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Versions: 00 draft-ivov-mmusic-latching

Network Working Group                                          H. Kaplan
Internet-Draft                                               Acme Packet
Intended status: Informational                                   E. Ivov
Expires: September 6, 2012                                         Jitsi
                                                                 D. Wing
                                                                   Cisco
                                                           March 5, 2012


      Latching: Hosted NAT Traversal (HNT) for Media in Real-Time
                             Communication
                    draft-kaplan-mmusic-latching-00

Abstract

   This document describes behavior of signalling intermediaries in RTC
   deployments, sometimes referred to as Session Border Controllers
   (SBCs), when performing Hosted NAT Traversal (HNT).  HNT is a set of
   mechanisms, such as media relaying and latching, that such
   intermediaries use to enable other RTC devices behind NATs to
   communicate with each other.  This document is non-normative, and is
   only written to explain HNT in order to provide a reference to the
   IETF community, as well as an informative description to
   manufacturers, and users.

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 September 6, 2012.

Copyright Notice

   Copyright (c) 2012 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



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   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 . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  Background . . . . . . . . . . . . . . . . . . . . . . . . . .  4
   4.  Impact on Signaling  . . . . . . . . . . . . . . . . . . . . .  5
   5.  Media Behavior, Latching . . . . . . . . . . . . . . . . . . .  6
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 10
   7.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 12
     7.1.  Normative References . . . . . . . . . . . . . . . . . . . 12
     7.2.  Informative References . . . . . . . . . . . . . . . . . . 12
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 13





























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1.  Introduction

   Network Address Translators (NATs) are widely used in the Internet by
   consumers and organizations.  Although specific NAT behaviors vary,
   this document uses the term "NAT" for devices that map any IPv4 or
   IPv6 address and transport port number to another IPv4 or IPv6
   address and transport port number.  This includes consumer NAPTs,
   Firewall-NATs, IPv4-IPv6 NATs, Carrier-Grade NATs, etc.

   Protocols like SIP [RFC3261], and others that try to use a more
   direct path for media than with signalling, are difficult to use
   across NATs.  They use IP addresses and transport port numbers
   encoded in bodies such as SDP [RFC4566]> as well as, in the case of
   SIP, various header fields.  Such addresses and ports are unusable
   unless all peers in a session are located behind the same NAT.

   Mechanisms such as STUN [RFC5389], TURN [RFC5766], and ICE [RFC5245],
   did not exist when protocols like SIP began being deployed.  Session
   Border Controllers (SBCs) that were already being used by SIP domains
   for other SIP and media-related purposes began to use proprietary
   mechanisms to enable SIP devices behind NATs to communicate across
   the NATs.

   The term often used for this behavior is Hosted NAT Traversal (HNT),
   although some manufacturers sometimes use other names such as "Far-
   end NAT Traversal" or "NAT assist" instead.  The systems which
   perform HNT are frequently SBCs as described in [RFC5853], although
   other systems such as media gateways and "media proxies" sometimes
   perform the same role.  For the purposes of this document, all such
   systems are referred to as SBCs, and the NAT traversal behavior is
   called HNT.

   As of this document's creation time, a vast majority of SIP domains
   use HNT to enable SIP devices to communicate across NATs, despite the
   publication of ICE.  There are many reasons for this, but those
   reasons are not relevant to this document's purpose and will not be
   discussed.  It is however worth pointing out that the current
   deployment levels of HNT and NATs themselves make an exclusive
   adoption of ICE highly unlikely in the foreseeable future.

   The purpose of this document is to describe the mechanisms often used
   for HNT at the SDP and media layer, in order to aid understanding the
   implications and limitations imposed by it.  Although the mechanisms
   used in HNT are not novel to experts, publication in an IETF document
   is useful as a means of providing common terminology and a reference
   for related documents.

   In no way does this document try to make a case for HNT or present it



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   as a solution that is somehow superior to alternatives such as ICE.

   It is also worth mentioning that there are purely signaling-layer
   components of HNT as well.  One such component is briefly described
   for SIP in [RFC5853], but that is not the focus of this document.
   The SIP signaling-layer component of HNT is typically more
   implementation-specific and deployment-specific than the SDP and
   media components.  For the purposes of this document it is hence
   assumed that signaling intermediaries handle traffic in way that
   allows protocols such as SIP to function correctly across the NATs.

   The rest of this document is going to focus primarily on use of HNT
   for SIP.  However, the mechanisms described here are relatively
   generic and are often used with other protocols, such as XMPP
   [RFC6120], MGCP, H.248/MEGACO, and H.323.


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].


3.  Background

   The general problems with NAT traversal for protocols such as SIP
   are:
   1.  The addresses and port numbers encoded in SDP bodies (or their
       equivalents) by NATed User Agents (UAs) are not usable across the
       Internet, because they represent the private addressing
       information of the UA rather than the addresses/ports that will
       be mapped to/from by the NAT.
   2.  The policies inherent in NATs, and explicit in Firewalls, are
       such that packets from outside the NAT cannot reach the UA until
       the UA sends packet out first.
   3.  Some NATs apply endpoint dependent filtering on incoming packets,
       as described in [RFC4787] and thus a UA may only be able to
       receive packets from the same remote peer IP:port as it sends
       packets out to.

   In order to overcome these issues, signaling intermediaries such as
   SIP SBCs on the public side of the NATs perform HNT for both
   signaling and media.  An example deployment model of HNT and SBCs is
   shown in Figure 1.






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                              +-----+       +-----+
                              | SBC |-------| SBC |
                              +-----+       +-----+
                               /                 \
                              /     Public Net    \
                             /                     \
                       +-----+                     +-----+
                       |NAT-A|                     |NAT-B|
                       +-----+                     +-----+
                         /                             \
                        / Private Net       Private Net \
                       /                                 \
                   +------+                            +------+
                   | UA-A |                            | UA-B |
                   +------+                            +------+


                    Figure 1: Logical Deployment Paths


4.  Impact on Signaling

   Along with codec and other media-layer information, session
   establishment signaling also conveys, potentially private and non-
   globally routable addressing information.  Signaling intermediaries
   would hence modify such information so that peer UAs are given the
   (public) addressing information of a media relay controlled by the
   intermediary.

   Quite often, the IP address of the newly introduced media relay may
   be the same as that of the signaling intermediary (e.g. the SIP SBC)
   or it may be a completely different one.  In almost all cases
   however, the new address would belong to the same IP address family
   as the one used for signaling, since it is known to work for that UA.

   The port numbers introduced in the signaling by the intermediary are
   typically allocated dynamically.  Allocation strategies are entirely
   implementation dependent and they often vary from one product to the
   next.

   The offer/answer media negotiation model [RFC3264] is such that once
   an offer is sent, the generator of the offer needs to be prepared to
   receive media on the advertised address/ports.  In practice such
   media may or may not be received, depending on the implementations
   participating in a given session, local policies, and call scenario.
   For example if a SIP SDP Offer originally came from a UA behind a
   NAT, the SIP SBC cannot send media to it until an SDP Answer is given
   to the UA and latching (Section 5) occurs.  Another example is when a



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   SIP SBC sends an SDP Offer in a SIP INVITE to a residential
   customer's UA and receives back SDP in a 18x response, the SBC may
   decide not to send media to that customer UA until a SIP 200 response
   for policy reasons, to prevent toll-fraud.


5.  Media Behavior, Latching

   An UA behind a NAT streams media from a private address:port set that
   once packets cross the NAT, will be mapped to a public set.  The UA
   however is not typically aware of the public mapping and would often
   advertise in the private address:port couple in signaling.  This way,
   when the signalling intermediary performing HNT receives the private
   addressing information from the UA it will not know what address/
   ports to send media to.  Therefore media relays used in HNT would
   often use a mechanism called "latching".

   Historically, "latching" only referred to the process by which SBCs
   "latch" onto UDP packets from a given UA for security purposes, and
   "symmetric-latching" is when the latched address:ports are used to
   send media back to the UA.  Today most people talk about them both as
   "latching", and thus this document does as well.

   The latching mechanism works as follows:
   1.  After receiving an offer from a NATed UA, a signaling
       intermediary located on the public Internet would allocate a set
       of IP address:ports on a media relay.  The set would then be
       advertised to the remote party so that it would use it for all
       media it wished to send toward the UA.
   2.  Next, after receiving an answer to its offer, the signaling
       server would allocate a second address:port set on the media
       relay.  It would advertise this second set to the UA and use it
       for all media traffic to and from the UA.
   3.  The media relay receives the media packets on the allocated
       ports, and uses their source address and port as a destination
       for all media bound in the opposite direction.  In other words,
       it "latches" or locks on these source address:port set.
   4.  This way all media streamed by the UA would be received on the
       second address:port set.  The source addresses and ports of the
       traffic would belong to the public interface of the NAT in front
       of the UA and anything that the relay sends there would find its
       way to it.
   5.  Similarly the source of the stream originating at the remote
       party would be latched upon and used for media going in that
       direction.
   6.  Latching is usually done only once per peer and not allowed to
       change or cause a re-latching until a new offer and answer get
       exchanged.



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   Figure 2 describes how latching occurs for SIP where HNT is provided
   by an SBC connected to two networks: 38.2.2/24 facing towards the UAC
   network and 38.1.1/24 facing towards the UAS network.


             10.0.0.1
             SIP UAC    NAT          38.2.2/24-SBC-38.1.1/24  SIP UAS
             -------    ---                    ---            -------
                |        |                      |                |
            1.  |-SIP INVITE+offer c=10.0.0.1-->|                |
                |        |                      |                |
            2.  |        |    (SBC allocates 38.1.1.2/22007      |
                |        |     for inbound RTP from UAS,         |
                |        |     and 38.2.2.4/36010 for            |
                |        |     inbound RTP from UAC)             |
                |        |                      |                |
            3.  |        |                      |-INVITE+offer-->|
                |        |                      | c=38.1.1.2/2207|
                |        |                      |                |
            4.  |        |                      |<-180 Ringing---|
                |        |                      |                |
                |        |                      |                |
            5.  |<---180 Ringing----------------|                |
                |        |                      |                |
            6.  |        |                      |<-200+answer----|
            7.  |<-200+answer, c=38.2.2.4/36010-| c=114.1.1.3    |
                |        |                      |                |
            8.  |--ACK------------------------->|                |
            9.  |        |                      |---ACK--------->|
                |        |                      |                |
           10.  |==RTP, dest=38.2.2.4/36010====>|                |
                |        |                      |                |
           11.  |        |                 (SBC latches to       |
                |        |                source IP address and  |
                |        |                port seen at (10))     |
                |        |                      |                |
           12.  |        |                      |<== RTP ========|
                |        |                      |                |
           13.  |<==RTP, to latched address=====|                |
                |        |                      |                |


           Figure 2: Latcing by a SIP SBC across two interfaces

   While XMPP implementations often rely on ICE to handle NAT traversal,
   there are some that also support a non-ICE transport called Raw UDP
   [XEP-0177].  Figure 3 describes how latching occurs for one such XMPP
   implementate where HNT is provided by an XMPP server on the public



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   internet.


   10.0.0.1     10.0.0.9/1.2.3.4            3.4.5.6             5.6.7.8
   XMPP Client1        NAT                XMPP Server       XMPP Client2
     -------          ---                    ---               -------
        |              |                      |                   |
    1.  |----session-initiate cand=10.0.0.1-->|                   |
        |              |                      |                   |
    2.  |<------------ack---------------------|                   |
        |              |                      |                   |
    3.  |              |    (Server allocates 3.4.5.6/2200        |
        |              |     for inbound RTP from Client2,        |
        |              |     and 3.4.5.6/3300 for                 |
        |              |     inbound RTP from Client1)            |
        |              |                      |                   |
    4.  |              |                      |-session-initiate->|
        |              |                      | cand=3.4.5.6/2200 |
        |              |                      |                   |
    5.  |              |                      |<-------ack--------|
        |              |                      |                   |
        |              |                      |                   |
    6.  |              |                      |<--session-accept--|
        |              |                      |   cand=5.6.7.8    |
        |              |                      |                   |
    7.  |              |                      |--------ack------->|
    8.  |<--session-accept cand=3.4.5.6/3300--|                   |
        |              |                      |                   |
    9.  |-------------ack-------------------->|                   |
        |              |                      |                   |
        |              |                      |                   |
   10.  |========RTP, dest=3.4.5.6/3300======>|                   |
        |              |                      |                   |
   11.  |              |              (XMPP server latches to     |
        |              |               src IP 1.2.3.4 and src     |
        |              |               port seen at (10))         |
        |              |                      |                   |
   12.  |              |                      |<== RTP ===========|
        |              |                      |                   |
   13.  |<======RTP, to latched address=======|                   |
        |              |                      |                   |


           Figure 3: Latcing by a SIP SBC across two interfaces

   The above is a general description, and some details vary between
   implementations or configuration settings.  For example, some
   intermediaries perform additional logic before latching on received



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   packet source information to prevent malicious attacks or latching
   erroneously to previous media senders - often called "rogue-rtp" in
   the industry.

   It is worth pointing out that latching is not an exclusively "server
   affair" and some clients may also use it in cases where they are
   configured with a public IP address and they are contacted by a NATed
   client with no other NAT traversal means.

   In order for latching to function correctly, the UA behind the NAT
   needs to support symmetric RTP.  That is, it needs to use the same
   ports for sending data as the ones it listens on for inbound packets.
   Today this is the case for with, for example, almost all SIP and XMPP
   clients.  Also UAs need to make sure they can begin sending media
   packets independently and without waiting for packets to arrive
   first.  In theory, it is possible that some UAs would not send
   packets out first; for example if a SIP session begins in 'inactive'
   or 'recvonly' SDP mode from the UA behind the NAT.  In practice,
   however, SIP sessions from regular UAs (the kind that one could find
   behind a NAT) virtually never begin in an inactive or recvonly mode,
   for obvious reasons.  The media direction would also be problematic
   if the SBC side indicated 'inactive' or 'sendonly' modes when it sent
   SDP to the UA.  However SBCs providing HNT would always be configured
   to avoid this.

   Given that, in order for latching to work properly, media relays need
   to begin receiving media before they start sending, it is possible
   for deadlocks to occur.  This can happen when the UAC and the UAS in
   a session are connected to different signalling intermediaries that
   both provide HNT.  In this case the media relays controled by the
   signalling servers could end up each waiting upon the other to
   initiate the streaming.  To prevent this relays would often attempt
   to start streaming toward the address:port sets provided in the
   offer/answer even before receiving any inbound traffic.  If the
   entity they are streaming to is another HNT performing server it
   would have provided its relay's public address and ports and the
   early stream would find its target.

   Although many SBCs only support UDP-based media latching, and in
   particular RTP/RTCP, many SBCs support TCP-based media latching as
   well.  TCP-based latching is more complicated, and involves forcing
   the UA behind the NAT to be the TCP client and sending the initial
   SYN-flagged TCP packet to the SBC (i.e., be the 'active' mode side of
   a TCP-based media session).  If both UAs of a TCP-based media session
   are behind NATs, then SBCs typically force both UAs to be the TCP
   clients, and the SBC splices the TCP connections together.  TCP
   splicing is a well-known technique, and described in [tcp-splicing].




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   HNT and latcing in particular are generally found to be working
   reliably but they do have obvious caveats.  The first one usually
   raised by IETF members is that UAs are not aware of it occurring.
   This makes it impossible for the mechanism to be used with protocols
   such as ICE that try various traversal techniques in an effort to
   choose the one the best suits a particular situation.  Overwriting
   address information in in offers and answers may actually completely
   prevent UAs from using ICE because of the ice-mismatch rules
   described in [RFC5245]

   The second issue raised by IETF members is that it causes media to go
   through a relay instead of directly over the IP-routed path between
   the two participating UAs.  While this adds obvious drawbacks such as
   reduced scalability and often increased latency, it is also
   considered a benefit by SBC administrators: if a customer pays for
   "phone" service, for example, the media is what is truly being paid
   for, and the administrators usually like to be able to detect that
   media is flowing correctly, evaluate its quality, know if and why it
   failed, etc.  Also in some cases routing media through operator
   controlled relays may route media over paths explicitly optimized for
   media and hence offer better performance than regular Internet
   routing.


6.  Security Considerations

   The security implications for HNT are complicated.  The mechanism
   itself needs to be concerned with latching to incorrect and possibly
   malicious sources.  A malicious source could, for example, attempt a
   resource exhaustion attack by flooding all possible media-latching
   UDP ports on the SBC in order to prevent calls from succeeding.  SBCs
   have various mechanisms to prevent this from happening., or alert an
   administrator, but a sufficiently sophisticated attacker may be able
   to bypass them for some time.  The most common example is typically
   referred to as "restricted-latching", whereby the SBC will not latch
   to any packets from a source public IP other than the one the SIP UA
   uses for SIP signaling.  In some cases the limitation may be loosened
   to allos media from a range of IPs belonging to the same network.
   This way the SBC simply ignores and does not latch onto packets
   coming from the attacker.  If the attacker knows the public source IP
   of the real SIP UA making a call, then they could still flood all of
   the SBC's public IPs and ports with packets spoofing that real SIP
   UA's public source IP.  However, this would only disturb media that
   IP (or range of IPs) rather than all calls that the SBC is servicing.

   A malicious source could send media packets to an SBC media-latching
   UDP port in the hopes of being latched-to for the purpose of
   receiving media for a given SIP session.  SBCs have various



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   mechanisms to prevent this as well.  Restricted latching for example
   would also help in this case since the attacker can't make the SBC
   send media packets back to themselves since the SBC will not latch
   onto the attackers packets.  There could still be an issue if the
   attacker happens to be either (1) in the IP routing path and thus can
   spoof the same IP as the real UA and get the media coming back, in
   which case the attacker hardly needs to attack at all to begin with,
   or (2) the attacker is behind the same NAT as the real SIP UA, in
   which case the attacker's packets will be latched-to by the SBC and
   the SBC will send media back to the attacker.  In this latter case,
   which is a corner-case to begin with, in practice the real SIP UA
   will end the call anyway, because the human won't hear anything and
   will hang up.  EXCEPT, if it's not a human but rather an answering
   machine, it may not hang up (though most answering machines do hang
   up when they don't get media).  The attacker could also redirect all
   media to the real SIP UA after receiving it, in which case the attack
   would likely remain undetected and succeed.  Naturally, SRTP
   [RFC3711] would prevent such an attack from being useful, and should
   be used independently of HNT.

   For SIP clients, HNT is usually transparent in the sense that the SIP
   UA does not know it occurs.  In certain cases it may be detectable,
   such as when ICE is supported by the SIP UA and the SBC modifies the
   default connection address and media port numbers in SDP, thereby
   disabling ICE due to the mismatch condition.  Even in that case,
   however, the SIP UA only knows a middlebox is relaying media, but not
   necessarily that it is performing latching/HNT.  [TODO: need to
   explain further]

   In order to perform HNT, the SBC has to modify SDP to and from the
   SIP UA behind a NAT, and thus the SIP UA cannot use S/MIME [RFC5751],
   and it cannot sign a sending request or verify a received request
   using [RFC4474] unless the SBC re-signs the request.  However it is
   sometimes argued that, neither S/MIME nor [RFC4474] are widely
   deployed and that this may not be a real concern.

   From a privacy perspective, media relaying is sometimes seen as a way
   of protecting one's IP address and not revealing it to the remote
   party.  That kind of IP address masking is often perceived as
   important.  However, this is no longer an exclusive advantage of HNT
   since it can also be accomplished by client-controlled relaying
   mechanisms such as TURN [RFC5766], if the client explicitly wishes to
   do so.


7.  References





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7.1.  Normative References

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

7.2.  Informative 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.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              June 2002.

   [RFC3489]  Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy,
              "STUN - Simple Traversal of User Datagram Protocol (UDP)
              Through Network Address Translators (NATs)", RFC 3489,
              March 2003.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [RFC4474]  Peterson, J. and C. Jennings, "Enhancements for
              Authenticated Identity Management in the Session
              Initiation Protocol (SIP)", RFC 4474, August 2006.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [RFC4787]  Audet, F. and C. Jennings, "Network Address Translation
              (NAT) Behavioral Requirements for Unicast UDP", BCP 127,
              RFC 4787, January 2007.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              April 2010.

   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              October 2008.

   [RFC5751]  Ramsdell, B. and S. Turner, "Secure/Multipurpose Internet
              Mail Extensions (S/MIME) Version 3.2 Message
              Specification", RFC 5751, January 2010.



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Internet-Draft    Hosted NAT Traversal for Media in RTC       March 2012


   [RFC5766]  Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN): Relay Extensions to Session
              Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.

   [RFC5853]  Hautakorpi, J., Camarillo, G., Penfield, R., Hawrylyshen,
              A., and M. Bhatia, "Requirements from Session Initiation
              Protocol (SIP) Session Border Control (SBC) Deployments",
              RFC 5853, April 2010.

   [RFC6120]  Saint-Andre, P., "Extensible Messaging and Presence
              Protocol (XMPP): Core", RFC 6120, March 2011.

   [RFC6189]  Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media
              Path Key Agreement for Unicast Secure RTP", RFC 6189,
              April 2011.

   [XEP-0177]
              Beda, J., Saint-Andre, P., Hildebrand, J., and S. Egan,
              "XEP-0177: Jingle Raw UDP Transport Method", XEP XEP-0177,
              December 2009.


Authors' Addresses

   Hadriel Kaplan
   Acme Packet
   100 Crosby Drive
   Bedford, MA  01730
   USA

   Email: hkaplan@acmepacket.com


   Emil Ivov
   Jitsi
   Strasbourg  67000
   France

   Email: emcho@jitsi.org












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   Dan Wing
   Cisco Systems, Inc.
   170 West Tasman Drive
   San Jose, CA  95134
   USA

   Email: dwing@cisco.com












































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