A Transport Independent Bandwidth Modifier for the Session Description Protocol (SDP)
RFC 3890
| Document | Type | RFC - Proposed Standard (September 2004) | |
|---|---|---|---|
| Author | Magnus Westerlund | ||
| Last updated | 2018-12-20 | ||
| Replaces | draft-westerlund-mmusic-sdp-bwparam | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
| Stream | WG state | WG Document | |
| Document shepherd | (None) | ||
| IESG | IESG state | RFC 3890 (Proposed Standard) | |
| Action Holders |
(None)
|
||
| Consensus boilerplate | Unknown | ||
| Telechat date | (None) | ||
| Responsible AD | Jon Peterson | ||
| Send notices to | (None) |
RFC 3890
Network Working Group M. Westerlund
Request for Comments: 3890 Ericsson
Category: Standards Track September 2004
A Transport Independent Bandwidth Modifier
for the Session Description Protocol (SDP)
Status of this Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2004).
Abstract
This document defines a Session Description Protocol (SDP) Transport
Independent Application Specific Maximum (TIAS) bandwidth modifier
that does not include transport overhead; instead an additional
packet rate attribute is defined. The transport independent bit-rate
value together with the maximum packet rate can then be used to
calculate the real bit-rate over the transport actually used.
The existing SDP bandwidth modifiers and their values include the
bandwidth needed for the transport and IP layers. When using SDP
with protocols like the Session Announcement Protocol (SAP), the
Session Initiation Protocol (SIP), and the Real-Time Streaming
Protocol (RTSP), and when the involved hosts has different transport
overhead, for example due to different IP versions, the
interpretation of what lower layer bandwidths are included is not
clear.
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RFC 3890 Bandwidth Modifier for SDP September 2004
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. The Bandwidth Attribute. . . . . . . . . . . . . . . . . 3
1.1.1. Conference Total . . . . . . . . . . . . . . . . 3
1.1.2. Application Specific Maximum . . . . . . . . . . 3
1.1.3. RTCP Report Bandwidth. . . . . . . . . . . . . . 4
1.2. IPv6 and IPv4. . . . . . . . . . . . . . . . . . . . . . 4
1.3. Further Mechanisms that Change the Bandwidth
Utilization. . . . . . . . . . . . . . . . . . . . . . . 5
1.3.1. IPsec. . . . . . . . . . . . . . . . . . . . . . 5
1.3.2. Header Compression . . . . . . . . . . . . . . . 5
2. Definitions. . . . . . . . . . . . . . . . . . . . . . . . . . 6
2.1. Glossary . . . . . . . . . . . . . . . . . . . . . . . . 6
2.2. Terminology. . . . . . . . . . . . . . . . . . . . . . . 6
3. The Bandwidth Signaling Problems . . . . . . . . . . . . . . . 6
3.1. What IP Version is Used. . . . . . . . . . . . . . . . . 6
3.2. Taking Other Mechanisms into Account . . . . . . . . . . 7
3.3. Converting Bandwidth Values. . . . . . . . . . . . . . . 8
3.4. RTCP Problems. . . . . . . . . . . . . . . . . . . . . . 8
3.5. Future Development . . . . . . . . . . . . . . . . . . . 9
3.6. Problem Conclusion . . . . . . . . . . . . . . . . . . . 9
4. Problem Scope. . . . . . . . . . . . . . . . . . . . . . . . . 10
5. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 10
6. Solution . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
6.1. Introduction . . . . . . . . . . . . . . . . . . . . . . 11
6.2. The TIAS Bandwidth Modifier. . . . . . . . . . . . . . . 11
6.2.1. Usage. . . . . . . . . . . . . . . . . . . . . . 11
6.2.2. Definition . . . . . . . . . . . . . . . . . . . 12
6.2.3. Usage Rules. . . . . . . . . . . . . . . . . . . 13
6.3. Packet Rate Parameter. . . . . . . . . . . . . . . . . . 13
6.4. Converting to Transport-Dependent Values . . . . . . . . 14
6.5. Deriving RTCP bandwidth. . . . . . . . . . . . . . . . . 15
6.5.1. Motivation for this Solution. . . . . . . . . . . 15
6.6. ABNF Definitions . . . . . . . . . . . . . . . . . . . . 16
6.7. Example. . . . . . . . . . . . . . . . . . . . . . . . . 16
7. Protocol Interaction . . . . . . . . . . . . . . . . . . . . . 17
7.1. RTSP . . . . . . . . . . . . . . . . . . . . . . . . . . 17
7.2. SIP. . . . . . . . . . . . . . . . . . . . . . . . . . . 17
7.3. SAP. . . . . . . . . . . . . . . . . . . . . . . . . . . 18
8. Security Considerations. . . . . . . . . . . . . . . . . . . . 18
9. IANA Considerations. . . . . . . . . . . . . . . . . . . . . . 18
10. Acknowledgments. . . . . . . . . . . . . . . . . . . . . . . . 19
11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 19
11.1. Normative References . . . . . . . . . . . . . . . . . . 19
11.2. Informative References . . . . . . . . . . . . . . . . . 19
12. Author's Address . . . . . . . . . . . . . . . . . . . . . . . 21
13. Full Copyright Statement . . . . . . . . . . . . . . . . . . . 22
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RFC 3890 Bandwidth Modifier for SDP September 2004
1. Introduction
This specification is structured in the following way: In this
section, some information regarding SDP bandwidth modifiers, and
different mechanisms that affect transport overhead are asserted. In
section 3, the problems found are described, including problems that
are not solved by this specification. In section 4 the scope of the
problems this specification solves is presented. Section 5 contains
the requirements applicable to the problem scope. Section 6 defines
the solution, which is a new bandwidth modifier, and a new maximum
packet rate attribute. Section 7 looks at the protocol interaction
for SIP, RTSP, and SAP. The security considerations are discussed in
section 8. The remaining sections are the necessary IANA
considerations, acknowledgements, reference list, author's address,
and copyright and IPR notices.
Today the Session Description Protocol (SDP) [1] is used in several
types of applications. The original application is session
information and configuration for multicast sessions announced with
Session Announcement Protocol (SAP) [5]. SDP is also a vital
component in media negotiation for the Session Initiation Protocol
(SIP) [6] by using the offer answer model [7]. The Real-Time
Streaming Protocol (RTSP) [8] also makes use of SDP to declare to the
client what media and codec(s) comprise a multi-media presentation.
1.1. The Bandwidth Attribute
In SDP [1] there exists a bandwidth attribute, which has a modifier
used to specify what type of bit-rate the value refers to. The
attribute has the following form:
b=<modifier>:<value>
Today there are four defined modifiers used for different purposes.
1.1.1. Conference Total
The Conference Total is indicated by giving the modifier "CT".
Conference total gives a maximum bandwidth that a conference session
will use. Its purpose is to decide if this session can co-exist with
any other sessions, defined in RFC 2327 [1].
1.1.2. Application Specific Maximum
The Application Specific maximum bandwidth is indicated by the
modifier "AS". The interpretation of this attribute is dependent on
the application's notion of maximum bandwidth. For an RTP
application, this attribute is the RTP session bandwidth as defined
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in RFC 3550 [4]. The session bandwidth includes the bandwidth that
the RTP data traffic will consume, including the lower layers, down
to the IP layer. Therefore, the bandwidth is in most cases
calculated over RTP payload, RTP header, UDP, and IP, defined in RFC
2327 [1].
1.1.3. RTCP Report Bandwidth
In RFC 3556 [9], two bandwidth modifiers are defined. These
modifiers, "RS" and "RR", define the amount of bandwidth that is
assigned for RTCP reports by active data senders and RTCP reports by
other participants (receivers), respectively.
1.2. IPv6 and IPv4
Today there are two IP versions, 4 [14] and 6 [13], used in parallel
on the Internet, creating problems. However, there exist a number of
possible transition mechanisms.
- The nodes which wish to communicate must share the IP version;
typically this is done by deploying dual-stack nodes. For
example, an IPv4 only host cannot communicate with an IPv6 only
host.
- If communication between nodes which do not share a protocol
version is required, use of a translation or proxying mechanism
would be required. Work is underway to specify such a mechanism
for this purpose.
------------------ ----------------------
| IPv4 domain | | IPv6 Domain |
| | ------------- | |
| ---------- |-|Translator |-| ---------- |
| |Server A| | | or proxy | | |Client B| |
| ---------- | ------------- | ---------- |
------------------ ----------------------
Figure 1. Translation or proxying between IPv6 and IPv4 addresses.
- IPv6 nodes belonging to different domains running IPv6, but
lacking IPv6 connectivity between them, solve this by tunneling
over the IPv4 net, see Figure 2. Basically, the IPv6 packets are
sent as payload in IPv4 packets between the tunneling end-points
at the edge of each IPv6 domain. The bandwidth required over the
IPv4 domain will be different from IPv6 domains. However, as the
tunneling is normally not performed by the application end-point,
this scenario can not usually be taken into consideration.
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--------------- --------------- ---------------
| IPv6 domain | | IPv4 domain | | IPv6 Domain |
| | |-------------| | |
| ---------- |--||Tunnel ||--| ---------- |
| |Server A| | |-------------| | |Client B| |
| ---------- | | | | ---------- |
--------------- --------------- --------------|
Figure 2. Tunneling through a IPv4 domain
IPv4 has a minimum header size of 20 bytes, while the fixed part of
the IPv6 header is 40 bytes.
The difference in header sizes means that the bit-rate required for
the two IP versions is different. The significance of the difference
depends on the packet rate and payload size of each packet.
1.3. Further Mechanisms that Change the Bandwidth Utilization
There exist a number of other mechanisms that also may change the
overhead at layers below media transport. We will briefly cover a
few of these here.
1.3.1. IPsec
IPsec [19] can be used between end points to provide confidentiality
through the application of the IP Encapsulating Security Payload
(ESP) [21] or integrity protection using the IP Authentication Header
(AH) [20] of the media stream. The addition of the ESP and AH
headers increases each packet's size.
To provide virtual private networks, complete IP packets may be
encapsulated between an end node and the private networks security
gateway, thus providing a secure tunnel that ensures confidentiality,
integrity, and authentication of the packet stream. In this case,
the extra IP and ESP header will significantly increase the packet
size.
1.3.2. Header Compression
Another mechanism that alters the actual overhead over links is
header compression. Header compression uses the fact that most
network protocol headers have either static or predictable values in
their fields within a packet stream. Compression is normally only
done on a per hop basis, i.e., on a single link. The normal reason
for doing header compression is that the link has fairly limited
bandwidth and significant gain in throughput is achieved.
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There exist several different header compression standards. For
compressing IP headers only, there is RFC 2507 [10]. For compressing
packets with IP/UDP/RTP headers, CRTP [11] was created at the same
time. More recently, the Robust Header Compression (ROHC) working
group has been developing a framework and profiles [12] for
compressing certain combinations of protocols, like IP/UDP, and
IP/UDP/RTP.
2. Definitions
2.1. Glossary
ALG - Application Level Gateway.
bps - bits per second.
RTSP - Real-Time Streaming Protocol, see [8].
SDP - Session Description Protocol, see [1].
SAP - Session Announcement Protocol, see [5].
SIP - Session Initiation Protocol, see [6].
TIAS - Transport Independent Application Specific maximum, a
bandwidth modifier.
2.2. Terminology
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 BCP 14, RFC 2119 [3].
3. The Bandwidth Signaling Problems
When an application wants to use SDP to signal the bandwidth required
for this application, some problems become evident due to the
inclusion of the lower layers in the bandwidth values.
3.1. What IP Version is Used
If one signals the bandwidth in SDP, for example, using "b=AS:" as an
RTP based application, one cannot know if the overhead is calculated
for IPv4 or IPv6. An indication of which protocol has been used when
calculating the bandwidth values is given by the "c=" connection
address line. This line contains either a multicast group address or
a unicast address of the data source or sink. The "c=" line's
address type may be assumed to be of the same type as the one used in
the bandwidth calculation, although no document specifying this point
seems to exist.
In cases of SDP transported by RTSP, this is even less clear. The
normal usage for a unicast on-demand streaming session is to set the
connection data address to a null address. This null address does
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have an address type, which could be used as an indication. However,
this is also not clarified anywhere.
Figure 1, illustrates a connection scenario between a streaming
server A and a client B over a translator. When B receives the SDP
from A over RTSP, it will be very difficult for B to know what the
bandwidth values in the SDP represent. The following possibilities
exist:
1. The SDP is unchanged and the "c=" null address is of type IPv4.
The bandwidth value represents the bandwidth needed in an IPv4
network.
2. The SDP has been changed by an Application Level Gateway (ALG).
The "c=" address is changed to an IPv6 type. The bandwidth value
is unchanged.
3. The SDP is changed and both "c=" address type and bandwidth value
is converted. Unfortunately, this can seldom be done, see 3.3.
In case 1, the client can understand that the server is located in an
IPv4 network and that it uses IPv4 overhead when calculating the
bandwidth value. The client can almost never convert the bandwidth
value, see section 3.3.
In case 2, the client does not know that the server is in an IPv4
network and that the bandwidth value is not calculated with IPv6
overhead. In cases where a client uses this value to determine if
its end of the network has sufficient resources the client will
underestimate the required bit-rate, potentially resulting in bad
application performance.
In case 3, everything works correctly. However, this case will be
very rare. If one tries to convert the bandwidth value without
further information about the packet rate, significant errors may be
introduced into the value.
3.2. Taking Other Mechanisms into Account
Section 1.2 and 1.3 lists a number of reasons, like header
compression and tunnels, that would change lower layer header sizes.
For these mechanisms there exist different possibilities to take them
into account.
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Using IPsec directly between end-points should definitely be known to
the application, thus enabling it to take the extra headers into
account. However the same problem also exists with the current SDP
bandwidth modifiers where a receiver is not able to convert these
values taking the IPsec headers into account.
It is less likely that an application would be aware of the existence
of a virtual private network. Thus the generality of the mechanism
to tunnel all traffic may prevent the application from even
considering whether it would be possible to convert the values.
When using header compression, the actual overhead will be less
deterministic, but in most cases an average overhead can be
determined for a certain application. If a network node knows that
some type of header compression is employed, this can be taken into
consideration. For RSVP [15], there exists an extension, RFC 3006
[16], that allows the data sender to inform network nodes about the
compressibility of the data flow. To be able to do this with any
accuracy, the compression factor and packet rate or size is needed,
as RFC 3006 provides.
3.3. Converting Bandwidth Values
If one would like to convert a bandwidth value calculated using IPv4
overhead to IPv6 overhead, the packet rate is required. The new
bandwidth value for IPv6 is normally "IPv4 bandwidth" + "packet rate"
* 20 bytes, where 20 bytes is the usual difference between IPv6 and
IPv4 headers. The overhead difference may be some other value in
cases when IPv4 options [14] or IPv6 extension headers [13] are used.
As converting requires the packet rate of the stream, this is not
possible in the general case. Many codecs have either multiple
possible packet/frame rates or can perform payload format
aggregation, resulting in many possible rates. Therefore, some extra
information in the SDP will be required. The "a=ptime:" parameter
may be a possible candidate. However, this parameter is normally
only used for audio codecs. Its definition [1] is that it is only a
recommendation, which the sender may disregard. A better parameter
is needed.
3.4. RTCP Problems
When RTCP is used between hosts in IPv4 and IPv6 networks over
translator, similar problems exist. The RTCP traffic going from the
IPv4 domain will result in a higher RTCP bit-rate than intended in
the IPv6 domain due to the larger headers. This may result in up to
a 25% increase in required bandwidth for the RTCP traffic. The
largest increase will be for small RTCP packets when the number of
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IPv4 hosts is much larger than the number of IPv6 hosts.
Fortunately, as RTCP has a limited bandwidth compared to RTP, it will
only result in a maximum of 1.75% increase of the total session
bandwidth when RTCP bandwidth is 5% of RTP bandwidth. The RTCP
randomization may easily result in short term effects of the same
magnitude, so this increase may be considered tolerable. The
increase in bandwidth will in most cases be less.
At the same time, this results in unfairness in the reporting between
an IPv4 and IPv6 node. In the worst case scenario, the IPv6 node may
report with 25% longer intervals.
These problems have been considered insignificant enough to not be
worth any complex solutions. Therefore, only a simple algorithm for
deriving RTCP bandwidth is defined in this specification.
3.5. Future Development
Today there is work in the IETF to design a new datagram transport
protocol suitable for real-time media. This protocol is called the
Datagram Congestion Control Protocol (DCCP). It will most probably
have a different header size than UDP, which is the protocol most
often used for real-time media today. This results in even more
possible transport combinations. This may become a problem if one
has the possibility of using different protocols, which will not be
determined prior to actual protocol SETUP. Thus, pre-calculating
this value will not be possible, which is one further motivation why
a transport independent bandwidth modifier is needed.
DCCP's congestion control algorithms will control how much bandwidth
can really be utilized. This may require further work with
specifying SDP bandwidth modifiers to declare the dynamic
possibilities of an application's media stream. For example, min and
max media bandwidth the application is capable of producing at all,
or for media codecs only capable of producing certain bit-rates,
enumerating possible rates. However, this is for future study and
outside the scope of the present solution.
3.6. Problem Conclusion
A shortcoming of the current SDP bandwidth modifiers is that they
also include the bandwidth needed for lower layers. It is in many
cases difficult to determine which lower layers and their versions
were included in the calculation, especially in the presence of
translation or proxying between different domains. This prevents a
receiver from determining if given bandwidth needs to be converted
based on the actual lower layers being used.
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Secondly, an attribute to give the receiver an explicit determination
of the maximum packet rate that will be used does not exist. This
value is necessary for accurate conversion of any bandwidth values if
the difference in overhead is known.
4. Problem Scope
The problems described in section 3 are common and effect application
level signaling using SDP, other signaling protocols, and also
resource reservation protocols. However, this document targets the
specific problem of signaling the bit-rate in SDP. The problems need
to be considered in other affected protocols and in new protocols
being designed. In the MMUSIC WG there is work on a replacement of
SDP called SDP-NG. It is recommended that the problems outlined in
this document be considered when designing solutions for specifying
bandwidth in the SDP-NG [17].
As this specification only targets carrying the bit-rate information
within SDP, it will have a limited applicability. As SDP information
is normally transported end-to-end by an application protocol, nodes
between the end-points will not have access to the bit-rate
information. It will normally only be the end points that are able
to take this information into account. An interior node will need to
receive the information through a means other than SDP, and that is
outside the scope of this specification.
Nevertheless, the bit-rate information provided in this specification
is sufficient for cases such as first-hop resource reservation and
admission control. It also provide information about the maximum
codec rate, which is independent of lower-level protocols.
This specification does NOT try to solve the problem of detecting
NATs or other middleboxes.
5. Requirements
The problems outlined in the preceding sections and with the above
applicability, should meet the following requirements:
- The bandwidth value SHALL be given in a way such that it can be
calculated for all possible combinations of transport overhead.
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6. Solution
6.1. Introduction
This chapter describes a solution for the problems outlined in this
document for the Application Specific (AS) bandwidth modifier, thus
enabling the derivation of the required bit-rate for an application,
or RTP session's data and RTCP traffic. The solution is based upon
the definition of a new Transport Independent Application Specific
(TIAS) bandwidth modifier and a new SDP attribute for the maximum
packet rate (maxprate).
The CT is a session level modifier and cannot easily be dealt with.
To address the problems with different overhead, it is RECOMMENDED
that the CT value be calculated using reasonable worst case overhead.
An example of how to calculate a reasonable worst case overhead is:
Take the overhead of the largest transport protocol (using average
size if variable), add that to the largest IP overhead that is
expected for use, plus the data traffic rate. Do this for every
individual media stream used in the conference and add them together.
The RR and RS modifiers [9] will be used as defined and include
transport overhead. The small unfairness between hosts is deemed
acceptable.
6.2. The TIAS Bandwidth Modifier
6.2.1. Usage
A new bandwidth modifier is defined to be used for the following
purposes:
- Resource reservation. A single bit-rate can be enough for use as
a resource reservation. Some characteristics can be derived from
the stream, codec type, etc. In cases where more information is
needed, another SDP parameter will be required.
- Maximum media codec rate. With the definition below of "TIAS",
the given bit-rate will mostly be from the media codec.
Therefore, it gives a good indication of the maximum codec bit-
rate required to be supported by the decoder.
- Communication bit-rate required for the stream. The "TIAS" value
together with "maxprate" can be used to determine the maximum
communication bit-rate the stream will require. Using session
level values or by adding all maximum bit-rates from the streams
in a session together, a receiver can determine if its
communication resources are sufficient to handle the stream. For
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example, a modem user can determine if the session fits his
modem's capabilities and the established connection.
- Determine the RTP session bandwidth and derive the RTCP bandwidth.
The derived transport dependent attribute will be the RTP session
bandwidth in case of RTP based transport. The TIAS value can also
be used to determine the RTCP bandwidth to use when using implicit
allocation. RTP [4] specifies that if not explicitly stated,
additional bandwidth, equal to 5% of the RTP session bandwidth,
shall be used by RTCP. The RTCP bandwidth can be explicitly
allocated by using the RR and RS modifiers defined in [9].
6.2.2. Definition
A new session and media level bandwidth modifier is defined:
b=TIAS:<bandwidth-value> ; see section 6.6 for ABNF definition.
The Transport Independent Application Specific Maximum (TIAS)
bandwidth modifier has an integer bit-rate value in bits per second.
A fractional bandwidth value SHALL always be rounded up to the next
integer. The bandwidth value is the maximum needed by the
application (SDP session level) or media stream (SDP media level)
without counting IP or other transport layers like TCP or UDP.
At the SDP session level, the TIAS value is the maximal amount of
bandwidth needed when all declared media streams are used. This MAY
be less than the sum of all the individual media streams values.
This is due to the possibility that not all streams have their
maximum at the same point in time. This can normally only be
verified for stored media streams.
For RTP transported media streams, TIAS at the SDP media level can be
used to derive the RTP "session bandwidth", defined in section 6.2 of
[4]. In the context of RTP transport, the TIAS value is defined as:
Only the RTP payload as defined in [4] SHALL be used in the
calculation of the bit-rate, i.e., excluding the lower layers
(IP/UDP) and RTP headers including RTP header, RTP header
extensions, CSRC list, and other RTP profile specific fields.
Note that the RTP payload includes both the payload format header
and the data. This may allow one to use the same value for RTP-
based media transport, non-RTP transport, and stored media.
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Note 1: The usage of bps is not in accordance with RFC 2327 [1].
This change has no implications on the parser, only the interpreter
of the value must be aware. The change is done to allow for better
resolution, and has also been used for the RR and RS bandwidth
modifiers, see [9].
Note 2: RTCP bandwidth is not included in the bandwidth value. In
applications using RTCP, the bandwidth used by RTCP is either 5% of
the RTP session bandwidth including lower layers or as specified by
the RR and RS modifiers [9]. A specification of how to derive the
RTCP bit-rate when using TIAS is presented in chapter 6.5.
6.2.3. Usage Rules
"TIAS" is primarily intended to be used at the SDP media level. The
"TIAS" bandwidth attribute MAY be present at the session level in
SDP, if all media streams use the same transport. In cases where the
sum of the media level values for all media streams is larger than
the actual maximum bandwidth need for all streams, it SHOULD be
included at session level. However, if present at the session level
it SHOULD be present also at the media level. "TIAS" SHALL NOT be
present at the session level unless the same transport protocols is
used for all media streams. The same transport is used as long as
the same combination of protocols is used, like IPv6/UDP/RTP.
To allow for backwards compatibility with applications of SDP that do
not implement "TIAS", it is RECOMMENDED to also include the "AS"
modifier when using "TIAS". The presence of a value including
lower-layer overhead, even with its problems, is better than none.
However, an SDP application implementing TIAS SHOULD ignore the "AS"
value and use "TIAS" instead when both are present.
When using TIAS for an RTP-transported stream, the "maxprate"
attribute, if possible to calculate, defined next, SHALL be included
at the corresponding SDP level.
6.3. Packet Rate Parameter
To be able to calculate the bandwidth value including the lower
layers actually used, a packet rate attribute is also defined.
The SDP session and media level maximum packet rate attribute is
defined as:
a=maxprate:<packet-rate> ; see section 6.6 for ABNF definition.
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The <packet-rate> is a floating-point value for the stream's maximum
packet rate in packets per second. If the number of packets is
variable, the given value SHALL be the maximum the application can
produce in case of a live stream, or for stored on-demand streams,
has produced. The packet rate is calculated by adding the number of
packets sent within a 1 second window. The maxprate is the largest
value produced when the window slides over the entire media stream.
In cases that this can't be calculated, i.e., a live stream, a
estimated value of the maximum packet rate the codec can produce for
the given configuration and content SHALL be used.
Note: The sliding window calculation will always yield an integer
number. However the attributes field is a floating-point value
because the estimated or known maximum packet rate per second may be
fractional.
At the SDP session level, the "maxprate" value is the maximum packet
rate calculated over all the declared media streams. If this can't
be measured (stored media) or estimated (live), the sum of all media
level values provides a ceiling value. Note: the value at session
level can be less then the sum of the individual media streams due to
temporal distribution of media stream's maximums. The "maxprate"
attribute MUST NOT be present at the session level if the media
streams use different transport. The attribute MAY be present if the
media streams use the same transport. If the attribute is present at
the session level, it SHOULD also be present at the media level for
all media streams.
"maxprate" SHALL be included for all transports where a packet rate
can be derived and TIAS is included. For example, if you use TIAS
and a transport like IP/UDP/RTP, for which the max packet rate
(actual or estimated) can be derived, then "maxprate" SHALL be
included. However, if either (a) the packet rate for the transport
cannot be derived, or (b) TIAS is not included, then, "maxprate" is
not required to be included.
6.4. Converting to Transport-Dependent Values
When converting the transport-independent bandwidth value (bw-value)
into a transport-dependent value including the lower layers, the
following MUST be done:
1. Determine which lower layers will be used and calculate the sum of
the sizes of the headers in bits (h-size). In cases of variable
header sizes, the average size SHALL be used. For RTP-transported
media, the lower layers SHALL include the RTP header with header
extensions, if used, the CSRC list, and any profile-specific
extensions.
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2. Retrieve the maximum packet rate from the SDP (prate = maxprate).
3. Calculate the transport overhead by multiplying the header sizes
by the packet rate (t-over = h-size * prate).
4. Round the transport overhead up to nearest integer in bits
(t-over = CEIL(t-over)).
5. Add the transport overhead to the transport independent bandwidth
value (total bit-rate = bw-value + t-over)
When the above calculation is performed using the "maxprate", the
bit-rate value will be the absolute maximum the media stream may use
over the transport assumed in the calculations.
6.5. Deriving RTCP Bandwidth
This chapter does not solve the fairness and possible bit-rate change
introduced by IPv4 to IPv6 translation. These differences are
considered small enough, and known solutions introduce code changes
to the RTP/RTCP implementation. This section provides a consistent
way of calculating the bit-rate to assign to RTCP, if not explicitly
given.
First the transport-dependent RTP session bit-rate is calculated, in
accordance with section 6.4, using the actual transport layers used
at the end point where the calculation is done. The RTCP bit-rate is
then derived as usual based on the RTP session bandwidth, i.e.,
normally equal to 5% of the calculated value.
6.5.1. Motivation for this Solution
Giving the exact same RTCP bit-rate value to both the IPv4 and IPv6
hosts will result in the IPv4 host having a higher RTCP sending rate.
The sending rate represents the number of RTCP packets sent during a
given time interval. The sending of RTCP is limited according to
rules defined in the RTP specification [4]. For a 100-byte RTCP
packet (including UDP/IPv4), the IPv4 sender has an approximately 20%
higher sending rate. This rate falls with larger RTCP packets. For
example, 300-byte packets will only give the IPv4 host a 7% higher
sending rate.
The above rule for deriving RTCP bandwidth gives the same behavior as
fixed assignment when the RTP session has traffic parameters giving a
large TIAS/maxprate ratio. The two hosts will be fair when the
TIAS/maxprate ratio is approximately 40 bytes/packet, given 100-byte
RTCP packets. For a TIAS/maxprate ratio of 5 bytes/packet, the IPv6
host will be allowed to send approximately 15-20% more RTCP packets.
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The larger the RTCP packets become, the more it will favor the IPv6
host in its sending rate.
The conclusions is that, within the normal useful combination of
transport-independent bit rates and packet rates, the difference in
fairness between hosts on different IP versions with different
overhead is acceptable. For the 20-byte difference in overhead
between IPv4 and IPv6 headers, the RTCP bandwidth actually used in a
unicast connection case will not be larger than approximately 1% of
the total session bandwidth.
6.6. ABNF Definitions
This chapter defines in ABNF from RFC 2234 [2] the bandwidth modifier
and the packet rate attribute.
The bandwidth modifier:
TIAS-bandwidth-def = "b" "=" "TIAS" ":" bandwidth-value CRLF
bandwidth-value = 1*DIGIT
The maximum packet rate attribute:
max-p-rate-def = "a" "=" "maxprate" ":" packet-rate CRLF
packet-rate = 1*DIGIT ["." 1*DIGIT]
6.7. Example
v=0
o=Example_SERVER 3413526809 0 IN IP4 server.example.com
s=Example of TIAS and maxprate in use
c=IN IP4 0.0.0.0
b=AS:60
b=TIAS:50780
t=0 0
a=control:rtsp://server.example.com/media.3gp
a=range:npt=0-150.0
a=maxprate:28.0
m=audio 0 RTP/AVP 97
b=AS:12
b=TIAS:8480
a=maxprate:10.0
a=rtpmap:97 AMR/8000
a=fmtp:97 octet-align;
a=control:rtsp://server.example.com/media.3gp/trackID=1
m=video 0 RTP/AVP 99
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b=AS:48
b=TIAS:42300
a=maxprate:18.0
a=rtpmap:99 MP4V-ES/90000
a=fmtp:99 profile-level-id=8;
config=000001B008000001B509000001010000012000884006682C2090A21F
a=control:rtsp://server.example.com/media.3gp/trackID=3
In this SDP example of a streaming session's SDP, there are two media
streams, one audio stream encoded with AMR and one video stream
encoded with the MPEG-4 Video encoder. AMR is used here to produce a
constant rate media stream and uses a packetization resulting in 10
packets per second. This results in a TIAS bandwidth rate of 8480
bits per second, and the claimed 10 packets per second. The video
stream is more variable. However, it has a measured maximum payload
rate of 42,300 bits per second. The video stream also has a variable
packet rate, despite the fact that the video is 15 frames per second,
where at least one instance in a second long window contains 18
packets.
7. Protocol Interaction
7.1. RTSP
The "TIAS" and "maxprate" parameters can be used with RTSP as
currently specified. To be able to calculate the transport dependent
bandwidth, some of the transport header parameters will be required.
There should be no problem for a client to calculate the required
bandwidth(s) prior to an RTSP SETUP. The reason is that a client
supports a limited number of transport setups. The one actually
offered to a server in a SETUP request will be dependent on the
contents of the SDP description. The "m=" line(s) will signal the
desired transport profile(s) to the client.
7.2. SIP
The usage of "TIAS" together with "maxprate" should not be different
from the handling of the "AS" modifier currently in use. The needed
transport parameters will be available in the transport field in the
"m=" line. The address class can be determined from the "c=" field
and the client's connectivity.
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7.3. SAP
In the case of SAP, all available information to calculate the
transport dependent bit-rate should be present in the SDP. The "c="
information gives the address family used for the multicast. The
transport layer, e.g., RTP/UDP, for each media is evident in the
media line ("m=") and its transport field.
8. Security Consideration
The bandwidth value that is supplied by the parameters defined here
can be altered, if not integrity protected. By altering the
bandwidth value, one can fool a receiver into reserving either more
or less bandwidth than actually needed. Reserving too much may
result in unwanted expenses on behalf of the user, while also
blocking resources that other parties could have used. If too little
bandwidth is reserved, the receiving user's quality may be effected.
Trusting a too-large TIAS value may also result in the receiver
rejecting the session due to insufficient communication and decoding
resources.
Due to these security risks, it is strongly RECOMMENDED that the SDP
be integrity protected and source authenticated so tampering can not
be performed, and the source can be trusted. It is also RECOMMENDED
that any receiver of the SDP perform an analysis of the received
bandwidth values to verify that they are reasonable expected values
for the application. For example, a single channel AMR-encoded voice
stream claiming to use 1000 kbps is not reasonable.
Please note that some of the above security requirements are in
conflict with that required to make signaling protocols using SDP
work through a middlebox, as discussed in the security considerations
of RFC 3303 [18].
9. IANA Considerations
This document registers one new SDP session and media level attribute
"maxprate", see section 6.3.
A new SDP [1] bandwidth modifier (bwtype) "TIAS" is also registered
in accordance with the rules requiring a standards-track RFC. The
modifier is defined in section 6.2.
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10. Acknowledgments
The author would like to thank Gonzalo Camarillo and Hesham Soliman
for their work reviewing this document. A very big thanks goes to
Stephen Casner for reviewing and helping fix the language, and
identifying some errors in the previous versions. Further thanks for
suggestion to improvements go to Colin Perkins, Geetha Srikantan, and
Emre Aksu.
The author would also like to thank all persons on the MMUSIC working
group's mailing list that have commented on this specification.
11. References
11.1. Normative References
[1] Handley, M. and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
[2] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", RFC 2234, November 1997.
[3] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[4] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", STD 64,
RFC 3550, July 2003.
11.2. Informative References
[5] Handley, M., Perkins, C., and E. Whelan, "Session Announcement
Protocol", RFC 2974, October 2000.
[6] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:
Session Initiation Protocol", RFC 3261, June 2002.
[7] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
[8] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming
Protocol (RTSP)", RFC 2326, April 1998.
[9] Casner, S., "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC 3556,
July 2003.
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[10] Degermark, M., Nordgren, B., and S. Pink, "IP Header
Compression", RFC 2507, February 1999.
[11] Casner, S. and V. Jacobson, "Compressing IP/UDP/RTP Headers for
Low-Speed Serial Links", RFC 2508, February 1999.
[12] Bormann, C., Burmeister, C., Degermark, M., Fukushima, H.,
Hannu, H., Jonsson, L-E., Hakenberg, R., Koren, T., Le, K., Liu,
Z., Martensson, A., Miyazaki, A., Svanbro, K., Wiebke, T.,
Yoshimura, T., and H. Zheng, "RObust Header Compression (ROHC):
Framework and four profiles: RTP, UDP, ESP, and uncompressed ",
RFC 3095, July 2001.
[13] Deering, S. and R. Hinden, "Internet Protocol, Version 6 (IPv6)
Specification", RFC 2460, December 1998.
[14] Postel, J., "Internet Protocol", STD 5, RFC 791, September 1981.
[15] Braden, R., Zhang, L., Berson, S., Herzog, S., and S. Jamin,
"Resource ReSerVation Protocol (RSVP) -- Version 1 Functional
Specification", RFC 2205, September 1997.
[16] Davie, B., Iturralde, C., Oran, D., Casner, S., and J.
Wroclawski, "Integrated Services in the Presence of Compressible
Flows", RFC 3006, November 2000.
[17] Kutscher, Ott, Bormann, "Session Description and Capability
Negotiation," Work in Progress, March 2003.
[18] Srisuresh, P., Kuthan, J., Rosenberg, J., Molitor, A., and A.
Rayhan, "Middlebox communication architecture and framework",
RFC 3303, August 2002.
[19] Kent, S. and R. Atkinson, "Security Architecture for the
Internet Protocol", RFC 2401, November 1998.
[20] Kent, S. and R. Atkinson, "IP Authentication Header", RFC 2402,
November 1998.
[21] Kent, S. and R. Atkinson, "IP Encapsulating Security Payload
(ESP)", RFC 2406, November 1998.
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12. Author's Address
Magnus Westerlund
Ericsson Research
Ericsson AB
Torshamnsgatan 23
SE-164 80 Stockholm, SWEDEN
Phone: +46 8 7190000
EMail: Magnus.Westerlund@ericsson.com
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RFC 3890 Bandwidth Modifier for SDP September 2004
13. Full Copyright Statement
Copyright (C) The Internet Society (2004).
This document is subject to the rights, licenses and restrictions
contained in BCP 78, and except as set forth therein, the authors
retain all their rights.
This document and the information contained herein are provided on an
"AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/S HE
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Acknowledgement
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Internet Society.
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