Internet DRAFT - draft-ietf-mmusic-latching
draft-ietf-mmusic-latching
Network Working Group E. Ivov
Internet-Draft Jitsi
Intended status: Informational H. Kaplan
Expires: December 19, 2014 Oracle
D. Wing
Cisco
June 17, 2014
Latching: Hosted NAT Traversal (HNT) for Media in Real-Time
Communication
draft-ietf-mmusic-latching-08
Abstract
This document describes 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 IETF community, as well as an
informative description to manufacturers, and users.
Latching, which is one of the components 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 latching mechanism over the
Internet and recommends other solutions such as the Interactive
Connectivity Establishment (ICE) protocol.
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."
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This Internet-Draft will expire on December 19, 2014.
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, Latching . . . . . . . . . . . . . . . . . . 6
5. Security Considerations . . . . . . . . . . . . . . . . . . . 11
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 13
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 13
8. References . . . . . . . . . . . . . . . . . . . . . . . . . 13
8.1. Key References . . . . . . . . . . . . . . . . . . . . . 13
8.2. Additional References . . . . . . . . . . . . . . . . . . 14
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 15
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.
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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
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) and 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),
although a number of 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 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 [RFC5245] and Traversal Using Relays
around NAT (TURN) [RFC5766] protocols.
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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.
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 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], Media Gateway Control Protocol (MGCP) [RFC3435], H.248/
MEGACO [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] 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: 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, signalling and media would
either both travel over 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 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
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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, 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, 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 setup,
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", and thus this document does as well.
The latching mechanism works as follows:
1. After receiving an offer from Alice (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 (UAS) so that he would use
those media relay address:port tuples for all media it 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 it's the 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
tuple.
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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
User Agent Client (UAC) network and 198.51.100/24 facing towards the
User Agent Server (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
packet source information to prevent malicious attacks or latching
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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 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 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, 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, as described in [tcp-splicing].
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HNT and latching in particular are generally found to be working
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 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.
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, 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 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
surface, such configurations are generally performed only on a case-
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by-case basis so that the specifics of individual deployments would
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 them 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 since 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 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 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 this 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, 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. [RFC3261] Section 26 provides an overview
of additional threats and solutions on monitoring and session
interception.
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
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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 a middle box 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, 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,
neither S/MIME or [RFC4474] are widely deployed, thus not being able
to sign/verify requests appear 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. IANA Considerations
This document has no actions for IANA.
Note to the RFC-Editor: please remove this section prior to
publication as an RFC.
7. 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.
8. References
8.1. Key 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.
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[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", XEP XEP-0177,
December 2009.
8.2. Additional References
[H.323] International Telecommunication Union, "Packet Based
Multimedia Communication Systems", Recommendation H.323,
July 2003.
[RFC3424] Daigle, L. and IAB, "IAB Considerations for UNilateral
Self-Address Fixing (UNSAF) Across Network Address
Translation", RFC 3424, November 2002.
[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.
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Internet-Draft Hosted NAT Traversal for Media in RTC June 2014
[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. 235-240, March 1999.
Authors' Addresses
Emil Ivov
Jitsi
Strasbourg 67000
France
Email: emcho@jitsi.org
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Hadriel Kaplan
Oracle
100 Crosby Drive
Bedford, MA 01730
USA
Email: hadriel.kaplan@oracle.com
Dan Wing
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134
USA
Email: dwing@cisco.com
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