One document matched: draft-ietf-mmusic-rtsp-nat-evaluation-05.xml
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<front>
<title abbrev="Evaluation of NAT Traversal for RTSP">The Evaluation of
Different Network Addres Translator (NAT) Traversal Techniques for Media
Controlled by Real-time Streaming Protocol (RTSP)</title>
<author fullname="Magnus Westerlund" initials="M." surname="Westerlund">
<organization>Ericsson</organization>
<address>
<postal>
<street>Farogatan 6</street>
<city>Stockholm</city>
<region/>
<code>SE-164 80</code>
<country>Sweden</country>
</postal>
<phone>+46 8 719 0000</phone>
<facsimile/>
<email>magnus.westerlund@ericsson.com</email>
<uri/>
</address>
</author>
<author fullname="Thomas Zeng" initials="T." surname="Zeng">
<organization/>
<address>
<postal>
<street/>
<city/>
<region/>
<code/>
<country/>
</postal>
<phone/>
<facsimile/>
<email>thomas.zeng@gmail.com</email>
<uri/>
</address>
</author>
<date day="7" month="May" year="2012"/>
<abstract>
<t>This document describes several Network Address Translator (NAT)
traversal techniques that was considered to be used by Real-time
Streaming Protocol (RTSP). Each technique includes a description on how
it would be used, the security implications of using it and any other
deployment considerations it has. There are also disussions on how NAT
traversal techniques relates to firewalls and how each technique can be
applied in different use cases. These findings where used when selecting
the NAT traversal for RTSP 2.0 standardized in the MMUSIC WG.</t>
</abstract>
</front>
<middle>
<section title="Introduction">
<t>Today there is a proliferate deployment of different flavors of
Network Address Translator (NAT) boxes that in many cases only loosely
follows <xref target="RFC3022">standards</xref><xref
target="RFC2663"/><xref target="RFC3424"/>]. NATs cause discontinuity in
<xref target="RFC3424">address realms</xref>, therefore an application
protocol, such as <xref target="RFC2326">Real-time Streaming Protocol
(RTSP)</xref><xref target="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 <xref
target="RFC0793">Transmission Control Protocol (TCP)</xref> for example,
may not be blocked by NATs, its media streams may be blocked by NATs.
This will occur unless special protocol provisions are added to support
NAT-traversal.</t>
<t>Like NATs, firewalls (FWs) are also middle boxes that need to be
considered. Firewalls helps 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 to go through with
minimal implementation effort. The minimal effort is to 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.</t>
<t>This document describes several NAT-traversal mechanisms for RTSP
controlled media streaming. These NAT solutions fall into the category
of "UNilateral Self-Address Fixing (UNSAF)" as defined in <xref
target="RFC3424"/> and quoted below:</t>
<t>"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."</t>
<t>Following the guidelines spelled out in RFC 3424, we describe the
required RTSP protocol extensions for each method, transition
strategies, and security concerns.</t>
<t>This document is capturing the evaluation done in the process to
recommend FW/NAT traversal methods for RTSP streaming servers based on
<xref target="RFC2326">RFC 2326</xref> as well as the <xref
target="I-D.ietf-mmusic-rfc2326bis">RTSP 2.0 core spec</xref>. The
evaluation is focused on NAT traversal for the media streams carried
over <xref target="RFC0768">User Datagram Protocol (UDP)</xref>. Where
<xref target="RFC3550">Real-time Transport Protocol (RTP)</xref> over
UDP being the main case for such usage. The findings should be
applicable to other protocols as long as they have similar
properties.</t>
<t>The resulting ICE-based RTSP NAT traversal mechanism is specified in
<xref target="I-D.ietf-mmusic-rtsp-nat">"A Network Address Translator
(NAT) Traversal mechanism for media controlled by Real-Time Streaming
Protocol (RTSP)"</xref>.</t>
<section anchor="sec-nat-intro" title="Network Address Translators">
<t>Readers are urged to refer to <xref target="RFC2663">"IP Network
Address Translator (NAT) Terminology and Considerations"</xref> for
information on NAT taxonomy and terminology. Traditional NAT is the
most common type of NAT device deployed. Readers may refer to <xref
target="RFC3022">"Traditional IP Network Address Translator
(Traditional NAT)"</xref> for detailed information on traditional NAT.
Traditional NAT has two main varieties -- Basic NAT and Network
Address/Port Translator (NAPT).</t>
<t>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 mapping 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)".</t>
<t>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. There exist a number of address and
port mapping behaviors described in more detail in Section 4.1 of
<xref target="RFC4787">"Network Address Translation (NAT) Behavioral
Requirements for Unicast UDP"</xref>.</t>
<t>NATs also have a filtering behavior on traffic arriving on the
external side. Such behavior effects how well different methods for
NAT traversal works through these NATs. See Section 5 of <xref
target="RFC4787">"Network Address Translation (NAT) Behavioral
Requirements for Unicast UDP"</xref> for more information on the
different types of filtering that have been identified.</t>
</section>
<section title="Firewalls">
<t>A firewall (FW) 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. public Internet. Many
organizations use firewalls to prevent privacy intrusions and
malicious attacks to corporate computing resources in the private
intranet <xref target="RFC2588"/>.</t>
<t>A comparison between NAT and FW is given below:</t>
<t><list style="numbers">
<t>A firewall must sit between two network administrative domains,
while NAT does not have to sit between two domains.</t>
<t>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.</t>
</list></t>
<t>It should be noted that many NAT devices intended for small
office/home office (SOHO) include both NATs and firewall
functionality.</t>
<t>In the rest of this memo we use the phrase "NAT traversal"
interchangeably with "FW traversal", "NAT/FW traversal" and
"NAT/Firewall traversal".</t>
</section>
<section title="Glossary">
<t><list hangIndent="6" style="hanging">
<t hangText="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.</t>
<t hangText="ICE:">Interactive Connectivity Establishment, see
<xref target="RFC5245"/>.</t>
<t hangText="DNS:">Domain Name Service</t>
<t hangText="DDOS:">Distributed Denial Of Service attacks</t>
<t hangText="NAT:">Network Address Translator, see <xref
target="RFC3022"/>.</t>
<t hangText="NAPT:">Network Address/Port Translator, see <xref
target="RFC3022"/>.</t>
<t hangText="RTP:">Real-time Transport Protocol, see <xref
target="RFC3550"/>.</t>
<t hangText="RTSP:">Real-Time Streaming Protocol, see <xref
target="RFC2326"/> and <xref
target="I-D.ietf-mmusic-rfc2326bis"/>.</t>
<t hangText="SDP:">Session Description Protocol, see <xref
target="RFC4566"/>.</t>
<t hangText="SSRC:">Synchronization source in RTP, see <xref
target="RFC3550"/>.</t>
</list></t>
</section>
<section title="Definitions">
<t>The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
<xref target="RFC2119">RFC 2119</xref>.</t>
</section>
</section>
<section title="Detecting the loss of NAT mappings">
<t>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 <xref target="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.</t>
<t>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.</t>
<t>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, there are likely some
middleboxes 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 reports. 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.</t>
<t>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 to be 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/FW,
or that the client is mal-functioning.</t>
</section>
<section anchor="req-section" title="Requirements on NAT-Traversal">
<t>This section considers the set of requirements for the evaulation of
RTSP NAT traversal solutions.</t>
<t>RTSP is a client-server protocol. Typically services providers deploy
RTSP servers in the public address realm. However, there are use cases
where the reverse is true: RTSP clients are connecting from public
address realm to RTSP servers behind home NATs. 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 first requirement should be to
solve the RTSP NAT traversal problem for RTSP servers deployed in a
public network, i.e. no NAT at the server side.</t>
<t>The list of feature requirements for RTSP NAT solutions are given
below:</t>
<t><list style="numbers">
<t>MUST work for all flavors of NATs, including NATs with address
and port restricted filtering.</t>
<t>MUST work for firewalls (subject to pertinent firewall
administrative policies), including those with ALGs.</t>
<t>SHOULD have minimal impact on clients in the open and not
dual-hosted. RTSP dual-hosting means that RTSP protocol and the
media protocol (e.g. RTP) are implemented on different computers
with different IP addresses.<list style="symbols">
<t>For instance, no extra delay from RTSP connection till
arrival of media</t>
</list></t>
<t>SHOULD be simple to use/implement/administer that people actually
turn them on<list style="symbols">
<t>Otherwise people will resort to TCP tunneling through
NATs</t>
<t>Address discovery for NAT traversal should take place behind
the scene, if possible</t>
</list></t>
<t>SHOULD authenticate dual-hosted client transport handler to
prevent DDOS attacks.</t>
</list>The last requirement addresses the Distributed
Denial-Of-Service (DDOS) threat, which relates to NAT traversal as
explained below.</t>
<t>During NAT traversal, when the RTSP server determines the media
destination (Address and port) for client, the result may be that the
public IP address of the RTP receiver host is different than the public
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 in
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 is for the RTSP server to disallow the cases where
the client's RTP receiver has a different public IP address than that of
the RTSP client. However, 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. Message authentication is a big challenge in the
current wired and wireless networking environment. It may be necessary
in the immediate future to deploy NAT traversal solutions that do not
have full message authentication, but provide upgrade path to add
authentication features in the future.</t>
<t/>
</section>
<section anchor="NAT-trav-tech" title="NAT Traversal Techniques">
<t>There exist 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 cost is also different.
They also vary in security levels. In the following sections, each
technique is outlined in details with discussions on the corresponding
advantages and disadvantages.</t>
<t>This section includes NAT traversal techniques that have not been
formally specified anywhere else. The overview section of this document
may be the only publicly available specification of some of the NAT
traversal techniques. However that is no real barrier against doing an
evaluation of the NAT traversal technique. Some other techniques are
currently (at the time of writing) in a state of flux due to ongoing
standardization work on these techniques or has been not been progressed
within IETF after the intiial evaluation in this document, e.g. RTP
No-Op <xref target="I-D.ietf-avt-rtp-no-op"/>.</t>
<section title="STUN">
<t/>
<section title="Introduction">
<t>STUN - "Simple Traversal of UDP Through Network Address
Translators" <xref target="RFC3489"/><xref target="RFC5389"/> is a
standardized protocol that allows a client to use secure means to
discover the presence of a NAT between himself 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. STUN client
sends a request to a STUN server and the server returns a response.
There are two types of STUN requests - Binding Requests, sent over
UDP, and Shared Secret Requests, sent over TLS over TCP.</t>
<t>The first version of <xref target="RFC3489">STUN</xref> included
categorization and parameterization of NATs. This was abandoned in
the updated version due to it being unreliable.</t>
</section>
<section title="Using STUN to traverse NAT without server modifications">
<t>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).</t>
<t>Limitations:</t>
<t><list style="symbols">
<t>The server must be located in either a public address realm
or the next hop external address realm in regards to the
client.</t>
<t>The client may only be located behind NATs that performing
Endpoint Independent or Address Dependent Mappings. Clients
behind NATs that do Address and Port Dependent Mappings cannot
use this method.</t>
</list>Method:</t>
<t>A RTSP client using RTP transport over UDP can use STUN to
traverse a NAT(s) in the following way:</t>
<t><list style="numbers">
<t>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.</t>
<t>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 <xref
target="RFC4566"/>. For example, each RTP session will in
general require two UDP ports, one for RTP, and one for
RTCP.</t>
<t>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.</t>
<t>Perform the RTSP SETUP for each media. In the transport
header the following parameter SHOULD be included with the given
values: <xref
target="I-D.ietf-mmusic-rfc2326bis">"dest_addr"</xref> or
"destination" + <xref target="RFC2326">"client_port"</xref> 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-continuous port numbers for RTP and RTCP. This requires
the new feature provided in the <xref
target="I-D.ietf-mmusic-rfc2326bis">update to RFC2326</xref>.
The server should respond with a transport header containing an
"src_addr" or "source parameter" + "server_port" with the RTP
and RTCP source IP address and port of the media stream.</t>
<t>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 may
be 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 <xref target="I-D.ietf-avt-rtp-no-op">RTP No-Op
packet</xref> to the streaming server's discard port (port
number 9). The drawback of using RTP 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.</t>
</list>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.</t>
<t>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 FW 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, FW packets
should be retransmitted a reasonable number of times.</t>
<t>For a "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, FW 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 FW
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).</t>
<t>This method is also brittle because it relies on that one can use
STUN to classify the NAT behavior. If the NAT changes the properties
of the existing mapping and filtering state for example due to load,
then the methods will fail.</t>
</section>
<section title="Embedding STUN in RTSP">
<t>This section outlines the adaptation and embedding of STUN within
RTSP. This enables STUN to be used to traverse any type of NAT,
including symmetric NATs. This would require protocol changes.</t>
<t>This NAT traversal solution has limitations:</t>
<t><list style="numbers">
<t>It does not work if both RTSP client and RTSP server are
behind separate NATs.</t>
<t>The RTSP server may, for security reasons, refuse to send
media streams to an IP different from the IP in the client RTSP
requests.</t>
</list></t>
<t>Deviations from STUN as defined in RFC 3489:</t>
<t><list style="numbers">
<t>We allow RTSP applications to have the option to perform STUN
"Shared Secret Request" through RTSP, via extension to RTSP;</t>
<t>We require STUN server to be co-located on RTSP server's
media output ports.</t>
</list>In order to allow binding discovery without authentication,
the STUN server embedded in RTSP application must ignore
authentication tag, and the STUN client embedded in RTSP application
must use dummy authentication tag.</t>
<t>If STUN server is co-located with RTSP server's media output
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 inside the NAT must initiate UDP
traffic. The STUN Bind 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.</t>
<t>Likewise, if an RTSP server is behind a NAT, then an embedded
STUN server must co-locate 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 that the server correctly can
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.</t>
<t>To minimize delay, we require that the RTSP server supporting
this option must inform its client the RTP and RTCP ports from where
the server intend to 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 <xref
target="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.</t>
<t>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.</t>
</section>
<section title="Discussion On Co-located STUN Server">
<t>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. A
common heuristics is the first byte in the packet, which works fine
between STUN and RTP or RTCP where the first byte happens to be
different, but may not work as well with other media transport
protocols.</t>
</section>
<section title="ALG considerations">
<t>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 public 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 update
address fields which contains addresses other than the NATs internal
address domain. In cases where the ALG modifies fields unnecessary
two alternatives exist:<list style="numbers">
<t>The usage of TLS to encrypt the RTSP TCP connection to
prevent the ALG from reading and modifying the RTSP
messages.</t>
<t>To turn off the STUN based NAT traversal mechanism</t>
</list>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.</t>
</section>
<section title="Deployment Considerations">
<t>For the non-embedded usage of STUN the following applies:</t>
<t>Advantages:</t>
<t><list style="symbols">
<t>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.</t>
<t>Using STUN does not require RTSP server modifications; it
only affects the client implementation.</t>
</list>Disadvantages:</t>
<t><list style="symbols">
<t>Requires a STUN server deployed in the public address
space.</t>
<t>Only works with NATs that perform endpoint independent and
address dependent mappings. Port and address dependent filtering
NATs create some issues.</t>
<t>Brittle to NATs changing the properties of the NAT mapping
and filtering.</t>
<t>Does not work with port and address dependent mapping NATs
without server modifications.</t>
<t>Will mostly not work if a NAT uses multiple IP addresses,
since RTSP server generally requires all media streams to use
the same IP as used in the RTSP connection to prevent becoming a
DDOS tool.</t>
<t>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.</t>
<t>Using STUN requires that RTSP servers and clients support the
updated RTSP specification, 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.</t>
</list>Transition:</t>
<t>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.</t>
<t>For the "Embedded STUN" method the following applies:</t>
<t>Advantages:</t>
<t><list style="symbols">
<t>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.</t>
<t>STUN has built-in message authentication features, which
makes it more secure. See next section for an in-depth security
discussion.</t>
<t>This solution works as long as there is only one RTSP end
point in the private address realm, regardless of the NAT's
type. There may even be multiple NATs (see figure 1 in
RFC3489).</t>
<t>Compares 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
have become available <xref target="STUN-IMPL"/>. 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.</t>
</list>Disadvantages:</t>
<t><list style="symbols">
<t>Some extensions to the RTSP core protocol, signaled by RTSP
feature tags, must be introduced.</t>
<t>Requires an embedded STUN server to co-locate on each of 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, the later to STUN
server.</t>
<t>Does not support use cases that requires the RTSP connection
and the media reception to happen at different addresses, unless
the servers sequrity policy is relaxed.</t>
<t>Interaction problems exist when a RTSP ALG is not aware of
STUN unless TLS is used to protect the RTSP messages.</t>
<t>Using STUN requires that RTSP servers and clients support the
updated RTSP specification, and they both agree to support the
NAT traversal feature.</t>
<t>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.</t>
</list>Transition:</t>
<t>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.</t>
<t/>
</section>
<section title="Security Considerations">
<t>To prevent RTSP server 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 that RTSP message originates from. The RTSP
server is only prepared to make an exception of 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 restriction means that STUN does not work for use cases where
RTSP and media transport goes to different address.</t>
<t>In terms of security property, 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 ability the to spoof the TCP connection carrying
RTSP messages.</t>
<t>Using STUN's support for message authentication and secure
transport of RTSP messages, attackers cannot modify STUN responses
or RTSP messages 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.</t>
</section>
</section>
<section title="ICE">
<section title="Introduction">
<t><xref target="RFC5245">ICE (Interactive Connectivity
Establishment)</xref> is a methodology for NAT traversal that has
been developed for SIP using SDP offer/answer. The basic idea is to
try, in a parallel fashion, all possible connection addresses that
an end point may have. This allows the end-point 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.</t>
<t>Here is how ICE works on a high level. End point A collects all
possible address 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 leads to, a STUN server is
installed. This STUN server only accepts STUN requests using the
correct authentication through the use of username and password.</t>
<t>End-point A then sends a request to establish connectivity with
end-point B, which includes all possible destinations to get the
media through too A. Note that each of A's published address/port
pairs has a STUN server co-located. B, in its 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 have successfully completed
are then indicated and selected.</t>
<t>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 are discovered and conveyed in
the STUN responses.</t>
</section>
<section title="Using ICE in RTSP">
<t>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.</t>
<t>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 been opened, or the deployment of
proxies, etc.</t>
<t>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 using a
public address and the client is either having a public address or
is behind NAT(s). This process is only intended to support PLAY
mode, i.e. media traffic flows from server to client.</t>
<t><list style="numbers">
<t>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 make a SDP indicate a resource on a
server park rather than a specific machine.</t>
<t>The client gathers addresses and puts together its candidate
for each media stream indicated in the session description.</t>
<t>In each SETUP request the client includes its candidates in a
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 rather than always successful is to recommended as
it is most likely to be a server in the public.</t>
<t>The server responds to the SETUP (200 OK) for each media
stream with its candidates. A server with a public address
usually only provides a single ICE candidate. Also here one
candidate is the server primary address.</t>
<t>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 willingly to
receive the media, thus protecting itself from performing
unknowingly an DoS attack.</t>
<t>Connectivity checks from the client promoting a candiadate
pair was successful. Thus no further SETUP requests are
necessary 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 (Goto 7). If the checks for the availble candidates
failed and If further candidates have been derived during the
connectivity checks, then those can be signalled in new
candidate lines in SETUP request updating the list (Goto 5).</t>
<t>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 (Goto 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.</t>
<t>Process completed media can be delivered. ICE candidates not
used may be released.</t>
</list></t>
<t>To keep media paths alive the client needs to periodically send
data to the server. This will be realized with STUN. RTCP sent by
client should be able to keep RTCP open but STUN will also be used
based on the same motivations as for ICE for SIP.</t>
</section>
<section title="Implementation burden of ICE">
<t>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/FW traversal methods in this document.</t>
<t>RTSP server implementation requirements are:</t>
<t><list style="symbols">
<t>STUN server features</t>
<t>limited STUN client features</t>
<t>SDP generation with more parameters.</t>
<t>RTSP error code for ICE extension</t>
</list>RTSP client implantation requirements are:</t>
<t><list style="symbols">
<t>Limited STUN server features</t>
<t>Limited STUN client features</t>
<t>RTSP error code and ICE extension</t>
</list></t>
</section>
<section title="Deployment Considerations">
<t>Advantages:</t>
<t><list style="symbols">
<t>Solves NAT connectivity discovery for basically all cases as
long as a TCP connection between them can be established. This
includes servers behind NATs. (Note that a proxy between address
domains may be required to get TCP through).</t>
<t>Improves defenses against DDOS attacks, as media receiving
client requires authentications, via STUN on its media reception
ports.</t>
</list>Disadvantages:</t>
<t><list style="symbols">
<t>Increases the setup delay with at least the amount of time it
takes for the server to perform its STUN requests.</t>
<t>Assumes that it is possible to de-multiplex between the
packets of the media protocol and STUN packets.</t>
<t>Has fairly high implementation burden put on both RTSP server
and client.</t>
</list></t>
</section>
<section title="Security Consideration">
<t>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 a correctly utilizing ICE in RTSP. In fact
ICE do help avoid the DDoS 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.</t>
</section>
</section>
<section title="Symmetric RTP">
<t/>
<section title="Introduction">
<t>Symmetric RTP 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. Symmetric RTP is similar to
connection-oriented traffic, where one side (e.g., the RTSP 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". </t>
<t>Specifically, when the RTSP server receives the "connect" RTP
packet (a.k.a. FW packet, since it is used to punch a hole in the
FW/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. However, it does require
server modifications. Unless there is built-in protection mechanism,
symmetric RTP is very vulnerable to DDOS attacks, because attackers
can simply forge the source IP & Port of the binding packet.
Using the rule for restriciting IP address to that one of the
signalling connection will need to be applied here also.</t>
</section>
<section title="Necessary RTSP extensions">
<t>To support symmetric RTP the RTSP signaling must be extended to
allow the RTSP client to indicate that it will use symmetric RTP.
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. Care must be
taken in choosing client's 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 collision may occur, the RTSP server must be able to
signal to 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 session it can simultaneously handle per
interface IP addresses.</t>
</section>
<section title="Deployment Considerations">
<t>Advantages:</t>
<t><list style="symbols">
<t>Works for all types of NATs, including those using multiple
IP addresses. (Requirement 1 in <xref
target="req-section"/>).</t>
<t>Have no interaction problems with any RTSP ALG changing the
client's information in the transport header.</t>
</list>Disadvantages:</t>
<t><list style="symbols">
<t>Requires modifications to both RTSP server and client.</t>
<t>Limited to work with servers that have an public IP
address.</t>
<t>The format of the RTP packet for "connection setup" (a.k.a FW
packet) is yet to be defined. One possibility is to use RTP
No-Op packet format in <xref
target="I-D.ietf-avt-rtp-no-op"/>.</t>
<t>Has the same security situation as STUN and will need to use
address restrictions.</t>
</list></t>
</section>
<section title="Security Consideration">
<t>Symmetric RTP's major security issue is that RTP streams can be
hijacked and directed towards any target that the attacker desires
unless address restricitons are used.</t>
<t>The most serious security problem is the deliberate attack with
the use of a RTSP client and symmetric RTP. The attacker uses RTSP
to setup a media session. Then it uses symmetric RTP with a spoofed
source address of the intended target of the attack. There is no
defense against this attack other than restricting the possible bind
address to be the same as the RTSP connection arrived on. This
prevents symmetric RTP to be used in use cases that require differet
addresses for media destination and signalling.</t>
<t>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 NAT-traversal RTP packets containing the
correct RTP SSRC to the correct address and port on the server. The
destination of the packets is set as the source IP and port in these
RTP packets.</t>
<t>Another variation of this attack is for a man in the middle to
modify the RTP binding packet being sent by a client to the server
by simply changing the source IP to the target one desires to
attack.</t>
<t>One can fend off the first attack by applying encryption to the
RTSP signaling transport. However, the second variation is
impossible to defend against. As a NAT re-writes the source IP and
port this cannot be authenticated, but authentication is required in
order to protect against this type of DOS attack.</t>
<t>Yet another issues is that these attacks also can be used to deny
the client the service he desire from the RTSP server completely.
For a man in the middle capable of reading the signalling traffic or
intercepting the binding packets can completely deny the client
service by modifying or originating binding packets of itself.</t>
<t>The random nonce used in the binding packet determines how well
symmetric RTP can fend off stream-hijacking performed by parties
that are not "man-in-the-middle". 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 randomizer. 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.</t>
<t>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 SHOULD implement functionality that blocks
ports that receive multiple FW packets (i.e. the packet that is sent
to the server for FW traversal) with different invalid SSRCs,
especially when they are coming from the same IP/Port.</t>
<t>To improve the security against attackers the random tag's length
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 tag.</t>
</section>
<section anchor="sym-rtp-var" title="A Variation to Symmetric RTP">
<t>Symmetric RTP requires a valid RTP format in the FW packet, which
is the first packet that the client sends to the server to set up
virtual RTP connection. There is currently no appropriate RTP packet
format for this purpose, although the No-Op format was a proposal to
fix the problem <xref target="I-D.ietf-avt-rtp-no-op"/>. However,
that work has been abandoned. There exists a RFC that discusses the
implication of different type of packets as keep-alives for RTP
<xref target="RFC6263"/> and its findings are very relevant to the
FW packet.</t>
<t>Meanwhile, there has been FW traversal techniques deployed in the
wireless streaming market place that use non-RTP messages as FW
packets. This section attempts to summarize a subset of those
solutions that happens to use a variation to the standard symmetric
RTP solution.</t>
<t>In this variation of symmetric RTP, the FW packet is a small UDP
packet that does not contain RTP header. Hence the solution can no
longer be called symmetric RTP, yet it employs the same technique
for FW traversal. In response to client's FW packet, RTSP server
sends back a similar FW packet as a confirmation so that the client
can stop the so called "connection phase" of this NAT traversal
technique. Afterwards, the client only has to periodically send FW
packets as keep-alive messages for the NAT mappings.</t>
<t>The server listens on its RTP-media output port, and tries to
decode any received UDP packet as FW packet. This is valid since an
RTSP server is not expecting RTP traffic from the RTSP client. Then,
it can correlate the FW packet with the RTSP client's session ID or
the client's SSRC, and record the NAT bindings accordingly. The
server then sends a FW packet as the response to the client.</t>
<t>The FW packet can contain the SSRC to identify the RTP stream,
and care must be taken if the packet is bigger than 12 bytes,
ensuring that it is distinctively different from RTP packets, whose
header size is 12 bytes.</t>
<t>RTSP signaling can be added to do the following:</t>
<t><list style="numbers">
<t>Enables or disables such FW message exchanges. When the
FW/NAT has an RTSP-aware ALG, it is possible to disable FW
message exchange and let ALG works out the address and port
mappings.</t>
<t>Configures the number of re-tries and the re-try interval of
the FW message exchanges.</t>
</list>Such FW packets may also contain digital signatures to
support three-way handshake based receiver authentications, so as to
prevent DDoS attacks described before.</t>
<t>This approach has the following advantages when compared with the
symmetric RTP approach:</t>
<t><list style="numbers">
<t>There is no need to define RTP payload format for FW
traversal, therefore it is simple to use, implement and
administer (Requirement 4 in <xref target="req-section"/>),
although a binding protocol must be defined.</t>
<t>When properly defined, this kind of FW message exchange can
also authenticate RTP receivers, so as to prevent DDoS attacks
for dual-hosted RTSP client. By dual-hosted RTSP client we mean
the kind that uses one "perceived" IP address for RTSP message
exchange, and a different "perceived" IP address for RTP
reception. (Requirement 5 in <xref target="req-section"/>).</t>
</list>This approach has the following disadvantages when compared
with the symmetric RTP approach:</t>
<t><list style="numbers">
<t>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, or use the same type of FW
messages also for RTCP endpoints.</t>
<t>The server's sender SSRC for the RTP stream must be signaled
in RTSP's SETUP response, in the Transport header of the RTSP
SETUP response.</t>
</list>A solution with a 3-way handshaking and its own FW packets
can be compared with ICE and have the following differencies:</t>
<t><list style="symbols">
<t>Only works for servers with public IP addresses compared to
any type of server</t>
<t>Is somewhat simpler to implement due to the avoidance of the
ICE prioritization and checkboard mechanisms.</t>
</list>However, a 3-way binding protocol is very similar to using
STUN in both directions as binding protocol. Using STUN would remove
the need for implementing a new protocol.</t>
</section>
</section>
<section title="Application Level Gateways">
<t/>
<section title="Introduction">
<t>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:</t>
<t><list style="numbers">
<t>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.</t>
<t>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.</t>
</list>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.</t>
</section>
<section title="Outline On how ALGs for RTSP work">
<t>In this section, we provide a step-by-step outline on how one
should go about writing an ALG to enable RTSP to traverse a NAT.</t>
<t><list style="numbers">
<t>Detect any SETUP request.</t>
<t>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, the ALG SHOULD NOT change the address. Ways to detect
that these methods are used are:<list style="symbols">
<t>For embedded STUN, it would be 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.</t>
<t>For non-embedded STUN and TURN based solutions: This can
in some case 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. However
if multiple NATs are used this detection may fail. Remapping
should only be done for addresses belonging to the NATs own
private address space.</t>
</list>Otherwise continue to the next step.</t>
<t>Create UDP mappings (client given IP/port <-> external
IP/port) where needed for all possible transport specification
in the transport header of the request found in (1). Enter the
public address and port(s) of these mappings in transport
header. Mappings SHALL be created with consecutive public port
number starting on an even number for RTP for each media stream.
Mappings SHOULD also be given a long timeout period, at least 5
minutes.</t>
<t>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.</t>
<t>If SETUP response settles on RTP over TCP or RTP over RTSP as
lower transport, do nothing: let TCP tunneling to take care of
NAT traversal. Otherwise go to next step.</t>
<t>The ALG SHOULD keep alive the UDP mappings belonging to the
an RTSP session as long as: RTSP messages with the session's ID
has been sent in the last timeout interval, or UDP messages are
sent on any of the UDP mappings during the last timeout
interval.</t>
<t>The ALG MAY remove a mapping as soon a TEARDOWN response has
been received for that media stream.</t>
</list></t>
</section>
<section title="Deployment Considerations">
<t>Advantage:</t>
<t><list style="symbols">
<t>No impact on either client or server</t>
<t>Can work for any type of NATs</t>
</list>Disadvantage:</t>
<t><list style="symbols">
<t>When deployed they are hard to update to reflect protocol
modifications and extensions. If not updated they will break the
functionality.</t>
<t>When end-to-end security is used the ALG functionality will
fail.</t>
<t>Can interfere with other type of traversal mechanisms, such
as STUN.</t>
</list>Transition:</t>
<t>An RTSP ALG will not be phased out in any automatically way. It
must be removed, probably through the removal of the NAT it is
associated with.</t>
</section>
<section title="Security Considerations">
<t>An ALG will not work when deployment of end-to-end RTSP signaling
security. Therefore deployment of ALG will likely result in that
clients located behind NATs will not use end-to-end security.</t>
</section>
</section>
<section title="TCP Tunneling">
<t/>
<section title="Introduction">
<t>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 that the client has to accept that real-time performance
may no longer be possible. TCP's problem of ensuring timely deliver
was 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.</t>
</section>
<section title="Usage of TCP tunneling in RTSP">
<t>The RTSP core specification <xref
target="I-D.ietf-mmusic-rfc2326bis"/> supports interleaving of media
data on the TCP connection that carries RTSP signaling. See section
14 in <xref target="I-D.ietf-mmusic-rfc2326bis"/> for how to perform
this type of TCP tunneling. There also exist another way of
transporting RTP over TCP defined in Appendix C.2. For signaling and
rules on how to establish the TCP connection in lieu of UDP, see
appendix C.2 in <xref target="I-D.ietf-mmusic-rfc2326bis"/>. This is
based on the framing of RTP over the TCP connection as described in
<xref target="RFC4571">RFC 4571</xref>.</t>
</section>
<section title="Deployment Considerations">
<t>Advantage: <list style="symbols">
<t>Works through all types of NATs where server is in the
open.</t>
</list></t>
<t>Disadvantage: <list style="symbols">
<t>Functionality needs to be implemented on both server and
client.</t>
<t>Will not always meet multimedia stream's real-time
requirements.</t>
</list>Transition:</t>
<t>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 price of loss
of real-time properties.</t>
</section>
<section title="Security Considerations">
<t>The TCP tunneling of RTP has no known security problem besides
those already presented in the RTSP specification. It is not
possible to get any amplification effect that is desired 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.</t>
</section>
</section>
<section title="TURN (Traversal Using Relay NAT)">
<t/>
<section title="Introduction">
<t><xref target="RFC5766">Traversal Using Relay NAT (TURN)</xref> is
a protocol for setting up traffic relays that allows 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 mapping.
On the external side this is limited to the source address/port pair
of the first packet arriving on the binding. After the first packet
has arrived the mapping is "locked down" to that address. 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 RTSP message arrives. 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 or also the RTSP connection is relayed through the
same TURN server.</t>
</section>
<section anchor="use-turn-rtsp" title="Usage of TURN with RTSP">
<t>To use a TURN server for NAT traversal, the following steps
should be performed. <list style="numbers">
<t>The RTSP client connects with 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 <xref
target="RFC6062">TURN for TCP</xref>.</t>
<t>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 continuous ranges.</t>
<t>The RTSP client uses the acquired address and port mappings
in the RTSP SETUP request using the destination header. Note
that the server is required to have a mechanism to verify that
it is allowed to send media traffic to the given address. The
server SHOULD include its RTP SSRC in the SETUP response.</t>
<t>Client requests that the Server starts playing. The server
starts sending media packet to the given destination address and
ports.</t>
<t>The first media packet to arrive at the TURN server on the
external port causes "lock down"; then TURN server forwards the
media packets to the RTSP client.</t>
<t>When media arrives at the client, the client should try to
verify that the media packets are from the correct RTSP server,
by matching the RTP SSRC of the packet. Source IP address of
this packet will be that of the TURN server and can therefore
not be used to verify that the correct source has caused lock
down.</t>
<t>If the client notices that some other source has caused lock
down on the TURN server, the client should create new bindings
and change the session transport parameters to reflect the new
bindings.</t>
<t>If the client pauses and media are not sent for about 75% of
the mapping timeout the client should use TURN to refresh the
bindings.</t>
</list></t>
</section>
<section title="Deployment Considerations">
<t>Advantages: <list style="symbols">
<t>Does not require any server modifications.</t>
<t>Works for any types of NAT as long as the server has public
reachable IP address.</t>
</list></t>
<t>Disadvantage:<list style="symbols">
<t>Requires another network element, namely the TURN server.</t>
<t>A TURN server for RTSP is may not scale since the number of
sessions it must forward is proportional to the number of client
media sessions.</t>
<t>TURN server becomes a single point of failure.</t>
<t>Since TURN forwards media packets, it necessarily introduces
delay.</t>
<t>An RTSP ALG MAY change the necessary destinations parameter.
This will cause the media traffic to be sent to the wrong
address.</t>
</list> Transition:</t>
<t>TURN is not intended to be phase-out completely, see chapter 11.2
of <xref target="RFC5766"/>. However the usage of TURN could be
reduced when the demand for having NAT traversal is reduced.</t>
</section>
<section title="Security Considerations">
<t>An eavesdropper of RTSP messages between the RTSP client and RTSP
server will be able to do a simple denial of service attack on the
media streams by sending messages to the destination address and
port present in the RTSP SETUP messages. If the attacker's message
can reach the TURN server before the RTSP server's message, the lock
down can be accomplished towards some other address. This will
result in that the TURN server will drop all the media server's
packets when they arrive. This can be accomplished with little risk
for the attacker of being caught, as it can be performed with a
spoofed source IP. The client may detect this attack when it
receives the lock down packet sent by the attacker as being
mal-formatted and not corresponding to the expected context. It will
also notice the lack of incoming packets. See bullet 7 in Section
<xref format="counter" target="use-turn-rtsp"/>.</t>
<t>The TURN server can also 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
starts 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.</t>
<t>The first attack can be made very hard by applying transport
security for the RTSP messages, which will hide the TURN servers
address and port numbers from any eavesdropper.</t>
<t>The second attack requires that the attacker have access to a
user account on the TURN server to be able set up the TURN mappings.
To prevent this attack the server shall verify that the target
destination accept this media stream.</t>
</section>
</section>
</section>
<section title="Firewalls">
<t>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.</t>
<t>The firewall will need to allow the media streams associated with a
RTSP session 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 streams will use. 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.</t>
<t>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 be able 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.</t>
</section>
<section title="Comparision of NAT traversal techniques">
<t>This section evaluates the techniques described above against the
requirements listed in section <xref target="req-section"/>.</t>
<t>In the following table, the columns correspond to the numbered
requirements. For instance, the column under R1 corresponds to the first
requirement in section <xref target="req-section"/>: MUST work for all
flavors of NATs. The rows represent the different FW traversal
techniques. SymRTP is short for symmetric RTP, "V.SymRTP" is short for
"variation of symmetric RTP" as described in section <xref
target="sym-rtp-var"/>.</t>
<t>A Summary of the requirements are:</t>
<t><list style="hanging">
<t hangText="R1">Work for all flavors of NATs</t>
<t hangText="R2">Most work with Firewalls, including them with
ALGs</t>
<t hangText="R3">Should have minimal impact on clients not behind
NATs</t>
<t hangText="R4">Should be simple to use, Implement and
administrate.</t>
<t hangText="R5">Should provide a mitigation against DDoS
attacks</t>
</list></t>
<figure>
<preamble/>
<artwork><![CDATA[-----------------------------------------------+
| R1 | R2 | R3 | R4 | R5 |
------------+------+------+------+------+------+
STUN | Yes | Yes | No | Maybe| No |
------------+------+------+------+------+------+
ICE | Yes | Yes | No | No | Yes |
------------+------+------+------+------+------+
SymRTP | Yes | Yes | Yes |Maybe | No |
------------+------+------+------+------+------+
V. SymRTP | Yes | Yes | Yes | Yes |future|
------------+------+------+------+------+------+
3-W SymRTP | Yes | Yes | Yes | Maybe| Yes |
------------+------+------+------+------+------+
TURN | Yes | Yes | No | No | Yes |
------------------------------------------------
]]></artwork>
<postamble/>
</figure>
<t anchor="Comparision-table">The different techniques was discussed in
the MMUSIC WG. It was established that the WG would pursue an ICE based
solution due to its generality and capability of handle also servers
delivering media from behind NATs. There has been some discussion if the
increased implementation burden of ICE is motivated compared to a 3-W
SymRTP solution for this generality.</t>
<t>The ICE based RTSP NAT traversal solution is specified in <xref
target="I-D.ietf-mmusic-rtsp-nat">"A Network Address Translator (NAT)
Traversal mechanism for media controlled by Real-Time Streaming Protocol
(RTSP)"</xref>.</t>
</section>
<section anchor="IANA" title="IANA Considerations">
<t>This document makes no request of IANA.</t>
<t>Note to RFC Editor: this section may be removed on publication as an
RFC.</t>
</section>
<section anchor="Security" title="Security Considerations">
<t>In preceding sessions we have discussed security merits of each and
every NAT/FW 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.</t>
<t>Each of the proposed solutions has security implications. Using STUN
will provide the same level of security as RTSP with out transport level
security and source authentications; as long as the server does not
grant a client request to send media to different IP addresses. Using
symmetric RTP 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 tag that the client uses to
prove its identity when creating the address bindings. This can be
solved in the variation of symmetric RTP (section 6.3.5) with
authentication features. The usage of an RTSP ALG does not increase in
itself the risk for session hijacking. However the deployment of ALGs as
sole mechanism for RTSP NAT traversal will prevent deployment of
encrypted end-to-end 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 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.</t>
</section>
<section anchor="Acknowledgements" title="Acknowledgements">
<t>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, and Colin Perkins. Thomas Zeng would also like to
give special thanks to Greg Sherwood of PacketVideo for his input into
this memo.</t>
<t>Section <xref target="sec-nat-intro"/> contains text originally
written for RFC 4787 by Francois Audet and Cullen Jennings.</t>
</section>
</middle>
<back>
<references title="Informative References">
<?rfc include="reference.RFC.2119"?>
<?rfc include='reference.RFC.0768'?>
<?rfc include='reference.RFC.0793'?>
<?rfc include='reference.RFC.2326'?>
<?rfc include='reference.RFC.3550'?>
<?rfc include='reference.RFC.3489'?>
<?rfc include='reference.RFC.3022'?>
<?rfc include='reference.RFC.3424'?>
<?rfc include='reference.RFC.2588'?>
<?rfc include='reference.RFC.2663'?>
<?rfc include='reference.RFC.4787'?>
<?rfc include='reference.RFC.4566'?>
<?rfc include='reference.I-D.ietf-mmusic-rfc2326bis'?>
<?rfc include='reference.RFC.5766'?>
<?rfc include='reference.RFC.4571'?>
<?rfc include='reference.I-D.ietf-avt-rtp-no-op'?>
<?rfc include='reference.RFC.5389'?>
<?rfc include='reference.RFC.5245'?>
<?rfc include='reference.RFC.6263'?>
<?rfc include='reference.RFC.6062'?>
<?rfc include='reference.I-D.ietf-mmusic-rtsp-nat'?>
<reference anchor="STUN-IMPL">
<front>
<title>Open Source STUN Server and Client,
http://www.vovida.org/applications/downloads/stun/index.html</title>
<author>
<organization/>
</author>
<date day="26" month="June" year="2007"/>
</front>
</reference>
</references>
</back>
</rfc>
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