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Versions: (draft-noel-soc-overload-rate-control) 00 01 02 03 04 05 06 07 08 09 10 RFC 7415

SOC Working Group                                             Eric Noel
Internet-Draft                                                AT&T Labs
Intended status: Standards Track                     Philip M. Williams
Expires: April 10, 2015                            BT Innovate & Design


                                                       October 10, 2014


              Session Initiation Protocol (SIP) Rate Control
                draft-ietf-soc-overload-rate-control-10.txt


Abstract

   The prevalent use of Session Initiation Protocol (SIP) in Next
   Generation Networks necessitates that SIP networks provide adequate
   control mechanisms to maintain transaction throughput by preventing
   congestion collapse during traffic overloads. Already a loss-based
   solution to remedy known vulnerabilities of the SIP 503 (service
   unavailable) overload control mechanism has been proposed. This
   document proposes a rate-based control scheme to complement the
   loss-based control scheme, using the same signaling.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   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."

   This Internet-Draft will expire on January 22, 2015.

Copyright Notice

   Copyright (c) 2014 IETF Trust and the persons identified as the
   document authors. All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents



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   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document. Please review these documents
   carefully, as they describe your rights and restrictions with
   respect to this document. Code Components extracted from this
   document must include Simplified BSD License text as described in
   Section 4.e of the Trust Legal Provisions and are provided without
   warranty as described in the Simplified BSD License.

   Table of Contents

   1. Introduction...................................................2
   2. Terminology....................................................3
   3. Rate-based algorithm scheme....................................3
      3.1. Overview..................................................3
      3.2. Via header field parameters for overload control..........4
      3.3. Client and server rate-control algorithm selection........5
      3.4. Server operation..........................................5
      3.5. Client operation..........................................6
         3.5.1. Default algorithm....................................6
         3.5.2. Priority treatment..................................10
         3.5.3. Optional enhancement: avoidance of resonance........12
   4. Example.......................................................13
   5. Syntax........................................................14
   6. Security Considerations.......................................14
   7. IANA Considerations...........................................14
   8. References....................................................15
      8.1. Normative References.....................................15
      8.2. Informative References...................................15
   Appendix A. Contributors.........................................16
   Appendix B. Acknowledgments......................................16

1. Introduction

   The use of SIP in large-scale Next Generation Networks requires that
   SIP-based networks provide adequate control mechanisms for handling
   traffic growth. In particular, SIP networks must be able to handle
   traffic overloads gracefully, maintaining transaction throughput by
   preventing congestion collapse.

   A promising SIP-based overload control solution has been proposed in
   [RFC7339]. That solution provides a communication scheme for
   overload control algorithms. It also includes a default loss-based
   overload control algorithm that makes it possible for a set of
   clients to limit offered load towards an overloaded server.



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   However, such a loss control algorithm is sensitive to variations in
   load so that any increase in load would be directly reflected by the
   clients in the offered load presented to the overloaded servers.
   More importantly, a loss-based control scheme cannot guarantee an
   upper bound on the clients offered load from the clients towards an
   overloaded server and requires frequent updates which may have
   implications for stability.

   In accordance with the framework defined in [RFC7339], this document
   proposes an alternate overload control, the rate-based overload
   control algorithm.  The rate-based control guarantees an upper bound
   on the rate, constant between server updates, of requests sent by
   clients towards an overloaded server. The tradeoff is in terms of
   algorithmic complexity, since the overloaded server is more likely
   to use a different target (maximum rate) for each client than the
   loss-based approach.

   The proposed rate-based overload control algorithm mitigates
   congestion in SIP networks while adhering to the overload signaling
   scheme in [RFC7339] and presenting a rate-based control as an
   optional alternative to the default loss-based control scheme in
   [RFC7339].


2. Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [RFC2119].

   Unless otherwise specified, all SIP entities described in this
   document are assumed to support this specification.



3. Rate-based algorithm scheme

3.1. Overview

   The server is the one protected by the overload control algorithm
   defined here, and the client is the one that throttles traffic
   towards the server.


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   Following the procedures defined in [RFC7339], the server and
   clients signal one another support for rate-based overload control.

   Then periodically, the server relies on internal measurements (e.g.,
   CPU utilization or queueing delay) to evaluate its overload state
   and estimate a target maximum SIP request rate in number of request
   per second (as opposed to target percent loss in the case of loss-
   based control).

   When in overload, the server uses the Via header field oc parameter
   [RFC7339] of SIP responses in order to inform the clients of its
   overload state and of the target maximum SIP request rate for that
   client.

   Upon receiving the oc parameter with a target maximum SIP request
   rate, each client throttles new SIP requests towards the overloaded
   server.


3.2. Via header field parameters for overload control

   The use of the Via header field oc parameter informs clients of the
   desired maximum rate. They are defined in [RFC7339] and summarized
   below:

   oc: Used by clients in SIP requests to indicate [RFC7339] support
   and by servers to indicate the load reduction amount in the loss
   algorithm, and the maximum rate, in messages per second, for the
   rate based algorithm described here. oc-algo: Used by clients in SIP
   requests to advertise supported overload control algorithms and by
   servers to notify clients of the algorithm in effect. Supported
   values: loss (default), rate (optional).

   oc-validity: Used by servers in SIP responses to indicate an
   interval of time (msec) that the load reduction should be in effect.
   A value of 0 is reserved for the server to stop overload control. A
   non-zero value is required in all other cases.

   oc-seq: A sequence number associated with the "oc" parameter.

   Consult Section 4 for an illustration of the Via header field oc
   parameter usage



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3.3. Client and server rate-control algorithm selection

   Per [RFC7339], new clients indicate supported overload control
   algorithms to servers by inserting oc and oc-algo, with the names of
   the supported algorithms, in the Via header field of SIP requests
   destined to servers.  The inclusion by the client of the token
   "rate" indicates that the client supports a rate based algorithm.
   Conversely, servers notify clients of the selected overload control
   algorithm through the oc-algo parameter in the Via header field of
   SIP responses to clients. The inclusion by the server of the token
   "rate" in the oc-algo parameter indicates that the rate-based
   algorithm has been selected by the server.

   Support of rate-based control MUST be indicated by clients including
   the token "rate" in the oc-algo list. Selection of rate-based
   control MUST be indicated by servers by setting oc-algo to the token
   "rate".


3.4. Server operation

   The actual algorithm used by the server to determine its overload
   state and estimate a target maximum SIP request rate is beyond the
   scope of this document.

   However, the server MUST periodically evaluate its overload state
   and estimate a target SIP request rate beyond which it would become
   overloaded. The server must determine how it will allocate the
   target SIP request rate among its client. The server may set the
   same rate for every client, or may set different rates for different
   clients.

   The maximum rate determined by the server for a client applies to
   the entire stream of SIP requests, even though throttling may only
   affect a particular subset of the requests, since as per [RFC7339]
   and REQ 13 of [RFC5390], request prioritization is a client's
   responsibility.

   When setting the maximum rate for a particular client, the server
   may need take into account the workload (e.g., CPU load per request)


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   of the distribution of message types from that client.  Furthermore,
   because the client may prioritize the specific types of messages it
   sends while under overload restriction, this distribution of message
   types may be different from (e.g., either higher or lower CPU load)
   the message distribution for that client under non-overload
   conditions

   Note that the "oc" parameter for the rate algorithm is an upper
   bound (in messages per second) on the traffic sent by the client to
   the server. The client may send traffic at a rate significantly
   lower than the upper bound for a variety of reasons

   In other words, when multiple clients are being controlled by an
   overloaded server, at any given time some clients may receive
   requests at a rate below their target (maximum) SIP request rate
   while others above that target rate. But the resulting request rate
   presented to the overloaded server will converge towards the target
   SIP request rate.

   Upon detection of overload and the determination to invoke overload
   controls, the server MUST follow the specifications in [RFC7339] to
   notify its clients of the allocated target SIP request rate and that
   rate-based control is in effect.

   The server MUST use the [RFC7339] "oc" parameter to send a target
   SIP request rate to each of its clients.

   When a client supports the default loss algorithm and not the rate
   algorithm, the client would be handled in the same way as in draft-
   ietf-soc-overload-control section 5.10.2.


3.5. Client operation

 3.5.1. Default algorithm

   In determining whether or not to transmit a specific message, the
   client may use any algorithm that limits the message rate to the
   "oc" parameter in units of messages per second. For ease of
   discussion, we define T = 1/["oc" parameter] as the target inter-SIP
   request interval. The algorithm may be strictly deterministic, or it



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   may be probabilistic. It may, or may not, have a tolerance factor,
   to allow for short bursts, as long as the long term rate remains
   below 1/T.

   The algorithm may have provisions for prioritizing traffic in
   accordance with REQ 13 of [RFC5390].

   If the algorithm requires other parameters (in addition to "T",
   which is 1/["oc" parameter]), they may be set autonomously by the
   client, or they may be negotiated between client and server
   independently of the SIP-based overload control solution.

   In either case, the coordination is out-of-scope for this document.
   The default algorithms presented here (one without provisions for
   prioritizing traffic, one with) are only examples.

   To throttle new SIP requests at the rate specified by the "oc"
   parameter sent by the server to its clients, the client MAY use the
   proposed default algorithm for rate-based control or any other
   equivalent algorithm that forward messages in conformance with the
   upper bound of 1/T messages per second.

   The default leaky bucket algorithm presented here is based on [ITU-T
   Rec. I.371] Appendix A.2.  The algorithm makes it possible for
   clients to deliver SIP requests at a rate specified by the "oc"
   parameter with tolerance parameter TAU (preferably configurable).

   Conceptually, the leaky bucket algorithm can be viewed as a finite
   capacity bucket whose real-valued content drains out at a continuous
   rate of 1 unit of content per time unit and whose content increases
   by the increment T for each forwarded SIP request. T is computed as
   the inverse of the rate specified by the "oc" parameter, namely T =
   1 / ["oc" parameter].

   Note that when the "oc" parameter is 0 with a non-zero oc-validity,
   then the client should reject 100% of SIP requests destined to the
   overload server. However, when the oc-validity value is 0, the
   client should immediately stop throttling.






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   If, at a new SIP request arrival, the content of the bucket is less
   than or equal to the limit value TAU, then the SIP request is
   forwarded to the server; otherwise, the SIP request is rejected.

   Note that the capacity of the bucket (the upper bound of the
   counter) is (T + TAU).

   The tolerance parameter TAU determines how close the long-term
   admitted rate is to an ideal control that would admit all SIP
   requests for arrival rates less than 1/T and then admit SIP requests
   precisely at the rate of 1/T for arrival rates above 1/T. In
   particular at mean arrival rates close to 1/T, it determines the
   tolerance to deviation of the inter-arrival time from T (the larger
   TAU the more tolerance to deviations from the inter-departure
   interval T).

   This deviation from the inter-departure interval influences the
   admitted rate burstiness, or the number of consecutive SIP requests
   forwarded to the server (burst size proportional to TAU over the
   difference between 1/T and the arrival rate).

   In situations where clients are configured with some knowledge about
   the server (e.g., operator pre-provisioning), it can be beneficial
   to choose a value of TAU based on how many clients will be sending
   requests to the server.

   Servers with a very large number of clients, each with a relatively
   small arrival rate, will generally benefit from a smaller value for
   TAU in order to limit queuing (and hence response times) at the
   server when subjected to a sudden surge of traffic from all clients.
   Conversely, a server with a relatively small number of clients, each
   with proportionally larger arrival rate, will benefit from a larger
   value of TAU.

   Once the control has been activated, at the arrival time of the k-th
   new SIP request, ta(k), the content of the bucket is provisionally
   updated to the value

   X' = X - (ta(k) - LCT)




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   where X is the value of the leaky bucket counter after arrival of
   the last forwarded SIP request, and LCT is the time at which the
   last SIP request was forwarded.

   If X' is less than or equal to the limit value TAU, then the new SIP
   request is forwarded and the leaky bucket counter X is set to X' (or
   to 0 if X' is negative) plus the increment T, and LCT is set to the
   current time ta(k). If X' is greater than the limit value TAU, then
   the new SIP request is rejected and the values of X and LCT are
   unchanged.

   When the first response from the server has been received indicating
   control activation (oc-validity>0), LCT is set to the time of
   activation, and the leaky bucket counter is initialized to the
   parameter TAU0 (preferably configurable) which is 0 or larger but
   less than or equal to TAU.

   TAU can assume any positive real number value and is not necessarily
   bounded by T.

   TAU=4*T is a reasonable compromise between burst size and throttled
   rate adaptation at low offered rates.

   Note that specification of a value for TAU and any communication or
   coordination between servers are beyond the scope of this document.

   A reference algorithm is shown below.

   No priority case:

   // T: inter-transmission interval, set to 1 / ["oc" parameter]
   // TAU: tolerance parameter
   // ta: arrival time of the most recent arrival received by the
   //     client
   // LCT: arrival time of last SIP request that was sent to the server
   //      (initialized to the first arrival time)
   // X: current value of the leaky bucket counter (initialized to
   //    TAU0)

   // After most recent arrival, calculate auxiliary variable Xp
   Xp = X - (ta - LCT);


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   if (Xp <= TAU) {
     // Transmit SIP request
     // Update X and LCT
     X = max (0, Xp) + T;
     LCT = ta;
   } else {
     // Reject SIP request
     // Do not update X and LCT
   }


 3.5.2. Priority treatment

   As with the loss-based algorithm of [RFC7339], a client implementing
   the rate-based algorithm also prioritizes messages into two or more
   categories of requests:
   Requests that are candidates for reduction and requests not subject
   to reduction (except under extenuating circumstances when there
   aren't any messages in the first category that can be reduced).

   Accordingly, the proposed leaky bucket implementation is modified to
   support priority using two thresholds for SIP requests in the set of
   request candidates for reduction. With two priorities, the proposed
   leaky bucket requires two thresholds TAU1 < TAU2:
     . All new requests would be admitted when the leaky bucket
        counter is at or below TAU1,
     . Only higher priority requests would be admitted when the leaky
        bucket counter is between TAU1 and TAU2,
     . All requests would be rejected when the bucket counter is at or
        above TAU2.
   This can be generalized to n priorities using n thresholds for n>2
   in the obvious way.

   With a priority scheme that relies on two tolerance parameters (TAU2
   influences the priority traffic, TAU1 influences the non-priority
   traffic), always set TAU1 < TAU2 (TAU is replaced by TAU1 and TAU2).
   Setting both tolerance parameters to the same value is equivalent to
   having no priority. TAU1 influences the admitted rate the same way
   as TAU does when no priority is set. And the larger the difference



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   between TAU1 and TAU2, the closer the control is to strict priority
   queueing.

   TAU1 and TAU2 can assume any positive real number value and are not
   necessarily bounded by T.

   Reasonable values for TAU0, TAU1 and TAU2 are: TAU0 = 0, TAU1 = 1/2
   * TAU2 and TAU2 = 10 * T.

   Note that specification of a value for TAU1 and TAU2 and any
   communication or coordination between servers are beyond the scope
   of this document.

   A reference algorithm is shown below.

   Priority case:

   // T: inter-transmission interval, set to 1 / ["oc" parameter]
   // TAU1: tolerance parameter of no-priority SIP requests
   // TAU2: tolerance parameter of priority SIP requests
   // ta: arrival time of the most recent arrival received by the
   //     client
   // LCT: arrival time of last SIP request that was sent to the server
   //      (initialized to the first arrival time)
   // X: current value of the leaky bucket counter (initialized to
   //    TAU0)

   // After most recent arrival, calculate auxiliary variable Xp
   Xp = X - (ta - LCT);

   if (AnyRequestReceived && Xp <= TAU1) || (PriorityRequestReceived &&
   Xp <= TAU2 && Xp > TAU1) {
     // Transmit SIP request
     // Update X and LCT
     X = max (0, Xp) + T;
     LCT = ta;
   } else {
     // Reject SIP request
     // Do not update X and LCT
   }



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 3.5.3. Optional enhancement: avoidance of resonance

   As the number of client sources of traffic increases or the
   throughput of the server decreases, the maximum rate admitted by
   each client needs to decrease, and therefore the value of T becomes
   larger. Under some circumstances, e.g., if the traffic arises very
   quickly simultaneously at many sources, the occupancies of each
   bucket can become synchronized, resulting in the admissions from
   each source being close in time and batched or very 'peaky' arrivals
   at the server, which not only gives rise to control instability, but
   also very poor delays and even lost messages. An appropriate term
   for this is 'resonance' [Erramilli].

   If the network topology is such that resonance can occur, then a
   simple way to avoid resonance is to randomize the bucket occupancy
   at two appropriate points: At the activation of control and whenever
   the bucket empties, as follows.

   After updating the value of the leaky bucket to X', generate a value
   u as follows:

     if X' > 0, then u = 0

     else if X' <= 0 then let u be set to a random value uniformly
                     distributed between -1/2 and +1/2
   Then (only) if the arrival is admitted, increase the bucket by an
   amount T + uT, which will therefore be just T if the bucket hadn't
   emptied, or lie between T/2 and 3T/2 if it had.

   This randomization should also be done when control is activated,
   i.e., instead of simply initializing the leaky bucket counter to
   TAU0, initialize it to TAU0 + uT, where u is uniformly distributed
   as above. Since activation would have been a result of response to a
   request sent by the client, the second term in this expression can
   be interpreted as being the bucket increment following that
   admission.

   This method has the following characteristics:

     . If TAU0 is chosen to be equal to TAU and all sources were to
        activate control at the same time due to an extremely high
        request rate, then the time until the first request admitted by
        each client would be uniformly distributed over [0,T];




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     . The maximum occupancy is TAU + (3/2)T, rather than TAU + T
        without randomization;

     . For the special case of 'classic gapping' where TAU=0, then the
        minimum time between admissions is uniformly distributed over
        [T/2, 3T/2], and the mean time between admissions is the same,
        i.e., T+1/R where R is the request arrival rate;

     . As high load randomization rarely occurs, so there is no loss
        of precision of the admitted rate, even though the randomized
        'phasing' of the buckets remains.


4. Example

   Adapting the example in section 6.2 of [RFC7339], where client P1
   sends requests to a downstream server P2:

            INVITE sips:user@example.com SIP/2.0

            Via: SIP/2.0/TLS p1.example.net;

             branch=z9hG4bK2d4790.1;received=192.0.2.111;

             oc;oc-algo="loss,rate"

            ...

            SIP/2.0 100 Trying

            Via: SIP/2.0/TLS p1.example.net;

             branch=z9hG4bK2d4790.1;received=192.0.2.111;

             oc=0;oc-algo="rate";oc-validity=0;

             oc-seq=1282321615.781

            ...

   In the messages above, the first line is sent by P1 to P2.  This
   line is a SIP request; because P1 supports overload control, it
   inserts the "oc" parameter in the topmost Via header field that it
   created. P1 supports two overload control algorithms: loss and rate.





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   The second line, a SIP response, shows the top most Via header field
   amended by P2 according to this specification and sent to P1.
   Because P2 also supports overload control, it chooses the rate-based
   scheme and sends that back to P1 in the oc-algo parameter. It uses
   oc-validity=0 to indicate no overload control. In this example oc=0,
   but oc could be any value as oc is ignored when oc-validity=0.

   At some later time, P2 starts to experience overload. It sends the
   following SIP message indicating P1 should send SIP requests at a
   rate no greater than or equal to 150 SIP requests per second and for
   a duration of 1,000 msec.

            SIP/2.0 180 Ringing

            Via: SIP/2.0/TLS p1.example.net;

             branch=z9hG4bK2d4790.1;received=192.0.2.111;

             oc=150;oc-algo="rate";oc-validity=1000;

             oc-seq=1282321615.782

             ...



5. Syntax

   This specification extends the existing definition of the Via header
   field parameters of [RFC3261] as follows:

   algo-list /= "rate"



6. Security Considerations

   Aside from the resonance concerns discussed in Section 3.5.3, this
   mechanism does not introduce any security concerns beyond the
   general overload-control security issues discussed in [RFC7339].
   Methods to mitigate the risk of resonance are discussed in Section
   3.5.3.

7. IANA Considerations

   Header Field Parameter Name Predefined Values Reference



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   _______________________________________________________

   Via          oc-algo          Yes             RFC7339 RFCOPRQ

   RFCOPRQ [NOTE TO RFC-EDITOR: Please replace with final RFC number of
   draft-ietf-soc-overload-rate-control]



8. References

8.1. Normative References

   [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
             Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
             A., Peterson, J., Sparks, R., Handley, M., and E.
             Schooler, "SIP: Session Initiation Protocol", RFC 3261,
             June 2002.

   [RFC5390] Rosenberg, J., "Requirements for Management of Overload in
             the Session Initiation Protocol", RFC 5390, December 2008

   [RFC7339]
             Gurbani, V., Hilt, V., Schulzrinne, H., "Session
             Initiation Protocol (SIP) Overload Control", RFC 7339,
             September 2014.


8.2. Informative References

    [ITU-T Rec. I.371]
             "Traffic control and congestion control in B-ISDN", ITU-T
             Recommendation I.371.


    [Erramilli]
             A. Erramilli and L. J. Forys, "Traffic Synchronization
             Effects In Teletraffic Systems", ITC-13, 1991.









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Appendix A.                 Contributors

   Significant contributions to this document were made by Janet Gunn.



Appendix B.                 Acknowledgments

   Many thanks for comments and feedback on this document to: Richard
   Barnes, Keith Drage, Vijay Gurbany, Volker Hilt, Christer Holmberg,
   Winston Hong, Peter Yee, and James Yu.

   This document was prepared using 2-Word-v2.0.template.dot.



   Authors' Addresses

   Eric Noel
   AT&T Labs
   200 S Laurel Avenue
   Middletown, NJ 07747
   USA

   Philip M. Williams
   BT Innovate & Design
   Ipswich, IP5 3RE
   UK



















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