the media content being
streamed.
End-to-end media encryption has become much more widespread in the
years since the IETF issued "Pervasive Monitoring Is an Attack"
[RFC7258] as a Best Current Practice, describing pervasive monitoring
as a much greater threat than previously appreciated. After the
Snowden disclosures, many content providers made the decision to use
HTTPS protection - HTTP over TLS - for most or all content being
delivered as a routine practice, rather than in exceptional cases for
content that was considered sensitive.
However, as noted in [RFC7258], there is no way to prevent pervasive
monitoring by an attacker, while allowing monitoring by a more benign
entity who only wants to use DPI to examine HTTP requests and
responses to provide a better user experience. If a modern encrypted
transport protocol is used for end-to-end media encryption,
intermediary streaming operators are unable to examine transport and
application protocol behavior. As described in Section 7.2, only an
intermediary explicitly authorized by the streaming media provider
who is to examine packet payloads, rather than intercepting packets
and examining them without authorization, can continue these
practices.
[RFC7258] said that "The IETF will strive to produce specifications
that mitigate pervasive monitoring attacks," so streaming operators
should expect the IETF's direction toward preventing unauthorized
monitoring of IETF protocols to continue for the foreseeable future.
8. Further Reading and References
The MOPS community maintains a list of references and resources for
further reading at this location:
* https://github.com/ietf-wg-mops/draft-ietf-mops-streaming-
opcons/blob/main/living-doc-mops-streaming-opcons.md
(https://github.com/ietf-wg-mops/draft-ietf-mops-streaming-
opcons/blob/main/living-doc-mops-streaming-opcons.md)
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Editor's note: The URL above might or might not be changed during
IESG Evaluation. See https://github.com/ietf-wg-mops/draft-ietf-
mops-streaming-opcons/issues/114 (https://github.com/ietf-wg-mops/
draft-ietf-mops-streaming-opcons/issues/114) for updates.
9. IANA Considerations
This document requires no actions from IANA.
10. Security Considerations
Security is an important matter for streaming media applications and
the topic of media encryption was explained in Section 7. This
document itself introduces no new security issues.
11. Acknowledgments
Thanks to Alexandre Gouaillard, Aaron Falk, Chris Lemmons, Dave Oran,
Eric Vyncke, Glenn Deen, Kyle Rose, Leslie Daigle, Linda Dunbar,
Lucas Pardue, Mark Nottingham, Matt Stock, Mike English, Renan
Krishna, Roni Even, Sanjay Mishra, Kiran Makhjani, Chris Lemmons,
Tommy Pauly, Will Law, Michael Scharf, Eric Vyncke, Erik Kline, Roman
Danyliw, Valery Smyslov, Robert Wilton, Lars Eggert, Zahed Sarker,
Warren Kumari, John Scudder, Martin Duke, and Nancy Cam-Winget for
very helpful suggestions, reviews and comments.
12. Informative References
[ABRSurvey]
Taani, B., Begen, A. C., Timmerer, C., Zimmermann, R., and
A. Bentaleb et al, "A Survey on Bitrate Adaptation Schemes
for Streaming Media Over HTTP", IEEE Communications
Surveys & Tutorials , 2019,
<https://ieeexplore.ieee.org/abstract/document/8424813>.
[ADFRAUD] Sadeghpour, S. and N. Vlajic, "Ads and Fraud: A
Comprehensive Survey of Fraud in Online Advertising",
Journal of Cybersecurity and Privacy 1, no. 4: 804-832. ,
16 December 2021, <https://doi.org/10.3390/jcp1040039>.
[BALANCING]
Berger, D. D., "Balancing Consumer Privacy with Behavioral
Targeting", 27 Santa Clara High Technology Law Journal,
Vol. 27 Issue 1 Article 2 , 2010,
<https://digitalcommons.law.scu.edu/chtlj/vol27/iss1/2/>.
[BAP] "The Coalition for Better Ads", n.d.,
<https://www.betterads.org/>.
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[BEHAVE] Yan, J., Liu, N., Wang, G., Zhang, W., Jiang, Y., and Z.
Chen, "How much can behavioral targeting help online
advertising?", WWW '09: Proceedings of the 18th
international conference on World wide webApril 2009 Pages
261-270 , 20 April 2009,
<https://dl.acm.org/doi/abs/10.1145/1526709.1526745>.
[BEHAVE2] Goldfarb, A. and C. E. Tucker, "Online advertising,
behavioral targeting, and privacy", Communications of the
ACMVolume 54Issue 5May 2011 pp 25-27 , 1 May 2011,
<https://dl.acm.org/doi/abs/10.1145/1941487.1941498>.
[CMAF-CTE] Law, W., "Ultra-Low-Latency Streaming Using Chunked-
Encoded and Chunked Transferred CMAF", October 2018,
<https://www.akamai.com/us/en/multimedia/documents/white-
paper/low-latency-streaming-cmaf-whitepaper.pdf>.
[CODASPY17]
Reed, A. and M. Kranch, "Identifying HTTPS-Protected
Netflix Videos in Real-Time", ACM CODASPY , March 2017,
<https://dl.acm.org/doi/10.1145/3029806.3029821>.
[CoDel] Nichols, K. and V. Jacobson, "Controlling Queue Delay",
Communications of the ACM, Volume 55, Issue 7, pp. 42-50 ,
July 2012.
[COPA18] Arun, V. and H. Balakrishnan, "Copa: Practical Delay-Based
Congestion Control for the Internet", USENIX NSDI , April
2018, <https://web.mit.edu/copa/>.
[CTA-2066] Consumer Technology Association, "Streaming Quality of
Experience Events, Properties and Metrics", March 2020,
<https://shop.cta.tech/products/streaming-quality-of-
experience-events-properties-and-metrics>.
[CTA-5004] CTA, "Common Media Client Data (CMCD)", September 2020,
<https://shop.cta.tech/products/web-application-video-
ecosystem-common-media-client-data-cta-5004>.
[CVNI] "Cisco Visual Networking Index: Forecast and Trends,
2017-2022 White Paper", 27 February 2019,
<https://www.cisco.com/c/en/us/solutions/collateral/
service-provider/visual-networking-index-vni/white-paper-
c11-741490.html>.
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[ELASTIC] De Cicco, L., Caldaralo, V., Palmisano, V., and S.
Mascolo, "ELASTIC: A client-side controller for dynamic
adaptive streaming over HTTP (DASH)", Packet Video
Workshop , December 2013,
<https://ieeexplore.ieee.org/document/6691442>.
[Encodings]
Apple, Inc, "HLS Authoring Specification for Apple
Devices", June 2020,
<https://developer.apple.com/documentation/
http_live_streaming/
hls_authoring_specification_for_apple_devices>.
[I-D.cardwell-iccrg-bbr-congestion-control]
Cardwell, N., Cheng, Y., Yeganeh, S. H., Swett, I., and V.
Jacobson, "BBR Congestion Control", Work in Progress,
Internet-Draft, draft-cardwell-iccrg-bbr-congestion-
control-02, 7 March 2022,
<https://datatracker.ietf.org/doc/html/draft-cardwell-
iccrg-bbr-congestion-control-02>.
[I-D.draft-pantos-hls-rfc8216bis]
Pantos, R., "HTTP Live Streaming 2nd Edition", Work in
Progress, Internet-Draft, draft-pantos-hls-rfc8216bis-11,
11 May 2022, <https://datatracker.ietf.org/doc/html/draft-
pantos-hls-rfc8216bis-11>.
[I-D.ietf-httpbis-cache]
Fielding, R. T., Nottingham, M., and J. Reschke, "HTTP
Caching", Work in Progress, Internet-Draft, draft-ietf-
httpbis-cache-19, 12 September 2021,
<https://datatracker.ietf.org/doc/html/draft-ietf-httpbis-
cache-19>.
[I-D.ietf-quic-datagram]
Pauly, T., Kinnear, E., and D. Schinazi, "An Unreliable
Datagram Extension to QUIC", Work in Progress, Internet-
Draft, draft-ietf-quic-datagram-10, 4 February 2022,
<https://datatracker.ietf.org/doc/html/draft-ietf-quic-
datagram-10>.
[I-D.ietf-quic-manageability]
Kuehlewind, M. and B. Trammell, "Manageability of the QUIC
Transport Protocol", Work in Progress, Internet-Draft,
draft-ietf-quic-manageability-17, 11 July 2022,
<https://datatracker.ietf.org/doc/html/draft-ietf-quic-
manageability-17>.
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[I-D.ietf-tcpm-rfc793bis]
Eddy, W. M., "Transmission Control Protocol (TCP)
Specification", Work in Progress, Internet-Draft, draft-
ietf-tcpm-rfc793bis-28, 7 March 2022,
<https://datatracker.ietf.org/doc/html/draft-ietf-tcpm-
rfc793bis-28>.
[IAB-ADS] "IAB", n.d., <https://www.iab.com/>.
[IABcovid] Arkko, J., Farrel, S., Kühlewind, M., and C. Perkins,
"Report from the IAB COVID-19 Network Impacts Workshop
2020", November 2020, <https://datatracker.ietf.org/doc/
draft-iab-covid19-workshop/>.
[ISOBMFF] "ISO/IEC 14496-12:2022 Information technology — Coding of
audio-visual objects — Part 12: ISO base media file
format", January 2022,
<https://www.iso.org/standard/83102.html>.
[Jacobson-Karels]
Jacobson, V. and M. Karels, "Congestion Avoidance and
Control", November 1988,
<https://ee.lbl.gov/papers/congavoid.pdf>.
[LL-DASH] DASH-IF, "Low-latency Modes for DASH", March 2020,
<https://dashif.org/docs/CR-Low-Latency-Live-r8.pdf>.
[Micro] Taher, T. M., Misurac, M. J., LoCicero, J. L., and D. R.
Ucci, "Microwave Oven Signal Interference Mitigation For
Wi-Fi Communication Systems", 2008 5th IEEE Consumer
Communications and Networking Conference 5th IEEE, pp.
67-68 , 2008.
[MMSP20] Durak, K. and et al, "Evaluating the performance of
Apple's low-latency HLS", IEEE MMSP , September 2020,
<https://ieeexplore.ieee.org/document/9287117>.
[MMSys11] Akhshabi, S., Begen, A. C., and C. Dovrolis, "An
experimental evaluation of rate-adaptation algorithms in
adaptive streaming over HTTP", ACM MMSys , February 2011,
<https://dl.acm.org/doi/10.1145/1943552.1943574>.
[MPEG-CMAF]
"ISO/IEC 23000-19:2020 Multimedia application format
(MPEG-A) - Part 19: Common media application format (CMAF)
for segmented media", March 2020,
<https://www.iso.org/standard/79106.html>.
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[MPEG-DASH]
"ISO/IEC 23009-1:2019 Dynamic adaptive streaming over HTTP
(DASH) - Part 1: Media presentation description and
segment formats", December 2019,
<https://www.iso.org/standard/79329.html>.
[MPEG-DASH-SAND]
"ISO/IEC 23009-5:2017 Dynamic adaptive streaming over HTTP
(DASH) - Part 5: Server and network assisted DASH (SAND)",
February 2017, <https://www.iso.org/standard/69079.html>.
[MPEG-TS] "H.222.0 : Information technology - Generic coding of
moving pictures and associated audio information:
Systems", 29 August 2018,
<https://www.itu.int/rec/T-REC-H.222.0>.
[MPEGI] Boyce, J. M. and et al, "MPEG Immersive Video Coding
Standard", Proceedings of the IEEE , n.d.,
<https://ieeexplore.ieee.org/document/9374648>.
[OReilly-HPBN]
"High Performance Browser Networking (Chapter 2: Building
Blocks of TCP)", May 2021,
<https://hpbn.co/building-blocks-of-tcp/>.
[PCC] Schwarz, S. and et al, "Emerging MPEG Standards for Point
Cloud Compression", IEEE Journal on Emerging and Selected
Topics in Circuits and Systems , March 2019,
<https://ieeexplore.ieee.org/document/8571288>.
[Port443] "Service Name and Transport Protocol Port Number
Registry", April 2021, <https://www.iana.org/assignments/
service-names-port-numbers/service-names-port-
numbers.txt>.
[RFC2001] Stevens, W., "TCP Slow Start, Congestion Avoidance, Fast
Retransmit, and Fast Recovery Algorithms", RFC 2001,
DOI 10.17487/RFC2001, January 1997,
<https://www.rfc-editor.org/rfc/rfc2001>.
[RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP
Payload Format Specifications", BCP 36, RFC 2736,
DOI 10.17487/RFC2736, December 1999,
<https://www.rfc-editor.org/rfc/rfc2736>.
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[RFC3135] Border, J., Kojo, M., Griner, J., Montenegro, G., and Z.
Shelby, "Performance Enhancing Proxies Intended to
Mitigate Link-Related Degradations", RFC 3135,
DOI 10.17487/RFC3135, June 2001,
<https://www.rfc-editor.org/rfc/rfc3135>.
[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,
<https://www.rfc-editor.org/rfc/rfc3168>.
[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, <https://www.rfc-editor.org/rfc/rfc3550>.
[RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
Conrad, "Stream Control Transmission Protocol (SCTP)
Partial Reliability Extension", RFC 3758,
DOI 10.17487/RFC3758, May 2004,
<https://www.rfc-editor.org/rfc/rfc3758>.
[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
Digits, Telephony Tones, and Telephony Signals", RFC 4733,
DOI 10.17487/RFC4733, December 2006,
<https://www.rfc-editor.org/rfc/rfc4733>.
[RFC5481] Morton, A. and B. Claise, "Packet Delay Variation
Applicability Statement", RFC 5481, DOI 10.17487/RFC5481,
March 2009, <https://www.rfc-editor.org/rfc/rfc5481>.
[RFC5594] Peterson, J. and A. Cooper, "Report from the IETF Workshop
on Peer-to-Peer (P2P) Infrastructure, May 28, 2008",
RFC 5594, DOI 10.17487/RFC5594, July 2009,
<https://www.rfc-editor.org/rfc/rfc5594>.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
<https://www.rfc-editor.org/rfc/rfc5681>.
[RFC5762] Perkins, C., "RTP and the Datagram Congestion Control
Protocol (DCCP)", RFC 5762, DOI 10.17487/RFC5762, April
2010, <https://www.rfc-editor.org/rfc/rfc5762>.
[RFC6190] Wenger, S., Wang, Y.-K., Schierl, T., and A.
Eleftheriadis, "RTP Payload Format for Scalable Video
Coding", RFC 6190, DOI 10.17487/RFC6190, May 2011,
<https://www.rfc-editor.org/rfc/rfc6190>.
Holland, et al. Expires 12 January 2023 [Page 42]
Internet-Draft Media Streaming Ops July 2022
[RFC6582] Henderson, T., Floyd, S., Gurtov, A., and Y. Nishida, "The
NewReno Modification to TCP's Fast Recovery Algorithm",
RFC 6582, DOI 10.17487/RFC6582, April 2012,
<https://www.rfc-editor.org/rfc/rfc6582>.
[RFC6817] Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind,
"Low Extra Delay Background Transport (LEDBAT)", RFC 6817,
DOI 10.17487/RFC6817, December 2012,
<https://www.rfc-editor.org/rfc/rfc6817>.
[RFC6843] Clark, A., Gross, K., and Q. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Delay Metric
Reporting", RFC 6843, DOI 10.17487/RFC6843, January 2013,
<https://www.rfc-editor.org/rfc/rfc6843>.
[RFC7258] Farrell, S. and H. Tschofenig, "Pervasive Monitoring Is an
Attack", BCP 188, RFC 7258, DOI 10.17487/RFC7258, May
2014, <https://www.rfc-editor.org/rfc/rfc7258>.
[RFC7414] Duke, M., Braden, R., Eddy, W., Blanton, E., and A.
Zimmermann, "A Roadmap for Transmission Control Protocol
(TCP) Specification Documents", RFC 7414,
DOI 10.17487/RFC7414, February 2015,
<https://www.rfc-editor.org/rfc/rfc7414>.
[RFC7510] Xu, X., Sheth, N., Yong, L., Callon, R., and D. Black,
"Encapsulating MPLS in UDP", RFC 7510,
DOI 10.17487/RFC7510, April 2015,
<https://www.rfc-editor.org/rfc/rfc7510>.
[RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms
for Real-Time Transport Protocol (RTP) Sources", RFC 7656,
DOI 10.17487/RFC7656, November 2015,
<https://www.rfc-editor.org/rfc/rfc7656>.
[RFC7661] Fairhurst, G., Sathiaseelan, A., and R. Secchi, "Updating
TCP to Support Rate-Limited Traffic", RFC 7661,
DOI 10.17487/RFC7661, October 2015,
<https://www.rfc-editor.org/rfc/rfc7661>.
[RFC8083] Perkins, C. and V. Singh, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", RFC 8083,
DOI 10.17487/RFC8083, March 2017,
<https://www.rfc-editor.org/rfc/rfc8083>.
Holland, et al. Expires 12 January 2023 [Page 43]
Internet-Draft Media Streaming Ops July 2022
[RFC8084] Fairhurst, G., "Network Transport Circuit Breakers",
BCP 208, RFC 8084, DOI 10.17487/RFC8084, March 2017,
<https://www.rfc-editor.org/rfc/rfc8084>.
[RFC8085] Eggert, L., Fairhurst, G., and G. Shepherd, "UDP Usage
Guidelines", BCP 145, RFC 8085, DOI 10.17487/RFC8085,
March 2017, <https://www.rfc-editor.org/rfc/rfc8085>.
[RFC8095] Fairhurst, G., Ed., Trammell, B., Ed., and M. Kuehlewind,
Ed., "Services Provided by IETF Transport Protocols and
Congestion Control Mechanisms", RFC 8095,
DOI 10.17487/RFC8095, March 2017,
<https://www.rfc-editor.org/rfc/rfc8095>.
[RFC8216] Pantos, R., Ed. and W. May, "HTTP Live Streaming",
RFC 8216, DOI 10.17487/RFC8216, August 2017,
<https://www.rfc-editor.org/rfc/rfc8216>.
[RFC8312] Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and
R. Scheffenegger, "CUBIC for Fast Long-Distance Networks",
RFC 8312, DOI 10.17487/RFC8312, February 2018,
<https://www.rfc-editor.org/rfc/rfc8312>.
[RFC8404] Moriarty, K., Ed. and A. Morton, Ed., "Effects of
Pervasive Encryption on Operators", RFC 8404,
DOI 10.17487/RFC8404, July 2018,
<https://www.rfc-editor.org/rfc/rfc8404>.
[RFC8446] Rescorla, E., "The Transport Layer Security (TLS) Protocol
Version 1.3", RFC 8446, DOI 10.17487/RFC8446, August 2018,
<https://www.rfc-editor.org/rfc/rfc8446>.
[RFC8622] Bless, R., "A Lower-Effort Per-Hop Behavior (LE PHB) for
Differentiated Services", RFC 8622, DOI 10.17487/RFC8622,
June 2019, <https://www.rfc-editor.org/rfc/rfc8622>.
[RFC8723] Jennings, C., Jones, P., Barnes, R., and A.B. Roach,
"Double Encryption Procedures for the Secure Real-Time
Transport Protocol (SRTP)", RFC 8723,
DOI 10.17487/RFC8723, April 2020,
<https://www.rfc-editor.org/rfc/rfc8723>.
[RFC8824] Minaburo, A., Toutain, L., and R. Andreasen, "Static
Context Header Compression (SCHC) for the Constrained
Application Protocol (CoAP)", RFC 8824,
DOI 10.17487/RFC8824, June 2021,
<https://www.rfc-editor.org/rfc/rfc8824>.
Holland, et al. Expires 12 January 2023 [Page 44]
Internet-Draft Media Streaming Ops July 2022
[RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for
Browser-Based Applications", RFC 8825,
DOI 10.17487/RFC8825, January 2021,
<https://www.rfc-editor.org/rfc/rfc8825>.
[RFC8834] Perkins, C., Westerlund, M., and J. Ott, "Media Transport
and Use of RTP in WebRTC", RFC 8834, DOI 10.17487/RFC8834,
January 2021, <https://www.rfc-editor.org/rfc/rfc8834>.
[RFC8835] Alvestrand, H., "Transports for WebRTC", RFC 8835,
DOI 10.17487/RFC8835, January 2021,
<https://www.rfc-editor.org/rfc/rfc8835>.
[RFC8999] Thomson, M., "Version-Independent Properties of QUIC",
RFC 8999, DOI 10.17487/RFC8999, May 2021,
<https://www.rfc-editor.org/rfc/rfc8999>.
[RFC9000] Iyengar, J., Ed. and M. Thomson, Ed., "QUIC: A UDP-Based
Multiplexed and Secure Transport", RFC 9000,
DOI 10.17487/RFC9000, May 2021,
<https://www.rfc-editor.org/rfc/rfc9000>.
[RFC9001] Thomson, M., Ed. and S. Turner, Ed., "Using TLS to Secure
QUIC", RFC 9001, DOI 10.17487/RFC9001, May 2021,
<https://www.rfc-editor.org/rfc/rfc9001>.
[RFC9002] Iyengar, J., Ed. and I. Swett, Ed., "QUIC Loss Detection
and Congestion Control", RFC 9002, DOI 10.17487/RFC9002,
May 2021, <https://www.rfc-editor.org/rfc/rfc9002>.
[RFC9065] Fairhurst, G. and C. Perkins, "Considerations around
Transport Header Confidentiality, Network Operations, and
the Evolution of Internet Transport Protocols", RFC 9065,
DOI 10.17487/RFC9065, July 2021,
<https://www.rfc-editor.org/rfc/rfc9065>.
[RFC9114] Bishop, M., Ed., "HTTP/3", RFC 9114, DOI 10.17487/RFC9114,
June 2022, <https://www.rfc-editor.org/rfc/rfc9114>.
[RFC9260] Stewart, R., Tüxen, M., and K. Nielsen, "Stream Control
Transmission Protocol", RFC 9260, DOI 10.17487/RFC9260,
June 2022, <https://www.rfc-editor.org/rfc/rfc9260>.
[SFRAME] "Secure Media Frames Working Group (Home Page)", n.d.,
<https://datatracker.ietf.org/doc/charter-ietf-sframe/>.
Holland, et al. Expires 12 January 2023 [Page 45]
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[SRT] Sharabayko, M., "Secure Reliable Transport (SRT) Protocol
Overview", 15 April 2020,
<https://datatracker.ietf.org/meeting/interim-2020-mops-
01/materials/slides-interim-2020-mops-01-sessa-april-
15-2020-mops-interim-an-update-on-streaming-video-
alliance>.
[Survey360o]
Yaqoob, A., Bi, T., and G. Muntean, "A Survey on Adaptive
360° Video Streaming: Solutions, Challenges and
Opportunities", IEEE Communications Surveys & Tutorials ,
July 2020, <https://ieeexplore.ieee.org/document/9133103>.
Authors' Addresses
Jake Holland
Akamai Technologies, Inc.
150 Broadway
Cambridge, MA 02144,
United States of America
Email: jakeholland.net@gmail.com
Ali Begen
Networked Media
Turkey
Email: ali.begen@networked.media
Spencer Dawkins
Tencent America LLC
United States of America
Email: spencerdawkins.ietf@gmail.com
Holland, et al. Expires 12 January 2023 [Page 46]