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Internet Engineering Task Force Integrated Services WG INTERNET-DRAFT S. Shenker/C. Partridge/R. Guerin draft-ietf-intserv-guaranteed-svc-06.txt Xerox/BBN/IBM 13 August 1996 Expires: 2/13/97 Specification of Guaranteed Quality of Service Status of this Memo This document is an Internet-Draft. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet-Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet- Drafts as reference material or to cite them other than as ``work in progress.'' To learn the current status of any Internet-Draft, please check the ``1id-abstracts.txt'' listing contained in the Internet- Drafts Shadow Directories on ftp.is.co.za (Africa), nic.nordu.net (Europe), munnari.oz.au (Pacific Rim), ds.internic.net (US East Coast), or ftp.isi.edu (US West Coast). This document is a product of the Integrated Services working group of the Internet Engineering Task Force. Comments are solicited and should be addressed to the working group's mailing list at int- serv@isi.edu and/or the author(s). This draft reflects minor changes from the IETF meeting in Los Angeles and comments received after circulating draft 5. Abstract This memo describes the network element behavior required to deliver a guaranteed service (guaranteed delay and bandwidth) in the Internet. Guaranteed service provides firm (mathematically provable) bounds on end-to-end datagram queueing delays. This service makes it possible to provide a service that guarantees both delay and bandwidth. This specification follows the service specification template described in [1]. Shenker/Partridge/Guerin Expires 2/13/97 [Page 1] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 Introduction This document defines the requirements for network elements that support guaranteed service. This memo is one of a series of documents that specify the network element behavior required to support various qualities of service in IP internetworks. Services described in these documents are useful both in the global Internet and private IP networks. This document is based on the service specification template given in [1]. Please refer to that document for definitions and additional information about the specification of qualities of service within the IP protocol family. End-to-End Behavior The end-to-end behavior provided by a series of network elements that conform to this document is an assured level of bandwidth that, when used by a policed flow, produces a delay-bounded service with no queueing loss for all conforming datagrams (assuming no failure of network components or changes in routing during the life of the flow). The end-to-end behavior conforms to the fluid model (described under Network Element Data Handling below) in that the delivered queueing delays do not exceed the fluid delays by more than the specified error bounds. More precisely, the end-to-end delay bound is [(b- M)/R*(p-R)/(p-r)]+(M+Ctot)/R+Dtot for p>R>=r, and (M+Ctot)/R+Dtot for r<=p<=R, (where b, r, p, M, R, Ctot, and Dtot are defined later in this document). NOTE: While the per-hop error terms needed to compute the end-to- end delays are exported by the service module (see Exported Information below), the mechanisms needed to collect per-hop bounds and make the end-to-end quantities Ctot and Dtot known to the applications are not described in this specification. These functions are provided by reservation setup protocols, routing protocols or other network management functions and are outside the scope of this document. The maximum end-to-end queueing delay (as characterized by Ctot and Dtot) and bandwidth (characterized by R) provided along a path will be stable. That is, they will not change as long as the end-to-end path does not change. Guaranteed service does not control the minimal or average delay of datagrams, merely the maximal queueing delay. Furthermore, to Shenker/Partridge/Guerin Expires 2/13/97 [Page 2] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 compute the maximum delay a datagram will experience, the latency of the path MUST be determined and added to the guaranteed queueing delay. (However, as noted below, a conservative bound of the latency can be computed by observing the delay experienced by any one packet). This service is subject to admission control. Motivation Guaranteed service guarantees that datagrams will arrive within the guaranteed delivery time and will not be discarded due to queue overflows, provided the flow's traffic stays within its specified traffic parameters. This service is intended for applications which need a firm guarantee that a datagram will arrive no later than a certain time after it was transmitted by its source. For example, some audio and video "play-back" applications are intolerant of any datagram arriving after their play-back time. Applications that have hard real-time requirements will also require guaranteed service. This service does not attempt to minimize the jitter (the difference between the minimal and maximal datagram delays); it merely controls the maximal queueing delay. Because the guaranteed delay bound is a firm one, the delay has to be set large enough to cover extremely rare cases of long queueing delays. Several studies have shown that the actual delay for the vast majority of datagrams can be far lower than the guaranteed delay. Therefore, authors of playback applications should note that datagrams will often arrive far earlier than the delivery deadline and will have to be buffered at the receiving system until it is time for the application to process them. This service represents one extreme end of delay control for networks. Most other services providing delay control provide much weaker assurances about the resulting delays. In order to provide this high level of assurance, guaranteed service is typically only useful if provided by every network element along the path (i.e. by both routers and the links that interconnect the routers). Moreover, as described in the Exported Information section, effective provision and use of the service requires that the set-up protocol or other mechanism used to request service provides service characterizations to intermediate routers and to the endpoints. Network Element Data Handling Requirements The network element MUST ensure that the service approximates the "fluid model" of service. The fluid model at service rate R is Shenker/Partridge/Guerin Expires 2/13/97 [Page 3] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 essentially the service that would be provided by a dedicated wire of bandwidth R between the source and receiver. Thus, in the fluid model of service at a fixed rate R, the flow's service is completely independent of that of any other flow. The flow's level of service is characterized at each network element by a bandwidth (or service rate) R and a buffer size B. R represents the share of the link's bandwidth the flow is entitled to and B represents the buffer space in the network element that the flow may consume. The network element MUST ensure that its service matches the fluid model at that same rate to within a sharp error bound. The definition of guaranteed service relies on the result that the fluid delay of a flow obeying a token bucket (r,b) and being served by a line with bandwidth R is bounded by b/R as long as R is no less than r. Guaranteed service with a service rate R, where now R is a share of bandwidth rather than the bandwidth of a dedicated line, approximates this behavior. Consequently, the network element MUST ensure that the queueing delay of any datagram be less than b/R+C/R+D, where C and D describe the maximal local deviation away from the fluid model. It is important to emphasize that C and D are maximums. So, for instance, if an implementation has occasional gaps in service (perhaps due to processing routing updates), D needs to be large enough to account for the time a datagram may lose during the gap in service. (C and D are described in more detail in the section on Exported Information). NOTE: Strictly speaking, this memo requires only that the service a flow receives is never worse than it would receive under this approximation of the fluid model. It is perfectly acceptable to give better service. For instance, if a flow is currently not using its share, R, algorithms such as Weighted Fair Queueing that temporarily give other flows the unused bandwidth, are perfectly acceptable (indeed, are encouraged). Links are not permitted to fragment datagrams as part of guaranteed service. Datagrams larger than the MTU of the link MUST be policed as nonconformant which means that they will be policed according to the rules described in the Policing section below. Invocation Information Guaranteed service is invoked by specifying the traffic (TSpec) and the desired service (RSpec) to the network element. A service request for an existing flow that has a new TSpec and/or RSpec SHOULD be treated as a new invocation, in the sense that admission control SHOULD be reapplied to the flow. Flows that reduce their TSpec Shenker/Partridge/Guerin Expires 2/13/97 [Page 4] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 and/or their RSpec (i.e., their new TSpec/RSpec is strictly smaller than the old TSpec/RSpec according to the ordering rules described in the section on Ordering below) SHOULD never be denied service. The TSpec takes the form of a token bucket plus a peak rate (p), a minimum policed unit (m), and a maximum datagram size (M). The token bucket has a bucket depth, b, and a bucket rate, r. Both b and r MUST be positive. The rate, r, is measured in bytes of IP datagrams per second, and can range from 1 byte per second to as large as 40 terabytes per second (or close to what is believed to be the maximum theoretical bandwidth of a single strand of fiber). Clearly, particularly for large bandwidths, only the first few digits are significant and so the use of floating point representations, accurate to at least 0.1% is encouraged. The bucket depth, b, is also measured in bytes and can range from 1 byte to 250 gigabytes. Again, floating point representations accurate to at least 0.1% are encouraged. The range of values is intentionally large to allow for the future bandwidths. The range is not intended to imply that a network element has to support the entire range. The peak rate, p, is measured in bytes of IP datagrams per second and has the same range and suggested representation as the bucket rate. The peak rate is the maximum rate at which the source and any reshaping points (reshaping points are defined below) may inject bursts of traffic into the network. More precisely, it is a requirement that for all time periods the amount of data sent cannot exceed M+pT where M is the maximum datagram size and T is the length of the time period. Furthermore, p MUST be greater than or equal to the token bucket rate, r. If the peak rate is unknown or unspecified, then p MUST be set to infinity. The minimum policed unit, m, is an integer measured in bytes. All IP datagrams less than size m will be counted, when policed and tested for conformance to the TSpec, as being of size m. The maximum datagram size, M, is the biggest datagram that will conform to the traffic specification; it is also measured in bytes. The flow MUST be rejected if the requested maximum datagram size is larger than the MTU of the link. Both m and M MUST be positive, and m MUST be less than or equal to M. The guaranteed service uses the general TOKEN_BUCKET_TSPEC parameter defined in Reference [8] to describe a data flow's traffic characteristics. The description above is of that parameter. The TOKEN_BUCKET_TSPEC is general parameter number Shenker/Partridge/Guerin Expires 2/13/97 [Page 5] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 127. Use of this parameter for the guaranteed service TSpec simplifies the use of guaranteed Service in a multi-service environment. The RSpec is a rate R and a slack term S, where R MUST be greater than or equal to r and S MUST be nonnegative. The RSpec rate can be bigger than the TSpec rate because higher rates will reduce queueing delay. The slack term signifies the difference between the desired delay and the delay obtained by using a reservation level R. This slack term can be utilized by the network element to reduce its resource reservation for this flow. When a network element chooses to utilize some of the slack in the RSpec, it MUST follow specific rules in updating the R and S fields of the RSpec; these rules are specified in the Ordering and Merging section. If at the time of service invocation no slack is specified, the slack term, S, is set to zero. No buffer specification is included in the RSpec because the network element is expected to derive the required buffer space to ensure no queueing loss from the token bucket and peak rate in the TSpec, the reserved rate and slack in the RSpec, the exported information received at the network element, i.e., Ctot and Dtot or Csum and Dsum, combined with internal information about how the element manages its traffic. The TSpec can be represented by three floating point numbers in single-precision IEEE floating point format followed by two 32-bit integers in network byte order. The first floating point value is the rate (r), the second floating point value is the bucket size (b), the third floating point is the peak rate (p), the first integer is the minimum policed unit (m), and the second integer is the maximum datagram size (M). The RSpec rate term, R, can also be represented using single- precision IEEE floating point. The Slack term, S, can be represented as a 32-bit integer. When r, b, p, and R terms are represented as IEEE floating point values, the sign bit MUST be zero (all values MUST be non-negative). Exponents less than 127 (i.e., 0) are prohibited. Exponents greater than 162 (i.e., positive 35) are discouraged, except for specifying a peak rate of infinity. Infinity is represented with an exponent of all ones (255) and a sign bit and mantissa of all zeroes. Exported Information Each guaranteed service module MUST export at least the following information. All of the parameters described below are characterization parameters. Shenker/Partridge/Guerin Expires 2/13/97 [Page 6] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 A network element's implementation of guaranteed service is characterized by two error terms, C and D, which represent how the element's implementation of the guaranteed service deviates from the fluid model. These two parameters have an additive composition rule. The error term C is the rate-dependent error term. It represents the delay a datagram in the flow might experience due to the rate parameters of the flow. An example of such an error term is the need to account for the time taken serializing a datagram broken up into ATM cells, with the cells sent at a frequency of 1/r. NOTE: It is important to observe that when computing the delay bound, parameter C is divided by the reservation rate R. This division is done because, as with the example of serializing the datagram, the effect of the C term is a function of the transmission rate. Implementors should take care to confirm that their C values, when divided by various rates, give appropriate results. Delay values that are not dependent on the rate SHOULD be incorporated into the value for the D parameter. The error term D is the rate-independent, per-element error term and represents the worst case non-rate-based transit time variation through the service element. It is generally determined or set at boot or configuration time. An example of D is a slotted network, in which guaranteed flows are assigned particular slots in a cycle of slots. Some part of the per-flow delay may be determined by which slots in the cycle are allocated to the flow. In this case, D would measure the maximum amount of time a flow's data, once ready to be sent, might have to wait for a slot. (Observe that this value can be computed before slots are assigned and thus can be advertised. For instance, imagine there are 100 slots. In the worst case, a flow might get all of its N slots clustered together, such that if a packet was made ready to send just after the cluster ended, the packet might have to wait 100-N slot times before transmitting. In this case one can easily approximate this delay by setting D to 100 slot times). If the composition function is applied along the entire path to compute the end-to-end sums of C and D (Ctot and Dtot) and the resulting values are then provided to the end nodes (by presumably the setup protocol), the end nodes can compute the maximal datagram queueing delays. Moreover, if the partial sums (Csum and Dsum) from the most recent reshaping point (reshaping points are defined below) downstream towards receivers are handed to each network element then these network elements can compute the buffer allocations necessary to achieve no datagram loss, as detailed in the section Guidelines for Implementors. The proper use and provision of this service requires that the quantities Ctot and Dtot, and the quantities Csum Shenker/Partridge/Guerin Expires 2/13/97 [Page 7] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 and Dsum be computed. Therefore, we assume that usage of guaranteed service will be primarily in contexts where these quantities are made available to end nodes and network elements. The error term C is measured in units of bytes. An individual element can advertise a C value between 1 and 2**28 (a little over 250 megabytes) and the total added over all elements can range as high as (2**32)-1. Should the sum of the different elements delay exceed (2**32)-1, the end-to-end error term MUST be set to (2**32)-1. The error term D is measured in units of one microsecond. An individual element can advertise a delay value between 1 and 2**28 (somewhat over two minutes) and the total delay added over all elements can range as high as (2**32)-1. Should the sum of the different elements delay exceed (2**32)-1, the end-to-end delay MUST be set to (2**32)-1. The guaranteed service is service_name 2. The RSpec parameter is numbered 130. Error characterization parameters C and D are numbered 131 and 132. The end-to-end composed values for C and D (Ctot and Dtot) are numbered 133 and 134. The since-last-reshaping point composed values for C and D (Csum and Dsum) are numbered 135 and 136. Policing There are two forms of policing in guaranteed service. One form is simple policing (hereafter just called policing to be consistent with other documents), in which arriving traffic is compared against a TSpec. The other form is reshaping, where an attempt is made to restore (possibly distorted) traffic's shape to conform to the TSpec, and the fact that traffic is in violation of the TSpec is discovered because the reshaping fails (the reshaping buffer overflows). Policing is done at the edge of the network. Reshaping is done at all heterogeneous source branch points and at all source merge points. A heterogeneous source branch point is a spot where the multicast distribution tree from a source branches to multiple distinct paths, and the TSpec's of the reservations on the various outgoing links are not all the same. Reshaping need only be done if the TSpec on the outgoing link is "less than" (in the sense described in the Ordering section) the TSpec reserved on the immediately upstream link. A source merge point is where the distribution paths or trees from two different sources (sharing the same reservation) merge. It is the responsibility of the invoker of the service (a Shenker/Partridge/Guerin Expires 2/13/97 [Page 8] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 setup protocol, local configuration tool, or similar mechanism) to identify points where policing is required. Reshaping may be done at other points as well as those described above. Policing MUST not be done except at the edge of the network. The token bucket and peak rate parameters require that traffic MUST obey the rule that over all time periods, the amount of data sent cannot exceed M+min[pT, rT+b-M], where r and b are the token bucket parameters, M is the maximum datagram size, and T is the length of the time period (note that when p is infinite this reduces to the standard token bucket requirement). For the purposes of this accounting, links MUST count datagrams which are smaller than the minimum policing unit to be of size m. Datagrams which arrive at an element and cause a violation of the the M+min[pT, rT+b-M] bound are considered non-conformant. At the edge of the network, traffic is policed to ensure it conforms to the token bucket. Non-conforming datagrams SHOULD be treated as best-effort datagrams. [If and when a marking ability becomes available, these non-conformant datagrams SHOULD be ''marked'' as being non-compliant and then treated as best effort datagrams at all subsequent routers.] Best effort service is defined as the default service a network element would give to a datagram that is not part of a flow and was sent between the flow's source and destination. Among other implications, this definition means that if a flow's datagram is changed to a best effort datagram, all flow control (e.g., RED [2]) that is normally applied to best effort datagrams is applied to that datagram too. NOTE: There may be situations outside the scope of this document, such as when a service module's implementation of guaranteed service is being used to implement traffic sharing rather than a quality of service, where the desired action is to discard non- conforming datagrams. To allow for such uses, implementors SHOULD ensure that the action to be taken for non-conforming datagrams is configurable. Inside the network, policing does not produce the desired results, because queueing effects will occasionally cause a flow's traffic that entered the network as conformant to be no longer conformant at some downstream network element. Therefore, inside the network, network elements that wish to police traffic MUST do so by reshaping traffic to the token bucket. Reshaping entails delaying datagrams until they are within conformance of the TSpec. Reshaping is done by combining a buffer with a token bucket and peak Shenker/Partridge/Guerin Expires 2/13/97 [Page 9] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 rate regulator and buffering data until it can be sent in conformance with the token bucket and peak rate parameters. (The token bucket regulator MUST start with its token bucket full of tokens). Under guaranteed service, the amount of buffering required to reshape any conforming traffic back to its original token bucket shape is b+Csum+(Dsum*r), where Csum and Dsum are the sums of the parameters C and D between the last reshaping point and the current reshaping point. Note that the knowledge of the peak rate at the reshapers can be used to reduce these buffer requirements (see the section on "Guidelines for Implementors" below). A network element MUST provide the necessary buffers to ensure that conforming traffic is not lost at the reshaper. If a datagram arrives to discover the reshaping buffer is full, then the datagram is non-conforming. Observe this means that a reshaper is effectively policing too. As with a policer, the reshaper SHOULD relegate non-conforming datagrams to best effort. [If marking is available, the non-conforming datagrams SHOULD be marked] NOTE: As with policers, it SHOULD be possible to configure how reshapers handle non-conforming datagrams. Note that while the large buffer makes it appear that reshapers add considerable delay, this is not the case. Given a valid TSpec that accurately describes the traffic, reshaping will cause little extra actual delay at the reshaping point (and will not affect the delay bound at all). Furthermore, in the normal case, reshaping will not cause the loss of any data. However, (typically at merge or branch points), it may happen that the TSpec is smaller than the actual traffic. If this happens, reshaping will cause a large queue to develop at the reshaping point, which both causes substantial additional delays and forces some datagrams to be treated as non-conforming. This scenario makes an unpleasant denial of service attack possible, in which a receiver who is successfully receiving a flow's traffic via best effort service is pre-empted by a new receiver who requests a reservation for the flow, but with an inadequate TSpec and RSpec. The flow's traffic will now be policed and possibly reshaped. If the policing function was chosen to discard datagrams, the best-effort receiver would stop receiving traffic. For this reason, in the normal case, policers are simply to treat non-conforming datagrams as best effort (and marking them if marking is implemented). While this protects against denial of service, it is still true that the bad TSpec may cause queueing delays to increase. NOTE: To minimize problems of reordering datagrams, reshaping Shenker/Partridge/Guerin Expires 2/13/97 [Page 10] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 points may wish to forward a best-effort datagram from the front of the reshaping queue when a new datagram arrives and the reshaping buffer is full. Readers should also observe that reclassifying datagrams as best effort (as opposed to dropping the datagrams) also makes support for elastic flows easier. They can reserve a modest token bucket and when their traffic exceeds the token bucket, the excess traffic will be sent best effort. A related issue is that at all network elements, datagrams bigger than the MTU of the network element MUST be considered non-conformant and SHOULD be classified as best effort (and will then either be fragmented or dropped according to the element's handling of best effort traffic). [Again, if marking is available, these reclassified datagrams SHOULD be marked.] Ordering and Merging TSpec's are ordered according to the following rules. TSpec A is a substitute ("as good or better than") for TSpec B if (1) both the token rate r and bucket depth b for TSpec A are greater than or equal to those of TSpec B; (2) the peak rate p is at least as large in TSpec A as it is in TSpec B; (3) the minimum policed unit m is at least as small for TSpec A as it is for TSpec B; and (4) the maximum datagram size M is at least as large for TSpec A as it is for TSpec B. TSpec A is "less than or equal" to TSpec B if (1) both the token rate r and bucket depth b for TSpec A are less than or equal to those of TSpec B; (2) the peak rate p in TSpec A is at least as small as the peak rate in TSpec B; (3) the minimum policed unit m is at least as large for TSpec A as it is for TSpec B; and (4) the maximum datagram size M is at least as small for TSpec A as it is for TSpec B. A merged TSpec may be calculated over a set of TSpecs by taking (1) the largest token bucket rate, (2) the largest bucket size, (3) the largest peak rate, (4) the smallest minimum policed unit, and (5) the largest maximum datagram size across all members of the set. This use of the word "merging" is similar to that in the RSVP protocol [10]; a merged TSpec is one which is adequate to describe the traffic from any one of constituent TSpecs. A summed TSpec may be calculated over a set of TSpecs by computing (1) the sum of the token bucket rates, (2) the sum of the bucket sizes, (3) the sum of the peak rates, (4) the smallest minimum policed unit, and (5) the maximum packet size parameter. Shenker/Partridge/Guerin Expires 2/13/97 [Page 11] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 The minimum of two TSpecs differs according to whether the TSpecs can be ordered. If one TSpec is less than the other TSpec, the smaller TSpec is the minimum. Otherwise, the minimum TSpec of two TSpecs is determined by comparing the respective values in the two TSpecs and choosing (1) the smaller token bucket rate, (2) the larger token bucket size (3) the smaller peak rate, (4) the smaller minimum policed unit, and (5) the smaller maximum packet size. The RSpec's are merged in a similar manner as the TSpecs, i.e. a set of RSpecs is merged onto a single RSpec by taking the largest rate R, and the smallest slack S. More precisely, RSpec A is a substitute for RSpec B if the value of reserved service rate, R, in RSpec A is greater than or equal to the value in RSpec B, and the value of the slack, S, in RSpec A is smaller than or equal to that in RSpec B. Each network element receives a service request of the form (TSpec, RSpec), where the RSpec is of the form (Rin, Sin). The network element processes this request and performs one of two actions: a. it accepts the request and returns a new Rspec of the form (Rout, Sout); b. it rejects the request. The processing rules for generating the new RSpec are governed by the delay constraint: Sout + b/Rout + Ctoti/Rout <= Sin + b/Rin + Ctoti/Rin, where Ctoti is the cumulative sum of the error terms, C, for all the network elements that are upstream of the current element, i. In other words, this element consumes (Sin - Sout) of slack and can use it to reduce its reservation level, provided that the above inequality is satisfied. Rin and Rout MUST also satisfy the constraint: r <= Rout <= Rin. When several RSpec's, each with rate Rj, j=1,2..., are to be merged at a split point, the value of Rout is the maximum over all the rates Rj, and the value of Sout is the minimum over all the slack terms Sj. NOTE: The various TSpec functions described above are used by applications which desire to combine TSpecs. It is important to observe, however, that the properties of the actual reservation are determined by combining the TSpec with the RSpec rate (R). Because the guaranteed reservation requires both the TSpec and the RSpec rate, there exist some difficult problems for shared Shenker/Partridge/Guerin Expires 2/13/97 [Page 12] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 reservations in RSVP, particularly where two or more source streams meet. Upstream of the meeting point, it would be desirable to reduce the TSpec and RSpec to use only as much bandwidth and buffering as is required by the individual source's traffic. (Indeed, it may be necessary if the sender is transmitting over a low bandwidth link). However, the RSpec's rate is set to achieve a particular delay bound (and is notjust a function of the TSpec), so changing the RSpec may cause the reservation to fail to meet the receiver's delay requirements. At the same time, not adjusting the RSpec rate means that "shared" RSVP reservations using guaranteed service will fail whenever the bandwidth available at a particular link is less than the receiver's requested rate R, even if the bandwidth is adequate to support the number of senders actually using the link. At this time, this limitation is an open problem in using the guaranteed service with RSVP. Guidelines for Implementors This section discusses a number of important implementation issues in no particular order. It is important to note that individual subnetworks are network elements and both routers and subnetworks MUST support the guaranteed service model to achieve guaranteed service. Since subnetworks typically are not capable of negotiating service using IP-based protocols, as part of providing guaranteed service, routers will have to act as proxies for the subnetworks they are attached to. In some cases, this proxy service will be easy. For instance, on leased line managed by a WFQ scheduler on the upstream node, the proxy need simply ensure that the sum of all the flows' RSpec rates does not exceed the bandwidth of the line, and needs to advertise the rate-based and non-rate-based delays of the link as the values of C and D. In other cases, this proxy service will be complex. In an ATM network, for example, it may require establishing an ATM VC for the flow and computing the C and D terms for that VC. Readers may observe that the token bucket and peak rate used by guaranteed service map directly to the Sustained Cell Rate, Burst Size, and Peak Cell Rate of ATM's Q.2931 QoS parameters for Variable Bit Rate traffic. The assurance that datagrams will not be lost is obtained by setting the router buffer space B to be equal to the token bucket b plus some Shenker/Partridge/Guerin Expires 2/13/97 [Page 13] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 error term (described below). Another issue related to subnetworks is that the TSpec's token bucket rates measure IP traffic and do not (and cannot) account for link level headers. So the subnetwork network elements MUST adjust the rate and possibly the bucket size to account for adding link level headers. Tunnels MUST also account for the additional IP headers that they add. For datagram networks, a maximum header rate can usually be computed by dividing the rate and bucket sizes by the minimum policed unit. For networks that do internal fragmentation, such as ATM, the computation may be more complex, since one MUST account for both per-fragment overhead and any wastage (padding bytes transmitted) due to mismatches between datagram sizes and fragment sizes. For instance, a conservative estimate of the additional data rate imposed by ATM AAL5 plus ATM segmentation and reassembly is ((r/48)*5)+((r/m)*(8+52)) which represents the rate divided into 48-byte cells multiplied by the 5-byte ATM header, plus the maximum datagram rate (r/m) multiplied by the cost of the 8-byte AAL5 header plus the maximum space that can be wasted by ATM segmentation of a datagram (which is the 52 bytes wasted in a cell that contains one byte). But this estimate is likely to be wildly high, especially if m is small, since ATM wastage is usually much less than 52 bytes. (ATM implementors should be warned that the token bucket may also have to be scaled when setting the VC parameters for call setup and that this example does not account for overhead incurred by encapsulations such as those specified in RFC 1483). To ensure no loss, network elements will have to allocate some buffering for bursts. If every hop implemented the fluid model perfectly, this buffering would simply be b (the token bucket size). However, as noted in the discussion of reshaping earlier, implementations are approximations and we expect that traffic will become more bursty as it goes through the network. However, as with shaping the amount of buffering required to handle the burstiness is bounded by b+Csum+Dsum*R. If one accounts for the peak rate, this can be further reduced to M + (b-M)(p-X)/(p-r) + (Csum/R + Dsum)X where X is set to r if (b-M)/(p-r) is less than Csum/R+Dsum and X is R if (b-M)/(p-r) is greater than or equal to Csum/R+Dsum and p>R; otherwise, X is set to p. This reduction comes from the fact that Shenker/Partridge/Guerin Expires 2/13/97 [Page 14] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 the peak rate limits the rate at which the burst, b, can be placed in the network. Conversely, if a non-zero slack term, Sout, is returned by the network element, the buffer requirements are increased by adding Sout to Dsum. While sending applications are encouraged to set the peak rate parameter and reshaping points are required to conform to it, it is always acceptable to ignore the peak rate for the purposes of computing buffer requirements and end-to-end delays. The result is simply an overestimate of the buffering and delay. As noted above, if the peak rate is unknown (and thus potentially infinite), the buffering required is b+Csum+Dsum*R. The end-to-end delay without the peak rate is b/R+Ctot/R+Dtot. The parameter D for each network element SHOULD be set to the maximum datagram transfer delay variation (independent of rate and bucket size) through the network element. For instance, in a simple router, one might compute the difference between the worst case and best case times it takes for a datagram to get through the input interface to the processor, and add it to any variation that may occur in how long it would take to get from the processor to the outbound link scheduler (assuming the queueing schemes work correctly). For datagramized weighted fair queueing, D is set to the link MTU divided by the link bandwidth, to account for the possibility that a packet arrives just as a maximum-sized packet begins to be transmitted, and that the arriving packet should have departed before the maximum-sized packet. For a frame-based, slotted system such as Stop and Go queueing, D is the maximum number of slots a datagram may have to wait before getting a chance to be transmitted. Note that multicasting may make determining D more difficult. In many subnets, ATM being one example, the properties of the subnet may depend on the path taken from the multicast sender to the receiver. There are a number of possible approaches to this problem. One is to choose a representative latency for the overall subnet and set D to the (non-negative) difference from that latency. Another is to estimate subnet properties at exit points from the subnet, since the exit point presumably is best placed to compute the properties of its path from the source. NOTE: It is important to note that there is no fixed set of rules about how a subnet determines its properties, and each subnet technology will have to develop its own set of procedures to accurately compute C and D and slack values. D is intended to be distinct from the latency through the network element. Latency is the minimum time through the device (the speed Shenker/Partridge/Guerin Expires 2/13/97 [Page 15] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 of light delay in a fiber or the absolute minimum time it would take to move a packet through a router), while parameter D is intended to bound the variability in non-rate-based delay. In practice, this distinction is sometimes arbitrary (the latency may be minimal) -- in such cases it is perfectly reasonable to combine the latency with D and to advertise any latency as zero. NOTE: It is implicit in this scheme that to get a complete guarantee of the maximum delay a packet might experience, a user of this service will need to know both the queueing delay (provided by C and D) and the latency. The latency is not advertised by this service but is a general characterization parameter (advertised as specified in [8]). However, even if latency is not advertised, this service can still be used. The simplest approach is to measure the delay experienced by the first packet (or the minimum delay of the first few packets) received and treat this delay value as an upper bound on the latency. The parameter C is the data backlog resulting from the vagaries of how a specific implementation deviates from a strict bit-by-bit service. So, for instance, for datagramized weighted fair queueing, C is set to M to account for packetization effects. If a network element uses a certain amount of slack, Si, to reduce the amount of resources that it has reserved for a particular flow, i, the value Si SHOULD be stored at the network element. Subsequently, if reservation refreshes are received for flow i, the network element MUST use the same slack Si without any further computation. This guarantees consistency in the reservation process. As an example for the use of the slack term, consider the case where the required end-to-end delay, Dreq, is larger than the maximum delay of the fluid flow system. The latter is obtained by setting R=r in the fluid delay formula (for stability, R>=r must be true), and is given by b/r + Ctot/r + Dtot. In this case the slack term is S = Dreq - (b/r + Ctot/r + Dtot). The slack term may be used by the network elements to adjust their local reservations, so that they can admit flows that would otherwise have been rejected. A network element at an intermediate network element that can internally differentiate between delay and rate Shenker/Partridge/Guerin Expires 2/13/97 [Page 16] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 guarantees can now take advantage of this information to lower the amount of resources allocated to this flow. For example, by taking an amount of slack s <= S, an RCSD scheduler [5] can increase the local delay bound, d, assigned to the flow, to d+s. Given an RSpec, (Rin, Sin), it would do so by setting Rout = Rin and Sout = Sin - s. Similarly, a network element using a WFQ scheduler can decrease its local reservation from Rin to Rout by using some of the slack in the RSpec. This can be accomplished by using the transformation rules given in the previous section, that ensure that the reduced reservation level will not increase the overall end-to-end delay. Evaluation Criteria The scheduling algorithm and admission control algorithm of the element MUST ensure that the delay bounds are never violated and datagrams are not lost, when a source's traffic conforms to the TSpec. Furthermore, the element MUST ensure that misbehaving flows do not affect the service given to other flows. Vendors are encouraged to formally prove that their implementation is an approximation of the fluid model. Examples of Implementation Several algorithms and implementations exist that approximate the fluid model. They include Weighted Fair Queueing (WFQ) [2], Jitter- EDD [3], Virtual Clock [4] and a scheme proposed by IBM [5]. A nice theoretical presentation that shows these schemes are part of a large class of algorithms can be found in [6]. Examples of Use Consider an application that is intolerant of any lost or late datagrams. It uses the advertised values Ctot and Dtot and the TSpec of the flow, to compute the resulting delay bound from a service request with rate R. Assuming R < p, it then sets its playback point to [(b-M)/R*(p-R)/(p-r)]+(M+Ctot)/R+Dtot. Security Considerations This memo discusses how this service could be abused to permit denial of service attacks. The service, as defined, does not allow denial of service (although service may degrade under certain circumstances). Shenker/Partridge/Guerin Expires 2/13/97 [Page 17] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 Appendix 1: Use of the Guaranteed service with RSVP The use of guaranteed service in conjunction with the RSVP resource reservation setup protocol is specified in reference [9]. This document gives the format of RSVP FLOWSPEC, SENDER_TSPEC, and ADSPEC objects needed to support applications desiring guaranteed service and gives information about how RSVP processes those objects. The RSVP protocol itself is specified in Reference [10]. References [1] S. Shenker and J. Wroclawski. "Network Element QoS Control Service Specification Template". Internet Draft, July 1996, <draft- ietf-intserv-svc-template-03.txt> [2] A. Demers, S. Keshav and S. Shenker, "Analysis and Simulation of a Fair Queueing Algorithm," in Internetworking: Research and Experience, Vol 1, No. 1., pp. 3-26. [3] L. Zhang, "Virtual Clock: A New Traffic Control Algorithm for Packet Switching Networks," in Proc. ACM SIGCOMM '90, pp. 19-29. [4] D. Verma, H. Zhang, and D. Ferrari, "Guaranteeing Delay Jitter Bounds in Packet Switching Networks," in Proc. Tricomm '91. [5] L. Georgiadis, R. Guerin, V. Peris, and K. N. Sivarajan, "Efficient Network QoS Provisioning Based on per Node Traffic Shaping," IBM Research Report No. RC-20064. [6] P. Goyal, S.S. Lam and H.M. Vin, "Determining End-to-End Delay Bounds in Heterogeneous Networks," in Proc. 5th Intl. Workshop on Network and Operating System Support for Digital Audio and Video, April 1995. [7] A.K.J. Parekh, A Generalized Processor Sharing Approach to Flow Control in Integrated Services Networks, MIT Laboratory for Information and Decision Systems, Report LIDS-TH-2089, February 1992. [8] S. Shenker and J. Wroclawski. "General Characterization Parameters for Integrated Service Network Elements", Internet Draft, July 1996, <draft-ietf-intserv-charac-02.txt> [9] J. Wroclawski, "Use of RSVP with IETF Integrated Services", Internet Draft, July 1996, <draft-ietf-intserv-rsvp-use-00.txt> [10] B. Braden, et. al. "Resource Reservation Protocol (RSVP) - Version 1 Functional Specification", Internet Draft, July 1996, <draft-ietf-rsvp-spec-13.txt> Shenker/Partridge/Guerin Expires 2/13/97 [Page 18] INTERNET-DRAFT draft-ietf-intserv-guaranteed-svc-06.txt 13 August 1996 Authors' Addresses: Scott Shenker Xerox PARC 3333 Coyote Hill Road Palo Alto, CA 94304-1314 email: shenker@parc.xerox.com 415-812-4840 415-812-4471 (FAX) Craig Partridge BBN 2370 Amherst St Palo Alto CA 94306 email: craig@bbn.com Roch Guerin IBM T.J. Watson Research Center Yorktown Heights, NY 10598 email: guerin@watson.ibm.com 914-784-7038 914-784-6318 (FAX) Shenker/Partridge/Guerin Expires 2/13/97 [Page 19]