Internet DRAFT - draft-ivov-mmusic-latching
draft-ivov-mmusic-latching
Network Working Group E. Ivov
Internet-Draft Jitsi
Intended status: Informational H. Kaplan
Expires: April 26, 2013 Acme Packet
D. Wing
Cisco
October 23, 2012
Latching: Hosted NAT Traversal (HNT) for Media in Real-Time
Communication
draft-ivov-mmusic-latching-03
Abstract
This document describes behavior of signalling intermediaries in RTC
deployments, sometimes referred to as Session Border Controllers
(SBCs), when performing Hosted NAT Traversal (HNT). HNT is a set of
mechanisms, such as media relaying and latching, that such
intermediaries use to enable other RTC devices behind NATs to
communicate with each other. This document is non-normative, and is
only written to explain HNT in order to provide a reference to the
IETF community, as well as an informative description to
manufacturers, and users.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
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Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on April 26, 2013.
Copyright Notice
Copyright (c) 2012 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
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Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
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the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Background . . . . . . . . . . . . . . . . . . . . . . . . . . 4
4. Impact on Signaling . . . . . . . . . . . . . . . . . . . . . 5
5. Media Behavior, Latching . . . . . . . . . . . . . . . . . . . 6
6. Security Considerations . . . . . . . . . . . . . . . . . . . 11
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 13
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 13
8.1. Normative References . . . . . . . . . . . . . . . . . . . 13
8.2. Informative References . . . . . . . . . . . . . . . . . . 13
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 14
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1. Introduction
Network Address Translators (NATs) are widely used in the Internet by
consumers and organizations. Although specific NAT behaviors vary,
this document uses the term "NAT" for devices that map any IPv4 or
IPv6 address and transport port number to another IPv4 or IPv6
address and transport port number. This includes consumer NAPTs,
Firewall-NATs, IPv4-IPv6 NATs, Carrier-Grade NATs, etc.
Protocols like SIP [RFC3261], and others that try to use a more
direct path for media than with signalling, are difficult to use
across NATs. They use IP addresses and transport port numbers
encoded in bodies such as SDP [RFC4566]> as well as, in the case of
SIP, various header fields. Such addresses and ports are unusable
unless all peers in a session are located behind the same NAT.
Mechanisms such as STUN [RFC5389], TURN [RFC5766], and ICE [RFC5245],
did not exist when protocols like SIP began being deployed. Session
Border Controllers (SBCs) that were already being used by SIP domains
for other SIP and media-related purposes began to use proprietary
mechanisms to enable SIP devices behind NATs to communicate across
the NATs.
The term often used for this behavior is Hosted NAT Traversal (HNT),
although some manufacturers sometimes use other names such as "Far-
end NAT Traversal" or "NAT assist" instead. The systems which
perform HNT are frequently SBCs as described in [RFC5853], although
other systems such as media gateways and "media proxies" sometimes
perform the same role. For the purposes of this document, all such
systems are referred to as SBCs, and the NAT traversal behavior is
called HNT.
As of this document's creation time, a vast majority of SIP domains
use HNT to enable SIP devices to communicate across NATs, despite the
publication of ICE. There are many reasons for this, but those
reasons are not relevant to this document's purpose and will not be
discussed. It is however worth pointing out that the current
deployment levels of HNT and NATs themselves make an exclusive
adoption of ICE highly unlikely in the foreseeable future.
The purpose of this document is to describe the mechanisms often used
for HNT at the SDP and media layer, in order to aid understanding the
implications and limitations imposed by it. Although the mechanisms
used in HNT are not novel to experts, publication in an IETF document
is useful as a means of providing common terminology and a reference
for related documents.
In no way does this document try to make a case for HNT or present it
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as a solution that is somehow better than alternatives such as ICE.
The mechanisms described here, popular as they may be, are not
necessarily considered best practice or recommended operation.
It is also worth mentioning that there are purely signaling-layer
components of HNT as well. One such component is briefly described
for SIP in [RFC5853], but that is not the focus of this document.
The SIP signaling-layer component of HNT is typically more
implementation-specific and deployment-specific than the SDP and
media components. For the purposes of this document it is hence
assumed that signaling intermediaries handle traffic in way that
allows protocols such as SIP to function correctly across the NATs.
The rest of this document is going to focus primarily on use of HNT
for SIP. However, the mechanisms described here are relatively
generic and are often used with other protocols, such as XMPP
[RFC6120], MGCP, H.248/MEGACO, and H.323.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
3. Background
The general problems with NAT traversal for protocols such as SIP
are:
1. The addresses and port numbers encoded in SDP bodies (or their
equivalents) by NATed User Agents (UAs) are not usable across the
Internet, because they represent the private addressing
information of the UA rather than the addresses/ports that will
be mapped to/from by the NAT.
2. The policies inherent in NATs, and explicit in Firewalls, are
such that packets from outside the NAT cannot reach the UA until
the UA sends packet out first.
3. Some NATs apply endpoint dependent filtering on incoming packets,
as described in [RFC4787] and thus a UA may only be able to
receive packets from the same remote peer IP:port as it sends
packets out to.
In order to overcome these issues, signaling intermediaries such as
SIP SBCs on the public side of the NATs perform HNT for both
signaling and media. An example deployment model of HNT and SBCs is
shown in Figure 1.
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+-----+ +-----+
| SBC |-------| SBC |
+-----+ +-----+
/ \
/ Public Net \
/ \
+-----+ +-----+
|NAT-A| |NAT-B|
+-----+ +-----+
/ \
/ Private Net Private Net \
/ \
+------+ +------+
| UA-A | | UA-B |
+------+ +------+
Figure 1: Logical Deployment Paths
4. Impact on Signaling
Along with codec and other media-layer information, session
establishment signaling also conveys, potentially private and non-
globally routable addressing information. Signaling intermediaries
would hence modify such information so that peer UAs are given the
(public) addressing information of a media relay controlled by the
intermediary.
Quite often, the IP address of the newly introduced media relay may
be the same as that of the signaling intermediary (e.g. the SIP SBC)
or it may be a completely different one. In almost all cases
however, the new address would belong to the same IP address family
as the one used for signaling, since it is known to work for that UA.
The port numbers introduced in the signaling by the intermediary are
typically allocated dynamically. Allocation strategies are entirely
implementation dependent and they often vary from one product to the
next.
The offer/answer media negotiation model [RFC3264] is such that once
an offer is sent, the generator of the offer needs to be prepared to
receive media on the advertised address/ports. In practice such
media may or may not be received, depending on the implementations
participating in a given session, local policies, and call scenario.
For example if a SIP SDP Offer originally came from a UA behind a
NAT, the SIP SBC cannot send media to it until an SDP Answer is given
to the UA and latching (Section 5) occurs. Another example is when a
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SIP SBC sends an SDP Offer in a SIP INVITE to a residential
customer's UA and receives back SDP in a 18x response, the SBC may
decide not to send media to that customer UA until a SIP 200 response
for policy reasons, to prevent toll-fraud.
5. Media Behavior, Latching
An UA behind a NAT streams media from a private address:port set that
once packets cross the NAT, will be mapped to a public set. The UA
however is not typically aware of the public mapping and would often
advertise in the private address:port couple in signaling. This way,
when the signalling intermediary performing HNT receives the private
addressing information from the UA it will not know what address/
ports to send media to. Therefore media relays used in HNT would
often use a mechanism called "latching".
Historically, "latching" only referred to the process by which SBCs
"latch" onto UDP packets from a given UA for security purposes, and
"symmetric-latching" is when the latched address:ports are used to
send media back to the UA. Today most people talk about them both as
"latching", and thus this document does as well.
The latching mechanism works as follows:
1. After receiving an offer from a NATed UA, a signaling
intermediary located on the public Internet would allocate a set
of IP address:ports on a media relay. The set would then be
advertised to the remote party so that it would use it for all
media it wished to send toward the UA.
2. Next, after receiving an answer to its offer, the signaling
server would allocate a second address:port set on the media
relay. It would advertise this second set to the UA and use it
for all media traffic to and from the UA.
3. The media relay receives the media packets on the allocated
ports, and uses their source address and port as a destination
for all media bound in the opposite direction. In other words,
it "latches" or locks on these source address:port set.
4. This way all media streamed by the UA would be received on the
second address:port set. The source addresses and ports of the
traffic would belong to the public interface of the NAT in front
of the UA and anything that the relay sends there would find its
way to it.
5. Similarly the source of the stream originating at the remote
party would be latched upon and used for media going in that
direction.
6. Latching is usually done only once per peer and not allowed to
change or cause a re-latching until a new offer and answer get
exchanged (e.g. in a subsequent call or after a SIP peer has gone
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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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192.0.2.1 198.51.100.33
Alice NAT 203.0.113/24-SBC-198.51.100/24 Bob
------- --- --- -------
| | | |
1. |--SIP INVITE+offer c=192.0.2.1--->| |
| | | |
2. | | (SBC allocates 198.51.100.2/22007 |
| | for inbound RTP from UAS, |
| | and 203.0.113.4/36010 for |
| | inbound RTP from UAC) |
| | | |
3. | | |---INVITE+offer---->|
| | |c=198.51.100.2/22007|
| | | |
4. | | |<---180 Ringing-----|
| | | |
| | | |
5. |<------180 Ringing----------------| |
| | | |
6. | | |<---200+answer------|
7. |<-200+answer,c=203.0.113.4/36010--| c=198.51.100.33 |
| | | |
8. |------------ACK------------------>| |
9. | | |-------ACK--------->|
| | | |
10. |=====RTP,dest=203.0.113.4/36010==>| |
| | | |
11. | | (SBC latches to |
| | source IP address and |
| | port seen at (10)) |
| | | |
12. | | |<====== RTP ========|
| | | |
13. |<=====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
XMPP Client1 NAT XMPP Server XMPP Client2
------- --- --- -------
| | | |
1. |----session-initiate cand=192.0.2.1--->| |
| | | |
2. |<------------ack-----------------------| |
| | | |
3. | | (Server allocates 203.0.113.9/2200 |
| | for inbound RTP from Client2, |
| | and 203.0.113.9/3300 for |
| | inbound RTP from Client1) |
| | | |
4. | | |--session-initiate->|
| | cand=203.0.113.9/2200|
| | | |
5. | | |<-------ack---------|
| | | |
| | | |
6. | | |<--session-accept---|
| | | cand=198.51.100.8 |
| | | |
7. | | |--------ack-------> |
8. |<-session-accept cand=203.0.113.9/3300-| |
| | | |
9. |-----------------ack------------------>| |
| | | |
| | | |
10. |======RTP, dest=203.0.113.9/3300======>| |
| | | |
11. | | (XMPP server latches to |
| | src IP 203.0.113.4 and |
| | src port seen at (10)) |
| | | |
12. | | |<====== RTP ========|
| | | |
13. |<======RTP, to latched address=========| |
| | | |
Figure 3: Latcing by a SIP SBC across two interfaces
The above is a general description, and some details vary between
implementations or configuration settings. For example, some
intermediaries perform additional logic before latching on received
packet source information to prevent malicious attacks or latching
erroneously to previous media senders - often called "rogue-rtp" in
the industry.
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It is worth pointing out that latching is not an exclusively "server
affair" and some clients may also use it in cases where they are
configured with a public IP address and they are contacted by a NATed
client with no other NAT traversal means.
In order for latching to function correctly, the UA behind the NAT
needs to support symmetric RTP. That is, it needs to use the same
ports for sending data as the ones it listens on for inbound packets.
Today this is the case for with, for example, almost all SIP and XMPP
clients. Also UAs need to make sure they can begin sending media
packets independently and without waiting for packets to arrive
first. In theory, it is possible that some UAs would not send
packets out first; for example if a SIP session begins in 'inactive'
or 'recvonly' SDP mode from the UA behind the NAT. In practice,
however, SIP sessions from regular UAs (the kind that one could find
behind a NAT) virtually never begin in an inactive or 'recvonly'
mode, for obvious reasons. The media direction would also be
problematic if the SBC side indicated 'inactive' or 'sendonly' modes
when it sent SDP to the UA. However SBCs providing HNT would always
be configured to avoid this.
Given that, in order for latching to work properly, media relays need
to begin receiving media before they start sending, it is possible
for deadlocks to occur. This can happen when the UAC and the UAS in
a session are connected to different signalling intermediaries that
both provide HNT. In this case the media relays controlled by the
signalling servers could end up each waiting upon the other to
initiate the streaming. To prevent this relays would often attempt
to start streaming toward the address:port sets provided in the
offer/answer even before receiving any inbound traffic. If the
entity they are streaming to is another HNT performing server it
would have provided its relay's public address and ports and the
early stream would find its target.
Although many SBCs only support UDP-based media latching, and in
particular RTP/RTCP, many SBCs support TCP-based media latching as
well. TCP-based latching is more complicated, and involves forcing
the UA behind the NAT to be the TCP client and sending the initial
SYN-flagged TCP packet to the SBC (i.e., be the 'active' mode side of
a TCP-based media session). If both UAs of a TCP-based media session
are behind NATs, then SBCs typically force both UAs to be the TCP
clients, and the SBC splices the TCP connections together. TCP
splicing is a well-known technique, and described in [tcp-splicing].
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 members is that UAs are not aware of it occurring.
This makes it impossible for the mechanism to be used with protocols
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such as ICE that try various traversal techniques in an effort to
choose the one the best suits a particular situation. Overwriting
address information in in offers and answers may actually completely
prevent UAs from using ICE because of the ice-mismatch rules
described in [RFC5245]
The second issue raised by IETF members is that it causes media to go
through a relay instead of directly over the IP-routed path between
the two participating UAs. While this adds obvious drawbacks such as
reduced scalability and often increased latency, it is also
considered a benefit by SBC administrators: if a customer pays for
"phone" service, for example, the media is what is truly being paid
for, and the administrators usually like to be able to detect that
media is flowing correctly, evaluate its quality, know if and why it
failed, etc. Also in some cases routing media through operator
controlled relays may route media over paths explicitly optimized for
media and hence offer better performance than regular Internet
routing.
6. Security Considerations
A common concern is that an SBC that implements HNT may latch to
incorrect and possibly malicious sources. A malicious source could,
for example, attempt a resource exhaustion attack by flooding all
possible media-latching UDP ports on the SBC in order to prevent
calls from succeeding. SBCs have various mechanisms to prevent this
from happening, or alert an administrator 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
widen the gap for potential attackers, such configurations are
generally performed only on a case-by-case basis so that the
specifics of individual deployments would be taken into account.
In 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 disturb media from that IP (or
range of IP addresses) rather than all calls that the SBC is
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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 packets. There could still be an issue if the
attacker happens to be either (1) in the IP routing path and thus can
spoof the same IP as the real UA and get the media coming back, in
which case the attacker hardly needs to attack at all to begin with,
or (2) the attacker is behind the same NAT as the 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 end the call anyway, because a human user
would not hear anything and will hang 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 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 should
be used independently of HNT. For example, in cases where end-to-end
encryption is used it would still be possible for an attacker to
hijack a session despite the use of SRTP and perform a denial of
service attack. However, media integrity would not be compromised.
Additionally 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.
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],
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and it cannot sign a sending request or verify a received request
using [RFC4474] unless the SBC re-signs the request. However it is
sometimes argued that, neither S/MIME nor [RFC4474] are widely
deployed and that this may not be a real concern.
From a privacy perspective, media relaying is sometimes seen as a way
of protecting one's IP address and not revealing it to the remote
party. That kind of IP address masking is often perceived as
important. However, this is no longer an exclusive advantage of HNT
since it can also be accomplished by client-controlled relaying
mechanisms such as TURN [RFC5766], if the client explicitly wishes to
do so.
7. Acknowledgements
The authors would like to thank Flemming Andreasen and Miguel A.
Garcia for their reviews and suggestions on improving this document.
8. References
8.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
8.2. Informative References
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
with Session Description Protocol (SDP)", RFC 3264,
June 2002.
[RFC3489] Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy,
"STUN - Simple Traversal of User Datagram Protocol (UDP)
Through Network Address Translators (NATs)", RFC 3489,
March 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for
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Authenticated Identity Management in the Session
Initiation Protocol (SIP)", RFC 4474, August 2006.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4787] Audet, F. and C. Jennings, "Network Address Translation
(NAT) Behavioral Requirements for Unicast UDP", BCP 127,
RFC 4787, January 2007.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245,
April 2010.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
October 2008.
[RFC5751] Ramsdell, B. and S. Turner, "Secure/Multipurpose Internet
Mail Extensions (S/MIME) Version 3.2 Message
Specification", RFC 5751, January 2010.
[RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
Relays around NAT (TURN): Relay Extensions to Session
Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.
[RFC5853] Hautakorpi, J., Camarillo, G., Penfield, R., Hawrylyshen,
A., and M. Bhatia, "Requirements from Session Initiation
Protocol (SIP) Session Border Control (SBC) Deployments",
RFC 5853, April 2010.
[RFC6120] Saint-Andre, P., "Extensible Messaging and Presence
Protocol (XMPP): Core", RFC 6120, March 2011.
[RFC6189] Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media
Path Key Agreement for Unicast Secure RTP", RFC 6189,
April 2011.
[XEP-0177]
Beda, J., Saint-Andre, P., Hildebrand, J., and S. Egan,
"XEP-0177: Jingle Raw UDP Transport Method", XEP XEP-0177,
December 2009.
Ivov, et al. Expires April 26, 2013 [Page 14]
Internet-Draft Hosted NAT Traversal for Media in RTC October 2012
Authors' Addresses
Emil Ivov
Jitsi
Strasbourg 67000
France
Email: emcho@jitsi.org
Hadriel Kaplan
Acme Packet
100 Crosby Drive
Bedford, MA 01730
USA
Email: hkaplan@acmepacket.com
Dan Wing
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134
USA
Email: dwing@cisco.com
Ivov, et al. Expires April 26, 2013 [Page 15]
