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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 17, 2013                            BT Innovate & Design


                                                       October 17, 2012


              Session Initiation Protocol (SIP) Rate Control
                draft-ietf-soc-overload-rate-control-03.txt


Abstract

   The prevalent use of Session Initiation Protocol (SIP) [RFC3261] in
   Next Generation Networks necessitates that SIP networks provide
   adequate control mechanisms to maintain transaction throughput by
   preventing congestion collapse during traffic overloads. Already
   [draft-ietf-soc-overload-control-09] proposes a loss-based solution
   to remedy known vulnerabilities of the [RFC3261] SIP 503 (service
   unavailable) overload control mechanism. This document proposes a
   rate-based control solution to complement the loss-based control
   defined in [draft-ietf-soc-overload-control-09].

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 April 17, 2012.

Copyright Notice

   Copyright (c) 2012 IETF Trust and the persons identified as the
   document authors. All rights reserved.





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   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (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....................................4
      3.1. Overview..................................................4
      3.2. Summary of via headers 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. Optional enhancement: avoidance of resonance........10
   4. Example.......................................................11
   5. Syntax........................................................12
   6. Security Considerations.......................................13
   7. IANA Considerations...........................................14
   8. References....................................................14
      8.1. Normative References.....................................14
      8.2. Informative References...................................14
   Appendix A. Contributors.........................................15
   Appendix B. Acknowledgments......................................15

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
   [draft-ietf-soc-overload-control-09]. That solution provides a
   communication scheme for overload control algorithms. It also
   includes a default loss-based overload control algorithm that makes



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   it possible for a set of clients to limit offered load towards an
   overloaded server.

   However, such 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 cannot guarantee clients to
   produce a bounded offered load from the clients towards an
   overloaded server and requires frequent updates which may have
   implications for stability.

   This document proposes extensions to [draft-ietf-soc-overload-
   control-09] to support a rate-based control that 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 must estimate
   a separate target for each client, rather than an overall loss
   percentage, equally applicable to all clients.
   The proposed rate-based overload control algorithm mitigates
   congestion in SIP networks while adhering to the overload signaling
   scheme in [draft-ietf-soc-overload-control-09] and presenting a rate
   based control as an optional alternative to the default loss-based
   control in [draft-ietf-soc-overload-control-09].

2. Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [RFC2119].

   The normative statements in this specification as they apply to SIP
   clients and SIP servers assume that both the SIP clients and SIP
   servers support this specification.  If, for instance, only a SIP
   client supports this specification and not the SIP server, then
   follows that the normative statements in this specification
   pertinent to the behavior of a SIP server do not apply to the server
   that does not support this specification.







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3. Rate-based algorithm scheme

3.1. Overview

   The server is what the overload control algorithm defined here
   protects and the client is what throttles traffic towards the
   server.

   Following the procedures defined in [draft-ietf-soc-overload-
   control-09], 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, queueing delay...) to evaluate its overload state
   and estimate a target SIP request rate (as opposed to target percent
   loss in the case of loss-based control).

   When in overload, the server uses [draft-ietf-soc-overload-control-
   09] via header oc parameters of SIP responses to inform the clients
   of its overload state and of the target SIP request rate.

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

3.2. Summary of via headers parameters for overload control

   oc: Used by SIP clients to indicate draft-ietf-soc-overload-control-
   09 support and by SIP servers to indicate the load reduction amount.

   oc parameters defined in draft-ietf-soc-overload-control-09 are
   summarized below:

   oc-algo: Used by SIP clients to advertise supported overload control
   algorithms and by SIP servers to notify clients of algorithm in
   effect. Support values: loss (default), rate (optional).

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

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



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   The use of the via header oc parameter(s) inform of the desired
   rate, but they don't explicitly ''inform clients of the overload
   state''.

3.3. Client and server rate-control algorithm selection

   Per [draft-ietf-soc-overload-control-09], new clients indicate
   supported overload control algorithms to servers by inserting oc and
   oc-algo, with the names of the supported algorithms, in Via header
   of SIP requests destined to servers.  The inclusion by the client of
   the string ''rate'' indicates that the client supports a rate based
   algorithm. Conversely, servers notify clients of selected overload
   control algorithm through the oc-algo parameter in the Via header of
   SIP responses to clients. The inclusion by the server of the string
   ''rate'' indicates that the rate based algorithm has been selected by
   the server.

   Support of rate-based control MUST be indicated by clients setting
   oc-algo to "rate". Selection of rate-based control MUST be indicated
   by servers by setting oc-algo to ''rate''.


3.4. Server operation

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

   However, the server MUST be able to evaluate periodically its
   overload state and estimate a target SIP request rate beyond which
   it would become overloaded. The server must allocate a portion of
   the target SIP request rate to each of its client. The server may
   set the same rate for every client, or may set different rates for
   different clients.

   The max 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 [draft-
   ietf-soc-overload-control-09] and REQ 13 of RFC 5390, request
   prioritization is the client responsibility. But when deriving this
   rate the server may need to take into account characteristics of the
   requests, and the effect of the client prioritization on the load it



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   receives, e.g. CPU utilization will depend upon the characteristics
   of the requests which would presumably allow the server to take in
   account prioritization.
   Note that the target SIP request rate is a max rate that may not be
   attained by the arrival rate at the client, and the server cannot
   assume that it will.

   Upon detection of overload, the server MUST follow the
   specifications in [draft-ietf-soc-overload-control-09] to notify its
   clients of the allocated target SIP   request rate.

   The server MUST use [draft-ietf-soc-overload-control-09] oc
   parameter to send a target SIP request rate to each of its clients.


3.5. Client operation

 3.5.1. Default algorithm

   To throttle new SIP requests at the rate specified in the oc value
   sent by the server to its clients, the client MAY use the proposed
   default algorithm for rate-based control or any other equivalent
   algorithm.

   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 in the oc value
   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 in the oc value, namely T = 1 /
   oc-value.

   Note that when the oc-value is 0 with a non-zero oc-validity, then
   the client should reject 100% of SIP requests destined to the
   overload server. However, when both oc-value and oc-validity are 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 rate at 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 burstyness, or the number consecutive SIP requests
   forwarded to the SIP server (burst size proportional to TAU over the
   difference between 1/T and the arrival rate).

   SIP 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 SIP server with a relatively small number of
   clients, each with proportionally larger arrival rate, will benefit
   from a larger value of TAU.

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

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

   where X is the content of the bucket 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 bucket content 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


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   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 occupancy of the bucket is initialized to the
   parameter TAU0 (preferably configurable) which is 0 or larger but
   less than or equal to TAU.

   Following [draft-ietf-soc-overload-control-09], the client is
   responsible for message priority and for maintaining two categories
   of requests: Requests candidate for reduction, 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
   requests candidate for reduction. With two priorities, the proposed
   Leaky bucket requires two thresholds TAU1 < TAU2:
     . All new requests would be admitted when the bucket fill is at
        or below TAU1,
     . Only higher priority requests would be admitted when the bucket
        fill is between TAU1 and TAU2,
     . All requests would be rejected when the bucket fill is 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 are set. And the larger
   the difference between TAU1 and TAU2, the closer to the control is
   to strict priority.

   TAU (or TAU1 and TAU2) can assume any positive real number value and
   is not necessarily bounded by T.




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   Note that specification of a value for TAU is beyond the scope of
   this document.

   A reference algorithm is shown below.

   No priority case:

   // T: emission interval, set to 1 / TargetRate
   // TAU:  tolerance parameter
   // ta: arrival time of last arrival
   // LCT:  arrival time of last conforming SIP request (initialized to
   //       the first arrival time)
   // X: current value of leaky bucket counter (initialized to 0)

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

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

   Priority case:

   // T: emission interval, set to 1 / TargetRate
   // TAU1:  tolerance parameter of no priority SIP requests
   // TAU2:  tolerance parameter of priority SIP requests
   // ta: arrival time of last arrival
   // LCT:  arrival time of last conforming SIP request (initialized to
   //       the first arrival time)
   // X: current value of leaky bucket counter (initialized to 0)

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




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   if (AnyRequestReceived && Xp <= TAU1 || PriorityRequestReceived &&
   Xp <= TAU2 && Xp > TAU1) {
     // Accept 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. Optional enhancement: avoidance of resonance

   As the number of client sources of traffic increases and 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 this can occur, then a simple
   way to avoid this 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 bucket occupancy to X', generate a value u as
   follows:

     if X' > 0, then u=0

     else if X' <= 0 then 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 bucket fill 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


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   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];

     . 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;

     . At 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 [draft-ietf-soc-overload-control-09] example in section 6.2
   where SIP 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-algo="rate";oc-validity=0;




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            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 that it
   created. P1 supports two overload control algorithms: loss and rate.

   The second line --- a SIP response --- shows the top most Via header
   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.

   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 seconds.

            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:

   oc          = "oc" EQUAL oc-value

   oc-value    = "NaN" / oc-num

   oc-num      = 1*DIGIT






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6. Security Considerations

   For completeness, draft-ietf-soc-overload-control-09 Security
   Considerations section is repeated here.

   Overload control mechanisms can be used by an attacker to conduct a
   denial-of-service attack on a SIP entity if the attacker can pretend
   that the SIP entity is overloaded. When such a forged overload
   indication is received by an upstream SIP client, it will stop
   sending traffic to the victim. Thus, the victim is subject to a
   denial-of-service attack.

   An attacker can create forged overload feedback by inserting itself
   into the communication between the victim and its upstream
   neighbors. The attacker would need to add overload feedback
   indicating a high load to the responses passed from the victim to
   its upstream neighbor. Proxies can prevent this attack by
   communicating via TLS. Since overload feedback has no meaning beyond
   the next hop, there is no need to secure the communication over
   multiple hops.

   Another way to conduct an attack is to send a message containing a
   high overload feedback value through a proxy that does not support
   this extension. If this feedback is added to the second Via headers
   (or all Via headers), it will reach the next upstream proxy. If the
   attacker can make the recipient believe that the overload status was
   created by its direct downstream neighbor (and not by the attacker
   further downstream) the recipient stops sending traffic to the
   victim. A precondition for this attack is that the victim proxy does
   not support this extension since it would not pass through overload
   control feedback otherwise.

   A malicious SIP entity could gain an advantage by pretending to
   support this specification but never reducing the amount of traffic
   it forwards to the downstream neighbor. If its downstream neighbor
   receives traffic from multiple sources which correctly implement
   overload control, the malicious SIP entity would benefit since all
   other sources to its downstream neighbor would reduce load.

   The solution to this problem depends on the overload control method.
   For rate-based and window-based overload control, it is very easy
   for a downstream entity to monitor if the upstream neighbor
   throttles traffic forwarded as directed. For percentage throttling
   this is not always obvious since the load forwarded depends on the
   load received by the upstream neighbor.




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7. IANA Considerations

   None.

8. References

8.1. Normative References

   [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.

8.2. Informative References

   [draft-ietf-soc-overload-control-09]
             Gurbani, V., Hilt, V., Schulzrinne, H., "Session
             Initiation Protocol (SIP) Overload Control", draft-ietf-
             soc-overload-control-09.


   [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: Volker
   Hilt.

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



   Authors' Addresses

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

   Philip M Williams
   BT Innovate & Design
   UK





















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