wdiff draft-ietf-mmusic-rtsp-nat-evaluation-05.txt draft-ietf-mmusic-rtsp-nat-evaluation-06.txt

Network Working Group M. Westerlund
Internet-Draft Ericsson
Intended status: Informational T. Zeng
Expires: November 8, 2012 May 7, 27, 2013 November 23, 2012

The Evaluation of Different Network Addres Address Translator (NAT) Traversal
Techniques for Media Controlled by Real-time Streaming Protocol (RTSP)
draft-ietf-mmusic-rtsp-nat-evaluation-05
draft-ietf-mmusic-rtsp-nat-evaluation-06

Abstract

This document describes several Network Address Translator (NAT)
traversal techniques that was were considered to be used for establishing
the RTP media flows controlled by the 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 discussions 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. 2.0.

Status of this Memo

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Table of Contents

1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 5
1.1. Network Address Translators . . . . . . . . . . . . . . . 6
1.2. Firewalls . . . . . . . . . . . . . . . . . . . . . . . . 7
1.3. Glossary . . . . . . . . . . . . . . . . . . . . . . . . . 7
1.4. Definitions . . . . . . . . . . . . . . . . . . . . . . . 8
2. Detecting the loss of NAT mappings . . . . . . . . . . . . . . 8
3. Requirements on NAT-Traversal . . . . . . . . . . . . . . . . 9
4. NAT Traversal Techniques . . . . . . . . . . . . . . . . . . . 10
4.1. Stand-Alone STUN . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
4.1.1. Introduction . . . . . . . . . . . . . . . . . . . . . 11
4.1.2. Using STUN to traverse NAT without server
modifications . . . . . . . . . . . . . . . . . . . . 11
4.1.3. Embedding STUN in RTSP ALG considerations . . . . . . . . . . . . . . . . 13
4.1.4. Discussion On Co-located STUN Server . . 14
4.1.4. Deployment Considerations . . . . . . . 14
4.1.5. ALG considerations . . . . . . . 14
4.1.5. Security Considerations . . . . . . . . . . . 15
4.1.6. Deployment Considerations . . . . 15
4.2. Server Embedded STUN . . . . . . . . . . 15
4.1.7. Security Considerations . . . . . . . . . 16
4.2.1. Introduction . . . . . . 17
4.2. ICE . . . . . . . . . . . . . . . 16
4.2.2. Embedding STUN in RTSP . . . . . . . . . . . . 18
4.2.1. Introduction . . . . 16
4.2.3. Discussion On Co-located STUN Server . . . . . . . . . 17
4.2.4. ALG considerations . . . . . . . . 18
4.2.2. Using ICE in RTSP . . . . . . . . . . 17
4.2.5. Deployment Considerations . . . . . . . . 19
4.2.3. Implementation burden of ICE . . . . . . 17
4.2.6. Security Considerations . . . . . . . 20
4.2.4. Deployment Considerations . . . . . . . . 19
4.3. ICE . . . . . . 21
4.2.5. Security Consideration . . . . . . . . . . . . . . . . 21
4.3. Symmetric RTP . . . . . 19
4.3.1. Introduction . . . . . . . . . . . . . . . . . 22
4.3.1. Introduction . . . . 19
4.3.2. Using ICE in RTSP . . . . . . . . . . . . . . . . . 22
4.3.2. Necessary RTSP extensions . 20
4.3.3. Implementation burden of ICE . . . . . . . . . . . . . 22
4.3.3. 21
4.3.4. Deployment Considerations . . . . . . . . . . . . . . 23
4.3.4. 22
4.3.5. Security Consideration . . . . . . . . . . . . . . . . 23
4.3.5. A Variation to Symmetric RTP . . . . 22
4.4. Latching . . . . . . . . . 24
4.4. Application Level Gateways . . . . . . . . . . . . . . . . 26 22
4.4.1. Introduction . . . . . . . . . . . . . . . . . . . . . 26 22
4.4.2. Outline On how ALGs for Necessary RTSP work extensions . . . . . . . . . . 27 . . . . 23
4.4.3. Deployment Considerations . . . . . . . . . . . . . . 28 23
4.4.4. Security Considerations . . . . Consideration . . . . . . . . . . . 28
4.5. TCP Tunneling . . . . . 24
4.5. A Variation to Latching . . . . . . . . . . . . . . . . . 28 25
4.5.1. Introduction . . . . . . . . . . . . . . . . . . . . . 28 25
4.5.2. Usage of TCP tunneling in Necessary RTSP extensions . . . . . . . . . . . . 29 . . 26
4.5.3. Deployment Considerations . . . . . . . . . . . . . . 29 27
4.5.4. Security Considerations . . . . . . . . . . . . . . . 29 27
4.6. TURN (Traversal Using Relay NAT) Three Way Latching . . . . . . . . . . . . . 30
4.6.1. Introduction . . . . . . . 27
4.6.1. Introduction . . . . . . . . . . . . . . 30 . . . . . . . 28
4.6.2. Usage of TURN with Necessary RTSP extensions . . . . . . . . . . . . . . . 30 28
4.6.3. Deployment Considerations . . . . . . . . . . . . . . 31
4.6.4. 28
4.7. Application Level Gateways . . . . . . . . . . . . . . . . 28
4.7.1. Introduction . . . . . . . . . . . . . . . . . . . . . 29
4.7.2. Outline On how ALGs for RTSP work . . . . . . . . . . 29
4.7.3. Deployment Considerations . . . . . . . . . . . . . . 30
4.7.4. Security Considerations . . . . . . . . . . . . . . . 32
5. Firewalls 31

4.8. TCP Tunneling . . . . . . . . . . . . . . . . . . . . . . 31
4.8.1. Introduction . . . . . 33
6. Comparision . . . . . . . . . . . . . . . . 31
4.8.2. Usage of NAT traversal techniques TCP tunneling in RTSP . . . . . . . . . . . 33
7. IANA . 31
4.8.3. Deployment Considerations . . . . . . . . . . . . . . 32
4.8.4. Security Considerations . . . . . . . . . . . . . . . 32
4.9. TURN (Traversal Using Relay NAT) . . . . . . . . . . . . . 32
4.9.1. Introduction . . . . . . . . . . . . . . . . . . . . . 32
4.9.2. Usage of TURN with RTSP . . . . . . . . . . . . . . . 33
4.9.3. Deployment Considerations . . . . . . . . . . . . . . 34
8.
4.9.4. Security Considerations . . . . . . . . . . . . . . . 35
5. Firewalls . . . . 34
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 35
10. Informative References
6. Comparison of NAT traversal techniques . . . . . . . . . . . . 36
7. IANA Considerations . . . . . . . . 35
Authors' Addresses . . . . . . . . . . . . . 38
8. Security Considerations . . . . . . . . . . . 37

1. . . . . . . . . 38
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 39
10. Informative References . . . . . . . . . . . . . . . . . . . . 39
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 41

1. Introduction

Today there is a proliferate deployment of different flavors of
Network Address Translator (NAT) boxes that in many cases only
loosely follows follow standards [RFC3022][RFC2663][RFC3424]].
[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, may is 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. NAT-traversal.

Like NATs, firewalls (FWs) Firewalls 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.

This document describes several NAT-traversal mechanisms for RTSP
controlled media streaming. These 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 FW/NAT 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]. Where [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.

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

Readers are urged to refer to "IP Network Address Translator (NAT)
Terminology and Considerations" [RFC2663] for information on NAT
taxonomy and terminology. Traditional NAT is the most common type of
NAT device deployed. Readers may refer to "Traditional IP Network
Address Translator (Traditional NAT)" [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 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)".

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
"Network Address Translation (NAT) Behavioral Requirements for
Unicast UDP" [RFC4787].

NATs also have a filtering behavior on traffic arriving on the
external side. Such behavior effects 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.

1.2. Firewalls

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 [RFC2588].

A comparison between NAT and FW Firewall is given below:

1. A firewall must sit between two network administrative domains,
while NAT does not have to sit between two 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/home office (SOHO) use include both NATs and firewall
functionality.

In the rest of this memo we use the phrase "NAT traversal"
interchangeably with "FW "Firewall traversal", "NAT/FW traversal" and "NAT/
Firewall "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].

DNS: Domain Name Service

DDOS:

DoS: Denial of Service

DDoS: Distributed Denial Of of Service attacks

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].

1.4. Definitions

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 RFC
2119 [RFC2119].

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, 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.

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, NAT/Firewall, or that the client is mal-functioning.

3. Requirements on NAT-Traversal

This section considers the set of requirements for the evaulation evaluation of
RTSP NAT traversal solutions.

RTSP is a client-server protocol. Typically services service 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.

The list of feature requirements for RTSP NAT solutions are given
below:

1. MUST Must work for all flavors of NATs, including NATs with address
and port restricted filtering.

2. MUST Must work for firewalls (subject to pertinent firewall
administrative policies), including those with ALGs.

3. SHOULD Should have minimal impact on clients in the open and 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

4. SHOULD Should be simple to use/implement/administer that so people actually
turn them on

* Otherwise people will resort to TCP tunneling through NATs

* Address discovery for NAT traversal should take place behind
the scene, automatically,
if possible

5. SHOULD Should authenticate dual-hosted client transport handler to
prevent DDOS DDoS attacks.

The last requirement addresses the Distributed Denial-Of-Service
(DDOS) 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 (address and port) for the 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 DDoS threat
that has significant amplification potentials because the RTP media
streams in general consist of large number of IP packets. DDOS 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 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 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 With the increased deployment of NAT
traversal process. Message authentication is middleboxes
by operators, i.e. carrier grade NAT (CGN), the reusing of a big challenge in public
IP address for many customers reduces the
current wired and wireless networking environment. It may be
necessary protection provided. Also
in the immediate future to deploy NAT traversal solutions
that do not some applications (e.g., centralized conferencing), dual-hosted
RTSP/RTP clients have full message authentication, but provide upgrade
path valid use cases. The key is how to add authentication features in
authenticate the messages exchanged during the future. NAT traversal process.

4. NAT Traversal Techniques

There exist 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 cost is costs are 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.

The main evaluation was done prior to 2007 and are based on what was
available then. 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 not a real
barrier against doing an evaluation of the NAT traversal technique.
Some other techniques have been recommended against or are currently (at the time of writing) in a state of
flux no longer
possible due to ongoing standardization work on these techniques works' outcome or has
been not been progressed their failure to
progress within IETF after the intiial 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

STUN - "Simple

Session Traversal of UDP Through Network Address Translators"
[RFC3489][RFC5389] 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 himself 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 requests messages - Binding Requests, sent over UDP, Requests
and Shared Secret Requests, sent
over TLS over TCP. 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. 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 performing
Endpoint Independent perform "Endpoint-
Independent" or Address Dependent "Address-Dependent" Mappings. Clients behind NATs
that do Address "Address and Port Dependent Port-Dependent" Mappings cannot use this
method. See [RFC4787] for full definition of these terms.

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 recommend 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 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 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-continuous 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 parameter"
"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 should periodically send
UDP traffic over all mappings needed for the session. For the
mapping carrying RTCP traffic the periodic RTCP traffic may be are
likely enough. For mappings carrying RTP traffic and for
mappings carrying RTCP packets at too low a frequency, keep-alive
messages
SHOULD 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
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 FW 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 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 hole-punching packets should be retransmitted a reasonable
number of times.

For a 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, FW 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 FW hole punching packets
MUST 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 also brittle because it relies on that assumes one can use STUN to
classify the NAT behavior. 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. Embedding STUN in ALG considerations

If a NAT supports RTSP

This section outlines the adaptation ALG (Application Level Gateway) and embedding is not
aware of the STUN within
RTSP. This enables STUN to be used to traverse any type of NAT,
including symmetric NATs. This would require protocol changes.

This NAT traversal solution has limitations:

1. It does not work if both RTSP option, service failure may happen,
because a 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 discovers its public IP address and port numbers,
and inserts them in its SETUP requests. When the client RTSP
requests.

Deviations from STUN as defined ALG processes
the SETUP request it may change the destination and port number,
resulting in RFC 3489: 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 unnecessarily
two alternatives exist:

1. We allow RTSP applications Use TLS to have encrypt the option to perform STUN
"Shared Secret Request" through RTSP, via extension RTSP TCP connection to RTSP; prevent the ALG
from reading and modifying the RTSP messages.

2. We require Turn off the STUN server to based NAT traversal mechanism

As it may be co-located on RTSP server's media
output ports.

In order difficult to allow binding discovery without authentication, determine why the STUN
server embedded in failure occurs, the usage
of TLS protected RTSP application must ignore authentication tag,
and message exchange at all times would avoid this
issue.

4.1.4. Deployment Considerations

For the Stand-Alone usage of STUN client embedded in RTSP application must use dummy
authentication tag.

If the following applies:

Advantages:

o STUN server is co-located with RTSP server's media output port, an
RTSP client using RTP transport over UDP a solution first used by SIP applications. As shown
above, with little or no changes, the RTSP application can use re-use
STUN to traverse ALL
types of NATs. In as a NAT traversal solution, avoiding the case pit-fall of port and address dependent mapping
NATs, the party inside the NAT must initiate UDP traffic. The STUN
Bind Request, being solving
a UDP packet itself, can serve as the traffic
initiating packet. Subsequently, both the problem twice.

o Using STUN Binding Response
packets and the RTP/RTCP packets can traverse the NAT, regardless of
whether the does not require RTSP server or modifications; it only
affects the RTSP client is behind NAT.

Likewise, if an RTSP server is behind implementation.

Disadvantages:

o Requires a NAT, then an embedded STUN server must co-locate on the RTSP client's RTCP port. Also it will
become deployed in the client that needs to disclose his destination public address
rather than the server so space.

o Only works with NATs that the server correctly can determine its
NAT external source perform endpoint independent and address for the media streams. In this case, we
assume that the client has
dependent mappings. Address and Port-Dependent filtering NATs
create some means of establishing TCP connection issues.

o Brittle to NATs changing the properties of the RTSP server behind NAT so as to exchange RTSP messages mapping and
filtering.

o Does not work with
the port and address dependent mapping NATs without
server modifications.

o Will mostly not work if a NAT uses multiple IP addresses, since
RTSP server.

To minimize delay, we servers generally require that all media streams to use the same
IP as used in the RTSP server supporting this
option must inform its client connection to prevent becoming a DDoS tool.

o Interaction problems exist when a RTSP-aware ALG interferes with
the RTP use of STUN for NAT traversal unless TLS secured RTSP message
exchange is used.

o Using STUN requires that RTSP servers and RTCP ports from where clients support the
server intend
updated RTSP specification [I-D.ietf-mmusic-rfc2326bis], because
it is no longer possible to send out guarantee that RTP and RTCP packets, respectively. This
can be done ports are
adjacent to each other, as required by using the "server_port" parameter in RFC2326, "client_port" and the
"src_addr" parameter in [I-D.ietf-mmusic-rfc2326bis]. Both are
"server_port" parameters in RFC2326.

Transition:

The usage of STUN can be phased out gradually as the RTSP Transport header. But in general this strategy will require
that one first do one SETUP request per media to learn the step of a
STUN capable server
ports, then perform or client should be to check the presence of
NATs. The removal of STUN checks, followed by a subsequent SETUP
to change capability in the client port and destination address implementations
will have to what was learned
during the STUN checks. wait until there is absolutely no need to use STUN.

4.1.5. Security Considerations

To be certain that RTCP works correctly prevent the RTSP end-point (server or
client) will be required to send and receive RTCP packets server from being used as Denial of Service (DoS)
attack tools the
same port.

4.1.4. Discussion On Co-located STUN Server

In order RTSP Transport header parameter "destination" and
"dest_addr" are generally not allowed to use STUN point to traverse "address and port dependent"
filtering or mapping NATs any IP address
other than the STUN server needs to be co-located with one the streaming RTSP message originates from. The RTSP 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 only prepared to make an exception to this rule when the first byte in client is
trusted (e.g., through the packet, which works fine
between STUN and RTP use of a secure authentication process, or RTCP where
through some secure method of challenging the first byte happens destination to be
different, but may verify
its willingness to accept the RTP traffic). Such a restriction means
that STUN in general does not work as well with other media transport
protocols.

4.1.5. ALG considerations

If a NAT supports for use cases where RTSP ALG (Application Level Gateway) and is not
aware of the media
transport go to different addresses.

STUN traversal option, service failure may happen,
because a client discovers its public IP combined with destination address and port numbers,
and inserts them in its SETUP requests, when the restricted RTSP ALG processes has the SETUP request it may change same
security properties as the destination and port number,
resulting in unpredictable behavior. An ALG should not update
address fields which contains addresses other than core RTSP. It is protected from being
used as a DoS attack tool unless the NATs internal
address domain. In cases where attacker has the ALG modifies fields unnecessary
two alternatives exist:

1. The usage of TLS ability to encrypt
spoof the RTSP TCP connection to prevent
the ALG from reading and modifying the carrying RTSP messages.

2. To 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

Using STUN's support for message exchange at all times would avoid this
issue.

4.1.6. Deployment Considerations

For the non-embedded usage authentication and secure transport
of RTSP messages, attackers cannot modify STUN the following applies:

Advantages:

o STUN is a solution first used by SIP applications. As shown
above, with little responses or no changes, RTSP application can re-use STUN
messages (TLS) to change media destination. This protects against
hijacking, however as a NAT traversal solution, avoiding client can be the pit-fall initiator of solving a
problem twice.

o Using STUN does not require an attack,
these mechanisms cannot securely prevent RTSP server modifications; it only
affects servers being used as
DoS attack tools.

4.2. Server Embedded STUN

4.2.1. Introduction

This Section describes an alternative to the client implementation.

Disadvantages:

o Requires a stand-alone STUN server deployed usage
in the public address space.

o Only works with NATs previous section that perform endpoint independent and address
dependent mappings. Port has quite significantly different
behavior.

4.2.2. Embedding STUN in RTSP

This section outlines the adaptation and address dependent filtering NATs
create some issues.

o Brittle embedding of STUN within
RTSP. This enables STUN to NATs changing the properties be used to traverse any type of the NAT mapping and
filtering.

o Does not work with port and NAT,
including address dependent and Port-Dependent mapping NATs without
server modifications.

o Will mostly NATs. This would
require RTSP level protocol changes.

This NAT traversal solution has limitations:

1. It does not work if a NAT uses multiple IP addresses, since both RTSP client and RTSP server generally requires all 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.

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 streams to use learn the same
IP as used in server
ports, then perform the RTSP connection to prevent becoming a DDOS tool.

o Interaction problems exist when STUN checks, followed by a RTSP-aware ALG interferes with subsequent SETUP
to change the client port and destination address to what was learned
during the use of STUN for NAT traversal unless TLS secured RTSP message
exchange is used.

o Using STUN requires checks.

To be certain that RTSP servers and clients support RTCP works correctly the
updated RTSP specification, because it is no longer possible end-point (server or
client) will be required to
guarantee that RTP send and receive RTCP ports are adjacent to each other, as
required by packets from the "client_port"
same port.

4.2.3. Discussion On Co-located STUN Server

In order to use STUN to traverse "address and "server_port" parameters in
RFC2326.

Transition:

The usage of port dependent"
filtering or mapping NATs the STUN can server needs to be phased out gradually as co-located with
the first step of a
STUN capable streaming server or client should media output ports. This creates a de-
multiplexing problem: we must be able to check the presence of
NATs. differentiate a STUN packet
from a media packet. This will be done based on heuristics. The removal of
existing STUN capability heuristics is the first byte in the client implementations
will have packet and the
Magic Cookie field (added in RFC5389), which works fine between STUN
and RTP or RTCP where the first byte happens to wait until there is absolutely no need be different. Thanks
to use STUN. 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. secure against hi-jacking attacks. See next section for an
in-depth security discussion.

o This solution works as long as there is only one RTSP end point endpoint in
the private address realm, regardless of the NAT's type. There
may even be multiple NATs (see figure Figure 1 in RFC3489). [RFC5389]).

o Compares 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 have become available [STUN-IMPL]. There is the
need to embed STUN in RTSP server and client, which require a de-multiplexer 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 co-locate 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 later latter to the
STUN server.

o Does not support use cases that requires require the RTSP connection and
the media reception to happen at different addresses, unless the
servers sequrity
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, 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.

4.1.7. Security Considerations

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.

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.

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
STUN capable machine can be to check the initiator presence of an attack,
these mechanisms cannot securely prevent RTSP servers being NATs for the
presently used as
DoS attack tools.

4.2. network connection. The removal of STUN capability in
the client implementations will have to wait until there is
absolutely no need to use STUN.

4.2.6. Security Considerations

See Stand-Alone STUN (Section 4.1.5).

4.3. ICE

4.2.1.

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 end point endpoint may have. be
reachable by. This allows the end-point 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 on at a high level. End point A collects all
possible address 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 leads 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 "destinations" [RFC5245] to get
the media through too to A. Note that each of A's published local address/port
pairs (host candidates and server reflexive base) has a STUN server
co-located. B, 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 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.

4.2.2.

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.

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 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 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.

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 make a have the SDP indicate a resource on a
server park rather than a specific machine.

2. The client gathers addresses and puts together its candidate candidates for
each media stream indicated in the session description.

3. In each SETUP request the client includes its candidates in a 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 rather than always successful is to recommended recommended, as it
is most likely to be a server in the public.

4. 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.

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 willingly to
receive the media, thus protecting itself from performing
unknowingly an DoS attack.

6. Connectivity checks from the client promoting a candiadate candidate pair
was
were 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 (Go to 7). If
the checks for the availble available candidates failed and If 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 (Goto (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 (Goto (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 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 client
should be able to keep RTCP open but STUN will also be used based on
the same motivations as for ICE for SIP.

4.2.3.

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/FW 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 implantation implementation requirements are:

o Limited STUN server features

o Limited STUN client features

o RTSP error code and ICE extension

4.2.4.

4.3.4. Deployment Considerations

Advantages:

o 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).

o Improves defenses against DDOS DDoS attacks, as 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.

4.2.5.

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 a correctly utilizing using ICE in RTSP. In fact ICE do 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.

4.3. Symmetric RTP

4.3.1.

4.4. Latching

4.4.1. Introduction

Symmetric RTP

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. Symmetric RTP Latching 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". It requires that all RTP/RTCP transport is
done symmetrical, i.e. Symmetric RTP [RFC4961].

Specifically, when the RTSP server receives the "connect" RTP latching packet
(a.k.a. FW hole-punching packet, since it is used to punch a hole in the FW/NAT
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. NATs, given that the server is
not behind a NAT. However, it does require server modifications.
Unless there is built-in protection mechanism,
symmetric RTP latching is very
vulnerable to DDOS DDoS attacks, because attackers can simply forge the
source IP & Port of the binding latching packet. Using the rule for restriciting
restricting IP address to that the one of the signalling signaling connection will
need to be applied here also.

4.3.2. 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 symmetric RTP Latching, the RTSP signaling must be extended to allow the
RTSP client to indicate that it will use symmetric RTP. 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. attacks or simply avoid mis-association
when multiple clients are behind the same NAT. Care must be taken in
choosing client's 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 collision 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 session sessions it can simultaneously handle per
interface IP addresses.

4.3.3.

4.4.3. Deployment Considerations

Advantages:

o Works for all types of NATs, including those using multiple IP
addresses. client-facing NATs. (Requirement 1 in
Section 3).

o Have 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 have an public IP address.

o The format of the RTP packet for "connection setup" (a.k.a FW
Latching packet) is yet to be defined. One possibility is to use
RTP No-Op packet format in [I-D.ietf-avt-rtp-no-op].

o Has SSRC management if RTP is used for latching due to risk for mis-
association of clients to RTSP sessions at the same server if SSRC
collision occurs.

o Has worse security situation as than STUN due to lack of STUN message
authentication and will need to use address restrictions.

4.3.4.

4.4.4. Security Consideration

Symmetric RTP's

Latching's major security issue is that RTP streams can be hijacked
and directed towards any target that the attacker desires unless
address restricitons 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 symmetric RTP. Latching. The attacker uses RTSP to setup a
media session. Then it uses symmetric RTP 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 bind address a latching
packet can come from to be the same as the RTSP TCP connection arrived on. are from.
This prevents symmetric RTP Latching to be used in use cases that require differet different
addresses for media destination and signalling.

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 Latching packets containing the correct RTP
SSRC to the correct address and port on the server. The
destination of the packets is set as RTSP server
will then use the source IP and port in these
RTP packets. 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 binding 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 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 DoS attack.

Yet another issues is that these attacks also can be used to deny the
client the service he desire it desires from the RTSP server completely. For a
man in the middle capable of reading the signalling signaling traffic or
intercepting the binding latching packets can completely deny the client
service by modifying or originating binding latching packets of itself.

The amount of random nonce used non-guessable material in the binding latching packet
determines how well
symmetric RTP Latching 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. 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 SHOULD could implement functionality that blocks packets
to ports or from sources that receive or send multiple FW Latching
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. 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 tag's length
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 tag.

4.3.5. material.

4.5. A Variation to Symmetric RTP

Symmetric RTP Latching

4.5.1. Introduction

Latching as described above requires the usage of a valid RTP format in
as the FW Latching packet, which is i.e. the first packet that the client sends
to the server to set up virtual RTP connection. There is currently existed no
appropriate RTP packet format for this purpose, although the No-Op
format was a proposal to fix the problem [I-D.ietf-avt-rtp-no-op].
However, that work has
been 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 FW format of the
Latching packet.

Meanwhile, there has been FW NAT/Firewall traversal techniques deployed
in the wireless streaming market place that use non-RTP messages as FW
Latching packets. This section attempts to summarize describes a variant based on a subset
of those solutions that happens to use a variation to alters the standard symmetric
RTP previously described Latching
solution.

4.5.2. Necessary RTSP extensions

In this variation of symmetric RTP, Latching, the FW Latching packet is a small UDP
packet that does not contain an 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
Latching packet, the RTSP server sends back a similar FW Latching 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 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 FW Latching packet. This is valid since an
RTSP server is not expecting RTP traffic from the RTSP client. Then,
it can correlate the FW 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 FW Latching packet as the response to the
client.

The FW Latching 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.

RTSP signaling can be added to do the following:

1. Enables Enable or disables disable such FW Latching message exchanges. When the FW/NAT
Firewall/NAT has an RTSP-aware ALG, it is possible to disable FW
Latching message exchange and let the ALG works work out the address
and port mappings.

2. Configures Configure the number of re-tries and the re-try interval of the
FW
Latching message exchanges.

Such FW packets may also contain digital signatures to support three-
way handshake based receiver authentications, so as to prevent DDoS
attacks described before.

4.5.3. Deployment Considerations

This approach has the following advantages when compared with the
symmetric RTP approach:
Latching approach (Section 4.4):

1. There is no need to define RTP payload format for FW Firewall
traversal, therefore it is simple to use, implement and
administer (Requirement 4 in Section 3), although instead a binding Latching
protocol must be defined.

2. When properly defined, this kind of FW message Latching packet 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 Section 3). receivers, to prevent hijacking attacks.

This approach has the following disadvantages when compared with the
symmetric RTP
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 FW messages Latching packets also for RTCP
endpoints.

2. The server's sender SSRC for the RTP stream must be signaled in
RTSP's SETUP response, in 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 unlikelier 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 an authentication mechanism are 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 messages is added to the exchange. The
client acknowledges the Transport header reception of the Latching confirmation
message and echoes back the server's nonce. Thus confirming that the
Latched address goes to a RTSP SETUP
response. 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 handshaking handshake and its own FW Latching packets can be
compared with ICE the ICE-based solution (Section 4.3) and have the
following differencies: differences:

o Only works for servers with public IP addresses compared to any
type of server

o Is somewhat May be simpler to implement due to the avoidance of the ICE
prioritization and checkboard check-board mechanisms.

However, a 3-way binding Latching protocol is very similar to using STUN in
both directions as binding Latching and verification protocol. Using STUN
would remove the need for implementing a new protocol.

4.4.

4.7. Application Level Gateways

4.4.1.
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 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.

4.4.2.

4.7.2. Outline On how ALGs for RTSP work

In this section, we provide a step-by-step outline on how one should 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 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 non-embedded stand alone 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. address then such a solution is in
use. However if multiple NATs are used this detection may
fail. Remapping should only be done for addresses belonging
to the NATs 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 specification specifications in the
transport header of the request found in (1). Enter the public external
address and port(s) of these mappings in transport header.
Mappings SHALL shall be created with consecutive public port number numbers
starting on an even number for RTP for each media stream.
Mappings SHOULD should also be given a long timeout period, at least 5
minutes.

4. When the SETUP response is received from the server server, the ALG MAY may
remove the unused UDP mappings, i.e. the ones not present in the
transport header. The session ID SHOULD 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 to take care of NAT
traversal. Otherwise go to next step.

6. The ALG SHOULD should keep alive the UDP mappings belonging to the an RTSP
session as long as: an RTSP messages with the session's ID has
been sent in the last timeout interval, or a UDP messages are has
been sent on any of the UDP mappings during the last timeout
interval.

7. The ALG MAY may remove a mapping as soon a TEARDOWN response has been
received for that media stream.

4.4.3.

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 used, the ALG functionality will fail.

o Can interfere with other type types of traversal mechanisms, such as
STUN.

Transition:

An RTSP ALG will not be phased out in any automatically automatic way. It must be
removed, probably through the removal of the NAT it is associated
with.

4.4.4.

4.7.4. Security Considerations

An ALG will not work when whit 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.

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.

An NAT with RTSP ALG that assigns mappings based on SETUP requests
could potentially become victim of an 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 signaling
security. Therefore deployment of ALG will likely result in that
clients located behind NATs will not use end-to-end security.

4.5. ALG.

4.8. TCP Tunneling

4.5.1.

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 that the
client has having to accept that real-time performance may no longer can be
possible. impacted.
TCP's problem of ensuring timely deliver delivery was one of the reasons why
RTP was developed. Problems that arise with TCP are: head-of-
line head-of-line
blocking, delay introduced by retransmissions, highly varying rate
due to the congestion control algorithm.

4.5.2. If sufficient amount of
buffering (several seconds) in the receiving client can be tolerated
then TCP clearly can work.

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 exist exists another way of
transporting RTP over TCP defined in Appendix C.2. 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].

4.5.3.

4.8.3. Deployment Considerations

Advantage:

o Works through all types of NATs where the RTSP server is in the open. 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.

4.5.4.

4.8.4. Security Considerations

The TCP tunneling of RTP has no known security problem problems 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.

4.6.

4.9. TURN (Traversal Using Relay NAT)

4.6.1.

4.9.1. Introduction

Traversal Using Relay NAT (TURN) [RFC5766] is a protocol for setting
up traffic relays that allows 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
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 the
RTSP message arrives. 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 or also
the RTSP connection is relayed through the same TURN server.

4.6.2.

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 continuous ranges.

3. 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
should include its RTP SSRC in the SETUP response.

4. Client The client requests that the Server server starts playing. The server
starts sending media packet packets to the given destination address and
ports.

5. 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.

6. 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 The 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.

7. 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.

8. If the client pauses and media are is not sent for about 75% of the
mapping timeout the client should use TURN to refresh the
bindings.

4.6.3.

4.9.3. Deployment Considerations

Advantages:

o Does not require any server modifications.

o Works for any types type of NAT as long as the RTSP server has public
reachable IP address.

Disadvantage:

o Requires another network element, namely the TURN server.

o 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.

o TURN server becomes a single point of failure.

o Since TURN forwards media packets, it necessarily introduces
delay.

o An RTSP ALG MAY may change the necessary destinations parameter. This
will cause the media traffic to be sent to the wrong address.

Transition:

TURN is not intended to be phase-out phased-out completely, see chapter 11.2 Section 19 of
[RFC5766]. However the usage of TURN could be reduced when the
demand for having NAT traversal is reduced.

4.6.4.

4.9.4. Security Considerations

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 dropping 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 mal-formed and not corresponding
to the expected context. It will also notice the lack of further
incoming packets. See bullet 7 in Section 4.6.2. 4.9.2.

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 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 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.

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.

The second attack requires that the attacker have 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 shall 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 where, the media streams stream packets will use. 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.

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.

6. Comparision Comparison of NAT traversal techniques

This section evaluates the techniques described above against the
requirements listed in section 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 Section 3: MUST must work for all flavors of NATs.
The rows represent the different FW NAT/Firewall traversal techniques.
SymRTP
Latch is short for symmetric RTP, "V.SymRTP" Latching, "V. Latch" is short for "variation of symmetric RTP"
Latching" as described in section Section 4.3.5. 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 Most

R2: Must work with Firewalls, including them those with ALGs

R3: Should have minimal impact on clients not behind NATs

R4 NATs, counted
in minimal number of additional RTTs

R4: Should be simple to use, Implement and administrate.

R5 administer.

R5: Should provide a mitigation against DDoS attacks

-----------------------------------------------+

The following considerations are also added to requirements:

C1: Will solution support both Clients and Servers behind NAT

C2: How Robust is the solution to changing NAT behaviors

------------+------+------+------+------+------+------+------+
| R1 | R2 | R3 | R4 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 | R5
------------+------+------+------+------+------+------+------+
V. Latch |
------------+------+------+------+------+------+
STUN Yes | Yes | 1 | Yes | No | Maybe| No |
------------+------+------+------+------+------+
ICE Yes |
------------+------+------+------+------+------+------+------+
3-W Latch | Yes | Yes | No 1.5 | Maybe| Yes | No | Yes |
------------+------+------+------+------+------+
SymRTP
------------+------+------+------+------+------+------+------+
ALG |(Yes) | Yes | Yes 0 | No | Yes |Maybe | No |
------------+------+------+------+------+------+
V. SymRTP | Yes | Yes
------------+------+------+------+------+------+------+------+
TCP Tunnel | Yes | Yes |future|
------------+------+------+------+------+------+
3-W SymRTP | Yes 1.5 | Yes | Yes | Maybe| No | Yes |
------------+------+------+------+------+------+
------------+------+------+------+------+------+------+------+
TURN | Yes | Yes | No 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 was 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 handle 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 has been were some discussion in the WG if the
increased implementation burden of ICE is sufficiently motivated
compared to a 3-W SymRTP 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].

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 sessions sections we have discussed security merits of each and
every NAT/FW 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 with out transport
level security and source authentications; authentications, as long as the server does
not grant a client request to send allow media to be sent to a different IP addresses. IP-address than the RTSP
client request was sent from. Using symmetric RTP 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 tag bits 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) Latching (Section 4.5) with authentication features. Still both
those variants of Latching is 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. The usage of an RTSP ALG does not increase 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 encrypted end-to-end encrypted RTSP signaling. The usage
of TCP tunneling has no known security problems. However 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 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, and Colin Perkins. Thomas Zeng would also
like to give special thanks to Greg Sherwood of PacketVideo for his
input into this memo.

Section 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-rfc2326bis]
Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
and M. Stiemerling, "Real Time Streaming Protocol 2.0
(RTSP)", draft-ietf-mmusic-rfc2326bis-29 draft-ietf-mmusic-rfc2326bis-30 (work in
progress), March July 2012.

[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-12
draft-ietf-mmusic-rtsp-nat-14 (work in progress),
May
November 2012.

[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
August 1980.

[RFC0793] Postel, J., "Transmission Control Protocol", STD 7,
RFC 793, September 1981.

[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.

[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.

[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.

[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://
www.vovida.org/applications/downloads/stun/index.html",
June 2007.

Authors' Addresses

Magnus Westerlund
Ericsson
Farogatan 6
Stockholm, SE-164 80
Sweden

Phone: +46 8 719 0000
Fax:
Email: magnus.westerlund@ericsson.com
URI:

Thomas Zeng

Phone:
Fax:
Email: thomas.zeng@gmail.com
URI: