Internet DRAFT - draft-alvestrand-rtp-sess-neutral
draft-alvestrand-rtp-sess-neutral
Network Working Group H. Alvestrand
Internet-Draft Google
Intended status: Informational June 17, 2012
Expires: December 19, 2012
Why RTP Sessions Should Be Content Neutral
draft-alvestrand-rtp-sess-neutral-01
Abstract
This document is not intended for publication as an RFC.
It gives the underpinning arguments for why the idea that RTP
sessions and MIME top level types are related is a deeply broken
paradigm, and that we need to get away from it.
These arguments are solely the opinion of the listed author.
Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
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
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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 December 19, 2012.
Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
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Provisions Relating to IETF Documents
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Stuff that can be carried over RTP . . . . . . . . . . . . . . 3
3. What the network can do to "help" a flow . . . . . . . . . . . 4
4. The definition of an RTP "session" . . . . . . . . . . . . . . 5
5. Proper and improper use of RTP sessions . . . . . . . . . . . 6
6. The Pernicious Effect of SDP on the Media Type System . . . . 8
7. The Mixer Fallacy . . . . . . . . . . . . . . . . . . . . . . 8
8. Corrective Actions . . . . . . . . . . . . . . . . . . . . . . 9
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
10. Security Considerations . . . . . . . . . . . . . . . . . . . 9
11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 10
12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 10
12.1. Normative References . . . . . . . . . . . . . . . . . . 10
12.2. Informative References . . . . . . . . . . . . . . . . . 10
Appendix A. Change log . . . . . . . . . . . . . . . . . . . . . 10
A.1. Version -00 to -01 . . . . . . . . . . . . . . . . . . . 11
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 11
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1. Introduction
The RTP universe of functionality can, for the purposes of this
argument, be reduced to two components: The RTP wire protocol
[RFC3550] (consisting of the RTP packet format, the RTCP reporting
format, and the handling rules for RTP sessions), and the SDP session
description language. For the purposes of this argument, the SDP
functionality for describing non-RTP sessions is ignored, as is the
ability to negotiate RTP sessions by other means than SDP.
This document argues that the RTP mechanisms of multiple RTP sessions
make sense for a lot of purposes, but does NOT make sense for a
mandated separation between different top-level MIME media types.
2. Stuff that can be carried over RTP
RTP, according to its own description an application layer framework
component, is a suitable protocol for framing data that needs to
travel across the network in a time-sensitive fashion, with the idea
that it is going to be presented at the receiving end in a time
sequence. Normally, the data (usually called "media") is streamed
across the network at a rate approximately equal to the speed at
which it is intended to be presented ("real time data").
Examples of data carried over RTP include:
o G.711 Audio - 64 Kbits/second, completely fixed bitrate
o GSM AMR Audio - 4.75 to 12.2 Kbits/second, variable bitrate
o OPUS audio compressed into near-incomprehensibility - 6 kbits/
second, variable bitrate
o OPUS audio carrying high fidelity music - 500 kbits/second,
variable bitrate
o QQVGA (160x120) video at 15 FPS in H.264 compression - 50 Kbits/
second, variable bitrate, lots of schemes for error concealment
and correction
o HD video at 1920x1080@60 in H.264 compression - 1.4 Mbits/second
o Real-time text (T.140) - very few bits/second
o [RFC4733] DTMF tone signalling - very few bits/second
Schemes designed to increase the reliability of data carried across
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RTP include:
o Forward error correction (FEC)
o Duplicated streams, codec-independent
o Duplicate sending of important information within the codec
o NAK-based resends signalled over RTCP
o Stream reset requests signalled over RTCP
Some of these are only applicable to media types (in particular,
"send me a new I-frame" doesn't make sense if you don't have
I-frames). Others can be used with any type of data.
3. What the network can do to "help" a flow
The network can apply various things to help the session data arrive
according to policy:
o Capacity reservation for specific flows
o Priority queueing, sending certain types of data faster than
others
o Filtering or blocking certain types of communication that the
managers deem inappropriate
The network can do these things in multiple ways, including so-called
"deep packet inspection", but the most common techniques require
being able to identify either the requested handling of the packets
(DiffServ using DSCP codepoints) or recognizing the flow based on its
5-tuple (source and destination address and port + protocol),
possibly correlating the 5-tuple with information carried to the
router through some kind of management interface (either connected to
the session setup protocol or managed via some other interface such
as RSVP/IntServ), and behaving accordingly.
All techniques have limitations; DSCP requires a certain trust in the
endpoints using the codepoints for "deserving traffic"; deep packet
inspection requires that packets be unencrypted, and stream control
requires that 5-tuples be related back to their putative purpose
either by heuristics or by being connected to management protocols.
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4. The definition of an RTP "session"
An RTP session is defined in RFC 3550 section 3:
"RTP Session: An association among a set of participants
communicating with RTP. A participant may be involved in multiple
RTP sessions at the same time. In a multimedia session, each medium
is typically carried in a separate RTP session with its own RTCP
packets unless the encoding itself multiplexes multiple media into a
single data stream. A participant distinguishes multiple RTP
sessions by reception of different sessions using different pairs of
destination transport addresses, where a pair of transport addresses
comprises one network address plus a pair of ports for RTP and RTCP.
All participants in an RTP session may share a common destination
transport address pair, as in the case of IP multicast, or the pairs
may be different for each participant, as in the case of individual
unicast network addresses and port pairs. In the unicast case, a
participant may receive from all other participants in the session
using the same pair of ports, or may use a distinct pair of ports for
each.
The distinguishing feature of an RTP session is that each maintains a
full, separate space of SSRC identifiers (defined next). The set of
participants included in one RTP session consists of those that can
receive an SSRC identifier transmitted by any one of the participants
either in RTP as the SSRC or a CSRC (also defined below) or in RTCP.
For example, consider a three- party conference implemented using
unicast UDP with each participant receiving from the other two on
separate port pairs. If each participant sends RTCP feedback about
data received from one other participant only back to that
participant, then the conference is composed of three separate point-
to-point RTP sessions. If each participant provides RTCP feedback
about its reception of one other participant to both of the other
participants, then the conference is composed of one multi-party RTP
session. The latter case simulates the behavior that would occur
with IP multicast communication among the three participants.
The RTP framework allows the variations defined here, but a
particular control protocol or application design will usually impose
constraints on these variations."
An RTP session is thus characterized by:
o A single SSRC space
o A single reporting space - all participants see all RTCP messages
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o Non overlapping transport addresses
As we can see here, it is not possible to tell from a single packet
whether it belongs to the same session as another packet or not; if
we observe two packets with the same source and destination
addresses, it seems safe to assume that they belong to the same
session, but for all other cases, deciding whether or not two packets
or packet streams are in the same session requires knowledge of the
configuration of the session.
5. Proper and improper use of RTP sessions
Section 5.2 of RFC 3550 gives the canonical statement of RTP session
(mis)use:
"In RTP, multiplexing is provided by the destination transport
address (network address and port number) which is different for each
RTP session. For example, in a teleconference composed of audio and
video media encoded separately, each medium SHOULD be carried in a
separate RTP session with its own destination transport address."
This sentence makes two very important leaps of faith:
o That distinguishing sessions by destination transport address is
necessary and sufficient
o That it is appropriate to give strong guidance about the
distribution of media streams across RTP sessions
Both of these are shaky.
As the cost of connecting ports has increased due to NATs, firewalls
and IPv4 exhaustion, there has been a strong push towards using fewer
ports, and indeed fewer 5-tuples, so that it is not uncommon to see
flows that can be distinguished only by source address; there have
also been proposals floated for putting multiple RTP sessions across
one 5-tuple [draft-westerlund-avtcore-transport-multiplexing].
The cost of ports is also one factor pushing towards multiple media
types in one RTP session; however, the more important underlying
challenge is that this distinction is neither necessary nor
sufficient to distinguish the cases in which RTP media streams want
to have differential treatment from the network, and thus need to
assign streams either to the same session (to guarantee the same
treatment) or to different sessions (to allow for differential
treatment).
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Consider the list of scenarios above, and imagine RTP being used for:
o A videoconference between 3 people who know each other well, using
low end equipment and barely-sufficient bandwidth pipes
o A Berlin Philharmonic concert broadcast featuring Brahms' "Tragic
Overture"
o A point-to-point transmission of a Manchester United vs Liverpool
football match
o A professor's lecture, with "talking head" presentation
simultaneous with slides, and opportunity for students to ask
questions
In each of these contexts, the tradeoff between audio and video is
different; in the Brahms case, the audio (which is the point of the
transmission) is likely to be transmitted at higher bandwidth than
the video, and if one of them has to have his bandwidth reduced, the
video should be reduced in quality before the audio is. In contrast,
in the football match, spectators care about seeing the action; as
long as they can understand the commentator's voice, the audio
quality is "good enough".
In the lecture case, quality of the lecturer's slides and voice is
critical; video from students is almost irrelevant to the larger
purpose.
A logical arrangement of media streams in RTP sessions would be to
group them by importance, and send them with appropriate traffic
engineering structuring; in the lecturer case, the slides and the
professor's voice would be carried in a high priority media stream,
while the professor's picture would have second priority, and voice
and video from students would be made available on an "if it works,
it works" basis. Someone may easily decide that the student feedback
track is not worth listening to, or remove the talking head of the
professor; it would be strange indeed to try to listen to the lecture
without viewing the supporting material.
This illustrates two points:
o The RTP session mechanism, using the 5-tuple as the unit of
differentiation, is a simple, effective and readily deployed
mechanism for separating streams that require different treatment
from the network in easily distinguished partitions.
o The assignment of media to such partitions is application
dependent, and the decision on how to group and how to prioritize
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needs to be taken by the application developer.
6. The Pernicious Effect of SDP on the Media Type System
In the list of reasons to argue against the inappropriate advice
quoted above from RFC 3550, its pernicious influence on the MIME type
system bears mentioning.
The MIME type system, as described in [RFC4288], consists of a two-
level hierarchy: A top level media type (text, audio, video,
application and so on), and a media subtype that identifies (to some
level of precision) the format of the data being carried.
The system has mostly been respected, with some types (for instance
PDF) forever being borderline between the various categories, but
over the years, a few types have been entered into the system with
their top level types being decided, not by the nature of their
content, but by *the type with which their proponents wished to have
them multiplexed in an RTP session*.
This includes the types that designate repair mechanism (audio/
parityfec, audio/red), timed data transfer (audio/clearmode) and that
ultimate triumph of expediency over cleanliness: audio/t140c, audio/
3gpp-tt and video/3gpp-tt: Text types registered as audio and video.
For each of these, there is a fairly natural fit in the normal MIME
hierarchy (application/ for the mechanism types and text/ for the
text types); the assignment of them to the "media" top level types
has been done as an expediency in order to get around the stultifying
results of the advice given in RFC 3550.
7. The Mixer Fallacy
One of the arguments in favour of the RFC 3550 separation has been
that a mixer can be deployed that knows nothing of the semantics of
the media streams; it can "just mix them".
This applies partly to exactly one type of application: The audio
conference bridge.
For a video mixer, it does not apply; external logic (such as
listening to the audio voume of the corresponding audio channel, or
explicit flow control) is needed to select the right video stream to
send out. And even for larger audio bridges, it is common to have
functions like floor control, remote mute and other participant
management tools in order to control the bridge - as soon as such
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tools are introduced, they are as relevant for a multi-media-type RTP
session as they are to a single-media-type RTP session.
8. Corrective Actions
There are not many protocol changes that really need to be taken to
solve this problem.
The basic mechanism of RTP is media type independent. There are some
RTCP issues with dealing with RTP flows of wildly varying bandwidth,
but as can be seen from the table of media types in the introduction,
this issue isn't solved by separating them; the bandwidth ranges of
the types overlap.
The thing that binds most in the current protocol suite is the
conservation of the inappropriate binding in the SDP media
description/negotiation format, where the MIME type is represented in
two pieces, one of which is tied to the RTP session rather than to
the payload type it is associated with, and there are fairly well-
understood ways to get around that, such as the BUNDLE grouping
extension [I-D.ietf-mmusic-sdp-bundle-negotiation]. Better designed
negotiation protocols would not have this problem at all.
In order to get out of the bind that SDP places us in, a change such
as BUNDLE should be adopted, and the IETF should record that the
advice from RFC 3550 is to be considered *advice*, not command: It is
sometimes appropriate to separate media streams according to top
level type, and sometimes not appropriate to do so. The application
is the one that needs to make this decision.
9. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
10. Security Considerations
This note does not discuss any change that the author thinks would
have any significant influence on the security of RTP traffic.
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11. Acknowledgements
This note has benefited greatly from exchanges with Colin Perkins,
whose unwavering support of a sharply differing viewpoint has served
to inform the arguments presented in this document. Magnus
Westerlund and Christer Holmberg also deserve special mention for
engaging constructively in the discussion.
12. References
12.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
12.2. Informative References
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
Using Session Description Protocol (SDP) Port Numbers",
draft-ietf-mmusic-sdp-bundle-negotiation-00 (work in
progress), February 2012.
[I-D.westerlund-avtcore-transport-multiplexing]
Westerlund, M. and C. Perkins, "Multiple RTP Session on a
Single Lower-Layer Transport",
draft-westerlund-avtcore-transport-multiplexing-01 (work
in progress), October 2011.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC4288] Freed, N. and J. Klensin, "Media Type Specifications and
Registration Procedures", BCP 13, RFC 4288, December 2005.
[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF
Digits, Telephony Tones, and Telephony Signals", RFC 4733,
December 2006.
Appendix A. Change log
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A.1. Version -00 to -01
Version number bump, since the debate is ongoing. A few nits fixed.
Added the "Mixer Fallacy" section. Updated reference to "bundle" to
new draft name.
This should be the last version, since the author is in the process
of working with the authors of
[I-D.westerlund-avtcore-transport-multiplexing] to achieve a jointly
agreeable text. Hopefully this will take lesss than 6 months.
Author's Address
Harald T. Alvestrand
Google
Kungsbron 2
Stockholm, 11122
Sweden
Email: harald@alvestrand.no
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