





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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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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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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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
   REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE
   INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR
   IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF
   THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
   WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

Intellectual Property

   The IETF takes no position regarding the validity or scope of any
   Intellectual Property Rights or other rights that might be claimed to
   pertain to the implementation or use of the technology described in
   this document or the extent to which any license under such rights
   might or might not be available; nor does it represent that it has
   made any independent effort to identify any such rights.  Information
   on the IETF's procedures with respect to rights in IETF Documents can
   be found in BCP 78 and BCP 79.

   Copies of IPR disclosures made to the IETF Secretariat and any
   assurances of licenses to be made available, or the result of an
   attempt made to obtain a general license or permission for the use of
   such proprietary rights by implementers or users of this
   specification can be obtained from the IETF on-line IPR repository at
   http://www.ietf.org/ipr.

   The IETF invites any interested party to bring to its attention any
   copyrights, patents or patent applications, or other proprietary
   rights that may cover technology that may be required to implement
   this standard.  Please address the information to the IETF at ietf-
   ipr@ietf.org.

Acknowledgement

   Funding for the RFC Editor function is currently provided by the
   Internet Society.







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