Latching: Hosted NAT Traversal (HNT) for Media in Real-Time Communication
RFC 7362
| Document | Type | RFC - Informational (September 2014) | |
|---|---|---|---|
| Authors | Emil Ivov , Hadriel Kaplan , Dan Wing | ||
| Last updated | 2018-12-20 | ||
| Replaces | draft-ivov-mmusic-latching | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Formats | |||
| Reviews |
GENART Last Call review
(of
-05)
by Elwyn Davies
Ready w/nits
|
||
| Additional resources | Mailing list discussion | ||
| Stream | WG state | Submitted to IESG for Publication | |
| Document shepherd | Ari Keränen | ||
| Shepherd write-up | Show Last changed 2014-02-11 | ||
| IESG | IESG state | RFC 7362 (Informational) | |
| Action Holders |
(None)
|
||
| Consensus boilerplate | Yes | ||
| Telechat date | (None) | ||
| Responsible AD | Alissa Cooper | ||
| Send notices to | (None) | ||
| IANA | IANA review state | Version Changed - Review Needed | |
| IANA action state | No IANA Actions |
RFC 7362
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.
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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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RFC 7362 Hosted NAT Traversal for Media in RTC September 2014
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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RFC 7362 Hosted NAT Traversal for Media in RTC September 2014
[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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RFC 7362 Hosted NAT Traversal for Media in RTC September 2014
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
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