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Versions: (draft-westerlund-mmusic-sdp-bwparam) 00 01 02 03 04 05 06 RFC 3890

Network Working Group                                  Magnus Westerlund
INTERNET-DRAFT                                                  Ericsson
Expires: October 2004                                     April 14, 2004






         A Transport Independent Bandwidth Modifier for the Session
                         Description Protocol (SDP).
                   <draft-ietf-mmusic-sdp-bwparam-06.txt>



Status of this memo


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   This document is a submission of the IETF MMUSIC WG. Comments should
   be directed to the MMUSIC WG mailing list, mmusic@ietf.org.



Abstract


   The existing Session Description Protocol (SDP) bandwidth modifiers
   and their values include the bandwidth needed also 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




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   IP versions, the interpretation of what lower layer bandwidths are
   included is not clear. This document defines an SDP bandwidth
   modifier (TIAS) 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.
















































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TABLE OF CONTENTS


1. Definitions.........................................................4
   1.1. Glossary.......................................................4
   1.2. Terminology....................................................4
2. Introduction........................................................4
   2.1. The Bandwidth Attribute........................................4
      2.1.1. Conference Total..........................................5
      2.1.2. Application Specific Maximum..............................5
      2.1.3. RTCP Report bandwidth.....................................5
   2.2. IPv6 and IPv4..................................................5
   2.3. Further Mechanisms that change the bandwidth utilization.......6
      2.3.1. IPSec.....................................................6
      2.3.2. Header Compression........................................7
3. The Bandwidth Signaling Problems....................................7
   3.1. What IP version is used........................................7
   3.2. Taking other mechanisms into account...........................8
   3.3. Converting bandwidth values....................................9
   3.4. RTCP problems..................................................9
   3.5. Future development............................................10
   3.6. Problem Conclusion............................................10
4. Problem Scope......................................................11
5. Requirements.......................................................11
6. Solution...........................................................11
   6.1. Introduction..................................................11
   6.2. The TIAS bandwidth modifier...................................12
      6.2.1. Usage....................................................12
      6.2.2. Definition...............................................13
      6.2.3. Usage Rules..............................................13
   6.3. Packet Rate parameter.........................................14
   6.4. Converting to Transport-Dependent values......................15
   6.5. Deriving RTCP bandwidth.......................................15
      6.5.1. Motivation for this solution.............................16
   6.6. ABNF definitions..............................................16
   6.7. Example.......................................................17
7. Protocol Interaction...............................................17
   7.1. RTSP..........................................................17
   7.2. SIP...........................................................18
   7.3. SAP...........................................................18
8. Security Consideration.............................................18
9. IANA Considerations................................................19
10. Acknowledgments...................................................19
11. Author's Addresses................................................19
12. References........................................................19
   12.1. Normative references.........................................19
   12.2. Informative References.......................................19









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1. Definitions


1.1. 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.



1.2. 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 RFC 2119 [3].



2. Introduction


   This specification is structured in the following way: In this
   section (2) 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
   consideration, acknowledgements, author's address, reference list,
   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.



2.1. The Bandwidth Attribute







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   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 modifiers defined which are used for different
   purposes.



2.1.1. Conference Total


   The Conference Total is indicated by giving the modifier "CT". The
   meaning of Conference total is to give 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].



2.1.2. 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 in RFC 3550
   [4]. The session bandwidth includes the bandwidth that the RTP data
   traffic will consume, including the lower layers down to IP layer.
   Therefore, the bandwidth is in most cases calculated over RTP
   payload, RTP header, UDP and IP. Defined in RFC 2327 [1].



2.1.3. 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.



2.2. IPv6 and IPv4


   Today there are two IP versions 4 [14] and 6 [13] used in parallel on
   the Internet. This creates problems and 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, using a translation or proxying mechanism




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      would be required.  Work is underway to specify one 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
      put 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.



     ---------------  ---------------  ---------------
     | 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.



2.3. 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 here shortly cover
   a few of these.



2.3.1. IPSec




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   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 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. The extra IP and
   ESP header will in this case significantly increase the packet size.



2.3.2. 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 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.


   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.



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, with for example "b=AS:" for 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 there seems to exist no specification
   pointing this out.




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   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
   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 "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 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 3.2.


   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 3.2.


   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 2.2 and 2.3 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.


   Using IPSec directly between end-points should definitely been known
   to the application, thus enabling it to take the extra headers into
   account. However the same problem exist with the current SDP





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   bandwidth modifiers that 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 to even consider this
   even if 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 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. Also 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 an
   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
   25% increase in required bandwidth for the RTCP traffic. The largest
   increase will be for small RTCP packets when the number of 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




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   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. The IPv6 node may report with 25% longer
   intervals, in the worst case.


   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 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 to use 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
   that really can 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
   include also the bandwidth needed for lower layers. It is in many
   cases difficult to determine which lower layers and their versions
   that 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.


   Secondly there exist no attribute to give the receiver an explicit
   determination of the maximum packet rate that will be used. This
   value is necessary for accurate conversion of any bandwidth values if
   the difference in overhead is known.





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4. Problem Scope


   The problems described in chapter 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 SDP-NG [17].


   As this specification only targets carrying the bit-rate information
   within SDP it will have a limited applicability. As SDP information
   normally is 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 other means 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 does 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


   A solution to the problems outlined in the preceding chapters 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.



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).




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   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 to 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 to use for
      resource reservation. Some characteristics can be derived from the
      stream, codec type, etc. In cases where more information is
      needed, then 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 on 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 through 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
      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 shall be used by RTCP
      equal to 5% of the RTP session bandwidth. The RTCP bandwidth can
      be explicitly allocated by using the RR and RS modifiers defined
      in [9].





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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 and other transport layers like TCP or UDP.


   At the SDP session level, the TIAS value is the maximal amount of
   bandwidth need when all declared media streams are used. This MAY be
   less than the sum of all the individual media streams values. This
   can be 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.


   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 uses the same transport. In cases when the




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   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.


   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 live stream, or for stored on-demand streams, has
   produced. The packet rate is calculated by adding together the number
   of packets sent within a 1 second long 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. for example 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. The
   reason is that 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




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   temporal distribution of media streams maximums. The "maxprate"
   attribute MUST NOT be present at 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.
   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 chapter gives only 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 chapter 6.4, using the actual transport layers used




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   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.
   With sending rate it is meant 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.
   The larger the RTCP packets become, the more it will favor the IPv6
   host in 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]





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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
   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 here used to produce a
   constant rate media stream and does use 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 42300 bits per second. The video also has variable packet
   rate, despite the fact that the video is 15 frames per second there
   where at least one instance when a second long window contained 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




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   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 to the
   client the desired transport profile(s).



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 available in the transport field in the
   "m=" line. The address class can be determined from the "c=" field
   and the client's connectivity.



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, if not integrity protected, be altered. By altering the
   bandwidth value one can fool a receiver to reserve either more or
   less bandwidth than actually needed. Reserving too much may result in
   unwanted expenses on behalf of the user and also blocking of
   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 no tampering can
   be performed and the source 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 and are what can be
   expected 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 what is required to make signaling protocols using SDP
   to work through a middlebox as discussed in the security
   considerations of RFC 3303 [18].






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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.



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 fixing the language and
   finding some errors in the draft. Further thanks for suggestion to
   improvements goes 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. Author's Addresses


      Magnus Westerlund         Tel:   +46 8 4048287
      Ericsson Research         Email: Magnus.Westerlund@ericsson.com
      Ericsson AB
      Torshamnsgatan 23
      SE-164 80 Stockholm, SWEDEN


12. References


12.1. Normative references


   [1]  M. Handley, V. Jacobson, "Session Description Protocol (SDP)",
        IETF RFC 2327, April 1998.
   [2]  D. Crocker and P. Overell, "Augmented BNF for syntax specifica-
        tions: ABNF," RFC 2234, Internet Engineering Task Force, Nov.
        1997.
   [3]  S. Bradner, "Key words for use in RFCs to Indicate Requirement
        Levels", RFC 2119, March 1997.
   [4]  H. Schulzrinne, et. al., "RTP: A Transport Protocol for Real-
        Time Applications", RFC 3550, Internet Engineering Task Force,
        July 2003.


12.2. Informative References


   [5]  M. Handley et al., "Session Announcement Protocol", IETF RFC
        2974, October 2000.
   [6]  J. Rosenberg, et. al., "SIP: Session Initiation Protocol", IETF
        RFC 3261, June 2002.




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   [7]  J. Rosenberg, H. Schulzrine, "An Offer/Answer Model with Session
        Description Protocol (SDP)", IETF RFC 3164, June 2002.
   [8]  H. Schulzrinne, et. al., "Real Time Streaming Protocol (RTSP)",
        IETF RFC 2326, April 1998.
   [9]  S. Casner, "SDP Bandwidth Modifiers for RTCP Bandwidth", IETF
        RFC 3556, Internet Engineering Task Force, July 2003.
   [10] M. Degermark, B. Nordgren, S. Pink, "IP Header Compression",
        IETF RFC 2507, February 1999.
   [11] S. Casner, V. Jacobson, "Compressing IP/UDP/RTP Headers for Low-
        Speed Serial Links", IETF RFC 2508, February 1999.
   [12] C. Bormann, et. al., "RObust Header Compression (ROHC):
        Framework and four profiles", IETF RFC 3095, July 2001.
   [13] S. Deering and R. Hinden, "Internet Protocol, Version 6 (IPv6)
        Specification", RFC 2460, Internet Engineering Task Force,
        December 1998.
   [14] J. Postel, "Internet Protocol", RFC 791, Internet Engineering
        Task Force, 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., et. al., "Integrated Services in the Presence of
        Compressible Flows," RFC 3006, Internet Engineering Task Force,
        November 2000.
   [17] Kutscher, Ott, Bormann, "Session Description and Capability
        Negotiation," IETF draft, work in progress, march 2003.
   [18] P. Srisuresh, J. Kuthan, J. Rosenberg, A. Molitor, A. Rayhan, "
        Middlebox communication architecture and framework," RFC 3303,
        Internet Engineering Task Force, August 2002.
   [19] S. Kent, R. Atkinson, "Security Architecture for the Internet
        Protocol.," RFC 2401, Internet Engineering Task Force, November
        1998.
   [20] S. Kent, R. Atkinson., "IP Authentication Header.," RFC 2402,
        Internet Engineering Task Force, November 1998.
   [21] S. Kent, R. Atkinson., "IP Encapsulating Security Payload
        (ESP).," RFC 2406, November 1998.


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.



Disclaimer of Validity


   This document and the information contained herein are provided on an
   "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE 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





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   INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
   WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.



Changes


   [Note to RFC Editor: Remove this section when publishing]


   The following changes have been done to this version compared to
   draft-ietf-mmusic-sdp-bwparam-05.txt


   - Removed any explicit naming of a translation mechanism.
   - Updated the ID boilerplate in accordance with RFC 3367, and RFC
      3668.
   - Clarified that there exist further mechanisms that effect the
      lower layers, like IPSec.
   - Added a problem conclusion section.




This Internet-Draft expires in October 2004.


































Westerlund                                                     [Page 21]

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