Network Working Group M. Westerlund
Internet-Draft Ericsson
Intended status: Informational T. Zeng
Expires: November 29, 2014
May 28, 2014
The Evaluation of Different Network Address Translator (NAT) Traversal
Techniques for Media Controlled by Real-time Streaming Protocol (RTSP)
draft-ietf-mmusic-rtsp-nat-evaluation-14
Abstract
This document describes several Network Address Translator (NAT)
traversal techniques that were considered to be used for establishing
the RTP media flows controlled by the Real-time Streaming Protocol
(RTSP). Each technique includes a description of how it would be
used, the security implications of using it and any other deployment
considerations it has. There are also discussions on how NAT
traversal techniques relate to firewalls and how each technique can
be applied in different use cases. These findings were used when
selecting the NAT traversal for RTSP 2.0, which is specified in a
separate document.
Status of This Memo
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provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
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time. It is inappropriate to use Internet-Drafts as reference
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This Internet-Draft will expire on November 29, 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
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Network Address Translators . . . . . . . . . . . . . . . 4
1.2. Firewalls . . . . . . . . . . . . . . . . . . . . . . . . 6
1.3. Glossary . . . . . . . . . . . . . . . . . . . . . . . . 6
2. Detecting the loss of NAT mappings . . . . . . . . . . . . . 7
3. Requirements on Solutions . . . . . . . . . . . . . . . . . . 8
4. NAT Traversal Techniques . . . . . . . . . . . . . . . . . . 9
4.1. Stand-Alone STUN . . . . . . . . . . . . . . . . . . . . 10
4.1.1. Introduction . . . . . . . . . . . . . . . . . . . . 10
4.1.2. Using STUN to traverse NAT without server
modifications . . . . . . . . . . . . . . . . . . . . 10
4.1.3. ALG considerations . . . . . . . . . . . . . . . . . 12
4.1.4. Deployment Considerations . . . . . . . . . . . . . . 13
4.1.5. Security Considerations . . . . . . . . . . . . . . . 14
4.2. Server Embedded STUN . . . . . . . . . . . . . . . . . . 14
4.2.1. Introduction . . . . . . . . . . . . . . . . . . . . 15
4.2.2. Embedding STUN in RTSP . . . . . . . . . . . . . . . 15
4.2.3. Discussion On Co-located STUN Server . . . . . . . . 16
4.2.4. ALG considerations . . . . . . . . . . . . . . . . . 16
4.2.5. Deployment Considerations . . . . . . . . . . . . . . 16
4.2.6. Security Considerations . . . . . . . . . . . . . . . 18
4.3. ICE . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
4.3.1. Introduction . . . . . . . . . . . . . . . . . . . . 18
4.3.2. Using ICE in RTSP . . . . . . . . . . . . . . . . . . 18
4.3.3. Implementation burden of ICE . . . . . . . . . . . . 20
4.3.4. Deployment Considerations . . . . . . . . . . . . . . 21
4.3.5. Security Consideration . . . . . . . . . . . . . . . 21
4.4. Latching . . . . . . . . . . . . . . . . . . . . . . . . 21
4.4.1. Introduction . . . . . . . . . . . . . . . . . . . . 21
4.4.2. Necessary RTSP extensions . . . . . . . . . . . . . . 22
4.4.3. Deployment Considerations . . . . . . . . . . . . . . 23
4.4.4. Security Consideration . . . . . . . . . . . . . . . 23
4.5. A Variation to Latching . . . . . . . . . . . . . . . . . 25
4.5.1. Introduction . . . . . . . . . . . . . . . . . . . . 25
4.5.2. Necessary RTSP extensions . . . . . . . . . . . . . . 25
4.5.3. Deployment Considerations . . . . . . . . . . . . . . 26
4.5.4. Security Considerations . . . . . . . . . . . . . . . 26
4.6. Three Way Latching . . . . . . . . . . . . . . . . . . . 27
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4.6.1. Introduction . . . . . . . . . . . . . . . . . . . . 27
4.6.2. Necessary RTSP extensions . . . . . . . . . . . . . . 27
4.6.3. Deployment Considerations . . . . . . . . . . . . . . 27
4.7. Application Level Gateways . . . . . . . . . . . . . . . 28
4.7.1. Introduction . . . . . . . . . . . . . . . . . . . . 28
4.7.2. Outline On how ALGs for RTSP work . . . . . . . . . . 28
4.7.3. Deployment Considerations . . . . . . . . . . . . . . 29
4.7.4. Security Considerations . . . . . . . . . . . . . . . 30
4.8. TCP Tunneling . . . . . . . . . . . . . . . . . . . . . . 30
4.8.1. Introduction . . . . . . . . . . . . . . . . . . . . 30
4.8.2. Usage of TCP tunneling in RTSP . . . . . . . . . . . 31
4.8.3. Deployment Considerations . . . . . . . . . . . . . . 31
4.8.4. Security Considerations . . . . . . . . . . . . . . . 31
4.9. TURN (Traversal Using Relay NAT) . . . . . . . . . . . . 31
4.9.1. Introduction . . . . . . . . . . . . . . . . . . . . 32
4.9.2. Usage of TURN with RTSP . . . . . . . . . . . . . . . 32
4.9.3. Deployment Considerations . . . . . . . . . . . . . . 33
4.9.4. Security Considerations . . . . . . . . . . . . . . . 34
5. Firewalls . . . . . . . . . . . . . . . . . . . . . . . . . . 34
6. Comparison of NAT traversal techniques . . . . . . . . . . . 35
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 37
8. Security Considerations . . . . . . . . . . . . . . . . . . . 37
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 38
10. Informative References . . . . . . . . . . . . . . . . . . . 38
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 40
1. Introduction
Today there is a proliferate deployment of different flavors of
Network Address Translator (NAT) boxes that in many cases only
loosely follow standards
[RFC3022][RFC2663][RFC3424][RFC4787][RFC5382]. NATs cause
discontinuity in address realms [RFC3424], therefore an application
protocol, such as Real-time Streaming Protocol (RTSP)
[RFC2326][I-D.ietf-mmusic-rfc2326bis], needs to deal with such
discontinuities caused by NATs. The problem is that, being a media
control protocol managing one or more media streams, RTSP carries
network address and port information within its protocol messages.
Because of this, even if RTSP itself, when carried over Transmission
Control Protocol (TCP) [RFC0793] for example, is not blocked by NATs,
its media streams may be blocked by NATs. This will occur unless
special protocol provisions are added to support NAT-traversal.
Like NATs, firewalls are also middle boxes that need to be
considered. Firewalls help prevent unwanted traffic from getting in
or out of the protected network. RTSP is designed such that a
firewall can be configured to let RTSP controlled media streams go
through with minimal implementation effort. The minimal effort is to
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implement an Application Level Gateway (ALG) to interpret RTSP
parameters. There is also a large class of firewalls, commonly home
firewalls, that uses a similar filtering behavior to what NAT has.
This type of firewalls can be handled using the same solution as
employed for NAT traversal instead of relying on ALGs.
This document describes several NAT-traversal mechanisms for RTSP
controlled media streaming. Many of these NAT solutions fall into
the category of "UNilateral Self-Address Fixing (UNSAF)" as defined
in [RFC3424] and quoted below:
"UNSAF is a process whereby some originating process attempts to
determine or fix the address (and port) by which it is known - e.g.
to be able to use address data in the protocol exchange, or to
advertise a public address from which it will receive connections."
Following the guidelines spelled out in RFC 3424, we describe the
required RTSP protocol extensions for each method, transition
strategies, and security concerns.
This document is capturing the evaluation done in the process to
recommend firewall/NAT traversal methods for RTSP streaming servers
based on RFC 2326 [RFC2326] as well as the RTSP 2.0 core spec
[I-D.ietf-mmusic-rfc2326bis]. The evaluation is focused on NAT
traversal for the media streams carried over User Datagram Protocol
(UDP) [RFC0768] with Real-time Transport Protocol (RTP) [RFC3550]
over UDP being the main case for such usage. The findings should be
applicable to other protocols as long as they have similar
properties.
At the time when the bulk of work on this document was done, a single
level of NAT was the dominant deployment for NATs, and multiple level
of NATs, including Carrier Grade NATs (CGNs) has been only partially
considered.
The resulting ICE-based RTSP NAT traversal mechanism is specified in
"A Network Address Translator (NAT) Traversal mechanism for media
controlled by Real-Time Streaming Protocol (RTSP)"
[I-D.ietf-mmusic-rtsp-nat].
1.1. Network Address Translators
We begin by reviewing two quotes from Section 3 in "Network Address
Translation (NAT) Behavioral Requirements for Unicast UDP" [RFC4787]
concering NATs and their terminology:
"Readers are urged to refer to [RFC2663] for information on NAT
taxonomy and terminology. Traditional NAT is the most common type of
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NAT device deployed. Readers may refer to [RFC3022] for detailed
information on traditional NAT. Traditional NAT has two main
varieties -- Basic NAT and Network Address/Port Translator (NAPT).
NAPT is by far the most commonly deployed NAT device. NAPT allows
multiple internal hosts to share a single public IP address
simultaneously. When an internal host opens an outgoing TCP or UDP
session through a NAPT, the NAPT assigns the session a public IP
address and port number, so that subsequent response packets from the
external endpoint can be received by the NAPT, translated, and
forwarded to the internal host. The effect is that the NAPT
establishes a NAT session to translate the (private IP address,
private port number) tuple to a (public IP address, public port
number) tuple, and vice versa, for the duration of the session. An
issue of relevance to peer-to-peer applications is how the NAT
behaves when an internal host initiates multiple simultaneous
sessions from a single (private IP, private port) endpoint to
multiple distinct endpoints on the external network. In this
specification, the term "NAT" refers to both "Basic NAT" and "Network
Address/Port Translator (NAPT)"."
"This document uses the term "address and port mapping" as the
translation between an external address and port and an internal
address and port. Note that this is not the same as an "address
binding" as defined in RFC 2663."
Note: In the above it would be more correct to use external
instead of public in the above text. The external IP address is
commonly a public one, but might be of other type if the NAT's
external side is in a private address domain.
In addition to the above quote there exists a number of address and
port mapping behaviors described in more detail in Section 4.1 of
"Network Address Translation (NAT) Behavioral Requirements for
Unicast UDP" [RFC4787] that are highly relevant to the discussion in
this document.
NATs also have a filtering behavior on traffic arriving on the
external side. Such behavior affects how well different methods for
NAT traversal works through these NATs. See Section 5 of "Network
Address Translation (NAT) Behavioral Requirements for Unicast UDP"
[RFC4787] for more information on the different types of filtering
that have been identified.
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1.2. Firewalls
A firewall is a security gateway that enforces certain access control
policies between two network administrative domains: a private domain
(intranet) and a external domain, e.g. Internet. Many organizations
use firewalls to prevent privacy intrusions and malicious attacks to
corporate computing resources in the private intranet [RFC2588].
A comparison between NAT and firewall is given below:
1. A firewall sits at security enforcement/protection points, while
NAT sits at borders between two address domains.
2. NAT does not in itself provide security, although some access
control policies can be implemented using address translation
schemes. The inherent filtering behaviours are commonly mistaken
for real security policies.
It should be noted that many NAT devices intended for Residential or
small office/home office (SOHO) use include both NATs and firewall
functionality.
In the rest of this memo we use the phrase "NAT traversal"
interchangeably with "firewall traversal", and "NAT/firewall
traversal".
1.3. Glossary
Address-Dependent Mapping: The NAT reuses the port mapping for
subsequent packets sent from the same internal IP address and
port to the same external IP address, regardless of the
external port. See [RFC4787].
Address and Port-Dependent Mapping: The NAT reuses the port mapping
for subsequent packets sent from the same internal IP address
and port to the same external IP address and port while the
mapping is still active. See [RFC4787].
ALG: Application Level Gateway, an entity that can be embedded in a
NAT or other middlebox to perform the application layer
functions required for a particular protocol to traverse the
NAT/middlebox.
Endpoint-Independent Mapping: The NAT reuses the port mapping for
subsequent packets sent from the same internal IP address and
port to any external IP address and port. See [RFC4787].
ICE: Interactive Connectivity Establishment, see [RFC5245].
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DNS: Domain Name Service
DoS: Denial of Service
DDoS: Distributed Denial of Service
NAT: Network Address Translator, see [RFC3022].
NAPT: Network Address/Port Translator, see [RFC3022].
RTP: Real-time Transport Protocol, see [RFC3550].
RTSP: Real-Time Streaming Protocol, see [RFC2326] and
[I-D.ietf-mmusic-rfc2326bis].
RTT: Round Trip Times.
SDP: Session Description Protocol, see [RFC4566].
SSRC: Synchronization source in RTP, see [RFC3550].
2. Detecting the loss of NAT mappings
Several NAT traversal techniques in the next chapter make use of the
fact that the NAT UDP mapping's external address and port can be
discovered. This information is then utilized to traverse the NAT
box. However any such information is only good while the mapping is
still valid. As the IAB's UNSAF document [RFC3424] points out, the
mapping can either timeout or change its properties. It is therefore
important for the NAT traversal solutions to handle the loss or
change of NAT mappings, according to RFC3424.
First, since NATs may also dynamically reclaim or readjust address/
port translations, "keep-alive" and periodic re-polling may be
required according to RFC 3424. Secondly, it is possible to detect
and recover from the situation where the mapping has been changed or
removed. The loss of a mapping can be detected when no traffic
arrives for a while. Below we will give some recommendation on how
to detect loss of NAT mappings when using RTP/RTCP under RTSP
control.
A RTP session normally has both RTP and RTCP streams. The loss of a
RTP mapping can only be detected when expected traffic does not
arrive. If a client does not receive data within a few seconds after
having received the "200 OK" response to a PLAY request, it may be
the result of a middlebox blocking the traffic. However, for a
receiver to be more certain to detect the case where no RTP traffic
was delivered due to NAT trouble, one should monitor the RTCP Sender
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reports if they are received and not also blocked. The sender report
carries a field telling how many packets the server has sent. If
that has increased and no RTP packets has arrived for a few seconds
it is likely the RTP mapping has been removed.
The loss of mapping for RTCP is simpler to detect. RTCP is normally
sent periodically in each direction, even during the RTSP ready
state. If RTCP packets are missing for several RTCP intervals, the
mapping is likely lost. Note that if neither RTCP packets nor RTSP
messages are received by the RTSP server for a while, the RTSP server
has the option to delete the corresponding RTP session, SSRC and RTSP
session ID, because either the client can not get through a middle
box NAT/firewall, or the client is mal-functioning.
3. Requirements on Solutions
This section considers the set of requirements for the evaluation of
RTSP NAT traversal solutions.
RTSP is a client-server protocol. Typically service providers deploy
RTSP servers on the Internet or otherwise reachable address realm.
However, there are use cases where the reverse is true: RTSP clients
are connecting from any address realm to RTSP servers behind NATs,
e.g. in a home. This is the case for instance when home surveillance
cameras running as RTSP servers intend to stream video to cell phone
users in the public address realm through a home NAT. In terms of
requirements, the primary requirement should be to solve the RTSP NAT
traversal problem for RTSP servers deployed in a network where the
server is on the external side of any NAT, i.e. server is not behind
a NAT.
The list of feature requirements for RTSP NAT solutions are given
below:
1. Must work for all flavors of NATs, including NATs with Address
and Port-Dependent Filtering.
2. Must work for firewalls (subject to pertinent firewall
administrative policies), including those with ALGs.
3. Should have minimal impact on clients not behind NATs and which
are not dual-hosted. RTSP dual-hosting means that the RTSP
signalling protocol and the media protocol (e.g. RTP) are
implemented on different computers with different IP addresses.
* For instance, no extra delay from RTSP connection till arrival
of media
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4. Should be simple to use/implement/administer so people actually
turn them on
* Otherwise people will resort to TCP tunneling through NATs
* Discovery of the address(es) assigned by NAT should happen
automatically, if possible
5. Should authenticate dual-hosted client transport handler to
prevent DDoS attacks.
The last requirement addresses the Distributed Denial-of-Service
(DDoS) threat, which relates to NAT traversal as explained below.
During NAT traversal, when the RTSP server determines the media
destination (address and port) for the client, the result may be that
the IP address of the RTP receiver host is different than the IP
address of the RTSP client host. This posts a DDoS threat that has
significant amplification potentials because the RTP media streams in
general consist of large number of IP packets. DDoS attacks occur if
the attacker fakes the messages in the NAT traversal mechanism to
trick the RTSP server into believing that the client's RTP receiver
is located on a separate host. For example, user A may use his RTSP
client to direct the RTSP server to send video RTP streams to
target.example.com in order to degrade the services provided by
target.example.com. Note a simple preventative measure commonly
deployed is for the RTSP server to disallow the cases where the
client's RTP receiver has a different IP address than that of the
RTSP client. With the increased deployment of NAT middleboxes by
operators, i.e. carrier grade NAT (CGN), the reuse of an IP address
on the NAT's external side by many customers reduces the protection
provided. Also in some applications (e.g., centralized
conferencing), dual-hosted RTSP/RTP clients have valid use cases.
The key is how to authenticate the messages exchanged during the NAT
traversal process.
4. NAT Traversal Techniques
There exists a number of potential NAT traversal techniques that can
be used to allow RTSP to traverse NATs. They have different features
and are applicable to different topologies; their costs are also
different. They also vary in security levels. In the following
sections, each technique is outlined with discussions on the
corresponding advantages and disadvantages.
The main evaluation was done prior to 2007 and is based on what was
available then. This section includes NAT traversal techniques that
have not been formally specified anywhere else. The overview section
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of this document may be the only publicly available specification of
some of the NAT traversal techniques. However that is not a real
barrier against doing an evaluation of the NAT traversal techniques.
Some other techniques have been recommended against or are no longer
possible due to standardization works' outcome or their failure to
progress within IETF after the initial evaluation in this document,
e.g. RTP No-Op [I-D.ietf-avt-rtp-no-op].
4.1. Stand-Alone STUN
4.1.1. Introduction
Session Traversal Utilities for NAT (STUN) [RFC5389] is a
standardized protocol that allows a client to use secure means to
discover the presence of a NAT between itself and the STUN server.
The client uses the STUN server to discover the address mappings
assigned by the NAT. STUN is a client-server protocol. The STUN
client sends a request to a STUN server and the server returns a
response. There are two types of STUN messages - Binding Requests
and Indications. Binding requests are used when determining a
client's external address and solicits a response from the STUN
server with the seen address.
The first version of STUN [RFC3489] included categorization and
parameterization of NATs. This was abandoned in the updated version
[RFC5389] due to it being unreliable and brittle. Some of the below
discussed methods are based on RFC3489 functionality which will be
called out and the downside of that will be part of the
characterization.
4.1.2. Using STUN to traverse NAT without server modifications
This section describes how a client can use STUN to traverse NATs to
RTSP servers without requiring server modifications. Note that this
method has limited applicability and requires the server to be
available in the external/public address realm in regards to the
client located behind a NAT(s).
Limitations:
o The server must be located in either a public address realm or the
next hop external address realm in regards to the client.
o The client may only be located behind NATs that perform "Endpoint-
Independent" or "Address-Dependent" Mappings. Clients behind NATs
that do "Address and Port-Dependent" Mappings cannot use this
method. See [RFC4787] for full definition of these terms.
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o Based on the discontinued middlebox classification of the replaced
STUN specification [RFC3489]. Thus brittle and unreliable.
Method:
A RTSP client using RTP transport over UDP can use STUN to traverse a
NAT(s) in the following way:
1. Use STUN to try to discover the type of NAT, and the timeout
period for any UDP mapping on the NAT. This is recommended to be
performed in the background as soon as IP connectivity is
established. If this is performed prior to establishing a
streaming session the delays in the session establishment will be
reduced. If no NAT is detected, normal SETUP should be used.
2. The RTSP client determines the number of UDP ports needed by
counting the number of needed media transport protocols sessions
in the multi-media presentation. This information is available
in the media description protocol, e.g. SDP [RFC4566]. For
example, each RTP session will in general require two UDP ports,
one for RTP, and one for RTCP.
3. For each UDP port required, establish a mapping and discover the
public/external IP address and port number with the help of the
STUN server. A successful mapping looks like: client's local
address/port <-> public address/port.
4. Perform the RTSP SETUP for each media. In the transport header
the following parameter should be included with the given values:
"dest_addr" [I-D.ietf-mmusic-rfc2326bis] or "destination" +
"client_port" [RFC2326] with the public/external IP address and
port pair for both RTP and RTCP. To be certain that this works
servers must allow a client to setup the RTP stream on any port,
not only even ports and with non-contiguous port numbers for RTP
and RTCP. This requires the new feature provided in the update
to RFC2326 [I-D.ietf-mmusic-rfc2326bis]. The server should
respond with a transport header containing an "src_addr" or
"source" + "server_port" parameters with the RTP and RTCP source
IP address and port of the media stream.
5. To keep the mappings alive, the client should periodically send
UDP traffic over all mappings needed for the session. For the
mapping carrying RTCP traffic the periodic RTCP traffic are
likely enough. For mappings carrying RTP traffic and for
mappings carrying RTCP packets at too low a frequency, keep-alive
messages should be sent. As keep alive messages, one could use
the RTP No-Op packet [I-D.ietf-avt-rtp-no-op] to the streaming
server's discard port (port number 9). The drawback of using RTP
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No-Op is that the payload type number must be dynamically
assigned through RTSP first. Otherwise STUN could be used for
the keep-alive as well as empty UDP packets.
If a UDP mapping is lost, the above discovery process must be
repeated. The media stream also needs to be SETUP again to change
the transport parameters to the new ones. This will cause a glitch
in media playback.
To allow UDP packets to arrive from the server to a client behind a
"Address Dependent" filtering NAT, the client must first send a UDP
packet to establish filtering state in the NAT. The client, before
sending a RTSP PLAY request, must send a so called hole-punching
packet (such as a RTP No-Op packet) on each mapping, to the IP
address given as the servers source address. To create minimum
problems for the server these UDP packets should be sent to the
server's discard port (port number 9). Since UDP packets are
inherently unreliable, to ensure that at least one UDP message passes
the NAT, hole-punching packets should be retransmitted a reasonable
number of times.
For an "Address and Port Dependent" filtering NAT the client must
send messages to the exact ports used by the server to send UDP
packets before sending a RTSP PLAY request. This makes it possible
to use the above described process with the following additional
restrictions: for each port mapping, hole-punching packets need to be
sent first to the server's source address/port. To minimize
potential effects on the server from these messages the following
type of hole punching packets must be sent. RTP: an empty or less
than 12 bytes UDP packet. RTCP: A correctly formatted RTCP RR or SR
message. The above described adaptations for restricted NATs will
not work unless the server includes the "src_addr" in the "Transport"
header (which is the "source" transport parameter in RFC2326).
This method is brittle because it assumes one can use STUN to
classify the NAT behavior, which was found to be problematic
[RFC5389]. If the NAT changes the properties of the existing mapping
and filtering state for example due to load, then the methods will
fail.
4.1.3. ALG considerations
If a NAT supports RTSP ALG (Application Level Gateway) and is not
aware of the STUN traversal option, service failure may happen,
because a client discovers its NAT external IP address and port
numbers, and inserts them in its SETUP requests. When the RTSP ALG
processes the SETUP request it may change the destination and port
number, resulting in unpredictable behavior. An ALG should not
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update address fields which contains addresses other than the NATs
internal address domain. In cases where the ALG modifies fields
unnecessarily two alternatives exist:
1. Use TLS to encrypt the RTSP TCP connection to prevent the ALG
from reading and modifying the RTSP messages.
2. Turn off the STUN based NAT traversal mechanism
As it may be difficult to determine why the failure occurs, the usage
of TLS protected RTSP message exchange at all times would avoid this
issue.
4.1.4. Deployment Considerations
For the Stand-Alone usage of STUN the following applies:
Advantages:
o STUN is a solution first used by SIP [RFC3261] based applications
(See section 1 and 2 of [RFC5389]). As shown above, with little
or no changes, the RTSP application can re-use STUN as a NAT
traversal solution, avoiding the pit-fall of solving a problem
twice.
o Using STUN does not require RTSP server modifications, assuming it
is a RTSP 2.0 compliant server; it only affects the client
implementation.
Disadvantages:
o Requires a STUN server deployed in the same address domain as the
server.
o Only works with NATs that perform endpoint independent and address
dependent mappings. Address and Port-Dependent filtering NATs
create some issues.
o Brittle to NATs changing the properties of the NAT mapping and
filtering.
o Does not work with port and address dependent mapping NATs without
server modifications.
o Will mostly not work if a NAT uses multiple IP addresses, since
RTSP servers generally require all media streams to use the same
IP as used in the RTSP connection to prevent becoming a DDoS tool.
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o Interaction problems exist when a RTSP-aware ALG interferes with
the use of STUN for NAT traversal unless TLS secured RTSP message
exchange is used.
o Using STUN requires that RTSP servers and clients support the
updated RTSP specification [I-D.ietf-mmusic-rfc2326bis], because
it is no longer possible to guarantee that RTP and RTCP ports are
adjacent to each other, as required by the "client_port" and
"server_port" parameters in RFC2326.
Transition:
The usage of STUN can be phased out gradually as the first step of a
STUN capable server or client should be to check the presence of
NATs. The removal of STUN capability in the client implementations
will have to wait until there is absolutely no need to use STUN.
4.1.5. Security Considerations
To prevent the RTSP server from being used as Denial of Service (DoS)
attack tools the RTSP Transport header parameter "destination" and
"dest_addr" are generally not allowed to point to any IP address
other than the one the RTSP message originates from. The RTSP server
is only prepared to make an exception to this rule when the client is
trusted (e.g., through the use of a secure authentication process, or
through some secure method of challenging the destination to verify
its willingness to accept the RTP traffic). Such a restriction means
that STUN in general does not work for use cases where RTSP and media
transport go to different addresses.
STUN combined with destination address restricted RTSP has the same
security properties as the core RTSP. It is protected from being
used as a DoS attack tool unless the attacker has the ability to
spoof the TCP connection carrying RTSP messages.
Using STUN's support for message authentication and secure transport
of RTSP messages, attackers cannot modify STUN responses or RTSP
messages (TLS) to change media destination. This protects against
hijacking, however as a client can be the initiator of an attack,
these mechanisms cannot securely prevent RTSP servers being used as
DoS attack tools.
4.2. Server Embedded STUN
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4.2.1. Introduction
This Section describes an alternative to the stand-alone STUN usage
in the previous section that has quite significantly different
behavior.
4.2.2. Embedding STUN in RTSP
This section outlines the adaptation and embedding of STUN within
RTSP. This enables STUN to be used to traverse any type of NAT,
including address and Port-Dependent mapping NATs. This would
require RTSP level protocol changes.
This NAT traversal solution has limitations:
1. It does not work if both RTSP client and RTSP server are behind
separate NATs.
2. The RTSP server may, for security reasons, refuse to send media
streams to an IP different from the IP in the client RTSP
requests.
Deviations from STUN as defined in RFC 5389:
1. The RTSP application must provision the client with an identity
and shared secret to use in the STUN authentication;
2. We require STUN server to be co-located on RTSP server's media
source ports.
If STUN server is co-located with RTSP server's media source port, an
RTSP client using RTP transport over UDP can use STUN to traverse ALL
types of NATs. In the case of port and address dependent mapping
NATs, the party on the inside of the NAT must initiate UDP traffic.
The STUN Binding Request, being a UDP packet itself, can serve as the
traffic initiating packet. Subsequently, both the STUN Binding
Response packets and the RTP/RTCP packets can traverse the NAT,
regardless of whether the RTSP server or the RTSP client is behind
NAT (however only one of the can be behind a NAT).
Likewise, if an RTSP server is behind a NAT, then an embedded STUN
server must be co-located on the RTSP client's RTCP port. Also it
will become the client that needs to disclose his destination address
rather than the server, so the server can correctly determine its NAT
external source address for the media streams. In this case, we
assume that the client has some means of establishing TCP connection
to the RTSP server behind NAT so as to exchange RTSP messages with
the RTSP server, potentially using a proxy or static rules.
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To minimize delay, we require that the RTSP server supporting this
option must inform the client about the RTP and RTCP ports from where
the server will send out RTP and RTCP packets, respectively. This
can be done by using the "server_port" parameter in RFC2326, and the
"src_addr" parameter in [I-D.ietf-mmusic-rfc2326bis]. Both are in
the RTSP Transport header. But in general this strategy will require
that one first do one SETUP request per media to learn the server
ports, then perform the STUN checks, followed by a subsequent SETUP
to change the client port and destination address to what was learned
during the STUN checks.
To be certain that RTCP works correctly the RTSP end-point (server or
client) will be required to send and receive RTCP packets from the
same port.
4.2.3. Discussion On Co-located STUN Server
In order to use STUN to traverse "address and port dependent"
filtering or mapping NATs the STUN server needs to be co-located with
the streaming server media output ports. This creates a de-
multiplexing problem: we must be able to differentiate a STUN packet
from a media packet. This will be done based on heuristics. The
existing STUN heuristics is the first byte in the packet and the
Magic Cookie field (added in RFC5389), which works fine between STUN
and RTP or RTCP where the first byte happens to be different. Thanks
to the magic cookie field it is unlikely that other protocols would
be mistaken for a STUN packet, but not assured.
4.2.4. ALG considerations
The same ALG traversal considerations as for Stand-Alone STUN applies
(Section 4.1.3).
4.2.5. Deployment Considerations
For the "Embedded STUN" method the following applies:
Advantages:
o STUN is a solution first used by SIP applications. As shown
above, with little or no changes, RTSP application can re-use STUN
as a NAT traversal solution, avoiding the pit-fall of solving a
problem twice.
o STUN has built-in message authentication features, which makes it
more secure against hi-jacking attacks. See next section for an
in-depth security discussion.
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o This solution works as long as there is only one RTSP endpoint in
the private address realm, regardless of the NAT's type. There
may even be multiple NATs (see Figure 1 in [RFC5389]).
o Compared to other UDP based NAT traversal methods in this
document, STUN requires little new protocol development (since
STUN is already a IETF standard), and most likely less
implementation effort, since open source STUN server and client
implementations are available [STUN-IMPL][PJNATH]. There is the
need to embed STUN in RTSP server and client, which require a de-
multiplexer between STUN packets and RTP/RTCP packets. There is
also a need to register the proper feature tags.
Disadvantages:
o Some extensions to the RTSP core protocol, likely signaled by RTSP
feature tags, must be introduced.
o Requires an embedded STUN server to be co-located on each of the
RTSP server's media protocol's ports (e.g. RTP and RTCP ports),
which means more processing is required to de-multiplex STUN
packets from media packets. For example, the de-multiplexer must
be able to differentiate a RTCP RR packet from a STUN packet, and
forward the former to the streaming server, and the latter to the
STUN server.
o Does not support use cases that require the RTSP connection and
the media reception to happen at different addresses, unless the
server's security policy is relaxed.
o Interaction problems exist when a RTSP ALG is not aware of STUN
unless TLS is used to protect the RTSP messages.
o Using STUN requires that RTSP servers and clients support the
updated RTSP specification [I-D.ietf-mmusic-rfc2326bis], and they
both agree to support the NAT traversal feature.
o Increases the setup delay with at least the amount of time it
takes to perform STUN message exchanges. Most likely an extra
SETUP sequence will be required.
Transition:
The usage of STUN can be phased out gradually as the first step of a
STUN capable machine can be to check the presence of NATs for the
presently used network connection. The removal of STUN capability in
the client implementations will have to wait until there is
absolutely no need to use STUN, i.e. no NATs or firewalls.
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4.2.6. Security Considerations
See Stand-Alone STUN (Section 4.1.5).
4.3. ICE
4.3.1. Introduction
ICE (Interactive Connectivity Establishment) [RFC5245] is a
methodology for NAT traversal that has been developed for SIP using
SDP offer/answer. The basic idea is to try, in a staggered parallel
fashion, all possible connection addresses that an endpoint may be
reachable by. This allows the endpoint to use the best available UDP
"connection" (meaning two UDP end-points capable of reaching each
other). The methodology has very nice properties in that basically
all NAT topologies are possible to traverse.
Here is how ICE works at a high level. End point A collects all
possible addresses that can be used, including local IP addresses,
STUN derived addresses, TURN addresses, etc. On each local port that
any of these address and port pairs lead to, a STUN server is
installed. This STUN server only accepts STUN requests using the
correct authentication through the use of a username and password.
End-point A then sends a request to establish connectivity with end-
point B, which includes all possible "destinations" [RFC5245] to get
the media through to A. Note that each of A's local address/port
pairs (host candidates and server reflexive base) has a STUN server
co-located. B in turn provides A with all its possible destinations
for the different media streams. A and B then uses a STUN client to
try to reach all the address and port pairs specified by A from its
corresponding destination ports. The destinations for which the STUN
requests successfully complete are then indicated and one is
selected.
If B fails to get any STUN response from A, all hope is not lost.
Certain NAT topologies require multiple tries from both ends before
successful connectivity is accomplished and therefore requests are
retransmitted multiple times. The STUN requests may also result in
that more connectivity alternatives (destinations) are discovered and
conveyed in the STUN responses.
4.3.2. Using ICE in RTSP
The usage of ICE for RTSP requires that both client and server be
updated to include the ICE functionality. If both parties implement
the necessary functionality the following steps could provide ICE
support for RTSP.
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This assumes that it is possible to establish a TCP connection for
the RTSP messages between the client and the server. This is not
trivial in scenarios where the server is located behind a NAT, and
may require some TCP ports be opened, or the deployment of proxies,
etc.
The negotiation of ICE in RTSP of necessity will work different than
in SIP with SDP offer/answer. The protocol interactions are
different and thus the possibilities for transfer of states are also
somewhat different. The goal is also to avoid introducing extra
delay in the setup process at least for when the server is not behind
a NAT in regards to the client, and the client is either having an
address in the same address domain, or is behind NAT(s) which can
address the address domain of the server. This process is only
intended to support PLAY mode, i.e. media traffic flows from server
to client.
1. The ICE usage begins in the SDP. The SDP for the service
indicates that ICE is supported at the server. No candidates can
be given here as that would not work with the on demand, DNS load
balancing, etc., that have the SDP indicate a resource on a
server park rather than a specific machine.
2. The client gathers addresses and puts together its candidates for
each media stream indicated in the session description.
3. In each SETUP request the client includes its candidates in an
ICE specific transport specification. This indicates for the
server the ICE support by the client. One candidate is the most
prioritized candidate and here the prioritization for this
address should be somewhat different compared to SIP. High
performance candidates is recommended rather than candidates with
the highest likelly hood of success, as it is more likely that a
server is not behind a NAT compared to a SIP user-agent.
4. The server responds to the SETUP (200 OK) for each media stream
with its candidates. A server not behind a NAT usually only
provides a single ICE candidate. Also here one candidate is the
server primary address.
5. The connectivity checks are performed. For the server the
connectivity checks from the server to the clients have an
additional usage. They verify that there is someone willing to
receive the media, thus preventing the server from unknowingly
performing a DoS attack.
6. Connectivity checks from the client promoting a candidate pair
were successful. Thus no further SETUP requests are necessary
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and processing can proceed with step 7. If another address than
the primary has been verified by the client to work, that address
may then be promoted for usage in a SETUP request (Go to 7). If
the checks for the available candidates failed and if further
candidates have been derived during the connectivity checks, then
those can be signalled in new candidate lines in a SETUP request
updating the list (Go to 5).
7. Client issues PLAY request. If the server also has completed its
connectivity checks for the promoted candidate pair (based on
username as it may be derived addresses if the client was behind
NAT) then it can directly answer 200 OK (Go to 8). If the
connectivity check has not yet completed it responds with a 1xx
code to indicate that it is verifying the connectivity. If that
fails within the set timeout, an error is reported back. Client
needs to go back to 6.
8. Process completed and media can be delivered. ICE candidates not
used may be released.
To keep media paths alive the client needs to periodically send data
to the server. This will be realized with STUN. RTCP sent by the
client should be able to keep RTCP open but STUN will also be used
based on the same motivations as for ICE for SIP.
4.3.3. Implementation burden of ICE
The usage of ICE will require that a number of new protocols and new
RTSP/SDP features be implemented. This makes ICE the solution that
has the largest impact on client and server implementations amongst
all the NAT/firewall traversal methods in this document.
RTSP server implementation requirements are:
o STUN server features
o Limited STUN client features
o SDP generation with more parameters.
o RTSP error code for ICE extension
RTSP client implementation requirements are:
o Limited STUN server features
o Limited STUN client features
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o RTSP error code and ICE extension
4.3.4. Deployment Considerations
Advantages:
o Solves NAT connectivity discovery for basically all cases as long
as a TCP connection between the client and server can be
established. This includes servers behind NATs. (Note that a
proxy between address domains may be required to get TCP through).
o Improves defenses against DDoS attacks, since a media receiving
client requires authentications, via STUN on its media reception
ports.
Disadvantages:
o Increases the setup delay with at least the amount of time it
takes for the server to perform its STUN requests.
o Assumes that it is possible to de-multiplex between the packets of
the media protocol and STUN packets.
o Has fairly high implementation burden put on both RTSP server and
client. However, several Open Source ICE implementations do
exist, such as [NICE][PJNATH].
4.3.5. Security Consideration
One should review the security consideration section of ICE and STUN
to understand that ICE contains some potential issues. However these
can be avoided by correctly using ICE in RTSP. An important factor
is to secure the signalling, i.e. use TLS between RTSP client and
server. In fact ICE does help avoid the DDoS attack issue with RTSP
substantially as it reduces the possibility for a DDoS using RTSP
servers to attackers that are on-path between the RTSP server and the
target and capable of intercepting the STUN connectivity check
packets and correctly send a response to the server.
4.4. Latching
4.4.1. Introduction
Latching [I-D.ietf-mmusic-latching] is a NAT traversal solution that
is based on requiring RTSP clients to send UDP packets to the
server's media output ports. Conventionally, RTSP servers send RTP
packets in one direction: from server to client. Latching is similar
to connection-oriented traffic, where one side (e.g., the RTSP
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client) first "connects" by sending a RTP packet to the other side's
RTP port, the recipient then replies to the originating IP and port.
This method is also referred to as "Late binding". It requires that
all RTP/RTCP transport is done symmetrical, i.e. Symmetric RTP
[RFC4961].
Specifically, when the RTSP server receives the latching packet
(a.k.a. hole-punching packet, since it is used to punch a hole in the
firewall/NAT and to aid the server for port binding and address
mapping) from its client, it copies the source IP and Port number and
uses them as delivery address for media packets. By having the
server send media traffic back the same way as the client's packet
are sent to the server, address mappings will be honored. Therefore
this technique works for all types of NATs, given that the server is
not behind a NAT. However, it does require server modifications.
Unless there is built-in protection mechanism, latching is very
vulnerable to both hijacking and becoming a tool in Distributed
Denail of Service (DDoS) attacks (See Security Considerations of
[I-D.ietf-mmusic-latching]), because attackers can simply forge the
source IP & Port of the latching packet. Using the rule for
restricting IP address to the one of the signaling connection will
need to be applied here also. However, that does not protect against
hijacking from another client behind the same NAT. This can become a
serious issue in deployments with CGNs.
4.4.2. Necessary RTSP extensions
To support Latching, the RTSP signaling must be extended to allow the
RTSP client to indicate that it will use Latching. The client also
needs to be able to signal its RTP SSRC to the server in its SETUP
request. The RTP SSRC is used to establish some basic level of
security against hijacking attacks or simply avoid mis-association
when multiple clients are behind the same NAT. Care must be taken in
choosing clients' RTP SSRC. First, it must be unique within all the
RTP sessions belonging to the same RTSP session. Secondly, if the
RTSP server is sending out media packets to multiple clients from the
same send port, the RTP SSRC needs to be unique amongst those
clients' RTP sessions. Recognizing that there is a potential that
RTP SSRC collisions may occur, the RTSP server must be able to signal
to a client that a collision has occurred and that it wants the
client to use a different RTP SSRC carried in the SETUP response or
use unique ports per RTSP session. Using unique ports limits an RTSP
server in the number of sessions it can simultaneously handle per
interface IP addresses.
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4.4.3. Deployment Considerations
Advantages:
o Works for all types of client-facing NATs. (Requirement 1 in
Section 3).
o Has no interaction problems with any RTSP ALG changing the
client's information in the transport header.
Disadvantages:
o Requires modifications to both RTSP server and client.
o Limited to work with servers that are not behind a NAT.
o The format of the RTP packet for "connection setup" (a.k.a
Latching packet) is not defined. One possibility considered was
to use RTP No-Op packet format in [I-D.ietf-avt-rtp-no-op], a
proposal which has been abandoned.
o SSRC management if RTP is used for latching due to risk for mis-
association of clients to RTSP sessions at the server if SSRC
collision occurs.
o Has significant security considerations (See Section 4.4.4), due
to lack of a strong authentication mechanism and will need to use
address restrictions.
4.4.4. Security Consideration
Latching's major security issue is that RTP streams can be hijacked
and directed towards any target that the attacker desires unless
address restrictions are used. In the case of NATs with multiple
clients on the inside of them, hijacking can still occur. This
becomes a significant threat in the context of carrier grade NATs
(CGN).
The most serious security problem is the deliberate attack with the
use of a RTSP client and Latching. The attacker uses RTSP to setup a
media session. Then it uses Latching with a spoofed source address
of the intended target of the attack. There is no defense against
this attack other than restricting the possible address a latching
packet can come from to the same as the RTSP TCP connection are from.
This prevents Latching to be used in use cases that require different
addresses for media destination and signalling. Even allowing only a
limited address range containing the signalling address from where
latching is allowed opens up a significant vulnerability as it is
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difficult to determine the address usage for the network the client
connects from.
A hijack attack can also be performed in various ways. The basic
attack is based on the ability to read the RTSP signaling packets in
order to learn the address and port the server will send from and
also the SSRC the client will use. Having this information the
attacker can send its own Latching packets containing the correct RTP
SSRC to the correct address and port on the server. The RTSP server
will then use the source IP and port from the Latching packet as the
destination for the media packets it sends.
Another variation of this attack is for a man in the middle to modify
the RTP latching packet being sent by a client to the server by
simply changing the source IP to the target one desires to attack.
One can fend off the snooping based attack by applying encryption to
the RTSP signaling transport. However, if the attacker is a man in
the middle modifying latching packets, the attack is impossible to
defend against other than through address restrictions. As a NAT re-
writes the source IP and (possibly) port this cannot be
authenticated, but authentication is required in order to protect
against this type of DoS attack.
Yet another issue is that these attacks also can be used to deny the
client the service it desires from the RTSP server completely. The
attacker modifies or originates its own latching packets with another
port than what the legit latching packets uses, which results in that
the media server sends the RTP/RTCP traffic to ports the client isn't
listening for RTP/RTCP on.
The amount of random non-guessable material in the latching packet
determines how well Latching can fend off stream-hijacking performed
by parties that are off the client to server network path, i.e. lacks
the capability to see the client's latching packets. This proposal
uses the 32-bit RTP SSRC field to this effect. Therefore it is
important that this field is derived with a non-predictable random
number generator. It should not be possible by knowing the algorithm
used and a couple of basic facts, to derive what random number a
certain client will use.
An attacker not knowing the SSRC but aware of which port numbers that
a server sends from can deploy a brute force attack on the server by
testing a lot of different SSRCs until it finds a matching one.
Therefore a server could implement functionality that blocks packets
to ports or from sources that receive or send multiple Latching
packets with different invalid SSRCs, especially when they are coming
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from the same IP/Port. Note that this mitigation in itself opens up
a new venue for DoS attacks against legit users trying to latch.
To improve the security against attackers the amount of random
material could be increased. To achieve a longer random tag while
still using RTP and RTCP, it will be necessary to develop RTP and
RTCP payload formats for carrying the random material.
4.5. A Variation to Latching
4.5.1. Introduction
Latching as described above requires the usage of a valid RTP format
as the Latching packet, i.e. the first packet that the client sends
to the server to establish a bi-directional transport flow for RTP
streams. There is currently no appropriate RTP packet format for
this purpose, although the RTP No-Op format was a proposal to fix the
problem [I-D.ietf-avt-rtp-no-op], however, that work was abandoned.
There exists a RFC that discusses the implication of different type
of packets as keep-alives for RTP [RFC6263] and its findings are very
relevant to the format of the Latching packet.
Meanwhile, there has been NAT/firewall traversal techniques deployed
in the wireless streaming market place that use non-RTP messages as
Latching packets. This section describes a variant based on a subset
of those solutions that alters the previously described Latching
solution.
4.5.2. Necessary RTSP extensions
In this variation of Latching, the Latching packet is a small UDP
packet that does not contain an RTP header. In response to the
client's Latching packet, the RTSP server sends back a similar
Latching packet as a confirmation so the client can stop the so
called "connection phase" of this NAT traversal technique.
Afterwards, the client only has to periodically send Latching packets
as keep-alive messages for the NAT mappings.
The server listens on its RTP-media output port, and tries to decode
any received UDP packet as Latching packet. This is valid since an
RTSP server is not expecting RTP traffic from the RTSP client. Then,
it can correlate the Latching packet with the RTSP client's session
ID or the client's SSRC, and record the NAT bindings accordingly.
The server then sends a Latching packet as the response to the
client.
The Latching packet can contain the SSRC to identify the RTP stream,
and care must be taken if the packet is bigger than 12 bytes,
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ensuring that it is distinctively different from RTP packets, whose
header size is 12 bytes.
RTSP signaling can be added to do the following:
1. Enable or disable such Latching message exchanges. When the
firewall/NAT has an RTSP-aware ALG, it is possible to disable
Latching message exchange and let the ALG work out the address
and port mappings.
2. Configure the number of re-tries and the re-try interval of the
Latching message exchanges.
4.5.3. Deployment Considerations
This approach has the following advantages when compared with the
Latching approach (Section 4.4):
1. There is no need to define RTP payload format for firewall
traversal, therefore it is simple to use, implement and
administer (Requirement 4 in Section 3), instead a Latching
protocol must be defined.
2. When properly defined, this kind of Latching packet exchange can
also authenticate RTP receivers, to prevent hijacking attacks.
This approach has the following disadvantages when compared with the
Latching approach:
1. RTP traffic is normally accompanied by RTCP traffic. This
approach needs to rely on RTCP RRs and SRs to enable NAT
traversal for RTCP endpoints, use RTP/RTCP Multiplexing
[RFC5761], or use the same type of Latching packets also for RTCP
endpoints.
2. The server's sender SSRC for the RTP stream or other session
Identity information must be signaled in RTSP's SETUP response,
in the Transport header of the RTSP SETUP response.
4.5.4. Security Considerations
Compared to the security properties of Latching this variant is
slightly improved. First of all it allows for a larger random field
in the Latching packets which makes it more unlikely for an off-path
attacker to succeed in a hi-jack attack. Secondly the confirmation
allows the client to know when Latching works and when it didn't and
thus restart the Latching process by updating the SSRC. Thirdly if
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an authentication mechanism is included in the latching packet
hijacking attacks can be prevented.
Still the main security issue remain that the RTSP server can't know
that the source address in the latching packet was coming from a RTSP
client wanting to receive media and not one that likes to direct the
media traffic to an DoS target.
4.6. Three Way Latching
4.6.1. Introduction
The three way Latching is an attempt to try to resolve the most
significant security issues for both previously discussed variants of
Latching. By adding a server request response exchange directly
after the initial latching the server can verify that the target
address present in the latching packet is an active listener and
confirm its desire to establish a media flow.
4.6.2. Necessary RTSP extensions
Uses the same RTSP extensions as the alternative latching method
(Section 4.5) uses. The extensions for this variant are only in the
format and transmission of the Latching packets.
The client to server latching packet is similar to the Alternative
Latching (Section 4.5), i.e. an UDP packet with some session
identifier and a random value. When the server responds to the
Latching packet with a Latching confirmation, it includes a random
value (Nonce) of its own in addition to echoing back the one the
client sent. Then a third message is added to the exchange. The
client acknowledges the reception of the Latching confirmation
message and echoes back the server's nonce. Thus confirming that the
Latched address goes to a RTSP client that initiated the latching and
is actually present at that address. The RTSP server will refuse to
send any media until the Latching Acknowledgement has been received
with a valid nonce.
4.6.3. Deployment Considerations
A solution with a 3-way handshake and its own Latching packets can be
compared with the ICE-based solution (Section 4.3) and have the
following differences:
o Only works for servers that are not behind a NAT.
o May be simpler to implement due to the avoidance of the ICE
prioritization and check-board mechanisms.
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However, a 3-way Latching protocol is very similar to using STUN in
both directions as Latching and verification protocol. Using STUN
would remove the need for implementing a new protocol.
4.7. Application Level Gateways
4.7.1. Introduction
An Application Level Gateway (ALG) reads the application level
messages and performs necessary changes to allow the protocol to work
through the middle box. However this behavior has some problems in
regards to RTSP:
1. It does not work when the RTSP protocol is used with end-to-end
security. As the ALG can't inspect and change the application
level messages the protocol will fail due to the middle box.
2. ALGs need to be updated if extensions to the protocol are added.
Due to deployment issues with changing ALGs this may also break
the end-to-end functionality of RTSP.
Due to the above reasons it is not recommended to use an RTSP ALG in
NATs. This is especially important for NATs targeted to home users
and small office environments, since it is very hard to upgrade NATs
deployed in home or SOHO (small office/home office) environment.
4.7.2. Outline On how ALGs for RTSP work
In this section, we provide a step-by-step outline on how one could
go about writing an ALG to enable RTSP to traverse a NAT.
1. Detect any SETUP request.
2. Try to detect the usage of any of the NAT traversal methods that
replace the address and port of the Transport header parameters
"destination" or "dest_addr". If any of these methods are used,
then the ALG should not change the address. Ways to detect that
these methods are used are:
* For embedded STUN, it would be to watch for a feature tag,
like "nat.stun". If any of those exists in the "supported",
"proxy-require", or "require" headers of the RTSP exchange.
* For stand alone STUN and TURN based solutions: This can be
detected by inspecting the "destination" or "dest_addr"
parameter. If it contains either one of the NAT's external IP
addresses or a public IP address then such a solution is in
use. However if multiple NATs are used this detection may
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fail. Remapping should only be done for addresses belonging
to the NAT's own private address space.
Otherwise continue to the next step.
3. Create UDP mappings (client given IP/port <-> external IP/port)
where needed for all possible transport specifications in the
transport header of the request found in (1). Enter the external
address and port(s) of these mappings in transport header.
Mappings shall be created with consecutive external port numbers
starting on an even number for RTP for each media stream.
Mappings should also be given a long timeout period, at least 5
minutes.
4. When the SETUP response is received from the server, the ALG may
remove the unused UDP mappings, i.e. the ones not present in the
transport header. The session ID should also be bound to the UDP
mappings part of that session.
5. If SETUP response settles on RTP over TCP or RTP over RTSP as
lower transport, do nothing: let TCP tunneling take care of NAT
traversal. Otherwise go to next step.
6. The ALG should keep the UDP mappings belonging to the RTSP
session as long as: an RTSP message with the session's ID has
been sent in the last timeout interval, or a UDP message has been
sent on any of the UDP mappings during the last timeout interval.
7. The ALG may remove a mapping as soon a TEARDOWN response has been
received for that media stream.
4.7.3. Deployment Considerations
Advantage:
o No impact on either client or server
o Can work for any type of NATs
Disadvantage:
o When deployed they are hard to update to reflect protocol
modifications and extensions. If not updated they will break the
functionality.
o When end-to-end security is used, the ALG functionality will fail.
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o Can interfere with other types of traversal mechanisms, such as
STUN.
Transition:
An RTSP ALG will not be phased out in any automatic way. It must be
removed, probably through the removal of the NAT it is associated
with.
4.7.4. Security Considerations
An ALG will not work with deployment of end-to-end RTSP signaling
security. Therefore deployment of ALG will likely result in clients
located behind NATs not using end-to-end security, or more likely
select a NAT traversal solution that allow for security.
The creation of an UDP mapping based on the signalling message has
some potential security implications. First of all if the RTSP
client releases its ports and another application are assigned these
instead it could receive RTP media as long as the mappings exist and
the RTSP server has failed to be signalled or notice the lack of
client response.
A NAT with RTSP ALG that assigns mappings based on SETUP requests
could potentially become victim of a resource exhaustion attack. If
an attacker creates a lot of RTSP sessions, even without starting
media transmission could exhaust the pool of available UDP ports on
the NAT. Thus only a limited number of UDP mappings should be
allowed to be created by the RTSP ALG.
4.8. TCP Tunneling
4.8.1. Introduction
Using a TCP connection that is established from the client to the
server ensures that the server can send data to the client. The
connection opened from the private domain ensures that the server can
send data back to the client. To send data originally intended to be
transported over UDP requires the TCP connection to support some type
of framing of the media data packets. Using TCP also results in the
client having to accept that real-time performance can be impacted.
TCP's problem of ensuring timely delivery was one of the reasons why
RTP was developed. Problems that arise with TCP are: head-of-line
blocking, delay introduced by retransmissions, highly varying rate
due to the congestion control algorithm. If sufficient amount of
buffering (several seconds) in the receiving client can be tolerated
then TCP clearly can work.
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4.8.2. Usage of TCP tunneling in RTSP
The RTSP core specification [I-D.ietf-mmusic-rfc2326bis] supports
interleaving of media data on the TCP connection that carries RTSP
signaling. See section 14 in [I-D.ietf-mmusic-rfc2326bis] for how to
perform this type of TCP tunneling. There also exists another way of
transporting RTP over TCP defined in Appendix C.2 in
[I-D.ietf-mmusic-rfc2326bis]. For signaling and rules on how to
establish the TCP connection in lieu of UDP, see appendix C.2 in
[I-D.ietf-mmusic-rfc2326bis]. This is based on the framing of RTP
over the TCP connection as described in RFC 4571 [RFC4571].
4.8.3. Deployment Considerations
Advantage:
o Works through all types of NATs where the RTSP server in not NATed
or at least reachable like it was not.
Disadvantage:
o Functionality needs to be implemented on both server and client.
o Will not always meet multimedia stream's real-time requirements.
Transition:
The tunneling over RTSP's TCP connection is not planned to be phased-
out. It is intended to be a fallback mechanism and for usage when
total media reliability is desired, even at the potential price of
loss of real-time properties.
4.8.4. Security Considerations
The TCP tunneling of RTP has no known security problems besides those
already presented in the RTSP specification. It is not possible to
get any amplification effect for denial of service attacks due to
TCP's flow control. A possible security consideration, when session
media data is interleaved with RTSP, would be the performance
bottleneck when RTSP encryption is applied, since all session media
data also needs to be encrypted.
4.9. TURN (Traversal Using Relay NAT)
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4.9.1. Introduction
Traversal Using Relay NAT (TURN) [RFC5766] is a protocol for setting
up traffic relays that allow clients behind NATs and firewalls to
receive incoming traffic for both UDP and TCP. These relays are
controlled and have limited resources. They need to be allocated
before usage. TURN allows a client to temporarily bind an address/
port pair on the relay (TURN server) to its local source address/port
pair, which is used to contact the TURN server. The TURN server will
then forward packets between the two sides of the relay.
To prevent DoS attacks on either recipient, the packets forwarded are
restricted to the specific source address. On the client side it is
restricted to the source setting up the allocation. On the external
side this is limited to the source address/port pair that have been
given permission by the TURN client creating the allocation. Packets
from any other source on this address will be discarded.
Using a TURN server makes it possible for a RTSP client to receive
media streams from even an unmodified RTSP server. However the
problem is those RTSP servers most likely restrict media destinations
to no other IP address than the one the RTSP message arrives from.
This means that TURN could only be used if the server knows and
accepts that the IP belongs to a TURN server and the TURN server
can't be targeted at an unknown address. Alternatively, both the
RTSP TCP connection as well as the RTP media is relayed through the
same TURN server.
4.9.2. Usage of TURN with RTSP
To use a TURN server for NAT traversal, the following steps should be
performed.
1. The RTSP client connects with the RTSP server. The client
retrieves the session description to determine the number of
media streams. To avoid the issue with having RTSP connection
and media traffic from different addresses also the TCP
connection must be done through the same TURN server as the one
in the next step. This will require the usage of TURN for TCP
[RFC6062].
2. The client establishes the necessary bindings on the TURN server.
It must choose the local RTP and RTCP ports that it desires to
receive media packets. TURN supports requesting bindings of even
port numbers and contiguous ranges.
3. The RTSP client uses the acquired address and port allocations in
the RTSP SETUP request using the destination header.
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4. The RTSP Server sends the SETUP reply, which must include the
transport headers src_addr parameter (source and port in RTSP
1.0). Note that the server is required to have a mechanism to
verify that it is allowed to send media traffic to the given
address.
5. The RTSP Client uses the RTSP Server's response to create TURN
permissions for the server's media traffic.
6. The client requests that the server starts playing. The server
starts sending media packets to the given destination address and
ports.
7. Media packets arrive at the TURN server on the external port; If
the packets match an established permission, the TURN server
forwards the media packets to the RTSP client.
8. If the client pauses and media is not sent for about 75% of the
mapping timeout the client should use TURN to refresh the
bindings.
4.9.3. Deployment Considerations
Advantages:
o Does not require any server modifications given that the server
includes the src_addr header in the SETUP response.
o Works for any type of NAT as long as the RTSP server has reachable
IP address that is not behind a NAT.
Disadvantage:
o Requires another network element, namely the TURN server.
o A TURN server for RTSP may not scale since the number of sessions
it must forward is proportional to the number of client media
sessions.
o TURN server becomes a single point of failure.
o Since TURN forwards media packets, it necessarily introduces
delay.
o An RTSP ALG may change the necessary destinations parameter. This
will cause the media traffic to be sent to the wrong address.
Transition:
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TURN is not intended to be phased-out completely, see Section 19 of
[RFC5766]. However the usage of TURN could be reduced when the
demand for having NAT traversal is reduced.
4.9.4. Security Considerations
The TURN server can become part of a denial of service attack towards
any victim. To perform this attack the attacker must be able to
eavesdrop on the packets from the TURN server towards a target for
the DoS attack. The attacker uses the TURN server to setup a RTSP
session with media flows going through the TURN server. The attacker
is in fact creating TURN mappings towards a target by spoofing the
source address of TURN requests. As the attacker will need the
address of these mappings he must be able to eavesdrop or intercept
the TURN responses going from the TURN server to the target. Having
these addresses, he can set up a RTSP session and start delivery of
the media. The attacker must be able to create these mappings. The
attacker in this case may be traced by the TURN username in the
mapping requests.
This attack requires that the attacker has access to a user account
on the TURN server to be able set up the TURN mappings. To prevent
this attack the RTSP server needs to verify that the ultimate target
destination accept this media stream. Which would require something
like ICE's connectivity checks being run between the RTSP server and
the RTSP client.
5. Firewalls
Firewalls exist for the purpose of protecting a network from traffic
not desired by the firewall owner. Therefore it is a policy decision
if a firewall will let RTSP and its media streams through or not.
RTSP is designed to be firewall friendly in that it should be easy to
design firewall policies to permit passage of RTSP traffic and its
media streams.
The firewall will need to allow the media streams associated with a
RTSP session to pass through it. Therefore the firewall will need an
ALG that reads RTSP SETUP and TEARDOWN messages. By reading the
SETUP message the firewall can determine what type of transport and
from where, the media stream packets will be sent. Commonly there
will be the need to open UDP ports for RTP/RTCP. By looking at the
source and destination addresses and ports the opening in the
firewall can be minimized to the least necessary. The opening in the
firewall can be closed after a TEARDOWN message for that session or
the session itself times out.
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The above possibilities for firewalls to inspect and respond to the
signalling are prevented if confidentiality protection is used for
the RTSP signalling, e.g. using the specified RTSP over TLS. This
results in that firewalls can't be actively opening pinholes for the
media streams based on the signalling. Instead other methods have to
be used to enable the transport flows for the media.
Simpler firewalls do allow a client to receive media as long as it
has sent packets to the target. Depending on the security level this
can have the same behavior as a NAT. The only difference is that no
address translation is done. To use such a firewall a client would
need to implement one of the above described NAT traversal methods
that include sending packets to the server to open up the mappings.
6. Comparison of NAT traversal techniques
This section evaluates the techniques described above against the
requirements listed in Section 3.
In the following table, the columns correspond to the numbered
requirements. For instance, the column under R1 corresponds to the
first requirement in Section 3: must work for all flavors of NATs.
The rows represent the different NAT/firewall traversal techniques.
Latch is short for Latching, "V. Latch" is short for "variation of
Latching" as described in Section 4.5. "3-W Latch" is short for the
Three Way Latching described in Section 4.6.
A Summary of the requirements are:
R1: Work for all flavors of NATs
R2: Must work with firewalls, including those with ALGs
R3: Should have minimal impact on clients not behind NATs, counted
in minimal number of additional RTTs
R4: Should be simple to use, Implement and administer.
R5: Should provide mitigation against DDoS attacks
The following considerations are also added to requirements:
C1: Will solution support both Clients and Servers behind NAT
C2: Is the solution robust to changing NAT behaviors
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------------+------+------+------+------+------+------+------+
| R1 | R2 | R3 | R4 | R5 | C1 | C2 |
------------+------+------+------+------+------+------+------+
STUN | No | Yes | 1 | Maybe| No | No | No |
------------+------+------+------+------+------+------+------+
Emb. STUN | Yes | Yes | 2 | Maybe| No | No | Yes |
------------+------+------+------+------+------+------+------+
ICE | Yes | Yes | 2.5 | No | Yes | Yes | Yes |
------------+------+------+------+------+------+------+------+
Latch | Yes | Yes | 1 | Maybe| No | No | Yes |
------------+------+------+------+------+------+------+------+
V. Latch | Yes | Yes | 1 | Yes | No | No | Yes |
------------+------+------+------+------+------+------+------+
3-W Latch | Yes | Yes | 1.5 | Maybe| Yes | No | Yes |
------------+------+------+------+------+------+------+------+
ALG |(Yes) | Yes | 0 | No | Yes | No | Yes |
------------+------+------+------+------+------+------+------+
TCP Tunnel | Yes | Yes | 1.5 | Yes | Yes | No | Yes |
------------+------+------+------+------+------+------+------+
TURN | Yes | Yes | 1 | No | Yes |(Yes) | Yes |
------------+------+------+------+------+------+------+------+
Figure 1: Comparison of fulfillment of requirements
Looking at Figure 1 one would draw the conclusion that using TCP
Tunneling or Three-Way Latching is the solutions that best fulfill
the requirements. The different techniques were discussed in the
MMUSIC WG. It was established that the WG would pursue an ICE based
solution due to its generality and capability of handling also
servers delivering media from behind NATs. TCP Tunneling is likely
to be available as an alternative, due to its specification in the
main RTSP specification. Thus it can be used if desired and the
potential downsides of using TCP is acceptable in particular
deployments. When it comes to Three-Way Latching it is a very
competitive technique given that you don't need support for RTSP
servers behind NATs. There were some discussion in the WG if the
increased implementation burden of ICE is sufficiently motivated
compared to a the Three-Way Latching solution for this generality.
In the end the authors believe that reuse of ICE, the greater
flexibility and anyway need to deploy a new solution was the decisive
factors.
The ICE based RTSP NAT traversal solution is specified in "A Network
Address Translator (NAT) Traversal mechanism for media controlled by
Real-Time Streaming Protocol (RTSP)" [I-D.ietf-mmusic-rtsp-nat].
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7. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
8. Security Considerations
In the preceding sections we have discussed security merits of the
different NAT/firewall traversal methods for RTSP discussed here. In
summary, the presence of NAT(s) is a security risk, as a client
cannot perform source authentication of its IP address. This
prevents the deployment of any future RTSP extensions providing
security against hijacking of sessions by a man-in-the-middle.
Each of the proposed solutions has security implications. Using STUN
will provide the same level of security as RTSP without transport
level security and source authentications, as long as the server does
not allow media to be sent to a different IP-address than the RTSP
client request was sent from. Using Latching will have a higher risk
of session hijacking or denial of service than normal RTSP. The
reason is that there exists a probability that an attacker is able to
guess the random bits that the client uses to prove its identity when
creating the address bindings. This can be solved in the variation
of Latching (Section 4.5) with authentication features. Still both
those variants of Latching are vulnerable against deliberate attack
from the RTSP client to redirect the media stream requested to any
target assuming it can spoof the source address. This security
vulnerability is solved by performing a Three-way Latching procedure
as discussed in Section 4.6. ICE resolves the binding vulnerability
of latching by using signed STUN messages, as well as requiring that
both sides perform connectivity checks to verify that the target IP
address in the candidate pair is both reachable and willing to
respond. ICE can however create a significant amount of traffic if
the number of candidate pairs are large. Thus pacing is required and
implementations should attempt to limit their number of candidates to
reduce the number of packets. If the signalling between the ICE
peers (RTSP client and Server) is not confidentiality and integrity
protected ICE is vulnerable to attacks where the candidate list is
manipulated. Lack of signalling security will also simplify spoofing
of STUN binding messages by revealing the secret used in signing.
The usage of an RTSP ALG does not in itself increase the risk for
session hijacking. However the deployment of ALGs as the sole
mechanism for RTSP NAT traversal will prevent deployment of end-to-
end encrypted RTSP signaling. The usage of TCP tunneling has no
known security problems. However, it might provide a bottleneck when
it comes to end-to-end RTSP signaling security if TCP tunneling is
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used on an interleaved RTSP signaling connection. The usage of TURN
has severe risk of denial of service attacks against a client. The
TURN server can also be used as a redirect point in a DDoS attack
unless the server has strict enough rules for who may create
bindings.
9. Acknowledgements
The author would also like to thank all persons on the MMUSIC working
group's mailing list that has commented on this document. Persons
having contributed in such way in no special order to this protocol
are: Jonathan Rosenberg, Philippe Gentric, Tom Marshall, David Yon,
Amir Wolf, Anders Klemets, Flemming Andreasen, Ari Keranen, Bill
Atwood, and Colin Perkins. Thomas Zeng would also like to give
special thanks to Greg Sherwood of PacketVideo for his input into
this memo.
Section 1.1 contains text originally written for RFC 4787 by Francois
Audet and Cullen Jennings.
10. Informative References
[I-D.ietf-avt-rtp-no-op]
Andreasen, F., "A No-Op Payload Format for RTP", draft-
ietf-avt-rtp-no-op-04 (work in progress), May 2007.
[I-D.ietf-mmusic-latching]
Ivov, E., Kaplan, H., and D. Wing, "Latching: Hosted NAT
Traversal (HNT) for Media in Real-Time Communication",
draft-ietf-mmusic-latching-05 (work in progress), May
2014.
[I-D.ietf-mmusic-rfc2326bis]
Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
and M. Stiemerling, "Real Time Streaming Protocol 2.0
(RTSP)", draft-ietf-mmusic-rfc2326bis-40 (work in
progress), February 2014.
[I-D.ietf-mmusic-rtsp-nat]
Goldberg, J., Westerlund, M., and T. Zeng, "A Network
Address Translator (NAT) Traversal Mechanism for Media
Controlled by Real-Time Streaming Protocol (RTSP)", draft-
ietf-mmusic-rtsp-nat-20 (work in progress), February 2014.
[NICE] "Libnice - The GLib ICE implementation,
http://nice.freedesktop.org/wiki/", May 2013.
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[PJNATH] "PJNATH - Open Source ICE, STUN, and TURN Library,
http://www.pjsip.org/pjnath/docs/html/", May 2013.
[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
August 1980.
[RFC0793] Postel, J., "Transmission Control Protocol", STD 7, RFC
793, September 1981.
[RFC2326] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
Streaming Protocol (RTSP)", RFC 2326, April 1998.
[RFC2588] Finlayson, R., "IP Multicast and Firewalls", RFC 2588, May
1999.
[RFC2663] Srisuresh, P. and M. Holdrege, "IP Network Address
Translator (NAT) Terminology and Considerations", RFC
2663, August 1999.
[RFC3022] Srisuresh, P. and K. Egevang, "Traditional IP Network
Address Translator (Traditional NAT)", RFC 3022, January
2001.
[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.
[RFC3424] Daigle, L. and IAB, "IAB Considerations for UNilateral
Self-Address Fixing (UNSAF) Across Network Address
Translation", RFC 3424, November 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.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
and RTP Control Protocol (RTCP) Packets over Connection-
Oriented Transport", RFC 4571, July 2006.
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[RFC4787] Audet, F. and C. Jennings, "Network Address Translation
(NAT) Behavioral Requirements for Unicast UDP", BCP 127,
RFC 4787, January 2007.
[RFC4961] Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
BCP 131, RFC 4961, July 2007.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, April
2010.
[RFC5382] Guha, S., Biswas, K., Ford, B., Sivakumar, S., and P.
Srisuresh, "NAT Behavioral Requirements for TCP", BCP 142,
RFC 5382, October 2008.
[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
"Session Traversal Utilities for NAT (STUN)", RFC 5389,
October 2008.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, April 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.
[RFC6062] Perreault, S. and J. Rosenberg, "Traversal Using Relays
around NAT (TURN) Extensions for TCP Allocations", RFC
6062, November 2010.
[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for
Keeping Alive the NAT Mappings Associated with RTP / RTP
Control Protocol (RTCP) Flows", RFC 6263, June 2011.
[STUN-IMPL]
"Open Source STUN Server and Client,
http://sourceforge.net/projects/stun/", May 2013.
Authors' Addresses
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Magnus Westerlund
Ericsson
Farogatan 6
Stockholm SE-164 80
Sweden
Phone: +46 8 719 0000
Email: magnus.westerlund@ericsson.com
Thomas Zeng
Email: thomas.zeng@gmail.com
Westerlund & Zeng Expires November 29, 2014 [Page 41]
