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Versions: 00 01 02                                                      
SIP -- Session Initiation Protocol                             D. Willis
Working Group                                                B. Campbell
Internet-Draft                                          dynamicsoft Inc.
Expires: August 13, 2003                                    Feb 12, 2003


   Session Initiation Protocol Extension to Assure Congestion Safety
                     draft-ietf-sip-congestsafe-01

Status of this Memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
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   This Internet-Draft will expire on August 13, 2003.

Copyright Notice

   Copyright (C) The Internet Society (2003).  All Rights Reserved.

Abstract

   The Session Initiation Protocol allows the use of UDP for transport
   of SIP messages.  The use of UDP inherently risks network congestion
   problems, as UDP itself does not define congestion prevention,
   avoidance, detection, or correction mechanisms.  This problem is
   aggravated by large SIP messages which fragment at the UDP level.
   Transport protocols in SIP are also negotiated on a per-hop basis, at
   the SIP level, so SIP proxies may convert from TCP to UDP and so
   forth.  This document defines what it means for SIP nodes to be
   congestion safe and specifies an extension by which a SIP User Agent
   may require that its requests are treated in a congestion safe
   manner.



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Table of Contents

   1.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  3

   2.  Background . . . . . . . . . . . . . . . . . . . . . . . . . .  3

   3.  Definition of Congestion Safety for SIP  . . . . . . . . . . .  3

   4.  Assuring Transitive Congestion Safety with Proxy-Require . . .  4

   5.  Responsible use of SIP over UDP  . . . . . . . . . . . . . . .  4
   5.1 Requirements For Use of SIP Over UDP . . . . . . . . . . . . .  6
   5.2 Pacing SIP Requests Over UDP . . . . . . . . . . . . . . . . .  6
   5.3 Proxy Rejects Request That Would Require UDP  Fragmentation  .  7
   5.4 Server Rejects Request Because Response Could Not Be Sent
       Safely . . . . . . . . . . . . . . . . . . . . . . . . . . . .  9

   6.  Syntax of Extensions and Changes to SIP Specifications . . . .  9

   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . .  9

   8.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 10

       Normative References . . . . . . . . . . . . . . . . . . . . . 11

       Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . 11

       Intellectual Property and Copyright Statements . . . . . . . . 12























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1. 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].

2. Background

   The Session Initiation Protocol RFC 3261 [4] provides application
   support over multiple transport protocols, including UDP and TCP.
   Transport negotiation is not "end to end" with SIP.  Instead, each
   SIP hop individually determines which transport to use.  For example,
   a User Agent (UA) may use TCP to talk to a proxy, that proxy my use
   UDP to talk to another proxy, and that second proxy may use SCTP to
   talk to a destination UA.

   UDP has inherent issues with congestion management.  The protocol has
   not explicit mechanisms for avoiding, detecting, or adapting to
   network congestion.  SIP attempts to deal with this in two ways:
   1.  Retransmission timers with exponential back offs.
   2.  Attempting to limit the size of transmissions over UDP to reduce
       the effects of fragmentation.

   This would appear to be an incomplete solution.  One solution might
   be to deprecate UDP entirely for SIP.  However, there is a large
   installed base using UDP, and there are legitimately places where UDP
   appears to be quite useful such as tiny mobile phones and in
   extremely high-volume proxies connecting over dedicated networks.

   As an alternative, this draft:
   1.  Defines what it means for a SIP node to be "congestion-safe".
   2.  Defines a mechanism whereby a congestion-safe UA may require that
       any proxy processing its requests be congestion safe.
   3.  Defines a mechanism whereby a proxy may reject a request that it
       would be forced to fragment, and in so doing inform the
       originating UA of relevant sizing parameters.
   4.  Defines a mechanism whereby a server may reject requests that
       would result in responses that might not be transmitted
       congestion-safely if the request itself was not received in a
       congestion-safe manner.

3. Definition of Congestion Safety for SIP

   A SIP node can be considered "congestion safe" if it never emits a
   request or response in a manner not known to be congestion safe.

   Requests may be considered congestion-safe if any one of the
   following criteria is met:



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   1.  The transport toward the next SIP hop is TCP, SCTP, or other
       transport providing congestion control and the next hop is known
       to be either a UA or a congestion-safe proxy.
   2.  The transport toward the next hop is UDP, the next hop is known
       to be a UA or congestion-safe proxy, and the network between the
       two is known to support congestion management at a lower layer.
       Note that this is an uncomoon case in typical Internet
       applications.
   3.  If the only available transport toward the next hop is UDP and
       the next hop is known to be a UA or congestion-safe proxy, the
       request MAY be transmitted over UDP or rejected by local policy.
       If the request is transmitted over UDP, the procedures described
       under the heading "Responsible use of SIP over UDP" in this
       document MUST be followed.

   Responses may be considered congestion-safe if any one of the
   following criteria is met:
   1.  The request was congestion-safe, as defined above.
   2.  The response is no larger than the request.

   The preceding uses the phrase "the next hop is known to be either a
   UA or a congestion-safe proxy." Such knowledge may be derived either
   through administrative configuration or through use of the
   Proxy-Require mechanism defined herein under the heading "Assuring
   Transitive Congestion Safety with Proxy-Require".

4. Assuring Transitive Congestion Safety with Proxy-Require

   SIP provides a mechanism whereby a user agent making a request can be
   assured that any proxy servicing that request support a specific
   extension or set of behavior.  To do so, the user agent includes a
   "Proxy-Require" header field with a value indicating a tag for the
   specific extension or behavior required.  There is an IANA
   registration process for these tags.  As per [4], proxies not
   recognizing a specific tag or unwilling to support the associated
   behavior reject a request referencing that tag with a 420 response,
   which has the semantic "Unsupported".

   We herein define a tag value of "congestion-safe".  A proxy
   forwarding a request containing a Proxy-Require with this tag value
   MUST manifest the property of congestion-safety as defined by this
   document.

5. Responsible use of SIP over UDP

   The fundamental problem with UDP is that it provides no feedback
   mechanism to allow a sender to pace its transmissions against the
   real performance of the network.  While this tends to have no



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   significant effect on extremely low-volume sender-receiver pairs, the
   impact of high-volume relationships on the network can be severe.
   Consider the following scenario, wherein the traffic between multiple
   UAs is funnelled through a single proxy-proxy relationship.

   Example of large-fan out/fan-in likely to encounter congestion:



         UA1----\                /----UA10
         UA2-----\              /-----UA11
         UA3------\            /------UA12
         UA4-------\          /-------UA13
         UA5--------P1------P2--------UA14
         UA6-------/          \-------UA15
         UA7------/            \------UA16
         UA8-----/              \-----UA17
         UA9----/                \----UA18


                                Figure 1

   In this scenario, any requests from UA(1..9) to UA(10..18) traverse
   the proxy-proxy link P1&lt-->P2.  Assuming current SIP practices, if
   this link is UDP and every UA emits a request simultaneously, each
   proxy will insert nine (one for each UA) requests, resulting in
   eighteen simultaneous requests on the P1&lt-->P2 link.  Each request
   may require retransmissions, and large requests may require
   fragmentation to fit the link MTU -- at the worst case, producing
   more than one hundred packets per request, or approximately 2,000
   simultaneously expressed packets in this scenario.  If the capacity
   of link P1&lt-->P2 is inadequate to deliver these messages within the
   SIP retransmission window, the originating UAs (or the proxies, if
   acting in transaction-stateful mode) generate retransmissions,
   further compounding the problem into a "retransmission storm".
   Real-world scenarios may scale far more seriously.  It is not
   unreasonable to assume that there may be tens of thousands of UAs on
   each side of the network.

   Clearly the best thing to do is to use a more sophisticated transport
   protocol (TCP, SCTP, etc.) between P1 and P2, and between each UA and
   its associated proxy.  If this is not feasible, it may be necessary
   to fall back to UDP.

   It should be noted that the fundamental problem not just between UAs
   and proxies, but whenever there is a high fan-out or fan-in ratio.
   If in the above example, each UA were behind a "residential proxy",
   the problem would occur in similar fashion.



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   One might propose that SIP ALWAYS use a congestion-controlled
   transport to talk to proxies, and only fall back to UDP when the next
   hop is a UA.  The primary problem with this approach is that in
   general, a SIP node does not and cannot know whether the next node is
   a UA or a proxy -- it is this ability to "insert" proxies into a
   sequence that provides much of the flexibility of SIP.  A secondary
   problem is that even if the next hop is a UA, some UAs are
   sufficienty high volume, and some links sufficiently narrow, that
   congestion might still result from the incautious use of UDP.

5.1 Requirements For Use of SIP Over UDP

   The previously described problems with the general use of SIP over
   UDP lead to the following two requirements for the use of UDP as a
   transport protocol for SIP:
   1.  Large messages MUST NOT be transmitted over UDP.  The SIP
       specification provides basic guidance for UAs.  Congestion-safe
       proxies MUST follow the procedures described below under the
       heading "Proxy Rejects Request That Would Require UDP
       Fragmentation." UAs MAY also make use of the MTU feedback
       techniques in that section.
   2.  Nodes sending requests over UDP MUST pace those requests as
       described under the heading "Pacing SIP requests over UDP."

   Response messages SHOULD be constrained to be smaller than the MTUs
   established for requests by the preceding mechanisms, and systems
   implementors should remain aware that SIP provides limited support
   for managing response sizes.  Further experience may indicate a need
   for further control over response handling.

5.2 Pacing SIP Requests Over UDP

   One simple way to describe the congestion problem is that UDP lets us
   send packets without knowing whether those packets are arriving.  The
   simplest approach to dealing with this at the application level is to
   send a request, then wait for some sort of response indicating that
   the request was received before sending anything else.  This produces
   an effect described by some as "ping-ponging" -- traffic bounces back
   and forth between two nodes like a ping-pong ball or tennis ball in a
   match.  Since there's only one ball in play between any two players
   at any given time, most of the potential for congestion cascades is
   eliminated.

   This pacing or serialization approach has the side-effect of
   significantly reducing the maximum throughput, as transmission occurs
   in only one direction at a time and there is at least a 2xRTT delay
   between transmissions.  More sophisticated algorithms such as those
   in TCP and SCTP have been developed to address this, and it would be



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   inappropriate to duplicate that work here.  Consequently, if greater
   efficiency is required than that provided by this simple approach,
   implementors should use TCP, SCTP, or another such protocol.  But if
   one absolutely must use UDP, this approach works, and is reasonably
   efficient in the most likely application of "edge proxy" to UA and
   other proxies with large fan-outs to individual low-volume nodes.

   SIP has two sorts of request transactions: "invite" and "non-invite"
   tranactions.  Invite transaction use a three way sequence of
   "request, response, acknowledgement" and may include a "provisional
   response" between the request and response steps.  Non-invite
   transactions use a two-way "request, response" sequence, and may also
   have a provisional response although that behavior has been
   deprecated.

   Congestion-safe use of SIP over UDP requires waiting for some sort of
   response to a request (or a timeout, which has backoff properties)
   before sending another request to that same destination.  A
   congestion-safe SIP node (UA or proxy) MUST NOT send a request to a
   given next-hop if there is an existing request to that destination
   which has not received some sort of response.  The existing
   transaction MUST either receive a response (final or provisional) or
   time-out before a new request can be made to that next-hop.

   This effectively requires congestion-safe proxies to act in a
   transaction-stateful manner on a per-next-hop destination basis, at
   least to the extent of tracking whether some sort of request is
   pending to each next-hop and correlating provisional and final
   responses to that request.

   Some may argue that this puts an excessive burden onto the SIP node,
   and that implementations that are "congestion-safe" per this
   specification will have reduced performance when used with UDP over a
   shared or public network.  We counter that congestion-safe transport
   protocols are readily available, and that network users which insist
   on using unsafe transports (such as UDP) MUST be responsible for
   assuring that they do not impede the function of other users of the
   network, even at the expense of reducing their own efficiency.  It is
   simply irresponsible to "blast away" at the network without regard
   for congestion or its impact on other users of the network.

5.3 Proxy Rejects Request That Would Require UDP  Fragmentation

   A proxy may be faced with a request to deliver a large message using
   UDP as a transport.  Fragmentation of such messages is problematic in
   several ways.  Loss of any fragment requires time-out and
   retransmission of the message.  The fragments are commonly
   transmitted out the interface at local interface (usually LAN) rates,



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   without awareness of intervening network conditions.  For these
   reason, we believe it in general a bad practice to send large
   requests over UDP.

   While the actual MTU of a link may not be known, common practice
   seems to indicate that the local interface MTU is likely to be a
   reasonable approximation.  Where the actual path MTU is known, that
   value should be used instead.

   When a congestion-safe SIP proxy processing a request determines that
   the next hop is reached via UDP, and that the request is larger than
   the effective MTU toward that hop and would consequently be
   fragmented, the proxy MUST reject that request with a 513 response.

   The base SIP specification provides minimal guidance on dealing with
   oversized requests.  There is an error response code, 513, with the
   semantic "request too large" that seems applicable.  However, SIP
   provides no guidance on how to indicate what size might be allowed.
   We define here two extension header fields that may be used in a 513
   response to indicate by the rejecting proxy the size of message
   allowed by that proxy.  The extension header field "Proxy-Max-Size"
   may be used to indicate the largest allowable request to the
   originating UA.  The extension header field "Proxy-Seen-Size" may be
   used to indicate the size of the rejected request as calculated by
   the rejecting proxy.  In both cases, the size value used indicates
   the SIP message size, which does not include IP or transport protocol
   overhead.

   A congestion-safe SIP proxy which rejects a request based on size
   SHOULD include a "Proxy-Max-Size" header field with a value
   indicating the largest size message allowed by this proxy on this
   link.  If a Proxy-Max-Size header field is sent, the proxy MUST also
   include a "Proxy-Seen-Size" header indicating the size of the request
   as seen at this proxy.

   A UA receiving a 513 response has the options of giving up, trying a
   smaller request, or trying a different set of proxies.  Should it
   choose to try a smaller request, it may estimate the size of the
   largest message that can be sent by taking the original request size,
   subtracting it from the value of the Proxy-Seen-Size header field,
   and subtracting that result from the value of the Proxy-max-Size
   header field.  Note that a UA SHOULD NOT repeatedly downsize and
   retry a request.  This technique is not an adequate replacement for
   TCP's Path MTU Discovery.  Any request that has been rejected more
   than once with a 513 SHOULD either be abandoned or re-issued over
   congestion-safe channels.





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5.4 Server Rejects Request Because Response Could Not Be Sent Safely

   A server receiving a SIP request generates a resposne to that
   request.  Delivery of this response may raise issues of
   congestion-safety.  Because SIP requires that responses traverse
   exactly the reverse of the route taken by the request (recorded in
   the Via: header fields values), the server has no options about
   routing the response.  If the request was delivered in a
   congestion-safe manner, it can be safely assumed that the response
   will also be returned in a congestion-safe manner, as it must
   traverse exactly this recorded route.  However, if the request was
   NOT received in a congestion-safe manner, the server cannot negotiate
   a congestion-safe path for the response, as the response must follow
   the path of the request.

   If the size of the generated response is less than the size of the
   received request, it may be reasonably assumed that since the request
   arrived intact, a response of equal or smaller size is likely to
   traverse the reverse of that path succesfully.  However, no such
   assumptions can be made about responses that are larger than the
   corresponding request.

   When a congestion-safe server generates a response to a request that
   is larger than the request and that request was not received over a
   congestion-safe channel, it cannot be assumed that the response can
   be safely transmitted.  An unsafe response cannot be transmitted by a
   congestion-safe server.  Instead the server MUST reject the request
   and return an error response using response code 514, which has the
   semantic of "Response Could Not Be Sent Safely".

   A UA receiving a 514 response to a request may either retry the
   request in a congestion-safe manner or abandon the request.

6. Syntax of Extensions and Changes to SIP Specifications

   The syntax for the Proxy-Max-Size header field is:

   Proxy-Max-Size = "Proxy-Max-Size" HCOLON 1*DIGIT

   The syntax for the Proxy-Seen-Size header field is:

   Proxy-Seen-Size = "Proxy-Seen-Size" HCOLON 1*DIGIT

7. IANA Considerations

   This document defines the SIP extension header fields
   "Proxy-Max-Size" and "Proxy-Seen-Size" ", which IANA will add to the
   registry of SIP header fields defined in [4].



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   This document also defines the SIP option tag "congestion-safe" which
   IANA will add to the registry of SIP option tags defined in [4].

   This document also defines the SIP response code 514, with the
   semantic "Response Cannot Be Sent Safely" which IANA will add to the
   registry of SIP response codes defined in [4] in the section for 5xx
   clase response codes.

   The following is the registration for the Proxy-Max-Size header
   field:

      RFC Number: RFCXXXX [Note to IANA: Fill in with the RFC number of
         this specification.]

      Header Field Name: Proxy-Max-Size

      Compact Form: none


   The following is the registration for the Proxy-Seen-Size header
   field:

      RFC Number: RFCXXXX [Note to IANA: Fill in with the RFC number of
         this specification.]

      Header Field Name: Proxy-Seen-Size

      Compact Form: none


   The following is the registration for the congestion-safe option tag:

      RFC Number: RFCXXXX [Note to IANA: Fill in with the RFC number of
         this specification.]

      Option Tag: congestion-safe

   The following is the registration for the SIP response code 514:

      RFC Number: RFCXXXX [Note to IANA: Fill in with the RFC number of
         this specification.]

      Response Code: 514    Response Cannot Be Sent Safely

8. Acknowledgements

   Robert Sparks and Jonathan Rosenberg argued with us vociferously over
   this topic and contributed substantial insight.



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Normative References

   [1]  Bradner, S., "The Internet Standards Process -- Revision 3", BCP
        9, RFC 2026, October 1996.

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

   [3]  Postel, J. and J. Reynolds, "Instructions to RFC Authors", RFC
        2223, October 1997.

   [4]  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.

   [5]  Mankin, A., Bradner, S., Mahy, R., Willis, D., Ott, J. and B.
        Rosen, "Change Process for the Session Initiation Protocol
        (SIP)", BCP 67, RFC 3427, December 2002.


Authors' Addresses

   Dean Willis
   dynamicsoft Inc.
   5100 Tennyson Parkway
   Suite 1200
   Plano, TX  75028
   US

   Phone: +1 972 473 5455
   EMail: dean.willis@softarmor.com
   URI:   http://www.dynamicsoft.com/


   Ben Campbell
   dynamicsoft Inc.
   5100 Tennyson Parkway
   Suite 1200
   Plano, TX  75028
   US

   Phone: +1 972 473 5452
   EMail: bcampbell@dynamicsoft.com
   URI:   http://www.dynamicsoft.com/







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   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
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