Extensible Bandwidth Attribute for SDP
draft-westerlund-mmusic-sdp-bw-attribute-01
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| Authors | Tomas Frankkila , Magnus Westerlund , Bo Burman | ||
| Last updated | 2012-04-24 | ||
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draft-westerlund-mmusic-sdp-bw-attribute-01
MMUSIC Working Group T. Frankkila
Internet-Draft M. Westerlund
Intended status: Standards Track B. Burman
Expires: October 26, 2012 Ericsson
April 24, 2012
Extensible Bandwidth Attribute for SDP
draft-westerlund-mmusic-sdp-bw-attribute-01
Abstract
Knowledge of what bandwidths the end-points intend to use is
important both for the other end-point and for resource allocation in
various types of networks. This is especially important for wireless
access networks which typically have quite limited resources. The
bandwidth attribute in Session Description Protocol (SDP), 'b=AS', is
today quite widely used to define the bandwidth that the end-points
intends to use, in various types of sessions. This document will
show that the existing bandwidth attribute, such as 'b=AS', although
widely used in todays scenarios, has limitations that make it hard or
even impossible for the end-points to express their intentions
accurately when it comes to bandwidth usage. To solve the identified
problems, this document defines a new extensible SDP bandwidth
attribute 'a=bw' which enables more detailed control over the
bandwidth declarations, request, and allocations. With the new
bandwidth attribute it is possible to define different scopes in the
session setup and then negotiate the bandwidth individually for each
scope.
Status of this Memo
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provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on October 26, 2012.
Copyright Notice
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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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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1. Requirements Language . . . . . . . . . . . . . . . . . . 4
2.2. Terminology . . . . . . . . . . . . . . . . . . . . . . . 5
3. Use Cases and Design Rationale . . . . . . . . . . . . . . . . 5
3.1. Existing Bandwidth Attribute . . . . . . . . . . . . . . . 6
3.1.1. Attribute Definition . . . . . . . . . . . . . . . . . 6
3.1.2. Offer/answer Procedure for the Existing Bandwidth
Attribute . . . . . . . . . . . . . . . . . . . . . . 7
3.1.3. End-point Behavior when Generating Traffic . . . . . . 7
3.2. Point-to-point Sessions using SDP offer/answer . . . . . . 8
3.2.1. Symmetric Point-to-point Sessions, Fixed-rate
Codecs . . . . . . . . . . . . . . . . . . . . . . . . 8
3.2.2. Symmetric Point-to-Point Sessions with
Rate-Adaptive Codec . . . . . . . . . . . . . . . . . 10
3.2.3. Symmetric Point-to-Point Sessions with Several
Rate-Adaptive Codecs . . . . . . . . . . . . . . . . . 11
3.2.4. Asymmetric Point-to-Point Sessions . . . . . . . . . . 12
3.3. Sessions with Multiple Streams . . . . . . . . . . . . . . 14
3.3.1. Multiple Streams . . . . . . . . . . . . . . . . . . . 14
3.4. User Experience and Bandwidth Negotiation . . . . . . . . 15
3.5. Summary of Findings . . . . . . . . . . . . . . . . . . . 15
4. Attribute Specification . . . . . . . . . . . . . . . . . . . 17
4.1. SDP Grammar . . . . . . . . . . . . . . . . . . . . . . . 17
4.2. Declarative Use . . . . . . . . . . . . . . . . . . . . . 21
4.3. Usage in Offer/Answer . . . . . . . . . . . . . . . . . . 21
4.4. Bucket Size Estimation . . . . . . . . . . . . . . . . . . 23
4.4.1. Sender Specified Token Bucket . . . . . . . . . . . . 24
4.4.2. Receiver Specified Token Bucket . . . . . . . . . . . 25
4.4.3. Bucket Adjustment in Middle Nodes . . . . . . . . . . 25
4.4.4. Network Policing . . . . . . . . . . . . . . . . . . . 25
4.4.5. Utilizing Network Feedback . . . . . . . . . . . . . . 26
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4.5. SDP Examples for Point-to-point Sessions . . . . . . . . . 26
4.5.1. Symmetric Fixed-rate Codecs . . . . . . . . . . . . . 26
4.5.2. Symmetric Rate-Adaptive Codec . . . . . . . . . . . . 26
4.5.3. Symmetric Several Rate-Adaptive Codecs . . . . . . . . 27
4.5.4. Asymmetric Session . . . . . . . . . . . . . . . . . . 28
4.5.5. Session with Retransmission . . . . . . . . . . . . . 29
4.6. SDP Examples with Sessions with Multiple Streams . . . . . 29
4.6.1. Multiple Streams . . . . . . . . . . . . . . . . . . . 29
4.6.2. Declarative Example with Stream Asymmetry . . . . . . 30
4.7. Interoperability Issues . . . . . . . . . . . . . . . . . 30
4.7.1. Interoperability with Existing Bandwidth Attribute . . 31
4.7.2. Interoperability with Existing Directional
Attribute . . . . . . . . . . . . . . . . . . . . . . 33
5. Rules and Recommendations for Extensions . . . . . . . . . . . 34
5.1. Directionality . . . . . . . . . . . . . . . . . . . . . . 34
5.2. Scope . . . . . . . . . . . . . . . . . . . . . . . . . . 34
5.3. Semantics . . . . . . . . . . . . . . . . . . . . . . . . 34
5.4. Values . . . . . . . . . . . . . . . . . . . . . . . . . . 34
6. Open Issues . . . . . . . . . . . . . . . . . . . . . . . . . 35
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 35
8. Security Considerations . . . . . . . . . . . . . . . . . . . 35
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 36
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 36
10.1. Normative References . . . . . . . . . . . . . . . . . . . 36
10.2. Informative References . . . . . . . . . . . . . . . . . . 37
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 37
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1. Introduction
This document looks at the issues of non-basic usage of RTP [RFC3550]
and analyzes how well the existing SDP [RFC4566] attribute 'b=AS' for
bandwidth negotiation performs in different scenarios.
This analysis is done by defining a number of use cases, containing
sessions with:
o single and multiple media types;
o symmetric and asymmetric media streams;
o single and multiple media sources, including multiple sources from
the same end-point;
o multiple end-points each having one or more media sources,
including applications that use multiple encodings of a particular
media.
It is shown that the existing bandwidth attributes 'b=AS' [RFC4566]
and 'b=TIAS' [RFC3890] has limitations which make it unclear or even
impossible for end-points and for resource allocation functions in
the network to determine how much bandwidth the service will use.
The analysis also provides the design rationale for the new bandwidth
attribute.
This document then proposes a general and extensible mechanism for
bandwidth negotiation that can be used for any type of session.
Interoperability with the existing mechanisms for bandwidth
negotiation is especially important since the existing bandwidth
attribute has a wide-spread usage.
This document also presents several examples for how the new
bandwidth attribute can be used in the session setup phase for
various types of sessions. The examples are derived for IP/UDP/RTP
transport although nothing should prevent using the new bandwidth
attribute also for other transport protocols.
2. Definitions
2.1. 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].
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2.2. Terminology
The following terms and abbreviations are used in this document:
Bandwidth: In this document, the bandwidth is defined as the IP
level bandwidth, i.e. including the network protocol (IPv4 or
IPv6) and transport protocol (TCP, UDP, RTP, etc) overhead. When
RTP is used then the RTCP bandwidth is handled separately from the
bandwidth used for RTP packets. Bandwidth in this context is in
the unit bits per second, not Hz.
Encoding: A particular encoding is the choice of the media encoder
(codec) that has been used to compress the media. Different
encodings result in the fidelity of that encoding through the
choice of sampling, bit-rate and other configuration parameters.
End-point: A single entity sending and/or receiving RTP packets. It
may be decomposed into several functional blocks, but as long as
it behaves a single RTP stack entity it is classified as a single
end-point.
Media stream: A sequence of RTP packets using a single SSRC that
together carry carries part or all of the content of a specific
Media Type from a specific sender source within a given RTP
session.
RTP session: An RTP session consists of one or more media streams
that have the same purpose. The typical example is to have one
RTP session per media type, i.e. that voice and video use
different RTP sessions (different ports) since they have different
purpose. It is however possible to have multiple streams in an
RTP session, for example when having both a stream for non-
redundant audio and another stream for re-transmissions of audio
packets. The fundamental definition of an RTP session is a single
SSRC space.
3. Use Cases and Design Rationale
This section describes a number of use cases where the existing
bandwidth attribute 'b=AS' is used for bandwidth definition. It also
discusses why the limitations of the existing bandwidth attribute
makes it hard for other end-points and resource allocation functions
to know or estimate how much bandwidth that will be used in the
ongoing session.
The analysis is made by defining a set of use cases. The first use
cases include fairly simple session types, i.e. point-to-point
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sessions with or without asymmetry. A few more complex use cases are
then analyzed. The last set up use cases reflect fairly advanced
session types, e.g. various variants of multiplexing and usage of
multiple media streams.
The discussion is then summarized and the design rationales for the
new bandwidth attribute are outlined.
3.1. Existing Bandwidth Attribute
The existing bandwidth modifier 'b=' defined in RFC 4566 [RFC4566] is
reviewed in this section.
3.1.1. Attribute Definition
The existing bandwidth attribute 'b=' is defined in Section 5.8 of
RFC 4566 [RFC4566]. The syntax is:
b=<bwtype>:<bandwidth>
where:
<bwtype> is either:
'AS' ("Application Specific"), which is the maximum bandwidth
as estimated by the application; or:
'CT' ("Conference Total"), which is the total bandwidth for all
media at all sites.
<bandwidth> is the bandwidth value in kilobits per second.
Bandwidth types have been defined for the negotiation of the RTCP
bandwidth using 'b=RS' and 'b=RR', RFC 3556 [RFC3556].
There is also a bandwidth type for negotiating the transport
independent application specific maximum bandwidth, 'b=TIAS', RFC
3890 [RFC3890]. This bandwidth type is similar to the 'b=AS'
bandwidth type, except that the overhead caused by the transport
protocol headers is not included.
One issue with the existing bandwidth attribute is that the syntax is
very limited since it only allows for defining new bandwidth types
(<bwtype>) and their respective single numerical value. This
limitation needs to be considered in the discussion below.
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3.1.2. Offer/answer Procedure for the Existing Bandwidth Attribute
"An Offer/Answer Model with the Session Description Protocol (SDP)"
[RFC3264] describes the offer/answer procedures for the existing
bandwidth attribute. For the SDP offer, it describes that the
bandwidth attribute indicates the desired bandwidth that the offerer
would like to receive. For the SDP answer, it describes that the
bandwidth attribute indicates the bandwidth that the answerer would
like the offerer to use when sending media. Thus, for offer/answer
negotiations, the bandwidth attribute indicates the bandwidth for the
receive direction of each end-point.
The solution presented in this document focuses primarily on
clarifying and assisting the Application Specific (AS) bandwidth.
[It is an open question to decide if and how to handle the RTCP
bandwidth negotiation, e.g. corresponding to b=RS and b=RR.]
[It is an open question to develop semantics for the transport
independent bandwidth negotiation, e.g. corresponding to b=TIAS.]
3.1.3. End-point Behavior when Generating Traffic
When an end-point is sending media then this can be done in many
different ways, depending on the choices the implementers have made.
Some end-points may send it's data in a fairly "nice and smooth"
media stream, which means that both the packet sizes and the packet
rates are more or less constant all the time. An example of a smooth
stream is when the end-point is encoding speech and is sending one
packet every 20 ms and when the packets are of equal size.
Other end-points may generate bursty streams, which have a large
peak-to-average ratio. An example of a bursty stream is when an end-
point is encoding video. Most of the time, the end-point is sending
packets with almost the same size and with constant packet rate.
However, it happens occasionally that the encoder generates much more
data for a frame, which may give a very large packet size. It may
even happen that the sender has to segment the data into several
packets, which may be transmitted in a burst, thereby causing a very
high peak rate.
Whether the stream is smooth or bursty makes a big difference for the
network and the policy control that usually applies in QoS controlled
networks. If the stream is too bursty, then a policy control
function may decide to drop packets that exceed the granted rate.
This will lead to degraded quality and reduced user satisfaction.
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The existing bandwidth attribute offers no mechanism to negotiate
what temporal variations that can be allowed for a stream. The only
available mechanism is to negotiate the maximum bandwidth, but there
is nothing that defines any kind of averaging window (or something
similar) that can be used to control the bandwidth variations from
the transmitted stream.
It is therefore proposed to use a Token Bucket model to describe the
bandwidth with two parameters, the token bucket rate and the bucket
size, see RFC 2212 [RFC2212].
3.2. Point-to-point Sessions using SDP offer/answer
The existing modifier for the application specific bandwidth 'b=AS'
is frequently used in the SDP offer/answer negotiation RFC 3264
[RFC3264] for setting up point-to-point sessions, for example for bi-
directional point-to-point VoIP or video telephony sessions. In this
section, the use of the legacy bandwidth modifier is reviewed for the
use in point-to-point sessions using SDP offer/answer.
3.2.1. Symmetric Point-to-point Sessions, Fixed-rate Codecs
This example below shows the SDP offer from end-point A for several
fixed-rate codecs, mu-law and A-law PCM/G.711 [G.711], AD-PCM/G.726
[G.726] and CS-ACELP/G.729 [G.729]. The codecs have different bit
rates. PCM encodes speech at 64 kbps. G.726 can encode speech at
four different rates, 64, 32, 24 and 16 kbps, but in this case it is
assumed that the 32 kbps variant is used. G.729 encodes speech at 8
kbps. The IP/UDP/RTP overhead with 20 ms packetization and IPv4
becomes 16 kbps in all cases giving 80, 48 and 24 kbps, respectively.
m=audio 49200 RTP/AVP 8 0 96 18
b=AS:80
a=rtpmap:96 G726-32/8000/1
a=ptime:20
a=maxptime:80
SDP offer for mu-law PCM, A-law PCM, G.726 and G.729 with IPv4
If end-point B accepts to use this codec then a likely SDP answer
would be:
m=audio 49400 RTP/AVP 8 0 96 18
b=AS:80
a=rtpmap:96 G726-32/8000/1
a=ptime:20
a=maxptime:80
SDP answer for mu-law PCM, A-law PCM, G.726 and G.729 with IPv4
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In this case, both end-points offer to receive 80 kbps. A resource
allocation function would thereby allocate 80 kbps in each direction.
However, if end-point B accepts to use one of the lower rate codecs,
for example G.729, but not the PCM codecs, then a likely SDP answer
would be:
m=audio 49400 RTP/AVP 18
b=AS:24
a=ptime:20
a=maxptime:80
SDP answer for G.729 with IPv4
This means that the offerer has offered to receive 80 kbps while the
answerer has offered to receive 24 kbps. In the direction A to B it
is clear that a resource allocation function should allocate 24 kbps.
However, in the direction B to A it is a little more unclear. On one
hand, end-point A has offered to receive 80 kbps. But, on the other
hand, end-point B has only indicated support for the G.729 codec and
its unknown if B can send with something in addition to G.729 from
A's offered set.
A resource allocation may also (incorrectly) conclude that end-point
B will also send maximum 24 kbps, since b=AS indicates 24 kbps. But,
since maxptime is 80 ms, this means that end-point B could very well
use application layer redundancy and encapsulate redundant frames
together with non-redundant frames, which would result in a bandwidth
exceeding 24 kbps. Even if maxptime would be 20 ms, end-point B
could still use application layer redundancy, if the non-redundant
and redundant frames are transmitted in different packets. This is
possible since end-point A has indicated that it is capable of
receiving 80 kbps. Hence, if the resource allocation function uses
the codec information and assumes that end-point B will send with
only 24 kbps, then this may cause packet losses and/or long delays.
It should be clear with this example that the current bandwidth
attribute, b=AS, can create ambiguities related to what bandwidth
that will be used in each direction. If the end-points and the
resource allocation functions make different interpretations then
there is a risk for either poor quality or wasted resources.
To solve this, a new bandwidth negotiation method should enable
negotiating different bandwidths for different codecs. If a codec
can be configured in several different ways, e.g. G.726 offers the
possibility to use four different static bit rates then this would
typically be negotiated using different RTP Payload Types. This
means that the solution needs to be capable of negotiating different
bandwidths for different Payload Types.
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3.2.2. Symmetric Point-to-Point Sessions with Rate-Adaptive Codec
This use case describes what might happen when using rate-adaptive
codecs in a session, for example AMR [AMR]. The rate adaptation
should adapt to a high bitrate when the operating conditions are
good, but should adapt to a low bitrate when the operating conditions
are degraded, e.g. due to congestion or bad coverage.
One example of the SDP offer-answer negotiation for rate-adaptive
codec is shown below.
m=audio 49200 RTP/AVP 97
b=AS:29
a=rtpmap:97 AMR/8000/1
a=fmtp:97 mode-change-capability=2; max-red=80
a=ptime:20
a=maxptime:100
SDP offer from end-point A for AMR and IPv4
The bandwidth attribute in the SDP indicates the bandwidth that the
offerer would like to receive, RFC 3264 [RFC3264].
m=audio 49100 RTP/AVP 97
b=AS:29
a=rtpmap:97 AMR/8000/1
a=fmtp:97 mode-change-capability=2; max-red=80
a=ptime:20
a=maxptime:100
SDP answer from end-point B also for AMR and IPv4
The bandwidth attribute in the SDP answer indicates the maximum
bandwidth that the answerer would like the offerer to use when
sending media, RFC 3264 [RFC3264].
In this case, it is clear that both end-points are prepared to
receive up to 29 kbps of media. Since AMR can adapt the rate for the
encoding, this means that the bandwidth can be reduced, e.g. to the
5.9 kbps mode, if congestion is detected. The existing bandwidth
attribute 'b=AS" is however only used to negotiate the maximum rate.
This means that there is nothing in the SDPs that describes how the
rate will be adapted. In some cases, usually for speech codec, it
might be possible to derive the lowest rate from the codec
information. However, there is no guarantee that the end-points will
adapt to this rate or whether it will stay at some higher rate. For
video codecs, there is usually no codec information at all that could
be used to determine how low rate the end-points will use. The
lowest usable rate for a video codec is generally not a video codec
limitation, but rather some end-user or service consideration on what
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is the lowest video quality that is still useful or acceptable in the
actual scenario.
This means that a resource allocation function has no information
which could be used to determine how the end-points will adapt during
periods of congestion. Hence the network does not know what to
assume from the end-points.
To solve this, a new bandwidth negotiation method should allow for
negotiating not only the highest rate but also the minimum rate that
is still useful.
3.2.3. Symmetric Point-to-Point Sessions with Several Rate-Adaptive
Codecs
Another example is when the originating end-point offers several
rate-adaptive codecs, with different bandwidths, and when the
answerer only support one or several of the lower-rate configurations
but not the configuration that uses the highest bandwidth. With the
legacy bandwidth modifier 'b=AS' it is only possible to indicate one
bandwidth for the whole RTP session, which means that the end-point
needs to indicate the highest bandwidth since this is the worst-case
scenario. An offer/answer for this case is shown below. The offerer
supports both AMR and AMR-WB AMR-WB [AMR-WB] and therefore indicates
the bandwidth needed for the AMR-WB configuration since it is higher
than for AMR. If the answerer does not support the AMR-WB codec then
it will have to remove this configuration from the SDP when creating
the SDP answer. This means that the answerer calculates the
bandwidth required for AMR instead of AMR-WB.
m=audio 49200 RTP/AVP 96 97
b=AS:41
a=rtpmap:96 AMR-WB/16000/1
a=fmtp:96 mode-change-capability=2; max-red=80
a=rtpmap:97 AMR/8000/1
a=fmtp:97 mode-change-capability=2; max-red=80
a=ptime:20
a=maxptime:100
SDP offer from end-point A for AMR-WB, AMR and IPv4
m=audio 49100 RTP/AVP 97
b=AS:29
a=rtpmap:97 AMR/8000/1
a=fmtp:97 mode-change-capability=2; max-red=80
a=ptime:20
a=maxptime:100
SDP answer from end-point B for AMR and IPv4 (AMR-WB is removed)
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Since the indicated bandwidth is for the receiving direction in this
example this means that:
o A must send media with a bandwidth not exceeding 29 kbps; and:
o B must send media with a bandwidth not exceeding 41 kbps.
This gives the same problem with ambiguous maximum rate as shown in
Section 3.2.1. In addition, since both AMR and AMR-WB are rate-
adaptive codecs, with different bit rates, they also have different
minimum rates. This means that a resource allocation would be
unaware about both the maximum bandwidth and the minimum (required)
bandwidth.
To solve this, a new bandwidth attribute should allow for negotiating
both maximum and minimum bitrates individually for each payload type.
For speech codecs, it is usually possible to derive the minimum rate
from the codec information. However, this is typically not possible
for video codecs since they only indicate the maximum encoding level.
For example, if end-point A offers to use H.264 level 3.0 H.264
[H.264] but end-point B is only capable of using level 1.2, then this
only limits the maximum bandwidth in the direction from A to B. In
the other direction, end-point A is still capable of receiving level
3.0.
3.2.4. Asymmetric Point-to-Point Sessions
The session setup for asymmetric streams is not always straight
forward. Lets say that one want to set up a session with 600 kbps in
the sending direction and 200 kbps in the receiving direction.
m=video 49200 RTP/AVP 96
b=AS:200
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42c00c
a=sendrecv
SDP offer to receive 200 kbps video
From this SDP, it can be determined that the end-point wants to
receive 200 kbps. There is some implicit information in the level
part of the profile-level-id for the H.264 example above, indicating
that the end-point can send using a higher bandwidth (up to 768
kbps), but it requires codec-specific knowledge to be able to extract
that implicit information. In this example, lets assume that the
sender does not even want to utilize the maximum allowed bandwidth
for the signaled level, but a slightly lower one, say 600 kbps. So
how is the answerer supposed to know that the offerer really wants to
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send up to 600 kbps, especially since not even the implicit level-
related can be used? There could be many reasons to use a lower
video bandwidth than the one defined as level maximum; limited
terminal performance in the send direction, a known network bandwidth
limitation, a bandwidth charging model that makes the user prefer a
lower bandwidth, etc.
One way to express the asymmetry is to set up different RTP sessions
for sending and receiving directions. An SDP offer for this might
be:
m=video 49200 RTP/AVP 96
b=AS:600
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42c00d
a=sendonly
m=video 49202 RTP/AVP 97
b=AS:200
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42c00c
a=recvonly
SDP offer with separate sessions for send and receive
If the answerer decides to accept this then the SDP answer might be:
m=video 49200 RTP/AVP 96
b=AS:600
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42c00d
a=recvonly
m=video 49202 RTP/AVP 97
b=AS:200
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42c00c
a=sendonly
SDP answer with separate sessions for send and receive
In this example, it is clear that the offerer can send video with 600
kpbs and receive video with up to 200 kbps. However, if the offer is
for different codecs, using different bandwidths, then one have the
same problem as described in Section 3.2.3.
Specifically for video, but possibly also for other media, it may
happen that different implementations send the media in different
ways. Some implementations may try to provide a fairly "smooth"
stream in terms of bandwidth variation over time, while other
implementations may give a very "bursty" stream.
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There also exist cases where opening additional RTP sessions just for
expressing asymmetric transmission bandwidths are not desirable.
3.3. Sessions with Multiple Streams
In this part of the analysis, it is assumed that an RTP session is
set up for multiple streams. This can be done in several ways and
for several reasons, as discussed in RTP Multiplexing Architecture
[I-D.westerlund-avtcore-multiplex-architecture].
3.3.1. Multiple Streams
The assumed usage here is a multi-party session, for example a video
conference using an RTP mixer. Some of the attendees are active and
their audio and video is distributed to the other users. Some
attendees are inactive and thus only receive media. In this example,
each end-point sends one video stream, but can receive up to four
simultaneous video streams, multiplexed as different SSRC in the same
RTP session. One or more central nodes (RTP Mixer) are used to help
facilitate the media transport between the participants, and are
involved in choosing the streams to be forwarded. Assume that there
is an aggregate bandwidth limit of 3 Mbps in the receive direction,
and that each received video stream should be limited to max 1 Mbps.
An SDP offer for the setting up a session with one video stream for
the sending direction and four video streams for the receiving
direction is shown below when using [I-D.westerlund-avtcore-max-ssrc]
to explicitly declare capability to handle multiple streams. In this
case, only the legacy 'b=AS' bandwidth attribute is used, valid only
for the aggregate.
m=video 49300 RTP/AVP 96
b=AS:3000
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42c016
a=max-recv-ssrc:* 4
SDP offer to receive multiple video streams
This example again highlights the asymmetry problem with the existing
bandwidth attribute, but it also highlights the lack of per-stream
bandwidth specification. This means that it is not possible to
declare the 1 Mbps bandwidth limit that should be used for each one
of the for streams in the receiving direction, which thus is a
desirable property of the new bandwidth attribute. Note also that in
this example, the 1 Mbps limit per stream cannot be fully utilized if
all four streams are used simultaneously.
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3.4. User Experience and Bandwidth Negotiation
Resource allocation is typically a compromise between perceived
quality and network utilization. From an end-user perspective, the
bandwidth for a service should be as high rate as possible, since
this should give the best user experience. However, from a network
perspective, one would like to minimize the rate, since this should
maximize the number of sessions that can be supported.
For some services, like conversational voice- and/or video-telephony,
one needs to ensure that the network is capable of delivering a
certain at least required rate, even when the network load is high.
This is needed to ensure user satisfaction, both in terms of quality
and end-to-end delay. This means that the end-points and the network
need to agree on what maximum bandwidth that can be used for the
session as well as some lowest useful "least required" bandwidth.
The current bandwidth modifier, 'b=AS', is used to negotiate the
maximum bandwidth. However, since it only allows for negotiate one
bandwidth it cannot be used to also negotiate a lower bandwidth
limit.
To solve this, a new bandwidth negotiation method should allow for
negotiating not only the highest rate but also the "at least
required" rate. To enable a negotiation between the end-point and
the network, a reasonable approach is that the end-point requests a
lower bandwidth limit and then the network indicate what "least
required" rate that was granted.
3.5. Summary of Findings
It should be clear from the above discussion that the current
bandwidth attribute is too limited to be used for all use cases and
that some extensions are needed.
The current bandwidth attribute, 'b=AS', is sufficient for simple
sessions but gives ambiguities when negotiating more advanced session
types. One of the drawbacks is that 'b=AS' only indicates the
desired bandwidth for the receiving direction but, if the answering
end-point wants to use a lower rate than what is offered, then there
is often no way for the resource allocation function to know what
bandwidth that will be used in the offerer's sending direction.
Implementers of end-points and resource allocation functions may try
to resolve this ambiguity by using other information available in the
SDP, e.g. codec-specific information. However, such information is
not always easily available, e.g. for video codecs.
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End-points may have to perform a second offer/answer negotiation to
resolve the ambiguity. This, obviously, has the drawbacks that the
SIP traffic is increased and that this takes some extra time. It is
also not guaranteed that the end-points will actually initiate a
second offer/answer negotiation.
The analysis above has also shown that the current bandwidth
attribute is insufficient to properly describe the session for multi-
stream scenarios.
The analysis above has also shown that the current bandwidth modifier
can be used to negotiate the maximum bit rate in bearers allocated in
some wireless networks, but it is insufficient for also negotiating a
lower, "least required", bandwidth limit.
Another problem with the existing bandwidth attribute is that the
syntax is very limited and does not allow for introducing extensions,
only additional identifiers with a single value each.
It is therefore proposed to define a new bandwidth attribute,
including a new syntax. The new bandwidth attribute should support:
Directionality: One need to be able to have different sets of
attributes values depending on direction.
Payload specific: With the new bandwidth attribute it should be
possible to specify different bandwidth values for different RTP
Payload types. This is because some codecs have different
characteristics and one may want to limit a specific codec and
payload configuration to a particular bandwidth. Especially
combined with codec negotiation there is a need to express
intentions and limitations on usage for that particular codec. In
addition, payload agnostic information is also needed.
Multiple streams: The new bandwidth attribute should support
bandwidth negotiation both for single streams and for multiple
streams. When multiple streams are used, the new bandwidth
attribute should allow for declaring both the bandwidth per stream
and the aggregated bandwidth.
Bandwidth specification method: To have a clear specification of
what any bit-rate values mean we propose that Token bucket
parameters should be used, i.e. bucket depth and bucket fill rate,
where appropriate for the semantics. If single values are to be
specified, a clear definition on how to derive that value must be
specified, including averaging intervals etc.
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Bandwidth semantics: It should be possible to negotiate different
types of bandwidths for each scope, including several bandwidth
properties in the same negotiation. It should, at least, be
possible to negotiate the highest bandwidth and a lower bandwidth
limit that indicates the lowest useful bandwidth to use the
related media. The least required bandwidth limit should ideally,
but need not necessarily, be guaranteed by the network and the
remote end-point(s).
Extensibility: The semantics need to be extensible, so that new
semantics can be defined in the future.
The existing bandwidth modifier, 'b=AS', is widely used today. The
existing SDP attributes for directionality, 'a=sendrecv',
'a=recvonly', 'a=sendonly' and 'a=inactive', are also widely used.
It is therefore important to ensure interworking between the new
bandwidth attribute and the mechanisms already existing in SDP.
4. Attribute Specification
This section proposes a new bandwidth attribute 'a=bw' that can be
used either as an extension to the already existing bandwidth
attribute 'b=AS' or replacing the existing bandwidth attribute. The
new bandwidth attribute includes semantics that allows for also
replacing the existing bandwidth attribute.
The syntax for the new bandwidth attribute is:
a=bw:<direction> <scope> <semantic>:<value>
where:
<direction> is the direction in which the scope and semantics
applies,
<scope> describes for what scope the definitions applies,
<semantic> is the actual bandwidth specification,
<value> in the form defined by the semantic used.
The new attribute is designed to allow for future extendability.
4.1. SDP Grammar
The ABNF RFC 5234 [RFC5234] for this attribute is the following:
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bw-attrib = "a=bw:" direction SP [req] scope SP
[req] semantics ":" values
direction = "send" / "recv" / "sendrecv" / direction-ext
scope = payloadType / scope-ext
payloadType = "pt=" ("*" / (PT-spec) *("," PT-spec))
PT-spec = PT-value / PT-value-range
PT-value = 1*3DIGIT
PT-value-range = PT-value "-" PT-value
req = "!"
semantics = "SMT" / "AMT" / "SLT" / "SLTR" / "ALT" / "ALTR" /
semantics-ext
values = token-bucket / value-ext
token-bucket = "tb=" br-value ":" bs-value
br-value = "*" / 1*15DIGIT ; Bucket Rate [bps]
bs-value = "*" / 1*15DIGIT ; Bucket Size [bytes]
direction-ext = token ; As defined in RFC 4566
scope-ext = 1*VCHAR ; As defined in RFC 5234
semantics-ext = token ; As defined in RFC 4566
value-ext = 0*(WSP / VCHAR) ; As defined in RFC 5234
ABNF for 'bw' attribute
The 'a=bw' attribute defines three possible directionalities for the
bandwidth:
send: In the send direction for SDP Offer/Answer agent or in case of
declarative use in relation to the device that is being configured
by the SDP.
recv: In the receiving direction for the SDP Offer/Answer agent
providing the SDP or in case of declarative use in relation to the
device that is being configured by the SDP.
sendrecv: The provided bandwidth values apply equally in send and
receive directions, i.e. the values configures the directions
symmetrically.
The directionality must be specified when the 'a=bw' attribute is
used. Only one directionality can be specified on each 'a=bw' line.
Special care must be taken to avoid conflicting definitions. For
example, if 'sendrecv' has been specified on one 'a=bw' line for a
scope, e.g. payload number 96, then the direction cannot be set to
'send' or 'recv' on another 'a=bw' line for the same scope. However,
it is allowed to specify directionality 'send' on one 'a=bw' line for
a scope and directionality 'recv' on another 'a=bw' line. This is
useful when the bandwidth is different in different directions.
Using 'sendrecv' as directionality on an 'a=bw' line is a shortcut in
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the sense that it is equivalent to using two separate 'a=bw' lines
where one uses 'send' and the other 'recv' but that otherwise are
semantically identical.
The scope indicates what is being configured by the bandwidth
semantics on this attribute line. Two different scopes are defined
based on payload type:
Payload Type: The bandwidth configuration applies to the specific
payload type value(s).
pt=*: Applies to all payload types being used.
Using pt=* indicates that the definitions apply to all payload
types being used. The scope may be a single payload type value,
e.g. pt=96. A list of payload type values can be created by using
a comma-separated list, e.g. pt=96,98,105. It is also possible to
specify a range of payload type values, e.g. pt=96-102, which
means that the definitions apply to all the payload type numbers
from 96 to 102. It is also possible to combine payload type
values, payload type lists and payload type ranges, e.g. pt=96,98-
102,104,105,110-113.
The scope parameter is extensible to allow for adding other scope
definitions in the future.
This specification defines six related semantics. All semantics
represent either the bandwidth consumption of a single stream or the
aggregate of streams as a token bucket defining a transmission
profile which the media sender must stay within. The token bucket
values are the token rate in bits per second and the bucket size in
bytes both provided as integers, see RFC 2212 [RFC2212]. The below
semantics includes the whole IP packet, for example IP, UDP, RTP
headers and RTP payload, as what shall be metered when determining if
the send pattern is within the profile. The token bucket definition
allows for wild cards enable to specify that one want a value as
token bucket, but has no proposed value.
The definitions of the semantics in more detail are:
SMT (Stream Maximum Token bucket): The maximum intended or allowed
bandwidth usage, including protocol overhead, for each individual
source (each SSRC) in an RTP session at the sender side specified
by a token bucket. The token bucket wild cards ("*") should not
be used for the SMT semantics since it should always be possible
to estimate the maximum bandwidth. This semantics is possible to
use with the scope for any payload type (pt=*) where it applies
independent of encoding and packetization, or for a specific or a
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set of payload type(s).
AMT (Aggregate Maximum Token bucket): The maximum intended or
allowed bandwidth usage for the sum of all sources (SSRCs) in an
RTP session according to the specified directionality at the media
sender specified by a token bucket. The 'sendrecv' directionality
parameter indicates equal token buckets in both directions, i.e.
the aggregate of streams sent to an end-point shall be within the
token bucket defined transmission profile, and the aggregate of
streams sent from that end-point shall also be within the same
token bucket profile at the sender. It can be used either to
express the maximum for one particular payload type, for a set of
payload types or for any payload type (pt=*). The token bucket
wild card ("*") should not be used for the AMT semantics since it
should always be possible to estimate the maximum bandwidth.
SLT (Stream Least required Token bucket): The least required
bandwidth, including IP protocol overhead, needed for the stream
for each individual source (each SSRC) in an RTP session as
specified by a token bucket at the sender. When using the SLT
semantic, the SMT semantic SHOULD also be specified for the same
direction and scope. If the SLT semantics is not defined then
this means that the least required bandwidth limit is zero. The
least required bandwidth is the minimum bandwidth that is
necessary for the service to work with usable quality.
SLTR (Stream Least required Token bucket Request): The request for
establishing the least required bandwidth, including protocol
overhead, needed for the stream for each individual source (each
SSRC) in an RTP session, as specified by a token bucket at the
stream sender. An end-point may use the SLTR semantics to request
to establish a least required bandwidth. An end-point using the
SLTR semantics may set the token bucket rate and/or the token
bucket size to "*" to indicate that the end-point has no
preference, but that it expects some network node or the answering
end-point to define the value(s). A network node answering to the
SLTR SHALL replace this with the SLT semantics to indicate the
least required bandwidth it sees necessary and which it has
attempted to guarantee. If the request is for certain specified
payload types, a network node that cannot grant bandwidth based on
payload types MAY replace those requested payload types with "*"
in the SLT response to indicate a payload type agnostic grant. An
end-point receiving an SDP with SLTR, i.e. where the network has
not replaced the SLTR semantics with any SLT semantics, SHOULD NOT
assume that the requested bandwidth is guaranteed.
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ALT (Aggregated Least required Token bucket): The least required
bandwidth, including protocol overhead, needed for the sum of all
sources (all SSRCs) in an RTP session as specified by a token
bucket at the stream sender. When using the ALT semantic the AMT
semantic SHOULD also be specified for the same direction and
scope. The directionality and payload type considerations for ALT
are the same as for AMT. If the ALT semantics is not defined then
this means that the least required bandwidth is zero.
ALTR (Aggregated Least required Token bucket Request): The request
for establishing a least required bandwidth, including protocol
overhead, needed for the sum of all sources (all SSRCs) in an RTP
session as specified by a token bucket at the media sender side.
The directionality and payload type considerations for ALTR are
the same as for SLTR. The ALTR semantics MUST only be used
together with AMT.
The SMT and AMT semantics, with or without SLT and ALT
respectively, may be used both symmetrically and in a particular
direction. They can be used either to express the maximum (and
minimum) for one particular payload type, for a set of payload
types or for any payload type (pt=*).
The required prefix ("!") is used when the direction, scope and
semantics is required be supported and understood by the SDP
consuming end-point.
4.2. Declarative Use
In declarative usage the SDP attribute is interpreted from the
perspective of the end-point being configured by the particular SDP.
An interpreter MAY ignore 'a=bw' attribute lines that contains
unknown scope or semantics that does not start with the required
("!") prefix. If a "required" prefix is present at an unknown scope
or semantics, the interpreter SHALL NOT use this SDP to configure the
end-point.
4.3. Usage in Offer/Answer
The offer/answer negotiation is performed for each 'a=bw' attribute
line individually with the scope and semantics immutable.
An offerer may use the 'a=bw' attribute(s) for some or all of the
offered media types. An answerer may remove the 'a=bw' attribute(s)
for the media types where it was used in the SDP offer.
The SDP may include an offer for an Aggregated Maximum Token bucket
(AMT) without specifying any Stream Token Buckets (SMTs) for any
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individual streams.
When using the 'a=bw' attribute to define the token bucket for a
certain scope then the offerer should define token buckets for all
scopes of the same type. For example, if the SDP offer includes
three payload types, e.g. 96, 97 and 98, and if a token bucket is
defined for payload type 96, then the offerer should also define
token buckets for the other payload types. This can be done either
by defining one token bucket each for payload type 97 and 98 or by
defining a common token bucket for payload type 97 and 98.
When the token bucket rate and size are declared in an offer for
directionality 'sendrecv' then this indicates the token bucket rate
and the token bucket sizes are the same in both directions. For
example, if the offered bandwidth is 1 Mbps, then the end-point
declares that it is capable of sending with a bandwidth up to 1 Mbps
and that it is capable of receiving with a bandwidth up to 1 Mbps.
If either the token bucket rate(s) or the token bucket sizes are
different in sending and receiving direction then 'sendrecv' cannot
be used. One should instead include two or more 'a=bw' lines with
the respective directionality, bandwidths and sizes.
When the token bucket parameters are declared in an SDP offer for
directionality 'send' then this indicates the token bucket parameters
the sender intends to use. The answerer may change this value, both
to increase it and to reduce it, see below.
When the token bucket parameters are declared in an SDP offer for
directionality 'recv' then this indicates that the largest envelope
for the token bucket parameters that the offerer thinks the media
sender shall use.
An agent understanding the 'a=bw' attribute and answering to an offer
including the 'a=bw' attribute SHOULD include the attribute in the
answer for all media types for which it was offered.
An answerer SHOULD ignore 'a=bw' attribute lines that contains
unknown scope or semantics that does not contain the required ("!")
prefix. If a "required" prefix is present at an unknown scope or
semantics, then the answerer SHALL reject the media description by
setting the port to 0 and copy the 'a=bw' attributes not understood
in the answer. In this case, 'a=bw' attributes that are understood
SHALL NOT be included in the answer.
If an answerer would like to add additional bandwidth configurations
using other directionality, scope, and semantics combination, then it
MAY do so by adding such definitions in the SDP answer.
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An agent may also divide an 'a=bw' offer into several 'a=bw' offers.
One example is when the SDP offer included an 'a=bw' offer with
directionality 'sendrecv', which indicates that the token bucket
parameters are the same in sending and receiving direction. If the
answerer would like to change the parameters for one or both
directions, so that the parameters are no longer the same for both
directions, then the answerer can include two 'a=bw' lines in the SDP
answer, one for sending direction and another for receiving
direction. In case an offered sendrecv media becomes a single
direction media then the sendrecv can be modified to that single
direction.
An agent responding to an offer will need to consider the
directionality and reverse them in the answer when responding to
media streams using unicast.
For media stream offers over unicast with directionality send, the
answerer SHALL reverse the directionality and indicate its reception
bandwidth capability, which may be lower or higher than what the
sender has indicated as its intended maximum.
For media stream offers over unicast with directionality receive, the
token bucket parameters indicate the upper limits. The answerer
SHALL reverse the directionality and may reduce the bandwidth when
producing the answer indicating the answerer intended maximum
transmission rate.
If the answerer removes one or several RTP Payload Types from the SDP
when creating the SDP answer then the corresponding 'a=bw' lines
SHOULD be removed as well. The answerer MAY however keep an 'a=bw'
line when the removed RTP Payload Type number is included within an
identified range or list of Payload Type numbers.
4.4. Bucket Size Estimation
In SDP bandwidth terms, the bucket size is a new parameter and what
value to use for it may be hard to understand for implementers of
this specification. This section therefore gives some guidelines on
how to set bucket size values.
A token bucket specifies an envelope for a transmission profile where
individual measurements have some impact if the media stream or
aggregate should be considered within the specified profile. The
semantics defined in this document only require that the media stream
is within the token bucket specification at the point emitting it
into the network. The network may add jitter causing the media
stream/aggregate to no longer be within the specified token bucket
profile.
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4.4.1. Sender Specified Token Bucket
A sender SHOULD base the choice of token bucket size on how it plans
to send data. That can in turn be decided from e.g. codec
configuration, intended number of encoded frames per packet (ptime),
network interface, maximum transmission unit (MTU), etc. In
practice, for the simplified case where the sender is designed to
send all packets with precisely even time spacing, the token bucket
size can be set to the maximum packet size and the bit-rate to the
long term highest bit-rate intended to be used.
However, for media streams that are more variable the bucket
parameters should be chosen so that the emitted traffic is not too
bursty measured over a shorter interval. Until the bucket is
drained, the media sender will be able to emit packets at or close to
the interface's maximum bit-rate. Long burst of packets at interface
speed becomes more sensitive to loss due to cross-traffic in
switching fabrics with small buffers. Due to this, a sender can
consider transmission scheduling to a rate lower than the interface
rate but higher than the token bucket average rate.
Let's consider the example of a large video intra frame consisting of
10 full MTU (let's assume 1500 bytes) packets which is 5 times the
size of the median frame size of two full MTU packets. The average
bit-rate may be 1 Mbps. If the token bucket was to be configured to
(1 Mbps, 1500) then that would imply that a new full MTU packet could
be emitted no more often than one packet every 12 ms. That would
require 120 ms to transmit the intra frame, which for a 25 frames per
second video is 3 frame intervals. Thus potentially inducing
significant playout jitter at a receiver. A token buffer
specification of (1 Mbps, 15000) would allow all 10 packets be sent
up to line speed. This could result in them being emitted every 1.2
ms over a 100 Mbps interface if there is no competing traffic. To
ensure that a 10 packet burst should be possible to transmit within
one frame interval of 40 ms, then the bucket depth needed is burst
size in bits, minus time interval times bucket fill rate, and the
resulting value converted back into bytes: (15000*8-0.04*1M) / 8 =
10000 bytes. The average bit-rate for this intra frame over a single
frame period becomes 4 Mbps. So the question is if bursts up to 4
Mbps should be allowed now and then as long as the average is within
1 Mbps, or if the sender has to transmit the intra using several
frame intervals, skipping the next frame(s) and hoping that the
receiver doesn't drop the intra frame as being too late. The sender
could also consider reducing the quality of the intra frame,
resulting in a reduced number of MTU required to transmit it.
A sender SHOULD avoid adding excessive safety margins to the sending
bucket size. A sender MAY add bucket size margins if it has
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knowledge of internal transmission timing variations, or if it knows
about packet handling outside the sender itself that will affect the
effective bucket size (as seen from a receiver) that is otherwise not
reflected in the conveyed bucket size figure.
4.4.2. Receiver Specified Token Bucket
With the semantics specified in this document, the intended media
receiver gets to provide token bucket parameters that specifies how
the sender should behave. The traffic received by the receiver (or
intermediate nodes) may no longer conform to the token bucket due to
jitter introduced by the network path between the sender and the
receiver. This document assumes that the receiver will have receiver
buffers for de-jittering that are significantly larger than the token
bucket parameters. This due to that a media unit like a video frame
may be transmitted over time using more data than the bucket depth
provides and instead spread it in time, transmitting each fragment
when the bucket is refilled enough for the next fragment to be sent.
A receiver's input to the sender's bit-rate limitation should be
based on known limitations such as the networks, decoding
capabilities etc. The bucket depth will control how bursty the
traffic can be beyond the long term average specified by the bucket
refill rate.
4.4.3. Bucket Adjustment in Middle Nodes
When there are media aware middle nodes on the media path between the
sender and receiver, those middle nodes may have to or want to apply
similar considerations as the original media sender and receiver. If
those middle nodes are aware of SDP and the new bandwidth attribute
from this specification, and have in-path SDP adjustment
capabilities, they could benefit from modifying the values to better
fit the actually available end-to-end media path capabilities. For
example, an RTP Media Translator can express what it actually is
going to deliver of the far end-point's media to an end-point instead
of that far end-point's provided values.
4.4.4. Network Policing
As the token bucket specified for the semantics in this document is
based on what the sender emit into the network, a policer should have
some margin allowing for network introduced jitter. The amount will
of course be dependent on the policer's location in relation to the
media sender.
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4.4.5. Utilizing Network Feedback
If the media uses RTP and when the media has been transmitted for
some time, the sender should have received a fair amount of RTCP
receiver reports from the receiver. The sender can from RTCP
estimate the observed network jitter at the receiver and may be able
to dynamically adjust the sender behavior such that the aggregate of
the sender behavior and the reported network jitter are fulfilling
the senders token bucket profile.
4.5. SDP Examples for Point-to-point Sessions
These SDP examples show how the new bandwidth attribute can be used.
The benefits, compared to the legacy bandwidth attribute, are also
highlighted.
The SDP examples included below are intentionally not complete. Only
the parts that are relevant for this description are included.
4.5.1. Symmetric Fixed-rate Codecs
This example shows the SDP offer for several fixed-rate codecs, mu-
law and A-law PCM, G.726 and G.728.
m=audio 49200 RTP/AVP 8 0 96 18
b=AS:80
a=rtpmap:96 G726-32/8000/1
a=bw:sendrecv pt=0,8 SMT:tb=80000:1000
a=bw:sendrecv pt=96 SMT:tb=48000:1000
a=bw:sendrecv pt=18 SMT:tb=24000:1000
a=ptime:20
a=maxptime:20
SDP offer for mu-law and A-law PCM and IPv4
The new bandwidth attribute offers the possibility to negotiate the
bandwidth individually for each codec. If the answerer removes a
codec when creating the answer then it is still known how much
bandwidth the other codecs will use. This means that the ambiguities
listed in Section 3.2.1 can be avoided.
4.5.2. Symmetric Rate-Adaptive Codec
This example shows the SDP negotiation for offering using the AMR
codec, AMR [AMR].
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m=audio 49200 RTP/AVP 97
b=AS:29
a=rtpmap:97 AMR/8000/1
a=fmtp:97 mode-change-capability=2; max-red=80
a=bw:sendrecv pt=97 SMT:tb=28800:200
a=bw:sendrecv pt=97 SLTR:tb=22400:200
a=ptime:20
a=maxptime:100
SDP offer from end-point A for AMR and IPv4
m=audio 49100 RTP/AVP 97
b=AS:29
a=rtpmap:97 AMR/8000/1
a=fmtp:97 mode-change-capability=2; max-red=80
a=bw:sendrecv pt=97 SMT:tb=28800:200
a=bw:sendrecv pt=97 SLT:tb=22400:200
a=ptime:20
a=maxptime:100
SDP answer from end-point B also for AMR and IPv4
Since the new bandwidth attribute offers a possibility to negotiate
both the maximum and the at least required bandwidth, it is possible
for both the other end-point and any resource allocation function to
know how the end-points will adapt when congestion is detected.
4.5.3. Symmetric Several Rate-Adaptive Codecs
This example shows how the new bandwidth attribute, 'a=bw', can be
used to negotiate the maximum and the least required bandwidths for
multiple rate-adaptive codecs, in this case for AMR and AMR-WB,
AMR-WB [AMR-WB]. For AMR, the highest codec mode is 12.2 kbps,
giving a maximum bandwidth of 28.8 kbps, and the at least required
mode is selected to be 5.9 kbps, giving a least required bandwidth of
22.4 kbps. For AMR-WB, the highest codec mode is 23.85 kbps, giving
a maximum bandwidth of 40.4 kbps, and the least required mode is 8.85
kbps, giving a least required bandwidth of 25.6 kbps.
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m=audio 49200 RTP/AVP 96 97
b=AS:41
a=rtpmap:96 AMR-WB/16000/1
a=fmtp:96 mode-change-capability=2; max-red=80
a=rtpmap:97 AMR/8000/1
a=fmtp:97 mode-change-capability=2; max-red=80
a=bw:sendrecv pt=96 SMT:tb=40400: 350
a=bw:sendrecv pt=96 SLTR:tb=25600:350
a=bw:sendrecv pt=97 SMT:tb=28800:200
a=bw:sendrecv pt=97 SLTR:tb=22400:200
a=ptime:20
a=maxptime:100
SDP offer from end-point A for AMR-WB, AMR and IPv4
m=audio 49100 RTP/AVP 97
b=AS:29
a=rtpmap:97 AMR/8000/1
a=fmtp:97 mode-change-capability=2; max-red=80
a=bw:sendrecv pt=97 SMT:tb=28800:200
a=bw:sendrecv pt=97 SLT:tb=22400:200
a=ptime:20
a=maxptime:100
SDP answer from end-point B for AMR and IPv4 (AMR-WB is removed)
In this case, it is clear when the answer is received that the
bandwidth needed for AMR applies to both directions. There is no
need for a send offer/answer negotiation to clarify that the
bandwidth applies also to end-point A's receiving direction.
Thereby, the issues listed in Section 3.2.3 are resolved.
4.5.4. Asymmetric Session
The following SDP example shows how to use the new bandwidth
attribute to offer asymmetric streams. In this case, the end-point
offers to send H.264 video with 1 Mbps while it is capable of
receiving H.264 with up to 3 Mbps. Note that this example does not
make use of the codec-specific H.264 level asymmetry signaling as
defined in RFC 6184 [RFC6184].
m=video 50324 RTP/AVP 96
b=AS:3000
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42c016
a=bw:send pt=96 SMT:tb=1000000:8192
a=bw:recv pt=96 SMT:tb=3000000:16384
SDP offer with asymmetric video bandwidth
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It should be clear from this example that the new bandwidth attribute
is useful when negotiating asymmetric sessions since it offers the
possibility to define the token bucket parameters for both sending
and receiving directions separately.
4.5.5. Session with Retransmission
This SDP example shows how the new bandwidth attribute, 'a=bw', can
be used for negotiating the bandwidth when the RTP Retransmission
Payload Format RFC 4588 [RFC4588] is used.
m=video 49170 RTP/AVPF 96 97
b=AS:500
a=rtpmap:96 MP4V-ES/90000
a=rtcp-fb:96 nack
a=fmtp:96 profile-level-id=8; config=01010000012000884006682C2090A21F
a=rtpmap:97 rtx/90000
a=fmtp:97 apt=96;rtx-time=3000
a=bw:send pt=* AMT:tb=500000:4096
a=bw:recv pt=* AMT:tb=500000:8192
SDP offer with aggregate bandwidth and RTP retransmission
In this case, it is beneficial to use the Aggregate Maximum Token
bucket semantics to allow the end-points to adapt the bandwidths used
for the original stream and for the retransmission stream during the
session. The end-point can send more original packets when the
packet loss rate is low. When the packet loss rate is high then the
end-point can use less bandwidth for the original packets and instead
allow for more retransmissions. It would also be possible to specify
separate limits for the original stream and the retransmission stream
by using a separate set of 'a=bw'-lines for pt=96 and pt=97.
4.6. SDP Examples with Sessions with Multiple Streams
4.6.1. Multiple Streams
The example below is based on the use case described in
Section 3.3.1. Only the negotiation for video is shown here.
m=video 49300 RTP/AVP 96
b=AS:3000
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42c01f
a=bw:send pt=* SMT:tb=1000000:1000
a=bw:recv pt=* SMT:tb=1000000:2000
a=bw:send pt=* AMT:tb=1000000:1000
a=bw:recv pt=* AMT:tb=3000000:6000
a=max-recv-ssrc:* 4
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SDP offer with both per-stream and aggregate bandwidth
With the new bandwidth attribute, it is possible to define the
bandwidth for each received stream independently from each other. In
this case, the SDP shows that the end-point is prepared to send
maximum 1 Mbps, and that the end-point is prepared to receive maximum
1 Mbps per stream. The SDP also shows that the end-point is prepared
to receive maximum 3 Mbps, aggregated for the up to four streams in
the receiving direction. Note that this implies that to receive more
than three streams, each stream's bandwidth must be reduced to comply
with the maximum aggregate.
4.6.2. Declarative Example with Stream Asymmetry
This example shows a declarative usage of the new bandwidth
attribute.
m=video 50324 RTP/AVP 96 97 98
a=rtpmap:96 H264/90000
a=rtpmap:97 H263-2000/90000
a=rtpmap:98 MP4V-ES/90000
a=max-recv-ssrc:96 2
a=max-recv-ssrc:* 5
a=bw:send pt=* SMT:tb=1200000:16384
a=bw:recv pt=96 SMT:tb=1500000:16384
a=bw:recv pt=97,98 SMT:tb=2500000:16384
a=bw:recv pt=* AMT:tb=8000000:65535
SDP offer with payload-specific per-stream bandwidth
In the above example, the outgoing single stream is limited to bucket
rate of 1.2 Mbps and bucket size of 16384 bytes. The up to 5
incoming streams can in total use maximum 8 Mbps bucket rate and with
a bucket size of 65535 bytes. However, the individual streams
maximum rate is depending on payload type. Payload type 96 (H.264)
is limited to 1.5 Mbps with a bucket size of 16384 bytes, while the
Payload types 97 (H.263) and 98 (MPEG-4) may use up top 2.5 Mbps with
a bucket size of 16384 bytes.
4.7. Interoperability Issues
The proposed new bandwidth attribute obviously has connections to the
bandwidth modifier 'b=AS' and the attributes defined for
directionality ('a=sendrecv', 'a=sendonly', 'a=recvonly' and
'a=inactive') defined in RFC 4566 [RFC4566]. It is therefore
important to properly analyze these relationships so that any
interoperability issues can be avoided.
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4.7.1. Interoperability with Existing Bandwidth Attribute
If the SDP includes both the 'b=AS' bandwidth modifier and 'a=bw'
bandwidth attribute then alignment may be necessary to avoid
confusion. This section gives some guidelines for such alignment.
It may however happen that some usage needs other alignments than
what is discussed below. If so, then those alignments need to be
considered on a case-by-case. The discussion below should therefore
not be seen as an exhaustive list.
In general, the bandwidths offered with 'b=AS' and 'a=bw' should be
aligned for the direction that applies for the 'b=AS' bandwidth
modifier. For 'sendrecv' and 'recvonly' sessions, 'b=AS' indicates
the bandwidth for the receiving direction. The b=AS is closest in
interpretation to the AMT semantic. If the stream maximum semantic
(SMT) is used then the sum of the bandwidths in the receive direction
may exceed the 'b=AS' bandwidth but the AMT should not exceed the
b=AS value.
If the session includes multiple streams, but if not all of the
streams will be active simultaneously, then 'b=AS' should indicate
the maximum bandwidth that will be used for the combinations of
streams that are active simultaneously, the same way AMT could be
used in such a session. This also means that the bandwidths offered
with 'a=bw' are accumulated for the combination of streams that are
active, and this aggregated bandwidth should not exceed the bandwidth
defined with 'b=AS'. Note however that it is possible and feasible
to specify an aggregate that is less than the sum of the maximum
bandwidth for the maximum amount of available streams. It may be
possible to use the maximum number of active streams with a lower
bandwidth than the maximum, or it may be possible to reduce the
active number of streams to stay within the bandwidth limit.
The SDP below gives an example of how this is done. In this example,
the intention is to use either the payload type pair (96, 97) or the
payload type pair (98, 99). The intention is however to, for
example, not pair payload types 96 and 98.
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m=video 50000 RTP/AVP 96 97 98 99 100
b=AS:1000
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42c00d
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42c00c
a=rtpmap:98 H264/90000
a=fmtp:98 profile-level-id=42c00d
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42c00c
a=rtpmap:100 H264/90000
a=fmtp:100 profile-level-id=42c00c
a=bw:sendrecv 96 SMT:tb=700000:4000
a=bw:recv 97 SMT:tb=300000:3000
a=bw:sendrecv 98 SMT:tb=500000:3000
a=bw:recv 99 SMT:tb=200000:2000
a=bw:send 100 SMT:tb=300000:1400
a=sendrecv
SDP offer with complex bandwidth relations
This session is bi-directional, as shown with the 'a=sendrecv'
attribute. The bandwidth offered with 'b=AS' therefore applies to
the receive direction. The 'b=AS' is then set based on the
combination of streams that gives the highest bandwidth, i.e. the
payload type pair (96, 97).
This means that the bandwidths offered with 'a=bw' are aligned with
the bandwidth offered with 'b=AS'.
If, on the other hand, the intention would be to use another
combination of payload types, for example (96, 98), then this would
add up to 1200 kbps, which would mean that the stream bandwidths
would not be aligned with the 'b=AS' bandwidth.
This shows that bandwidths for 'sendrecv' and 'recv' directions are
added together when determining the bandwidth for the combined
streams.
If the offer is "complex", for example offering multiple streams for
both speech and video, possibly with many different codecs, (and
therefore uses 'a=bw' together with the 'b=AS' bandwidth modifier)
and if the answerer wants to change this into a "simple" session
(e.g. plain simple VoIP with only one RTP payload type for codec X)
then the answerer may remove the 'a=bw' lines when creating the
answer. It may therefore happen that the answer includes only 'b=AS'
bandwidth modifier in the SDP answer. However, if the offer does not
include any 'b=AS' line then it is recommended to maintain the 'a=bw'
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lines also in the answer, even for "simple" sessions. This means
that the offerer cannot rely on the existence of 'a=bw' in the
answer.
4.7.2. Interoperability with Existing Directional Attribute
Since the 'a=bw' attribute includes a parameter for directionality it
is important to clarify the relationship to the already existing
directional attributes in SDP ('sendrecv', 'sendonly', 'recvonly' and
'inactive'). In general, one can say that:
o The SDP attribute indicates the directionality for the session.
o The 'a=bw' attribute defines the directionality for the bandwidth
for streams within the session.
o The SDP attribute for directionality has precedence over the
'a=bw' parameter for directionality when it comes to the media
that is actually being transmitted.
At session setup time, it is therefore acceptable to define streams
with other directionality than what is shown with the SDP attribute
for directionality. However, when media is transmitted, then the SDP
attribute for directionality has to be followed. An example of this
is shown below.
m=video 5000 RTP/AVP 96 97 98
b=AS:1000
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=42c00d
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42c00c
a=bw:sendrecv 96 SMT:tb=700000:4000
a=bw:recv 97 SMT:tb=200000:3000
a=bw:send 97 SMT:tb=300000:1400
a=recvonly
SDP offer specifying bandwidth in 'inactive' direction
This means that three bandwidths are defined at session setup:
o one stream (PT=96) for 700 kbps bi-directional video;
o one stream (PT=97) for 200 kbps receive-only video; and:
o one stream (PT=97) for 300 kbps send-only video.
However, since 'a=recvonly' is defined then this means that the end-
point is, at the session setup time, only willing to receive media
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even though the SDP contains bandwidth declarations also for the
sending direction. This allows for setting up streams that are
effectively inactive in one or both directions from the beginning of
the session and then enabling them later in the session.
This can be compared with the case when one defines one or more
codecs, even if the session starts up as 'inactive'.
5. Rules and Recommendations for Extensions
The a=bw attribute is defined to be extensible and this section
discusses the extension points that are available.
5.1. Directionality
The current specification defines send, recv and sendrecv. In case
some new directionality behavior is needed that doesn't match the
existing, a new one could be defined. This should be avoided unless
a clear need for a new directionality is found.
5.2. Scope
It is expected that there will be a need to extend the bandwidth
scope. This document only defines two scope types, session and
payload type, and there is very likely other desirable scopes that
will be defined in the future. Possible examples of scopes are those
applying to a specific SSRC, a particular end-point, or a class of
end-points.
5.3. Semantics
This is the extension point that is expected to be frequently used in
the future. A major proliferation of semantics is not good for
interoperability, but it is likely that bandwidth shortcomings or
missing functionalities will be discovered in the future. Thus
defining new semantics gives maximum flexibility to define the
meaning of the provided value(s), the format of the values and how to
interpret the directionality and scope values.
5.4. Values
This document only defines token buckets as values. In case fewer or
more parameters are needed to express a particular semantics, new
value formats can be defined. Defining new value formats should be
done with some consideration of generality and reuse so that future
semantics can also use the new value format, with the target to try
to minimize the number of different formats.
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6. Open Issues
This document contain a few open issues:
1. Multicast behavior needs to be specified.
2. It is an open question to decide if and how to handle the RTCP
bandwidth negotiation, e.g. corresponding to b=RS and b=RR.
3. It is an open question to develop semantics for the transport
independent bandwidth negotiation, e.g. corresponding to b=TIAS.
4. It is an open question what rules and recommendations there
should be for extensions to this memo.
7. IANA Considerations
Following the guidelines in RFC 4566 [RFC4566] and in RFC 3550
[RFC3550], the IANA is requested to register:
1. The bw attribute as defined in Section 4.1.
2. The bw attribute directionality registry rules
3. The bw attribute scope registry rules.
4. The bw attribute semantics registry rules.
5. The bw attribute values registry rules.
This section will be filled out in future versions of this document.
8. Security Considerations
Excessive bandwidth allocation can consume all the resources, much
more than what the end-point(s) intend to use. So, if a session
allocates an unnecessarily high bandwidth then this will likely mean
that some other users cannot be admitted, or that they cannot get QoS
guaranteed resources that they requested and have to use best effort.
It can also happen that the session itself is rejected, if the end-
points try to allocate resources that are not available. Allocating
too little bandwidth is likely to negatively impact the perceived
media quality or entirely prevent reception of requested media.
The above shows that the bandwidth attribute is a potential vector
for attacks both from malicious end-points or third party attackers
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that attempts to modify the attribute to impact the system to
allocate unnecessary resources, deny end-points service, reduce
quality for end-points or incur cost on users.
To prevent third party attacks the signalling should be source
authenticated and integrity protected to prevent any on or off-path
attacker from injecting or modifying the SDP. Malicious end-points
can't as easily be protected against using crypto, instead behavior
analysis and preventing such a malicious end-point from having
serious impact on other end-points are needed.
9. Acknowledgements
10. References
10.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2212] Shenker, S., Partridge, C., and R. Guerin, "Specification
of Guaranteed Quality of Service", RFC 2212,
September 1997.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3556] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth",
RFC 3556, July 2003.
[RFC3890] Westerlund, M., "A Transport Independent Bandwidth
Modifier for the Session Description Protocol (SDP)",
RFC 3890, September 2004.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC5234] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", STD 68, RFC 5234, January 2008.
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10.2. Informative References
[AMR] "3GPP TS 26.090, "Adaptive Multi-Rate (AMR) speech codec;
Transcoding functions".", June 1999.
[AMR-WB] "3GPP TS 26.190, "Adaptive Multi-Rate - Wideband (AMR-WB)
speech codec; Transcoding functions".", April 2001.
[G.711] "ITU-T Recommendation G.711, "Pulse Code Modulation (PCM)
of Voice Frequencies".", November 1988.
[G.726] "ITU-T Recommendation G.726, "40, 32, 24, 16 kbit/s
Adaptive Differential Pulse Code Modulation (ADPCM)".",
December 1990.
[G.729] "ITU-T Recommendation G.729, "Coding of speech at 8 kbit/s
using conjugate-structure algebraic-code-excited linear
prediction (CS-ACELP)".", March 1996.
[H.264] "ITU-T Recommendation H.264, "Advanced video coding for
generic audiovisual services".", May 2003.
[I-D.westerlund-avtcore-max-ssrc]
Westerlund, M., Burman, B., and F. Jansson, "Multiple
Synchronization sources (SSRC) in RTP Session Signaling",
draft-westerlund-avtcore-max-ssrc-00 (work in progress),
October 2011.
[I-D.westerlund-avtcore-multiplex-architecture]
Westerlund, M., Burman, B., and C. Perkins, "RTP
Multiplexing Architecture",
draft-westerlund-avtcore-multiplex-architecture-01 (work
in progress), March 2012.
[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
July 2006.
[RFC6184] Wang, Y., Even, R., Kristensen, T., and R. Jesup, "RTP
Payload Format for H.264 Video", RFC 6184, May 2011.
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Authors' Addresses
Tomas Frankkila
Ericsson
Laboratoriegrand 11
SE-971 28 Lulea
Sweden
Phone: +46 10 714 30 20
Email: tomas.frankkila@ericsson.com
Magnus Westerlund
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 82 87
Email: magnus.westerlund@ericsson.com
Bo Burman
Ericsson
Farogatan 6
SE-164 80 Kista
Sweden
Phone: +46 10 714 13 11
Email: bo.burman@ericsson.com
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