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RFC 7362

Informational
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Latching: Hosted NAT Traversal (HNT) for Media in Real-Time Communication

 


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Internet Engineering Task Force (IETF)                           E. Ivov
Request for Comments: 7362                                         Jitsi
Category: Informational                                        H. Kaplan
ISSN: 2070-1721                                                   Oracle
                                                                 D. Wing
                                                                   Cisco
                                                          September 2014


                  Latching: Hosted NAT Traversal (HNT)
                  for Media in Real-Time Communication

Abstract

   This document describes the behavior of signaling intermediaries in
   Real-Time Communication (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 Internet community and an
   informative description to manufacturers and users.

   Latching, which is one of the HNT components, has a number of
   security issues covered here.  Because of those, and unless all
   security considerations explained here are taken into account and
   solved, the IETF advises against use of the latching mechanism over
   the Internet and recommends other solutions, such as the Interactive
   Connectivity Establishment (ICE) protocol.

Status of This Memo

   This document is not an Internet Standards Track specification; it is
   published for informational purposes.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Not all documents
   approved by the IESG are a candidate for any level of Internet
   Standard; see Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://www.rfc-editor.org/info/rfc7362.

Page 2 
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
   (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.  Background  . . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.  Impact on Signaling . . . . . . . . . . . . . . . . . . . . .   5
   4.  Media Behavior and Latching . . . . . . . . . . . . . . . . .   6
   5.  Security Considerations . . . . . . . . . . . . . . . . . . .  11
   6.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  13
   7.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  14
     7.1.  Normative References  . . . . . . . . . . . . . . . . . .  14
     7.2.  Informative References  . . . . . . . . . . . . . . . . .  14

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 NATs,
   firewall/NATs, IPv4-IPv6 NATs, Carrier-Grade NATs (CGNs) [RFC6888],
   etc.

   The Session Initiation Protocol (SIP) [RFC3261], and others that try
   to use a more direct path for media than with signaling, are
   difficult to use across NATs.  These protocols use IP addresses and
   transport port numbers encoded in bodies such as the Session
   Description Protocol (SDP) [RFC4566] and, 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 Session Traversal Utilities for NAT (STUN)
   [RFC5389], Traversal Using Relays around NAT (TURN) [RFC5766], and
   Interactive Connectivity Establishment (ICE) [RFC5245] did not exist

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   when protocols like SIP began being deployed.  Some mechanisms, such
   as the early versions of STUN [RFC3489], had started appearing, but
   they were unreliable and suffered a number of issues typical for
   UNilateral Self-Address Fixing (UNSAF), as described in [RFC3424].
   For these and other reasons, 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.  These mechanisms
   are often transparent to endpoints and rely on a dynamic address and
   port discovery technique called "latching".

   The term often used for this behavior is "Hosted NAT Traversal
   (HNT)"; a number of manufacturers sometimes use other names such as
   "Far-end NAT Traversal" or "NAT assist" instead.  The systems that
   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.

   At the time of this document's publication, 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 make the complete
   extinction of this practice 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 well known in the community, publication in an IETF
   document is useful as a means of providing common terminology and a
   reference for related documents.

   This document does not attempt to make a case for HNT or present it
   as a solution that is somehow better than alternatives such as ICE.
   Due to the security issues presented in Section 5, the latching
   mechanism is considered inappropriate for general use on the Internet
   unless all security considerations are taken into account and solved.
   The IETF is instead advising for the use of the Interactive
   Connectivity Establishment (ICE) [RFC5245] and Traversal Using Relays
   around NAT (TURN) [RFC5766] protocols.

   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.

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   SIP uses numerous expressive primitives for message routing.  As a
   result, the HNT component for SIP 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 a way that allows protocols such as
   SIP to function correctly across the NATs.

   The rest of this document focuses primarily on the use of HNT for
   SIP.  However, the mechanisms described here are relatively generic
   and are often used with other protocols such as the Extensible
   Messaging and Presence Protocol (XMPP) [RFC6120], Media Gateway
   Control Protocol (MGCP) [RFC3435], Megaco/H.248 [RFC5125], and H.323
   [H.323].

2.  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 network 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 packets out first.

   3.  Some NATs apply endpoint-dependent filtering on incoming packets,
       as described in [RFC4787]; 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: Signaling and Media Flows in a Common Deployment Scenario

3.  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.

   In typical deployments, the media relay and signaling intermediary
   (i.e., the SBC) are co-located, thereby sharing the same IP address.
   Also, the address of the media relay would typically belong to the
   same IP address family as the one used for signaling (as it is known
   to work for that UA).  In other words, signaling and media would both
   travel over either IPv4 or IPv6.

   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 the 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 4) occurs.  Another example
   is, when a SIP SBC sends an SDP offer in a SIP INVITE to a

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   residential customer's UA and receives back SDP in a 18x response,
   the SBC may decide, for policy reasons, not to send media to that
   customer UA until a SIP 200 response has been received (e.g., to
   prevent toll fraud).

4.  Media Behavior and Latching

   An UA that is behind a NAT would stream media from an address and a
   port number (an address:port tuple) that are only valid in its local
   network.  Once packets cross the NAT, that address:port tuple will be
   mapped to a public one.  The UA, however, is not typically aware of
   the public mapping and would often advertise the private address:port
   tuple in signaling.  This way, while a session is still being set up,
   the signaling intermediary is not yet aware what addresses and ports
   the caller and the callee would end up using for media traffic: it
   has only seen them advertise the private addresses they use behind
   their respective NATs.  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:port tuples are used
   to send media back to the UA.  Today, most people talk about them
   both as "latching"; thus, this document does as well.

   The latching mechanism works as follows:

   1.  After receiving an offer from Alice (User Agent Client (UAC)
       located behind a NAT), a signaling intermediary located on the
       public Internet would allocate a set of IP address:port tuples on
       a media relay.  The set would then be advertised to Bob (User
       Agent Server (UAS)) so that he would use those media relay
       address:port tuples for all media he wished to send toward Alice
       (UAC).

   2.  Next, after receiving from Bob (UAS) an answer to its offer, the
       signaling server would allocate a second address:port set on the
       media relay.  In its answer to Alice (UAC), the SBC will replace
       Bob's address:port with this second set.  This way, Alice will
       send media to this media relay address:port.

   3.  The media relay receives the media packets on the allocated ports
       and uses their respective source address:ports as a destination
       for all media bound in the opposite direction.  In other words,
       it "latches" or locks on these source address:port tuples.

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   4.  This way, when Alice (UAC) streams media toward the media relay,
       it would be received on the second address:port tuple.  The
       source address:port of her traffic would belong to the public
       interface of Alice's NAT, and anything that the relay sends back
       to that address:port would find its way to Alice.

   5.  Similarly, the source of the media packets that Bob (UAS) is
       sending 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 (e.g., in a subsequent call or after a SIP peer has
       gone on and off hold).  The reasons for such restrictions are
       mostly related to security: once a session has started, a user
       agent is not expected to suddenly start streaming from a
       different port without sending a new offer first.  A change may
       indicate an attempt to hijack the session.  In some cases,
       however, a port change may be caused by a re-mapping in a NAT
       device standing between the SBC and the UA.  More advanced SBCs
       may therefore allow some level of flexibility on the re-latching
       restrictions while carefully considering the potential security
       implications of doing so.

   Figure 2 describes how latching occurs for SIP where HNT is provided
   by an SBC connected to two networks: 203.0.113/24 facing towards the
   UAC network and 198.51.100/24 facing towards the UAS network.

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   192.0.2.1   192.0.2.9/203.0.113.4                   198.51.100.33
      Alice         NAT       203.0.113.9-SBC-198.51.100.2     Bob
     -------        ---                   ---                -------
        |            |                     |                       |
    1.  |--SIP INVITE+offer c=192.0.2.1--->|                       |
        |            |                     |                       |
    2.  |            |   (SBC allocates 198.51.100.2:22007         |
        |            |    for inbound RTP from Bob)                |
        |            |                     |                       |
    3.  |            |                     |-----INVITE+offer----->|
        |            |                     |  c=198.51.100.2:22007 |
        |            |                     |                       |
    4.  |            |                     |<------180 Ringing-----|
        |            |                     |                       |
        |            |                     |                       |
    5.  |<------180 Ringing----------------|                       |
        |            |                     |                       |
    6.  |            |                     |<------200+answer------|
        |            |                     |                       |
    7.  |            |   (SBC allocates 203.0.113.9:36010          |
        |            |    for inbound RTP from Alice)              |
        |            |                     |                       |
    8.  |<-200+answer,c=203.0.113.9:36010--|  c=198.51.100.33      |
        |            |                     |                       |
    9.  |------------ACK------------------>|                       |
   10.  |            |                     |----------ACK--------->|
        |            |                     |                       |
   11.  |=====RTP,dest=203.0.113.9:36010==>|                       |
        |            |                     |                       |
   12.  |            |                (SBC latches to              |
        |            |               source IP address and         |
        |            |               port seen at (11))            |
        |            |                     |                       |
   13.  |            |                     |<======= RTP ==========|
        |            |                     |dest:198.51.100.2:22007|
   14.  |<=====RTP, to latched address=====|                       |
        |            |                     |                       |


           Figure 2: Latching 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 XMPP
   Jingle Raw UDP Transport Method [XEP-0177].  Figure 3 describes how
   latching occurs for one such XMPP implementation where HNT is
   provided by an XMPP server on the public Internet.

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   192.0.2.1  192.0.2.9/203.0.113.4        203.0.113.9      198.51.100.8
      Romeo           NAT                  XMPP Server            Juliet
      -----           ---                      ---                 -----
        |              |                        |                     |
    1.  |----session-initiate cand=192.0.2.1--->|                     |
        |              |                        |                     |
    2.  |<------------ack-----------------------|                     |
        |              |                        |                     |
    3.  |              |      (Server allocates 203.0.113.9:2200      |
        |              |       for inbound RTP from Juliet)           |
        |              |                        |                     |
    4.  |              |                        |--session-initiate-->|
        |              |                        |cand=203.0.113.9:2200|
        |              |                        |                     |
    5.  |              |                        |<--------ack---------|
        |              |                        |                     |
        |              |                        |                     |
    6.  |              |                        |<---session-accept---|
        |              |                        |  cand=198.51.100.8  |
        |              |                        |                     |
    7.  |              |                        |---------ack-------->|
        |              |                        |                     |
    8.  |              |      (Server allocates  203.0.113.9:3300     |
        |              |       for inbound RTP from Romeo)            |
        |              |                        |                     |
    9.  |<-session-accept cand=203.0.113.9:3300-|                     |
        |              |                        |                     |
   10.  |-----------------ack------------------>|                     |
        |              |                        |                     |
        |              |                        |                     |
   11.  |======RTP, dest=203.0.113.9:3300======>|                     |
        |              |                        |                     |
   12.  |              |               (XMPP server latches to        |
        |              |                src IP 203.0.113.4 and        |
        |              |                src port seen at (11))        |
        |              |                        |                     |
   13.  |              |                        |<======= RTP ========|
        |              |                        |dest=203.0.113.9:2200|
   14.  |<======RTP, to latched address=========|                     |
        |              |                        |                     |


        Figure 3: Latching by an XMPP Server 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 exclusively a "server
   affair", and some clients may also use it in cases where they are
   configured with a public IP address and 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 with almost all SIP and XMPP clients.  Also,
   UAs need to make sure they can begin sending media packets
   independently 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 '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 signaling intermediaries that
   both provide HNT.  In this case, the media relays controlled by the
   signaling 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 tuples 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 (in
   particular, RTP/RTCP), many SBCs support TCP-based media latching as
   well.  TCP-based latching is more complicated; it 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, as described in [TCP-SPLICING].

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   HNT and latching, in particular, are generally found to work
   reliably, but they do have obvious caveats.  The first one usually
   raised by IETF participants 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 that best suits a particular situation.
   Overwriting address information 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 participants 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 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
   the 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.

5.  Security Considerations

   A common concern is that an SBC (or an XMPP server -- all security
   considerations apply to both) that implements HNT may latch to
   incorrect and possibly malicious sources.  The ICE [RFC5245]
   protocol, for example, provides authentication tokens (conveyed in
   the ice-ufrag and ice-pwd attributes) that allow the identity of a
   peer to be confirmed before engaging in media exchange with her.
   Without such authentication, a malicious source could 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 to alert an
   administrator when it does.  Still, 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 address
   other than the one the SIP UA uses for SIP signaling.  This way, the
   SBC simply ignores and does not latch onto packets coming from the
   attacker.  In some cases, the limitation may be loosened to allow
   media from a range of IP addresses belonging to the same network in
   order to allow for use cases such as decomposed UAs and various forms
   of third-party call control.  However, since relaxing the
   restrictions in such a way may provide attackers with a larger attack

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   surface, such configurations are generally performed only on a case-
   by-case basis so that the specifics of individual deployments can be
   taken into account.

   All of the above problems would still arise if the attacker knows the
   public source IP of the UA that is actually making the call.  This
   would allow attackers to still flood all of the SBC's public IP
   addresses and ports with packets spoofing that SIP UA's public source
   IP address.  However, this would only impact media from that IP (or
   range of IP addresses) 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
   mechanisms to prevent this as well.  Restricted latching, for
   example, would also help in this case because the attacker can't make
   the SBC send media packets back to themselves since the SBC will not
   latch onto the attacker's media packets, not having seen the
   corresponding signaling packets first.  There could still be an issue
   if the attacker happens to be either (1) in the IP routing path where
   it can thus 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) behind the same NAT as the legitimate 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 the latter case,
   which may be of particular concern with Carrier-Grade NATs, the
   legitimate SIP UA will likely end the call anyway when a human user
   who does not hear anything hangs up.  In the case of a non-human call
   participant, such as an answering machine, this may not happen
   (although many such automated UAs would also hang up when they do not
   receive any 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.  Again, this would be of
   particular concern with larger-scale NATs serving many different
   endpoints, such as Carrier-Grade NATs.  The larger the number of
   devices fronted by a NAT is, the more use cases would vary, and the
   more the number of possible attack vectors would grow.

   Naturally, Secure RTP (SRTP) [RFC3711] would help mitigate such
   threats and, if used with the appropriate key negotiation mechanisms,
   would protect the media from monitoring while in transit.  It should
   therefore be used independently of HNT.  Section 26 of [RFC3261]
   provides an overview of additional threats and solutions on
   monitoring and session interception.

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   With SRTP, if the SBC that performs the latching is actually
   participating in the SRTP key exchange, then it would simply refuse
   to latch onto a source unless it can authenticate it.  Failing to
   implement and use SRTP would represent a serious threat to users
   connecting from behind Carrier-Grade NATs [RFC6888] and is considered
   a harmful practice.

   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 that a middlebox is relaying media but
   not necessarily that it is performing latching/HNT.

   In order to perform HNT, the SBC has to modify SDP to and from the
   SIP UA behind a NAT; thus, the SIP UA cannot use S/MIME [RFC5751],
   and it cannot sign a sending request, or verify a received request
   using the SIP Identity mechanism [RFC4474] unless the SBC re-signs
   the request.  However, neither S/MIME nor SIP Identity are widely
   deployed; thus, not being able to sign/verify requests appears not to
   be a concern at this time.

   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.

6.  Acknowledgements

   The authors would like to thank Flemming Andreasen, Miguel A.
   Garcia, Alissa Cooper, Vijay K. Gurbani, Ari Keranen, and Paul
   Kyzivat for their reviews and suggestions on improving this document.

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7.  References

7.1.  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.

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

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

   [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.

   [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.

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

7.2.  Informative References

   [H.323]    International Telecommunication Union, "Packet-based
              multimedia communication systems", ITU-T Recommendation
              H.323, December 2009.

   [RFC3424]  Daigle, L. and IAB, "IAB Considerations for UNilateral
              Self-Address Fixing (UNSAF) Across Network Address
              Translation", RFC 3424, November 2002.

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   [RFC3435]  Andreasen, F. and B. Foster, "Media Gateway Control
              Protocol (MGCP) Version 1.0", RFC 3435, January 2003.

   [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.

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

   [RFC5125]  Taylor, T., "Reclassification of RFC 3525 to Historic",
              RFC 5125, February 2008.

   [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.

   [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.

   [RFC6888]  Perreault, S., Yamagata, I., Miyakawa, S., Nakagawa, A.,
              and H. Ashida, "Common Requirements for Carrier-Grade NATs
              (CGNs)", BCP 127, RFC 6888, April 2013.

   [TCP-SPLICING]
              Maltz, D. and P. Bhagwat, "TCP Splice for application
              layer proxy performance", Journal of High Speed Networks
              Vol. 8, No. 3, 1999, pp. 225-240, March 1999.

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Authors' Addresses

   Emil Ivov
   Jitsi
   Strasbourg  67000
   France

   EMail: emcho@jitsi.org


   Hadriel Kaplan
   Oracle
   100 Crosby Drive
   Bedford, MA  01730
   USA

   EMail: hadrielk@yahoo.com


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

   EMail: dwing@cisco.com