RFC1363 日本語訳

1363 A Proposed Flow Specification. C. Partridge. September 1992. (Format: TXT=59214 bytes) (Status: INFORMATIONAL)
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Network Working Group                                     C. Partridge
Request for Comments: 1363                                         BBN
                                                        September 1992

コメントを求めるワーキンググループC.ヤマウズラ要求をネットワークでつないでください: 1363 BBN1992年9月

                     A Proposed Flow Specification

提案された流れ仕様

Status of this Memo

このMemoの状態

   This memo provides information for the Internet community.  It does
   not specify an Internet standard.  Distribution of this memo is
   unlimited.

このメモはインターネットコミュニティのための情報を提供します。 それはインターネット標準を指定しません。 このメモの分配は無制限です。

Abstract

要約

   A flow specification (or "flow spec") is a data structure used by
   internetwork hosts to request special services of the internetwork,
   often guarantees about how the internetwork will handle some of the
   hosts' traffic.  In the future, hosts are expected to have to request
   such services on behalf of distributed applications such as
   multimedia conferencing.

流れ仕様(または、「流れ仕様」)はインターネットワークの特殊業務、しばしばインターネットワークがどうホストの何らかの交通を扱うかに関する保証を要求するのにインターネットワークホストによって使用されたデータ構造です。 将来、ホストはマルチメディア会議などの分配されたアプリケーションを代表してそのようなサービスを要求しなければならないと予想されます。

   The flow specification defined in this memo is intended for
   information and possible experimentation (i.e., experimental use by
   consenting routers and applications only).  This RFC is a product of
   the Internet Research Task Force (IRTF).

このメモで定義された流れ仕様は情報と可能な実験(すなわち、同意しているルータとアプリケーションだけによる実験用)のために意図します。 このRFCはインターネットResearch Task Force(IRTF)の製品です。

Introduction

序論

   The Internet research community is currently studying the problems of
   supporting a new suite of distributed applications over
   internetworks.  These applications, which include multimedia
   conferencing, data fusion, visualization, and virtual reality, have
   the property that they require the distributed system (the collection
   of hosts that support the applications along with the internetwork to
   which they are attached) be able to provide guarantees about the
   quality of communication between applications.  For example, a video
   conference may require a certain minimum bandwidth to be sure that
   the video images are delivered in a timely way to all recipients.

インターネット研究団体は現在、インターネットワークの上の分配されたアプリケーションの新しい一組を支えるという問題を研究しています。 彼らは分散システム(それらが付けているインターネットワークに伴うアプリケーションを支持するホストの収集)を必要とします。これらのアプリケーション(マルチメディア会議を含んでいます)(データ融合、想像、およびバーチャルリアリティ)には特性がある、それ、アプリケーションのコミュニケーションの品質に関して保証を与えることができてください。 例えば、テレビ会議システムは、ある一定の最小の帯域幅がビデオ画像がすべての受取人へのタイムリーな方法で送られるのを確信しているのを必要とするかもしれません。

   One way for the distributed system to provide guarantees is for hosts
   to negotiate with the internetwork for rights to use a certain part
   of the internetwork's resources.  (An alternative is to have the
   internetwork infer the hosts' needs from information embedded in the
   data traffic each host injects into the network.  Currently, it is
   not clear how to make this scheme work except for a rather limited
   set of traffic classes.)

分散システムが保証を与える1つの方法はホストがインターネットワークのリソースのある部分を使用する権利のためにインターネットワークと交渉することです。 (代替手段はインターネットワークに各ホストがネットワークを注ぐデータ通信量に埋め込まれた情報からホストの必要性を推論させることです。 現在、どのようにかなり限られたセットの交通以外のこの計画仕事をクラスにするかは明確ではありません。)

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   There are a number of ways to effect a negotiation.  For example a
   negotiation can be done in-band or out-of-band.  It can also be done
   in advance of sending data (possibly days in advance), as the first
   part of a connection setup, or concurrently with sending (i.e., a
   host starts sending data and starts a negotiation to try to ensure
   that it will allowed to continue sending).  Insofar as is possible,
   this memo is agnostic with regard to the variety of negotiation that
   is to be done.

交渉に作用する多くの方法があります。 例えば、バンドかバンドの外で交渉できます。 また、接続設定の最初の部分としてデータ(ことによると何日も早い)を送る前か同時に発信でそれができます(すなわち、ホストは、データを送り始めて、発信し続けているのが許容されて、そうするのを保証しようとするために交渉を始めます)。 このメモは可能であるようにその程度において、することになっている交渉のバラエティーに関して不可知論者です。

   The purpose of this memo is to define a data structure, called a flow
   specification or flow spec, that can be used as part of the
   negotiation to describe the type of service that the hosts need from
   the internetwork.  This memo defines the format of the fields of the
   data structure and their interpretation.  It also briefly describes
   what purpose the different fields fill, and discusses why this set of
   fields is thought to be both necessary and sufficient.

このメモの目的はホストがインターネットワークから必要とするサービスのタイプについて説明するのに交渉の一部として使用できる流れ仕様か流れ仕様と呼ばれるデータ構造を定義することです。 このメモはデータ構造と彼らの解釈の分野の書式を定義します。 それは、また、異なった分野がどんな目的をいっぱいにするかを簡潔に説明して、なぜ必要であって、かつこのセットの分野が十分であると考えられるかを議論します。

   It is important to note that the goal of this flow spec is to able to
   describe *any* flow requirement, both for guaranteed flows and for
   applications that simply want to give hints to the internetwork about
   their requirements.

どんな*流れ要件、保証された流れ、およびそれらの要件に関して単にヒントをインターネットワークに与えたがっているアプリケーションのための*についても説明できるのにこの流れ仕様の目標があることに注意するのは重要です。

Format of the Flow Spec

流れ仕様の形式

       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |              Version          |    Maximum Transmission Unit  |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |      Token Bucket Rate        |        Token Bucket Size      |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |  Maximum Transmission Rate    |     Minimum Delay Noticed     |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |     Maximum Delay Variation   |        Loss Sensitivity       |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |     Burst Loss Sensitivity    |          Loss Interval        |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |    Quality of Guarantee       |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | バージョン| マキシマム・トランスミッション・ユニット| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 象徴バケツレート| 象徴バケツサイズ| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 最大の通信速度| 最小の遅れに気付きました。| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 最大の遅れ変化| 損失感度| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 損失感度を押し破いてください。| 損失間隔| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 保証の品質| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Discussion of the Flow Spec

流れ仕様の議論

   The flow spec indicates service requirements for a single direction.
   Multidirectional flows will need to request services in both
   directions (using two flow specs).

流れ仕様はただ一つの指示のためのサービス要件を示します。 Multidirectional流れは、両方の方向にサービスを要求する必要があるでしょう(2つの流れ仕様を使用して)。

   To characterize a unidirectional flow, the flow spec needs to do four
   things.

単方向の流動を特徴付けるために、流れ仕様は、4つのことをする必要があります。

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   First, it needs to characterize how the flow's traffic will be
   injected into the internetwork.  If the internetwork doesn't know
   what to expect (is it a gigabit-per-second flow or a three kilobit-
   per-second flow?) then it is difficult for the internetwork to make
   guarantees.  (Note the word "difficult" rather than "impossible."  It
   may be possible to statistically manage traffic or over-engineer the
   network so well that the network can accept almost all flows, without
   setup.  But this problem looks far harder than asking the sender to
   approximate its behavior so the network can plan.)  In this flow
   spec, injected traffic is characterized as having a sustainable rate
   (the token bucket rate) a peak rate (the maximum transmission rate),
   and an approximate burst size (the token bucket size).  A more
   precise definition of each of these fields is given below.  The
   characterization is based, in part, on the work done in [1].

まず最初に、それは、流れの交通がどうインターネットワークに注がれるかを特徴付ける必要があります。 インターネットワークが、何を予想したらよいかを知らないなら(それは、1秒あたりのギガビット流動かそれとも1 2番目あたり1つの3キロビット流動ですか?)、インターネットワークが保証をするのは、難しいです。 (「不可能である」というよりもむしろ「難しい」状態で単語に注意してください。 ネットワークがほとんどすべての流れを受け入れることができるくらいよく統計的に交通を管理するか、またはネットワークを過剰設計するのが可能であるかもしれません、セットアップなしで。 しかし、この問題はネットワークが計画されることができるように振舞いに近似するように送付者に頼むよりはるかに困難に見えます。) この流れ仕様では、注入された交通は持続可能なレート(象徴バケツレート)に、ピークレート(最大の通信速度)、および大体の放出量(象徴バケツサイズ)を持っているとして特徴付けられます。 それぞれのこれらの分野の、より正確な定義を以下に与えます。 特殊化は[1]で行われた仕事に一部基づいています。

   Second, the flow spec needs to characterize sensitivity to delay.
   Some applications are more sensitive than others.  At the same time,
   the internetwork will likely have a choice of routes with various
   delays available from the source to destination.  For example, both
   routes using satellites (which have very long delays) and routes
   using terrestrial lines (which will have shorter delays) may be
   available.  So the sending host needs to indicate the flow's
   sensitivity to delay.  However, this field is only advisory.  It only
   tells the network when to stop trying to reduce the delay - it does
   not specify a maximum acceptable delay.

2番目に、流れ仕様は、延着するように感度を特徴付ける必要があります。 いくつかのアプリケーションが他のものより敏感です。 同時に、インターネットワークには、ルートの様々な遅れが利用可能のソースから目的地までの選択がおそらくあるでしょう。 例えば、衛星を使用するルート(非常に長い遅れを持っている)と地球の線を使用するルート(より少しな遅れを持つ)の両方が利用可能であるかもしれません。 それで、送付ホストは、延着するように流れの感度を示す必要があります。 しかしながら、この分野は唯一の状況報告です。 それは、遅れを減少させようとするのをいつ止めるかをネットワークに言うだけです--それは最大の許容できる遅れを指定しません。

   There are two problems with allowing applications to specify the
   maximum acceptable delay.

アプリケーションが最大の許容できる遅れを指定するのを許容することに関する2つの問題があります。

   First, observe that an application would probably be happy with a
   maximum delay of 100 ms between the US and Japan but very unhappy
   with a delay of 100 ms within the same city.  This observation
   suggests that the maximum delay is actually variable, and is a
   function of the delay that is considered achievable.  But the
   achievable delay is largely determined by the geographic distance
   between the two peers, and this sort of geographical information is
   usually not available from a network.  Worse yet, the advent of
   mobile hosts makes such information increasingly hard to provide.  So
   there is reason to believe that applications may have difficulty
   choosing a rational maximum delay.

まず最初に、アプリケーションがたぶん米国と日本の間の100msの最大の遅れに満足ですが、同じ都市の中で100msの遅れに非常に不満であることを観測してください。 この観測は最大の遅れが実際に可変であり、達成可能であると考えられる遅れの機能であると示唆します。 しかし、2人の同輩の間の地理的な距離に従って、達成可能な遅れは主に決定しています、そして、通常、この種類の地理的な情報はネットワークから利用可能ではありません。 まだより悪くて、モバイルホストの到来は、そのような情報を提供するのをますます困難にします。 それで、アプリケーションが合理的な最大の遅れを選ぶのに苦労するかもしれないと信じる理由があります。

   The second problem with maximum delays is that they are an attempt to
   quantify what performance is acceptable to users, and an application
   usually does not know what performance will be acceptable its user.
   For example, a common justification for specifying a maximum
   acceptable delay is that human users find it difficult to talk to
   each other over a link with more than about 100 ms of delay.
   Certainly such delays can make the conversation less pleasant, but it

最大の遅れに関する2番目の問題はそれらが通常、ユーザにとって、どんな性能が許容しているか、そして、アプリケーションを定量化する試みが、どんな性能を許容するかを知らないということであるということです。そのユーザ。 例えば、最大の許容できる遅れを指定するための一般的な正当化は人間のユーザが、遅れのおよそ100以上msとのリンクの上に互いに話すのが難しいのがわかるということです。 しかし、確かに、そのような遅れで会話が、より快くなくなる場合がある、それ

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   is still possible to converse when delays are several seconds long,
   and given a choice between no connection and a long delay, many users
   will pick the delay.  (The phone call may involve an important matter
   that must be resolved.)

接続がなくて長時間の遅延の選択を遅れが長さ数秒であるとき、話すのにおいてまだ可能であり、考えて、多くのユーザが遅れを選ぶでしょう。 (電話は決議しなければならない要件にかかわるかもしれません。)

   As part of specifying a flow's delay sensitivity, the flow spec must
   also characterize how sensitive the flow is to the distortion of its
   data stream.

流れの遅れ感度を指定する一部、また仕様が特徴付けなければならない流れとして、流れがデータのひずみにどれくらい敏感であるかは流れます。

   Packets injected into a network according to some pattern will not
   normally come out of the network still conforming to the pattern.
   Instead, the pattern will have been distorted by queueing effects in
   the network.  Since there is reason to believe that it may make
   network design easier to continue to allow the networks slightly
   distort traffic patterns, it is expected that those applications
   which are sensitive to distortion will require their hosts to use
   some amount of buffering to reshape the flow back into its original
   form.  It seems reasonable to assume that buffer space is not
   infinite and that a receiving system will wish to limit the amount of
   buffering that a single flow can use.

通常、何らかのパターンによると、ネットワークに注がれたパケットはまだパターンに従っているネットワークから出て来ないでしょう。 代わりに、パターンはネットワークの待ち行列効果によって歪められてしまうでしょう。 ずっとネットワークをわずかにより許容しやすいネットワークデザインがそれでトラフィック・パターンを歪めるかもしれないと信じる理由があるので、ひずみに敏感なそれらのアプリケーションが、彼らのホストが原型への流れを造り直すのにいくらかの量のバッファリングを使用するのを必要とすると予想されます。 バッファ領域が無限でなく、受電方式がただ一つの流れが使用できるそれをバッファリングする量を制限したくなると仮定するのは妥当に思えます。

   The amount of buffer space required for removing distortion at the
   receiving system is determined by the variation in end-to-end
   transmission delays for data sent over the flow.  If the transmission
   delay is a mean delay, D, plus or minus a variance, V, the receiving
   system needs buffer space equivalent to 2 * V * the transmission
   rate.  To see why this is so, consider two packets, A and B, sent T
   time units apart which must be delivered to the receiving application
   T time units apart.  In the worst case, A arrives after a delay of
   D-V time units (the minimum delay) and B arrives after a delay of D+V
   time units (the maximum delay).  The receiver cannot deliver B until
   it arrives, which is T + 2 * V time units after A.  To ensure that A
   is delivered T time units before B, A must be buffered for 2 * V time
   units.  The delay variance field is the value of 2 * V, and allows
   the receiver to indicate how much buffering it is willing to provide.

受電方式でひずみを取り除くのに必要であるバッファ領域の量は流れの上に送られたデータのために終わりから終わりへのトランスミッション遅れの変化によって測定されます。 トランスミッション遅れが意地悪な遅れ、Dであるなら、そのうえか変化、Vを引いて、受電方式は2*に同等なバッファ領域を必要とします。通信速度対* これがなぜそうであるかを確認するには、2つのパケット(AとB)がTタイム・ユニット離れて発信したと考えてください(Tタイム・ユニット離れて、受信アプリケーションに渡さなければなりません)。 最悪の場合には、D-Vタイム・ユニット(最小の遅れ)とBの遅れがD+Vタイム・ユニット(最大の遅れ)の遅れの後に到着した後にAは到着します。 受信機は到着するまでBを渡すことができません(A.ToがBのTタイム・ユニット前にAを渡して、Aを2*Vタイム・ユニットバッファリングしなければならないのを確実にした後にT+2つの*Vタイム・ユニットです)。 遅れ変化の分野は、2*Vの値であり、受信機が、どのくらい提供するかを構わないそれをバッファリングするのが、思っている示すのを許容します。

   A third function of the flow spec is to signal sensitivity to loss of
   data.  Some applications are more sensitive to the loss of their data
   than other applications.  Some real-time applications are both
   sensitive to loss and unable to wait for retransmissions of data.
   For these particularly sensitive applications, hosts may implement
   forward error correction on a flow to try to absolutely minimize
   loss.  The loss fields allow hosts to request loss properties
   appropriate for the application's requirements.

流れ仕様の3番目の機能はデータの喪失に感度に合図することです。 いくつかのアプリケーションが他のアプリケーションよりそれらのデータの損失に敏感です。 いくつかのリアルタイムのアプリケーションは、損失に敏感であって、かつデータの「再-トランスミッション」を待つことができません。 これらの特に敏感なアプリケーションのために、ホストは損失を絶対に最小にしようとする流れで前進型誤信号訂正を実行するかもしれません。 損失分野で、ホストはアプリケーションの要件に、適切な損失の特性を要求できます。

   Finally, it is expected that the internetwork may be able to provide
   a range of service guarantees.  At the best, the internetwork may be
   asked to guarantee (with tight probability bounds) the quality of

最終的に、インターネットワークがさまざまなサービス保証を提供できるかもしれないと予想されます。 せいぜい、インターネットワークが品質を保証するように(きつい確率領域で)頼まれるかもしれません。

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   service it will provide.  Or the internetwork may simply be asked to
   ensure that packets sent over the flow take a terrestrial path.  The
   quality of guarantee field indicates what type of service guarantee
   the application desires.

それが提供するサービス。 または、インターネットワークが、流れの上に送られたパケットが地球の経路を取るのを保証するように単に頼まれるかもしれません。 保証分野の品質は、どんなタイプのサービスがアプリケーション願望を保証するかを示します。

Definition of Individual Fields

個々の分野の定義

General Format of Fields

分野の一般形式

   With a few exceptions, fields of the flow spec are expressed using a
   common 16-bit format.  This format has two forms.  The first form is
   shown below.

若干の例外はあるものの、流れ仕様の分野は、一般的な16ビットの形式を使用することで言い表されます。 この形式には、2つのフォームがあります。最初の書式は以下に示されます。

               0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
              +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
              |0|  Exponent   |     Value     |
              +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |0| 解説者| 値| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   In this format, the first bit is 0, followed by 7 bits of an exponent
   (E), and an 8-bit value (V).  This format encodes a number, of the
   form V * (2**E).  This representation was chosen to allow easy
   representation of a wide range of values, while avoiding over-precise
   representations.

この形式では、最初のビットは、解説者(E)の7ビットが支えた0、8ビット値(V)です。 この形式はフォームV*の数(2**E)をコード化します。 この表現は、さまざまな値の簡単な表現を許容するために過剰正確な表現を避けている間、選ばれました。

   In some case, systems will not wish to request a precise value but
   rather simply indicate some sensitivity.  For example, a virtual
   terminal application like Telnet will likely want to indicate that it
   is sensitive to delay, but it may not be worth expressing particular
   delay values for the network to try to achieve.  For these cases,
   instead of a number, the field in the flow spec will take the
   following form:

いつかの場合では、システムは、正確な値を要求しますが、かなり単に何らかの感度を示したくならないでしょう。 例えば、Telnetのような仮想端末アプリケーションは、延着するのが敏感であることをおそらく示したくなるでしょうが、それは、ネットワークが試みるように特定の遅れ値を言い表す、達成するために価値がないかもしれません。 これらのケースのために、数の代わりに、流れ仕様の分野は以下の形を取るでしょう:

               0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
              +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
              |1|   Well-defined Constant     |
              +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |1| 明確な定数| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   The first bit of the field is one, and is followed by a 15-bit
   constant.  The values of the constants for given fields are defined
   below.  Any additional values can be requested from the Internet
   Assigned Numbers Authority (IANA).

分野の最初のビットの1であり、15ビットの定数はあとに続いています。 与えられた分野への定数の値は以下で定義されます。 インターネットAssigned民数記Authority(IANA)からどんな加算値も要求できます。

   Version Field

バージョン分野

      This field is a 16-bit integer in Internet byte order.  It is the
      version number of the flow specification.  The version number of
      the flow specification defined in this document is 1.  The IANA is
      responsible for assigning future version numbers for any proposed

この分野はインターネットバイトオーダーで16ビットの整数です。 それは流れ仕様のバージョン番号です。 本書では定義された流れ仕様のバージョン番号は1です。 IANAはいずれの将来のバージョン番号が提案した割り当てに責任があります。

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      revisions of this flow specification.

この流れ仕様の改正。

      This field does not use the general field format.

この分野は一般的なフィールド形式を使用しません。

   Maximum Transmission Unit (MTU)

マキシマム・トランスミッション・ユニット(MTU)

      A 16-bit integer in Internet byte order which is the maximum
      number of bytes in the largest possible packet to be transmitted
      over this flow.

この流れの上の伝えられるべき可能な限り大きいパケットのバイトの最大数であるインターネットバイトオーダーにおける16ビットの整数。

      This field does not use the general field format.

この分野は一般的なフィールド形式を使用しません。

      The field serves two purposes.

分野は2つの目的に役立ちます。

      It is a convenient unit for expressing loss properties.  Using the
      default MTU of the internetwork is inappropriate since the
      internetwork have very large MTU, such the 64Kbytes of IP, but
      applications and hosts may be sensitive to losses of far less than
      an MTU's amount of data -- for example, a voice application would
      be sensitive to a loss of several consecutive small packets.

それは、損失の特性を示すための便利なユニットです。 アプリケーションとホストは遠いことの1未満MTUのデータ量の損失に敏感であるかもしれません--インターネットワークには非常に大きいMTU、IPのそのような64Kbytesがあるので、インターネットワークのデフォルトMTUを使用するのは不適当ですが、例えば、音声アプリケーションはいくつかの連続した小型小包の損失に敏感でしょう。

      The MTU also bounds the amount of time that a flow can transmit,
      uninterrupted, on a shared media.

また、MTUは流れが伝わることができるとぎれない時間に共有されたメディアでバウンドします。

      Similarly, the loss rates of links that suffer bit errors will
      vary dramatically based on the MTU size.

同様に、噛み付いている誤りを受けるリンクの損失率はMTUサイズに基づいてガラッと変わるでしょう。

   Token Bucket Rate

象徴バケツレート

      The token bucket rate is one of three fields used to define how
      traffic will be injected into the internetwork by the sending
      application.  (The other two fields are the token bucket size and
      the maximum transmission rate.)

象徴バケツレートは交通が送付アプリケーションでどうインターネットワークに注がれるかを定義するのに使用される3つの分野の1つです。 (他の2つの分野が、象徴バケツサイズと最大の通信速度です。)

      The token rate is the rate at which tokens (credits) are placed
      into an imaginary token bucket.  For each flow, a separate bucket
      is maintained.  To send a packet over the flow, a host must remove
      a number of credits equal to the size of the packet from the token
      bucket.  If there are not enough credits, the host must wait until
      enough credits accumulate in the bucket.

象徴レートは象徴(掛ける)が想像する象徴バケツの中に置かれる速度です。 各流れにおいて、別々のバケツは維持されます。 流れの上にパケットを送るために、ホストは象徴バケツからパケットのサイズと等しい多くのクレジットを取り除かなければなりません。 十分なクレジットがなければ、十分なクレジットがバケツの中に蓄積するまで、ホストは待たなければなりません。

      Note that the fact that the rate is expressed in terms of a token
      bucket rate does not mean that hosts must implement token buckets.
      Any traffic management scheme that yields equivalent behavior is
      permitted.

レートが象徴バケツレートで表されるという事実が、ホストが象徴バケツを実行しなければならないことを意味しないことに注意してください。 同等な振舞いをもたらすどんな輸送管理計画も受入れられます。

      The field is in the general field format and counts the number of
      byte credits (i.e., right to send a byte) per second which are

分野は、一般的なフィールド形式にはあって、そうする1秒あたりのバイトクレジット(すなわち、1バイト送るために正しい)の数を数えます。

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      deposited into the token bucket.  The value must be a number (not
      a well-known constant).

象徴バケツの中に預けられます。 値は数でなければなりません(周知の定数でない)。

      The value zero is slightly special.  It is used to indicate that
      the application is not making a request for bandwidth guarantees.
      If this field is zero, then the Token Bucket Size must also be
      zero, and the type of guarantee requested may be no higher than
      predicted service.

値ゼロはわずかに特別です。 それは、アプリケーションが帯域幅保証を求める要求をしていないのを示すのに使用されます。 この分野がゼロであるなら、また、Token Bucket Sizeはゼロと、予測されるより高いノーがサービスであったかもしれないなら要求された保証のタイプであるに違いありません。

   Token Bucket Size

象徴バケツサイズ

      The token bucket size controls the maximum amount of data that the
      flow can send at the peak rate.  More formally, if the token
      bucket size is B, and the token bucket rate is R, over any
      arbitrarily chosen interval T in the life of the flow, the amount
      of data that the flow sends cannot have exceeded B + (R * T)
      bytes.

象徴バケツサイズは流れがピークレートで送ることができる最大のデータ量を制御します。 より正式に、象徴バケツサイズがBであり、象徴バケツレートがRであるなら、流れの人生におけるどんな任意に選ばれた間隔Tにわたっても流れが送るデータ量はB+(R*T)バイトを超えていたはずがありません。

      The token bucket is filled at the token bucket rate.  The bucket
      size limits how many credits the flow may store.  When the bucket
      is full, new credits are discarded.

象徴バケツは象徴バケツレートでいっぱいにされます。 バケツサイズは流れが格納するかもしれないいくつのクレジットを制限するか。 バケツが完全であるときに、新しいクレジットは捨てられます。

      The field is in the general field format and indicates the size of
      the bucket in bytes.  The value must be a number.

分野は、一般的なフィールド形式にはあって、バイトでバケツのサイズを示します。 値は数でなければなりません。

      Note that the bucket size must be greater than or equal to the MTU
      size.

バケツサイズがそうしなければならないというメモはそう以上です。MTUサイズ。

      Zero is a legal value for the field and indicates that no credits
      are saved.

ゼロは、分野への法定価格であり、クレジットが全く救われないのを示します。

   Maximum Transmission Rate

最大の通信速度

      The maximum transmission rate limits how fast packets may be sent
      back to back from the host.  Consider that if the token bucket is
      full, it is possible for the flow to send a series of back-to-back
      packets equal to the size of the token bucket.  If the token
      bucket size is large, this back-to-back run may be long enough to
      significantly inhibit multiplexing.

速いパケットがあるかもしれない最大の通信速度限界は背中合わせにホストから送りました。 象徴バケツが完全であるなら、流れが象徴バケツのサイズと等しい一連の背中合わせのパケットを送るのが、可能であると考えてください。 象徴バケツサイズが大きいなら、この連続の走行は多重送信するのをかなり禁止できるくらい長いかもしれません。

      To limit this effect, the maximum transmission rate bounds how
      fast successive packets may be placed on the network.

この効果を制限するために、速い連続したパケットがネットワークに置かれるかもしれない最大の通信速度はバウンドしています。

      One can think of the maximum transmission rate control as being a
      form of a leaky bucket.  When a packet is sent, a number of
      credits equal to the size of the packet is placed into an empty
      bucket, which drains credits at the maximum transmission rate.  No
      more packets may be sent until the bucket has emptied again.

人は水漏れするバケツのフォームであるとして最大の通信速度コントロールを考えることができます。 パケットを送るとき、パケットのサイズと等しい多くのクレジットを空のバケツの中に置きます。バケツは最大の通信速度でクレジットを消耗させます。 バケツが再び空になるまで、それ以上のパケットを全く送らないかもしれません。

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      The maximum transmission rate is the rate at which the bucket is
      emptied.  The field is in the general field format and indicates
      the size of the bucket in bytes.  The value must be a number and
      must be greater than or equal to the token bucket rate.

最大の通信速度はバケツが空にされているレートです。 分野は、一般的なフィールド形式にはあって、バイトでバケツのサイズを示します。 値は、数でなければならなく、象徴バケツが、より評価するということでなければなりません。

      Note that the MTU size can be used in conjunction with the maximum
      transmission rate to bound how long an individual packet blocks
      other transmissions.  The MTU specifies the maximum time an
      individual packet may take.  The Maximum Transmission Rate, limits
      the frequency with which packets may be placed on the network.

MTUサイズがどれくらい長い間バウンドするかに最大の通信速度に関連して使用されて、個々のパケットが他のトランスミッションを妨げるということであるかもしれないことに注意してください。 MTUは個々のパケットがかかるかもしれない最大の時間を指定します。 Maximum Transmission Rate、パケットがネットワークに置かれるかもしれない頻度を制限します。

   Minimum Delay Noticed

最小の遅れに気付きました。

      The minimum delay noticed field tells the internetwork that the
      host and application are effectively insensitive to improvements
      in end-to-end delay below this value.  The network is encouraged
      to drive the delay down to this value but need not try to improve
      the delay further.

最小の遅れは、分野が、事実上、ホストとアプリケーションがこの値の下で終わりから終わりへの遅れにおける改良に鈍感であるとインターネットワークに言うのに気付きました。 ネットワークは、遅れをこの値まで追い立てるよう奨励されますが、さらに遅れを改良しようとする必要はありません。

      The field is in the general field format.

一般的なフィールド形式には分野があります。

      If expressed as a number it is the number of microseconds of delay
      below which the host and application do not care about
      improvements.  Human users only care about delays in the
      millisecond range but some applications will be computer to
      computer and computers now have clock times measured in a handful
      of nanoseconds.  For such computers, microseconds are an
      appreciable time.  For this reason, this field measures in
      microseconds, even though that may seem small.

数として言い表されるなら、それはホストとアプリケーションが改良を心配しないマイクロセカンドの遅れの数です。 いくつかのアプリケーションがコンピュータへのコンピュータになるでしょう、そして、人間のユーザはミリセカンド範囲で遅れを心配するだけですが、コンピュータで、一握りのナノ秒以降、現在、時計回を測定します。 そのようなコンピュータに関しては、マイクロセカンドはかなりの時間です。 この理由で、それは小さく見えるかもしれませんが、この分野はマイクロセカンドのときに測定します。

      If expressed as a well-known constant (first bit set), two field
      values are accepted:

周知の定数(最初のビットはセットした)として言い表すなら、2つの分野値を受け入れます:

         0 - the application is not sensitive to delay

0--アプリケーションは、遅らせるために敏感ではありません。

         1 - the application is moderately delay sensitive
             e.g., avoid satellite links where possible).

1、--、アプリケーションが遅れ適度に敏感であり、例えば、可能であるところで衛星中継を避けてください、)

   Maximum Delay Variation

最大の遅れ変化

      If a receiving application requires data to be delivered in the
      same pattern that the data was transmitted, it may be necessary
      for the receiving host to briefly buffer data as it is received so
      that the receiver can restore the old transmission pattern.  (An
      easy example of this is a case where an application wishes to send
      and transmit data such as voice samples, which are generated and
      played at regular intervals.  The regular intervals may be
      distorted by queueing effects in the network and the receiver may

受信アプリケーションが、データが同じパターンを果たされるのを必要とするなら、それのようにデータが送られて、受信ホストが簡潔にデータをバッファリングするのが必要であることは、受信機が古いトランスミッションパターンを復元できるように、受け取られています。 (この簡単な例はアプリケーションが声のサンプルなどのデータを送って、送りたがっているケースであり、どれが発生して、一定の間隔一定の間隔で上演されられるかはネットワークの待ち行列効果で歪められるかもしれなくて、受信機はそのようなケースであるかもしれません。

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      have to restore the regular spacing.)

通常のスペースを回復するために、持っています。)

      The amount of buffer space that the receiving host is willing to
      provide determines the amount of variation in delay permitted for
      individual packets within a given flow.  The maximum delay
      variation field makes it possible to tell the network how much
      variation is permitted.  (Implementors should note that the
      restrictions on the maximum transmission rate may cause data
      traffic patterns to be distorted before they are placed on the
      network, and that this distortion must be accounted for in
      determining the receiver buffer size.)

受信ホストが提供しても構わないと思っているバッファ領域の量は個々のパケットのために与えられた流れの中で受入れられた遅れの変化の量を測定します。 最大の遅れ変化分野で、どのくらいの変化が受入れられるかをネットワークに言うのが可能になります。 (作成者はそれらがネットワークに置かれる前に最大の通信速度における制限でデータトラフィック・パターンを歪めるかもしれなくて、このひずみは決定における受信機バッファサイズを説明しなければならないことに注意するべきです。)

      The field is in the general field format and must be a number.  It
      is the difference, in microseconds, between the maximum and
      minimum possible delay that a packet will experience.  (There is
      some question about whether microsecond units are too large.  At a
      terabit per second, one microsecond is a megabit.  Presumably if a
      host is willing to receive data at terabit speeds it is willing to
      provide megabits of buffer space.)

分野は、一般的なフィールド形式にはあって、数であるに違いありません。 それはパケットが経験する最大の、そして、最小の可能な遅れの間のマイクロセカンドの違いです。 (マイクロセカンド単位が大き過ぎるかどうかに関する何らかの質問があります。 1秒あたり1つのテラビットでは、1マイクロセカンドはメガビットです。 おそらく、ホストが、テラビット速度でデータを受け取っても構わないと思っているなら、それは、バッファ領域のメガビットを提供しても構わないと思っています。)

      The value of 0, meaning the receiving host will not buffer out
      delays, is acceptable but the receiving host must still have
      enough buffer space to receive a maximum transmission unit sized
      packet from the sending host.  Note that it is expected that a
      value of 0 will make it unlikely that a flow can be established.

0の値(受信ホストが遅れからバッファリングしない意味)は許容できますが、受信ホストには、送付ホストからマキシマム・トランスミッション・ユニットの大きさで分けられたパケットを受けることができるくらいのバッファ領域がまだなければなりません。 0の値で流れを確立できるのがありそうもなくなると予想されることに注意してください。

   Loss Sensitivity

損失感度

      This field indicates how sensitive the flow's traffic is to
      losses.  Loss sensitivity can be expressed in one of two ways:
      either as a number of losses of MTU-sized packets in an interval,
      or simply as a value indicating a level of sensitivity.

この分野は、流れの交通が損失にどれくらい敏感であるかを示します。 2つの方法の1つで損失感度を言い表すことができます: 間隔のMTUサイズのパケットの損害件数として、または、単に感度のレベルを示す値として。

      The field is in the general field format.

一般的なフィールド形式には分野があります。

      If the value is a number, then the value is the number of MTU-
      sized packets that may be lost out of the number of MTU-sized
      packets listed in the Loss Interval field.

値が数であるなら、値はLoss Interval分野に記載されたMTUサイズのパケットの数から失われるかもしれないMTUの大きさで分けられたパケットの数です。

      If the value is a well-known constant, then one of two values is
      permitted:

値が周知の定数であるなら、2つの値の1つは受入れられます:

         0 - the flow is insensitive to loss

0--流れは損失に神経が鈍いです。

         1 - the flow is sensitive to loss (where possible
             choose the path with the lowest loss rate).

1--流れは損失に敏感です(可能であるところで最も低い損失率がある経路を選んでください)。

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   Burst Loss Sensitivity

損失感度を押し破いてください。

      This field states how sensitive the flow is to losses of
      consecutive packets.  The field enumerates the maximum number of
      consecutive MTU-sized packets that may be lost.

この分野は、流れが連続したパケットの損失にどれくらい敏感であるかを述べます。 分野は失われるかもしれない連続したMTUサイズのパケットの最大数を列挙します。

      The field is in the general field format.

一般的なフィールド形式には分野があります。

      If the value is a number, then the value is the number of
      consecutive MTU-sized packets that may be lost.

値が数であるなら、値は失われるかもしれない連続したMTUサイズのパケットの数です。

      If the value is a well-known constant, then the value 0 indicates
      that the flow is insensitive to burst loss.

値が周知の定数であるなら、値0は、流れが損失を押し破くために神経が鈍いのを示します。

      Note that it is permissible to set the loss sensitivity field to
      simply indicate sensitivity to loss, and set a numerical limit on
      the number of consecutive packets that can be lost.

損失感度分野に単に感度を損失に示すように設定するのが許されていることに注意してください、そして、失われる場合がある連続したパケットの数における数字の限界を設定してください。

   Loss Interval

損失間隔

      This field determines the period over which the maximum number of
      losses per interval are measured.  In other words, given any
      arbitrarily chosen interval of this length, the number of losses
      may not exceed the number in the Loss Sensitivity field.

この分野は1間隔あたりの最大の損害件数が測定される期間を決定します。 言い換えれば、この長さのどんな任意に選ばれた間隔も考えて、損害件数はLoss Sensitivity分野で数を超えないかもしれません。

      The field is in the general field format.

一般的なフィールド形式には分野があります。

      If the Loss Sensitivity field is a number, then this field must
      also be a number and must indicate the number of MTU-sized packets
      which constitutes a loss interval.

Loss Sensitivity分野が数であるなら、この分野は、また、数でなければならなく、損失間隔を構成するMTUサイズのパケットの数を示さなければなりません。

      If the Loss Sensitivity field is not a number (i.e., is a well-
      known constant) then this field must use the well-known constant
      of 0 (i.e., first bit set, all other bits 0) indicating that no
      loss interval is defined.

Loss Sensitivity分野が数(すなわち、よく知られている定数である)でないなら、損失間隔が全く定義されないのを示しながら、この分野は0(すなわち、最初のビットセット、他のすべてのビット0)の周知の定数を使用しなければなりません。

   Quality of Guarantee

保証の品質

      It is expected that the internetwork will likely have to offer
      more than one type of guarantee.

インターネットワークがおそらく1つ以上のタイプの保証を提供しなければならないと予想されます。

      There are two unrelated issues related to guarantees.

保証に関連する関係ない2冊があります。

      First, it may not be possible for the internetwork to make a firm
      guarantee.  Consider a path through an internetwork in which the
      last hop is an Ethernet.  Experience has shown (e.g., some of the
      IETF conferencing experiments) that an Ethernet can often give
      acceptable performance, but clearly the internetwork cannot

First, it may not be possible for the internetwork to make a firm guarantee. Consider a path through an internetwork in which the last hop is an Ethernet. Experience has shown (e.g., some of the IETF conferencing experiments) that an Ethernet can often give acceptable performance, but clearly the internetwork cannot

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      guarantee that the Ethernet will not saturate at some time during
      a flow's lifetime.  Thus it must be possible to distinguish
      between flows which cannot tolerate the small possibility of a
      failure (and thus must guaranteed at every hop in the path) and
      those that can tolerate islands of uncertainty.

guarantee that the Ethernet will not saturate at some time during a flow's lifetime. Thus it must be possible to distinguish between flows which cannot tolerate the small possibility of a failure (and thus must guaranteed at every hop in the path) and those that can tolerate islands of uncertainty.

      Second, there is some preliminary work (see [2]) that suggests
      that some applications will be able to adapt to modest variations
      in internetwork performance and that network designers can exploit
      this flexibility to allow better network utilization.  In this
      model, the internetwork would be allowed to deviate slightly from
      the promised flow parameters during periods of load.  This class
      of service is called predicted service (to distinguish it from
      guaranteed service).

Second, there is some preliminary work (see [2]) that suggests that some applications will be able to adapt to modest variations in internetwork performance and that network designers can exploit this flexibility to allow better network utilization. In this model, the internetwork would be allowed to deviate slightly from the promised flow parameters during periods of load. This class of service is called predicted service (to distinguish it from guaranteed service).

      The difference between predicted service and service which cannot
      be perfectly guaranteed (e.g., the Ethernet example mentioned
      above) is that the imperfect guarantee makes no statistical
      promises about how it might mis-behave.  In the worst case, the
      imperfect guarantee will not work at all, whereas predicted
      service will give slightly degraded service.  Note too that
      predicted service assumes that the routers and links in a path all
      cooperate (to some degree) whereas an imperfect guarantee states
      that some routers or links will not cooperate.

The difference between predicted service and service which cannot be perfectly guaranteed (e.g., the Ethernet example mentioned above) is that the imperfect guarantee makes no statistical promises about how it might mis-behave. In the worst case, the imperfect guarantee will not work at all, whereas predicted service will give slightly degraded service. Note too that predicted service assumes that the routers and links in a path all cooperate (to some degree) whereas an imperfect guarantee states that some routers or links will not cooperate.

      The field is a 16-bit field in Internet byte order.  There are six
      legal values:

The field is a 16-bit field in Internet byte order. There are six legal values:

         0 - no guarantee is required (the host is simply expressing
             desired performance for the flow)

0 - no guarantee is required (the host is simply expressing desired performance for the flow)

         100 (hex) - an imperfect guarantee is requested.

100 (hex) - an imperfect guarantee is requested.

         200 (hex) - predicted service is requested and if unavailable,
                     then no flow should be established.

200 (hex) - predicted service is requested and if unavailable, then no flow should be established.

         201 (hex) - predicted service is requested but an imperfect
                     guarantee is acceptable.

201 (hex) - predicted service is requested but an imperfect guarantee is acceptable.

         300 (hex) - guaranteed service is requested and if a firm
                     guarantee cannot be given, then no flow should be
                     established.

300 (hex) - guaranteed service is requested and if a firm guarantee cannot be given, then no flow should be established.

         301 (hex) - guaranteed service is request and but an imperfect
                     guarantee is acceptable.

301 (hex) - guaranteed service is request and but an imperfect guarantee is acceptable.

      It is expected that asking for predicted service or permitting an
      imperfect guarantee will substantially increase the chance that a

It is expected that asking for predicted service or permitting an imperfect guarantee will substantially increase the chance that a

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      flow request will be accepted.

flow request will be accepted.

Possible Limitations in the Proposed Flow Spec

Possible Limitations in the Proposed Flow Spec

   There are at least three places where the flow spec is arguably
   imperfect, based on what we currently know about flow reservation.
   In addition, since this is a first attempt at a flow spec, readers
   should expect modifications as we learn more.

There are at least three places where the flow spec is arguably imperfect, based on what we currently know about flow reservation. In addition, since this is a first attempt at a flow spec, readers should expect modifications as we learn more.

   First, the loss model is not perfect.  Simply stating that an
   application is sensitive to loss and to burst loss is a rather crude
   indication of sensitivity.  However, explicitly enumerating loss
   requirements within a cycle is also an imperfect mechanism.  The key
   problem with the explicit values is that not all packets sent over a
   flow will be a full MTU in size.  Expressed another way, the current
   flow spec expects that an MTU-sized packet will be the unit of error
   recovery.  If flows send packets in a range of sizes, then the loss
   bounds may not be very useful.  However, the thought of allowing a
   flow to request a set of loss models (one per packet size) is
   sufficiently painful that I've limited the flow to one loss profile.
   Further study of loss models is clearly needed.

First, the loss model is not perfect. Simply stating that an application is sensitive to loss and to burst loss is a rather crude indication of sensitivity. However, explicitly enumerating loss requirements within a cycle is also an imperfect mechanism. The key problem with the explicit values is that not all packets sent over a flow will be a full MTU in size. Expressed another way, the current flow spec expects that an MTU-sized packet will be the unit of error recovery. If flows send packets in a range of sizes, then the loss bounds may not be very useful. However, the thought of allowing a flow to request a set of loss models (one per packet size) is sufficiently painful that I've limited the flow to one loss profile. Further study of loss models is clearly needed.

   Second, the minimum delay sensitivity field limits a flow to stating
   that there is one point on a performance sensitivity curve below
   which the flow is no longer interested in improved performance.  It
   may be that a single point is insufficient to fully express a flow's
   sensitivity.  For example, consider a flow for supporting part of a
   two-way voice conversation.  Human users will notice improvements in
   delay down to a few 10s of milliseconds.  However, the key point of
   sensitivity is the delay at which normal conversation begins to
   become awkward (about 100 milliseconds).  By allowing only one
   sensitivity point, the flow spec forces the flow designer to either
   ask for the best possible delay (e.g, a few 10's of ms) to try to get
   maximum performance from the network, or state a sensitivity of about
   95 ms, and accept the possibility that the internetwork will not try
   to improve delay below that value, even if it could (and even though
   the user would notice the improvement).  My expectation is that a
   simple point is likely to be easier to deal with than attempting to
   enumerate two (or three or four) points in the sensitivity curve.

Second, the minimum delay sensitivity field limits a flow to stating that there is one point on a performance sensitivity curve below which the flow is no longer interested in improved performance. It may be that a single point is insufficient to fully express a flow's sensitivity. For example, consider a flow for supporting part of a two-way voice conversation. Human users will notice improvements in delay down to a few 10s of milliseconds. However, the key point of sensitivity is the delay at which normal conversation begins to become awkward (about 100 milliseconds). By allowing only one sensitivity point, the flow spec forces the flow designer to either ask for the best possible delay (e.g, a few 10's of ms) to try to get maximum performance from the network, or state a sensitivity of about 95 ms, and accept the possibility that the internetwork will not try to improve delay below that value, even if it could (and even though the user would notice the improvement). My expectation is that a simple point is likely to be easier to deal with than attempting to enumerate two (or three or four) points in the sensitivity curve.

   Third, the models for service guarantees is still evolving and it is
   by no means clear that the service choices provided are the correct
   set.

Third, the models for service guarantees is still evolving and it is by no means clear that the service choices provided are the correct set.

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How an Internetwork is Expected to Handle a Flow Spec

How an Internetwork is Expected to Handle a Flow Spec

   There are at least two parts to the issue of how an internetwork is
   expected to handle a flow spec.  The first part deals with how the
   flow spec is interpreted so that the internetwork can find a route
   which will allow the internetwork to match the flow's requirements.
   The second part deals with how the network replies to the host's
   request.

There are at least two parts to the issue of how an internetwork is expected to handle a flow spec. The first part deals with how the flow spec is interpreted so that the internetwork can find a route which will allow the internetwork to match the flow's requirements. The second part deals with how the network replies to the host's request.

   The precise mechanism for setting up a flow, given a flow spec, is a
   large topic and beyond the scope of this memo.  The purpose of the
   next few paragraphs is simply to sketch an argument that this flow
   spec is sufficient to the requirements of the setup mechanisms known
   to the author.

The precise mechanism for setting up a flow, given a flow spec, is a large topic and beyond the scope of this memo. The purpose of the next few paragraphs is simply to sketch an argument that this flow spec is sufficient to the requirements of the setup mechanisms known to the author.

   The key problem in setting up a flow is determining if there exist
   one or more routes from the source to the destination(s) which might
   be able to support the quality of service requested.  Once one has a
   route (or set of candidate routes) one can take whatever actions may
   be appropriate to confirm that the route is actually viable and to
   cause the flow's data to follow that route.

The key problem in setting up a flow is determining if there exist one or more routes from the source to the destination(s) which might be able to support the quality of service requested. Once one has a route (or set of candidate routes) one can take whatever actions may be appropriate to confirm that the route is actually viable and to cause the flow's data to follow that route.

   There are a number of ways to find a route.  One might try to build a
   route on the fly by establishing the flow hop-by-hop (as ST-II does)
   or one might consult a route server which provides a set of candidate
   source routes derived from a routing database.  However, whatever
   system is used, some basic information about the flow needs to be
   provided to the routing system.  This information is:

There are a number of ways to find a route. One might try to build a route on the fly by establishing the flow hop-by-hop (as ST-II does) or one might consult a route server which provides a set of candidate source routes derived from a routing database. However, whatever system is used, some basic information about the flow needs to be provided to the routing system. This information is:

      * How much bandwidth the flow may require.  There's no point
        in routing a flow that expects to send at over 10 megabits per
        second via a T1 (1.5 megabit per second) link.

* How much bandwidth the flow may require. There's no point in routing a flow that expects to send at over 10 megabits per second via a T1 (1.5 megabit per second) link.

      * How delay sensitive the application is.  One does not wish
        to route a delay-sensitive application over a satellite link,
        unless the satellite link is the only possible route from here
        to there.

* How delay sensitive the application is. One does not wish to route a delay-sensitive application over a satellite link, unless the satellite link is the only possible route from here to there.

      * How much error can be tolerated.  Can we send this flow over
        our microwave channel on a rainy day or is a more reliable link
        required?

* How much error can be tolerated. Can we send this flow over our microwave channel on a rainy day or is a more reliable link required?

      * How firm the guarantees need to be.  Can we put an Ethernet
        in as one of the hops?

* How firm the guarantees need to be. Can we put an Ethernet in as one of the hops?

      * How much delay variation is tolerated.  Again, can an Ethernet
        be included in the path?  Does the routing system need to worry
        if the addition of this flow will cause a few routers to run

* How much delay variation is tolerated. Again, can an Ethernet be included in the path? Does the routing system need to worry if the addition of this flow will cause a few routers to run

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        at close to capacity?  (A side note: we assume that the routers
        are running with priority queueing systems, so running the router
        close to capacity doesn't mean that all flows get long and
        variable delays.  Rather, running close to capacity means that
        high priority flows will be unaffected, and low priority flows
        will get hit with a lot of delay and variation.)

at close to capacity? (A side note: we assume that the routers are running with priority queueing systems, so running the router close to capacity doesn't mean that all flows get long and variable delays. Rather, running close to capacity means that high priority flows will be unaffected, and low priority flows will get hit with a lot of delay and variation.)

   The flow spec provides all of this information.  So it seems
   plausible to assume it provides enough information to make routing
   decisions at setup time.

The flow spec provides all of this information. So it seems plausible to assume it provides enough information to make routing decisions at setup time.

   The flow spec was designed with the expectation that the network
   would give a yes or no reply to a request for a guaranteed flow.

The flow spec was designed with the expectation that the network would give a yes or no reply to a request for a guaranteed flow.

   Some researchers have suggested that the negotiation to set up a flow
   might be an extended negotiation, in which the requesting host
   initially requests the best possible flow it could desire and then
   haggles with the network until they agree on a flow with properties
   that the network can actually provide and the application still finds
   useful.  This notion bothers me for at least two reasons.  First, it
   means setting up a flow is a potentially long process.  Second, the
   general problem of finding all possible routes with a given set of
   properties is a version of the traveling salesman problem, and I
   don't want to embed traveling salesman algorithms into a network's
   routing system.

Some researchers have suggested that the negotiation to set up a flow might be an extended negotiation, in which the requesting host initially requests the best possible flow it could desire and then haggles with the network until they agree on a flow with properties that the network can actually provide and the application still finds useful. This notion bothers me for at least two reasons. First, it means setting up a flow is a potentially long process. Second, the general problem of finding all possible routes with a given set of properties is a version of the traveling salesman problem, and I don't want to embed traveling salesman algorithms into a network's routing system.

   The model used in designing this flow spec was that a system would
   ask for the minimum level of service that was deemed acceptable and
   the network would try to find a route that met that level of service.
   If the network is unable to achieve the desired level of service, it
   refuses the flow, otherwise it accepts the flow.

The model used in designing this flow spec was that a system would ask for the minimum level of service that was deemed acceptable and the network would try to find a route that met that level of service. If the network is unable to achieve the desired level of service, it refuses the flow, otherwise it accepts the flow.

The Flow Spec as a Return Value

The Flow Spec as a Return Value

   This memo does not specify the data structures that the network uses
   to accept or reject a flow.  However, the flow spec has been designed
   so that it can be used to return the type of service being
   guaranteed.

This memo does not specify the data structures that the network uses to accept or reject a flow. However, the flow spec has been designed so that it can be used to return the type of service being guaranteed.

   If the request is being accepted, the minimum delay field could be
   set to the guaranteed or predicted delay, and the quality of
   guarantee field could be set to no guarantee (0), imperfect guarantee
   (100 hex), predicted service (200 hex), or guaranteed service (300
   hex).

If the request is being accepted, the minimum delay field could be set to the guaranteed or predicted delay, and the quality of guarantee field could be set to no guarantee (0), imperfect guarantee (100 hex), predicted service (200 hex), or guaranteed service (300 hex).

   If the request is being rejected, the flow spec could be modified to
   indicate what type of flow the network believes it could accept e.g.,
   the traffic shape or delay characteristics could be adjusted or the

If the request is being rejected, the flow spec could be modified to indicate what type of flow the network believes it could accept e.g., the traffic shape or delay characteristics could be adjusted or the

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   type of guarantee lowered).  Note that this returned flow spec would
   likely be a hint, not a promised offer of service.

type of guarantee lowered). Note that this returned flow spec would likely be a hint, not a promised offer of service.

Why Type of Service is not Good Enough

Why Type of Service is not Good Enough

   The flow spec proposed in this memo takes the form of a set of
   parameters describing the properties and requirements of the flow.
   An alternative approach which is sometimes mentioned (and which is
   currently incorporated into IP) is to use a Type of Service (TOS)
   value.

The flow spec proposed in this memo takes the form of a set of parameters describing the properties and requirements of the flow. An alternative approach which is sometimes mentioned (and which is currently incorporated into IP) is to use a Type of Service (TOS) value.

   The TOS value is an integer (or bit pattern) whose values have been
   predefined to represent requested quality of services.  Thus, a TOS
   of 47 might request service for a flow using up to 1 gigabit per
   second of bandwidth with a minimum delay sensitivity of 100
   milliseconds.

The TOS value is an integer (or bit pattern) whose values have been predefined to represent requested quality of services. Thus, a TOS of 47 might request service for a flow using up to 1 gigabit per second of bandwidth with a minimum delay sensitivity of 100 milliseconds.

   TOS schemes work well if the different quality of services that may
   be requested are both enumerable and reasonably small.
   Unfortunately, these conditions do not appear to apply to future
   internetworks.  The range of possible bandwidth requests alone is
   huge.  Combine this range with several gradations of delay
   requirements, and widely different sensitivities to errors and the
   set of TOS values required becomes extremely large.  (At least one
   person has suggested to the author that perhaps a TOS field combined
   with a bandwidth parameter might be appropriate.  In other words, a
   two parameter model.  That's a tempting idea but my gut feeling is
   that it is not quite sufficient so I'm proposing a more complete
   parametric model.)

TOS schemes work well if the different quality of services that may be requested are both enumerable and reasonably small. Unfortunately, these conditions do not appear to apply to future internetworks. The range of possible bandwidth requests alone is huge. Combine this range with several gradations of delay requirements, and widely different sensitivities to errors and the set of TOS values required becomes extremely large. (At least one person has suggested to the author that perhaps a TOS field combined with a bandwidth parameter might be appropriate. In other words, a two parameter model. That's a tempting idea but my gut feeling is that it is not quite sufficient so I'm proposing a more complete parametric model.)

   Another reason to prefer parametric service is optimization issues.
   A key issue in flow setup is trying to design the the routing system
   to optimize its management of flows.  One can optimize on a number of
   criteria.  A good example of an optimization problem is the following
   question (expressed by Isidro Castineyra of BBN):

Another reason to prefer parametric service is optimization issues. A key issue in flow setup is trying to design the the routing system to optimize its management of flows. One can optimize on a number of criteria. A good example of an optimization problem is the following question (expressed by Isidro Castineyra of BBN):

     "Given a request to establish a flow, how can the internetwork
     accept that request in such a way as to maximize the chance that
     the internetwork will also be able to accept the next flow
     request?"

"Given a request to establish a flow, how can the internetwork accept that request in such a way as to maximize the chance that the internetwork will also be able to accept the next flow request?"

   The optimization goal here is call-completion - maximizing the chance
   that requests to establish flows will succeed.  One might
   alternatively try to maximize revenue (if one is charging for flows).

The optimization goal here is call-completion - maximizing the chance that requests to establish flows will succeed. One might alternatively try to maximize revenue (if one is charging for flows).

   The internetwork is presumably in a better position to do
   optimizations if it has more information about the flow's expected
   behavior.  For example, if a TOS system says only that a flow is

The internetwork is presumably in a better position to do optimizations if it has more information about the flow's expected behavior. For example, if a TOS system says only that a flow is

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   delay sensitive, the routing system must seek out the most direct
   route for the flow.  But if the routing system is told that the flow
   is sensitive only to delays over 100 milliseconds, there may be a
   number of routes other than the most direct route which can satisfy
   this delay, thus leaving the most direct route available for a later
   flow which needs a far lower delay.

delay sensitive, the routing system must seek out the most direct route for the flow. But if the routing system is told that the flow is sensitive only to delays over 100 milliseconds, there may be a number of routes other than the most direct route which can satisfy this delay, thus leaving the most direct route available for a later flow which needs a far lower delay.

   In fairness, it should be noted that a danger of a parametric model
   is that it is very easy to have too many parameters.  The yearn to
   optimize can be overdone.  The goal of this flow spec is to enumerate
   just enough parameters that it appears that essential needs can be
   expressed, and the internetwork has some information it can use to
   try to manage the flows.  Features that would simply be nice or
   useful to have (but not essential) are left out to keep the parameter
   space small.

In fairness, it should be noted that a danger of a parametric model is that it is very easy to have too many parameters. The yearn to optimize can be overdone. The goal of this flow spec is to enumerate just enough parameters that it appears that essential needs can be expressed, and the internetwork has some information it can use to try to manage the flows. Features that would simply be nice or useful to have (but not essential) are left out to keep the parameter space small.

An Implication of the Flow Spec

An Implication of the Flow Spec

   It is important to observe that the there are fields in the flow spec
   that are based on information from the sender (such as rate
   information) and fields in the flow spec that are based on
   information from the receiver (such as delay variation).  There are
   also fields that may sender and receiver to negotiate in advance.
   For example, the acceptable loss rate may depend on whether the
   sender and receiver both support the same type of forward error
   correction.  The delay sensitivity for a voice connection may depend,
   in part, on whether both sender and receiver support echo cancelling.

It is important to observe that the there are fields in the flow spec that are based on information from the sender (such as rate information) and fields in the flow spec that are based on information from the receiver (such as delay variation). There are also fields that may sender and receiver to negotiate in advance. For example, the acceptable loss rate may depend on whether the sender and receiver both support the same type of forward error correction. The delay sensitivity for a voice connection may depend, in part, on whether both sender and receiver support echo cancelling.

   The implication is that the internetwork must permit the sender and
   receiver to communicate in advance of setting up a flow, because a
   flow spec can only be defined once both sender and receiver have had
   their say.  In other words, a reserved flow should not be the only
   form of communication.   There must be some mechanism to perform a
   short exchange of messages in preparation for setting up a flow.

The implication is that the internetwork must permit the sender and receiver to communicate in advance of setting up a flow, because a flow spec can only be defined once both sender and receiver have had their say. In other words, a reserved flow should not be the only form of communication. There must be some mechanism to perform a short exchange of messages in preparation for setting up a flow.

   (Another aside: it has been suggested that perhaps the solution to
   this problem is to have the sender establish a flow with an
   incomplete flow spec, and when the receiver gets the flow spec, have
   the receiver send the completed flow spec back along the flow, so the
   internetwork can "revise" the flow spec according to the receiver's
   desires.  I have two problems with this approach.  First, it is
   entirely possible that the receiver's information may lead the
   internetwork to conclude that the flow established by the sender is
   no good.  For example, the receiver may indicate it has a smaller
   tolerance for delay variation than expected and force the flow to be
   rerouted over a completely different path.  Second, if we try to
   avoid having the receiver's information cause the flow to fail, then
   we have to over-allocate the flow's during the preliminary setup.

(Another aside: it has been suggested that perhaps the solution to this problem is to have the sender establish a flow with an incomplete flow spec, and when the receiver gets the flow spec, have the receiver send the completed flow spec back along the flow, so the internetwork can "revise" the flow spec according to the receiver's desires. I have two problems with this approach. First, it is entirely possible that the receiver's information may lead the internetwork to conclude that the flow established by the sender is no good. For example, the receiver may indicate it has a smaller tolerance for delay variation than expected and force the flow to be rerouted over a completely different path. Second, if we try to avoid having the receiver's information cause the flow to fail, then we have to over-allocate the flow's during the preliminary setup.

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   But over allocating the resources requested may lead us to choose
   better quality paths than we need for this flow.  In other words, our
   attempts to optimize use of the network will fail.)

But over allocating the resources requested may lead us to choose better quality paths than we need for this flow. In other words, our attempts to optimize use of the network will fail.)

Advance Reservations and Flow Duration

Advance Reservations and Flow Duration

   The primary purpose of a flow specification is to provide information
   to the internetwork so the internetwork can properly manage the
   proposed flow's traffic in the context of other traffic in the
   internetwork.  One question is whether the flow should give the
   network information about when the flow is expected to start and how
   long the flow is expected to last.

The primary purpose of a flow specification is to provide information to the internetwork so the internetwork can properly manage the proposed flow's traffic in the context of other traffic in the internetwork. One question is whether the flow should give the network information about when the flow is expected to start and how long the flow is expected to last.

   Announcing when a flow will start is generally of interest for
   advance reservations.  (If the flow is not be reserved substantially
   in advance, the presentation of the flow spec to the internetwork can
   be taken as an implicit request for a flow, now.)  It is my view that
   advance reservation is a distinct problem from the describing the
   properties of a flow.  Advanced reservations will require some
   mechanism to maintain information in the network about flows which
   are not currently active but are expected to be activated at some
   time in the future.  I anticipate this will require some sort of
   distributed database to ensure that information about advanced
   reservations is not accidentally lost if parts of the internetwork
   crash.  In other words, advance reservations will require
   considerable additional supporting baggage that it would probably be
   better to keep out of the average flow spec.

Announcing when a flow will start is generally of interest for advance reservations. (If the flow is not be reserved substantially in advance, the presentation of the flow spec to the internetwork can be taken as an implicit request for a flow, now.) It is my view that advance reservation is a distinct problem from the describing the properties of a flow. Advanced reservations will require some mechanism to maintain information in the network about flows which are not currently active but are expected to be activated at some time in the future. I anticipate this will require some sort of distributed database to ensure that information about advanced reservations is not accidentally lost if parts of the internetwork crash. In other words, advance reservations will require considerable additional supporting baggage that it would probably be better to keep out of the average flow spec.

   Deciding whether a flow spec should contain information about how
   long the flow is expected to run is a harder decision to make.
   Clearly if we anticipate that the internetwork will support advance
   reservations, it will be necessary for elements of the internetwork
   to predict their traffic load, so they can ensure that advance
   reservations are not compromised by new flow requests.  However,
   there is a school of thought that believes that estimating future
   load from current behavior of existing flows is more accurate than
   anything the flows may have declared in their flow specs.  For this
   reason, I've left a duration field out of the flow spec.

Deciding whether a flow spec should contain information about how long the flow is expected to run is a harder decision to make. Clearly if we anticipate that the internetwork will support advance reservations, it will be necessary for elements of the internetwork to predict their traffic load, so they can ensure that advance reservations are not compromised by new flow requests. However, there is a school of thought that believes that estimating future load from current behavior of existing flows is more accurate than anything the flows may have declared in their flow specs. For this reason, I've left a duration field out of the flow spec.

Examples

Examples

   To illustrate how the flow spec values might be used, this section
   presents three example flow specs.

To illustrate how the flow spec values might be used, this section presents three example flow specs.

   Telnet

Telnet

      For the first example, consider using the flow spec to request
      service for an existing application: Telnet.  Telnet is a virtual

For the first example, consider using the flow spec to request service for an existing application: Telnet. Telnet is a virtual

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      terminal protocol, and one can think of it as stringing a virtual
      wire across the network between the user's terminal and a remote
      host.

terminal protocol, and one can think of it as stringing a virtual wire across the network between the user's terminal and a remote host.

      Telnet has proved a very successful application without a need to
      reserve bandwidth: the amount of data sent over any Telnet
      connection tends to be quite small.  However, Telnet users are
      often quite sensitive to delay, because delay can affect the time
      it takes to echo characters.  This suggests that a Telnet
      connection might benefit from asking the internetwork to avoid
      long delay paths.  It could so so using the following flow spec
      (for both directions):

Telnet has proved a very successful application without a need to reserve bandwidth: the amount of data sent over any Telnet connection tends to be quite small. However, Telnet users are often quite sensitive to delay, because delay can affect the time it takes to echo characters. This suggests that a Telnet connection might benefit from asking the internetwork to avoid long delay paths. It could so so using the following flow spec (for both directions):

      Version=1
      MTU=80 [40 bytes of overhead + 40 bytes user data]
      Token Bucket Rate=0/0/0 [don't want a guarantee]
      Token Bucket Size=0/0/0
      Maximum Transmission Rate=0/0/0
      Maximum Delay Noticed=1/1 [constant = delay sensitive]
      Maximum Delay Variation=0/0/0 [not a concern]
      Loss Sensitivity=1/0 [don't worry about loss]
      Burst Loss Sensitivity=1/0
      Loss Interval=1/0
      Quality of Guarantee=1/0 [just asking]

Version=1 MTU=80 [40 bytes of overhead + 40 bytes user data] Token Bucket Rate=0/0/0 [don't want a guarantee] Token Bucket Size=0/0/0 Maximum Transmission Rate=0/0/0 Maximum Delay Noticed=1/1 [constant = delay sensitive] Maximum Delay Variation=0/0/0 [not a concern] Loss Sensitivity=1/0 [don't worry about loss] Burst Loss Sensitivity=1/0 Loss Interval=1/0 Quality of Guarantee=1/0 [just asking]

      It is worth noting that Telnet's flow spec is likely to be the
      same for all instantiations of a Telnet connection.  As a result,
      there may be some optimizations possible (such as just tagging
      Telnet packets as being subject to the well-known Telnet flow
      spec).

It is worth noting that Telnet's flow spec is likely to be the same for all instantiations of a Telnet connection. As a result, there may be some optimizations possible (such as just tagging Telnet packets as being subject to the well-known Telnet flow spec).

   A Voice Flow

A Voice Flow

      Now consider transmitting voice over the Internet.  Currently,
      good quality voice can be delivered at rates of 32Kbit/s or
      16Kbit/s.  Assuming the rate is 32Kbit/s and voice samples are 16
      bit samples packaged into UDP datagrams (for a data rate of about
      60 Kbyte/s), a flow spec might be:

Now consider transmitting voice over the Internet. Currently, good quality voice can be delivered at rates of 32Kbit/s or 16Kbit/s. Assuming the rate is 32Kbit/s and voice samples are 16 bit samples packaged into UDP datagrams (for a data rate of about 60 Kbyte/s), a flow spec might be:

      Version=1
      MTU=30 [2 byte sample in UDP datagram]
      Token Bucket Rate=0/10/59 [60.4 Kbytes/s]
      Token Bucket Size=0/0/30 [save enough to send immediately
                                after pauses]
      Maximum Transmission Rate=0/10/59 [peak same as mean]
      Maximum Delay Noticed=0/10/100 [100 ms]
      Maximum Delay Variation=0/10/10 [keep variation low]
      Loss Sensitivity=1/1 [loss sensitive]

Version=1 MTU=30 [2 byte sample in UDP datagram] Token Bucket Rate=0/10/59 [60.4 Kbytes/s] Token Bucket Size=0/0/30 [save enough to send immediately after pauses] Maximum Transmission Rate=0/10/59 [peak same as mean] Maximum Delay Noticed=0/10/100 [100 ms] Maximum Delay Variation=0/10/10 [keep variation low] Loss Sensitivity=1/1 [loss sensitive]

Partridge                                                      [Page 18]

RFC 1363             A Proposed Flow Specification        September 1992

Partridge [Page 18] RFC 1363 A Proposed Flow Specification September 1992

      Burst Loss Sensitivity=0/0/5 [keep bursts small]
      Loss Interval=1/0
      Quality of Guarantee=1/201 [predicted service and I'll accept
                                  worse]

Burst Loss Sensitivity=0/0/5 [keep bursts small] Loss Interval=1/0 Quality of Guarantee=1/201 [predicted service and I'll accept worse]

   A Variable Bit-Rate Video Flow

A Variable Bit-Rate Video Flow

      Variable bit-rate video transmissions vary the rate at which they
      send data according to the amount of the video image that has
      changed between frames.  In this example, we consider a one-way
      broadcast of a picture.  If we assume 30 frames a second and that
      a full frame is about 1 megabit of data, and that on average about
      10% of the frame changes, but in the worst case the entire frame
      changes, the flow spec might be:

Variable bit-rate video transmissions vary the rate at which they send data according to the amount of the video image that has changed between frames. In this example, we consider a one-way broadcast of a picture. If we assume 30 frames a second and that a full frame is about 1 megabit of data, and that on average about 10% of the frame changes, but in the worst case the entire frame changes, the flow spec might be:

      Version=1
      MTU=4096 [big so we can put lots of bits in each packet]
      Token Bucket Rate=0/20/1 [8 Mbits/s]
      Token Bucket Size=0/17/2 [2 Mbits/s]
      Maximum Transmission Rate=0/20/30 [30 Mbits/s]
      Maximum Delay Noticed=1/1 [somewhat delay sensitive]
      Maximum Delay Variation=0/10/1 [no more than one second of
                                      buffering]
      Loss Sensitivity=0/0/1 [worst case, one loss per frame]
      Burst Loss Sensitivity=0/0/1 [no burst errors please]
      Loss Interval=0/0/33 [one frame in MTU sized packets]
      Quality of Guarantee=1/300 [guaranteed service only]

Version=1 MTU=4096 [big so we can put lots of bits in each packet] Token Bucket Rate=0/20/1 [8 Mbits/s] Token Bucket Size=0/17/2 [2 Mbits/s] Maximum Transmission Rate=0/20/30 [30 Mbits/s] Maximum Delay Noticed=1/1 [somewhat delay sensitive] Maximum Delay Variation=0/10/1 [no more than one second of buffering] Loss Sensitivity=0/0/1 [worst case, one loss per frame] Burst Loss Sensitivity=0/0/1 [no burst errors please] Loss Interval=0/0/33 [one frame in MTU sized packets] Quality of Guarantee=1/300 [guaranteed service only]

      The token bucket is sized to be two frames of data, and the bucket
      rate will fill the bucket every 250 ms.  The expectation is that
      full scene changes will be rare and that a fast rate with a large
      bucket size should accommodate even a series of scene changes.

The token bucket is sized to be two frames of data, and the bucket rate will fill the bucket every 250 ms. The expectation is that full scene changes will be rare and that a fast rate with a large bucket size should accommodate even a series of scene changes.

   Disclaimer

Disclaimer

      In all cases, these examples are simply to sketch the use of the
      flow spec.  The author makes no claims that the actual values used
      are the correct ones for a particular application.

In all cases, these examples are simply to sketch the use of the flow spec. The author makes no claims that the actual values used are the correct ones for a particular application.

Security Considerations

Security Considerations

   Security considerations definitely exist.  For example, one might
   assume that users are charged for guaranteed flows.  In that case,
   some mechanism must exist to ensure that a flow request (including
   flow spec) is authenticated.  However I believe that such issues have
   to be dealt with as part of designing a negotiation protocol, and are
   not part of designing the flow spec data structure.

Security considerations definitely exist. For example, one might assume that users are charged for guaranteed flows. In that case, some mechanism must exist to ensure that a flow request (including flow spec) is authenticated. However I believe that such issues have to be dealt with as part of designing a negotiation protocol, and are not part of designing the flow spec data structure.

Partridge                                                      [Page 19]

RFC 1363             A Proposed Flow Specification        September 1992

Partridge [Page 19] RFC 1363 A Proposed Flow Specification September 1992

Acknowledgements

Acknowledgements

   I'd like to acknowledge the tremendous assistance of Steve Deering,
   Scott Shenker and Lixia Zhang of XEROX PARC in writing this RFC.
   Much of this flow spec was sketched out in two long meetings with
   them at PARC.  Others who have offered notable advice and comments
   include Isidro Castineyra, Deborah Estrin, and members of the End-
   to-End Research Group chaired by Bob Braden.  All ideas that prove
   misbegotten are the sole responsibility of the author.  This work was
   funded under DARPA Contract No. MDA903-91-D-0019.  The views
   expressed in this document are not necessarily those of the Defense
   Advanced Research Projects Agency.

I'd like to acknowledge the tremendous assistance of Steve Deering, Scott Shenker and Lixia Zhang of XEROX PARC in writing this RFC. Much of this flow spec was sketched out in two long meetings with them at PARC. Others who have offered notable advice and comments include Isidro Castineyra, Deborah Estrin, and members of the End- to-End Research Group chaired by Bob Braden. All ideas that prove misbegotten are the sole responsibility of the author. This work was funded under DARPA Contract No. MDA903-91-D-0019. The views expressed in this document are not necessarily those of the Defense Advanced Research Projects Agency.

References

References

   1. Parekh, A., "A Generalized Processor Sharing Approach
      to Flow Control in Integrated Services Networks",
      MIT Laboratory for Information and Decision Systems,
      Report No. LIDS-TH-2089.

1. Parekh、A.、「統合サービスネットワークにおけるフロー制御への一般化されたプロセッサ共有アプローチ」、情報のためのMIT研究所、および決定システムはいいえと報告します。 ふた、2089番目

   2. Clark, D., Shenker, S., and L. Zhang, "Supporting Real-Time
      Applications in an Integrated Services Packet Network:
      Architecture and Mechanism", Proceedings of ACM SIGCOMM '92,
      August 1992.

2. クラーク、D.、Shenker、S.、およびL.チャン、「統合サービスパケット網におけるリアルタイムのアプリケーションをサポートします:」 「アーキテクチャとメカニズム」、ACM SIGCOMM92年、8月1992の議事

Author's Address

作者のアドレス

   Craig Partridge
   BBN
   824 Kipling St
   Palo Alto, CA  94301

クレイグヤマウズラBBN824キップリング・Stパロアルト、カリフォルニア 94301

   Phone: 415-325-4541

以下に電話をしてください。 415-325-4541

   EMail: craig@aland.bbn.com

メール: craig@aland.bbn.com

Partridge                                                      [Page 20]

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