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rfc:rfc7604

Internet Engineering Task Force (IETF) M. Westerlund Request for Comments: 7604 Ericsson Category: Informational T. Zeng ISSN: 2070-1721 PacketVideo Corp

                                                        September 2015
          Comparison of Different NAT Traversal Techniques
  for Media Controlled by the Real-Time Streaming Protocol (RTSP)

Abstract

 This document describes several Network Address Translator (NAT)
 traversal techniques that were considered to be used for establishing
 the RTP media flows controlled by the Real-Time Streaming Protocol
 (RTSP).  Each technique includes a description of how it would be
 used, the security implications of using it, and any other deployment
 considerations it has.  There are also discussions on how NAT
 traversal techniques relate to firewalls and how each technique can
 be applied in different use cases.  These findings were used when
 selecting the NAT traversal for RTSP 2.0, which is specified in a
 separate document.

Status of This Memo

 This document is not an Internet Standards Track specification; it is
 published for informational purposes.
 This document is a product of the Internet Engineering Task Force
 (IETF).  It represents the consensus of the IETF community.  It has
 received public review and has been approved for publication by the
 Internet Engineering Steering Group (IESG).  Not all documents
 approved by the IESG are a candidate for any level of Internet
 Standard; see Section 2 of RFC 5741.
 Information about the current status of this document, any errata,
 and how to provide feedback on it may be obtained at
 http://www.rfc-editor.org/info/rfc7604.

Westerlund & Zeng Informational [Page 1] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

Copyright Notice

 Copyright (c) 2015 IETF Trust and the persons identified as the
 document authors.  All rights reserved.
 This document is subject to BCP 78 and the IETF Trust's Legal
 Provisions Relating to IETF Documents
 (http://trustee.ietf.org/license-info) in effect on the date of
 publication of this document.  Please review these documents
 carefully, as they describe your rights and restrictions with respect
 to this document.  Code Components extracted from this document must
 include Simplified BSD License text as described in Section 4.e of
 the Trust Legal Provisions and are provided without warranty as
 described in the Simplified BSD License.

Table of Contents

 1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   4
   1.1.  Network Address Translators . . . . . . . . . . . . . . .   5
   1.2.  Firewalls . . . . . . . . . . . . . . . . . . . . . . . .   6
   1.3.  Glossary  . . . . . . . . . . . . . . . . . . . . . . . .   7
 2.  Detecting the Loss of NAT Mappings  . . . . . . . . . . . . .   8
 3.  Requirements on Solutions . . . . . . . . . . . . . . . . . .   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.  ALG Considerations  . . . . . . . . . . . . . . . . .  14
     4.1.4.  Deployment Considerations . . . . . . . . . . . . . .  14
     4.1.5.  Security Considerations . . . . . . . . . . . . . . .  15
   4.2.  Server Embedded STUN  . . . . . . . . . . . . . . . . . .  16
     4.2.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  16
     4.2.2.  Embedding STUN in RTSP  . . . . . . . . . . . . . . .  16
     4.2.3.  Discussion on Co-located STUN Server  . . . . . . . .  17
     4.2.4.  ALG Considerations  . . . . . . . . . . . . . . . . .  17
     4.2.5.  Deployment Considerations . . . . . . . . . . . . . .  18
     4.2.6.  Security Considerations . . . . . . . . . . . . . . .  19
   4.3.  ICE . . . . . . . . . . . . . . . . . . . . . . . . . . .  19
     4.3.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  19
     4.3.2.  Using ICE in RTSP . . . . . . . . . . . . . . . . . .  20
     4.3.3.  Implementation Burden of ICE  . . . . . . . . . . . .  21
     4.3.4.  ALG Considerations  . . . . . . . . . . . . . . . . .  22
     4.3.5.  Deployment Considerations . . . . . . . . . . . . . .  22
     4.3.6.  Security Considerations . . . . . . . . . . . . . . .  23

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   4.4.  Latching  . . . . . . . . . . . . . . . . . . . . . . . .  23
     4.4.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  23
     4.4.2.  Necessary RTSP Extensions . . . . . . . . . . . . . .  24
     4.4.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  25
     4.4.4.  Deployment Considerations . . . . . . . . . . . . . .  25
     4.4.5.  Security Considerations . . . . . . . . . . . . . . .  26
   4.5.  A Variation to Latching . . . . . . . . . . . . . . . . .  27
     4.5.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  27
     4.5.2.  Necessary RTSP Extensions . . . . . . . . . . . . . .  28
     4.5.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  28
     4.5.4.  Deployment Considerations . . . . . . . . . . . . . .  28
     4.5.5.  Security Considerations . . . . . . . . . . . . . . .  29
   4.6.  Three-Way Latching  . . . . . . . . . . . . . . . . . . .  29
     4.6.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  29
     4.6.2.  Necessary RTSP Extensions . . . . . . . . . . . . . .  29
     4.6.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  30
     4.6.4.  Deployment Considerations . . . . . . . . . . . . . .  30
     4.6.5.  Security Considerations . . . . . . . . . . . . . . .  30
   4.7.  Application Level Gateways  . . . . . . . . . . . . . . .  31
     4.7.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  31
     4.7.2.  Outline on How ALGs for RTSP Work . . . . . . . . . .  31
     4.7.3.  Deployment Considerations . . . . . . . . . . . . . .  32
     4.7.4.  Security Considerations . . . . . . . . . . . . . . .  33
   4.8.  TCP Tunneling . . . . . . . . . . . . . . . . . . . . . .  33
     4.8.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  33
     4.8.2.  Usage of TCP Tunneling in RTSP  . . . . . . . . . . .  34
     4.8.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  34
     4.8.4.  Deployment Considerations . . . . . . . . . . . . . .  34
     4.8.5.  Security Considerations . . . . . . . . . . . . . . .  35
   4.9.  Traversal Using Relays around NAT (TURN)  . . . . . . . .  35
     4.9.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  35
     4.9.2.  Usage of TURN with RTSP . . . . . . . . . . . . . . .  36
     4.9.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  37
     4.9.4.  Deployment Considerations . . . . . . . . . . . . . .  37
     4.9.5.  Security Considerations . . . . . . . . . . . . . . .  37
 5.  Firewalls . . . . . . . . . . . . . . . . . . . . . . . . . .  38
 6.  Comparison of NAT Traversal Techniques  . . . . . . . . . . .  39
 7.  Security Considerations . . . . . . . . . . . . . . . . . . .  41
 8.  Informative References  . . . . . . . . . . . . . . . . . . .  42
 Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .  45
 Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  46

Westerlund & Zeng Informational [Page 3] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

1. Introduction

 Today there is a proliferating deployment of different types of
 Network Address Translator (NAT) boxes that in many cases only
 loosely follow standards [RFC3022] [RFC2663] [RFC3424] [RFC4787]
 [RFC5382].  NATs cause discontinuity in address realms [RFC3424];
 therefore, an application protocol, such as the Real-Time Streaming
 Protocol (RTSP) [RFC2326] [RTSP], 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 the
 Transmission Control Protocol (TCP) [RFC793], for example, is not
 blocked by NATs, its media streams may be blocked by NATs.  This will
 occur unless special protocol provisions are added to support NAT
 traversal.
 Like NATs, firewalls are also middleboxes that need to be considered.
 Firewalls help prevent unwanted traffic from getting in or out of the
 protected network.  RTSP is designed such that a firewall can be
 configured to let RTSP-controlled media streams go through with
 limited implementation effort.  The effort needed is to implement an
 Application Level Gateway (ALG) to interpret RTSP parameters.  There
 is also a large class of firewalls, commonly home firewalls, that use
 a filtering behavior that appears to be the same as what NATs have.
 This type of firewall will be successfully traversed using the same
 solution as employed for NAT traversal, instead of relying on an RTSP
 ALG.  Therefore, firewalls will also be discussed and some important
 differences highlighted.
 This document describes several NAT traversal mechanisms for RTSP-
 controlled media streaming.  Many of these NAT solutions fall into
 the category of "UNilateral Self-Address Fixing (UNSAF)" as defined
 in [RFC3424] and quoted below:
    [UNSAF] is a process whereby some originating process attempts to
    determine or fix the address (and port) by which it is known -
    e.g.  to be able to use address data in the protocol exchange, or
    to advertise a public address from which it will receive
    connections.
 Following the guidelines spelled out in RFC 3424, we describe the
 required RTSP extensions for each method, transition strategies, and
 security concerns.  The transition strategies are a discussion of how
 and if the method encourages a move towards not having any NATs on
 the path.

Westerlund & Zeng Informational [Page 4] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

 This document is capturing the evaluation done in the process to
 recommend firewall/NAT traversal methods for RTSP streaming servers
 based on [RFC2326] as well as the RTSP 2.0 core specification [RTSP].
 The evaluation is focused on NAT traversal for the media streams
 carried over the User Datagram Protocol (UDP) [RFC768] with RTP
 [RFC3550] over UDP being the main case for such usage.  The findings
 should be applicable to other protocols as long as they have similar
 properties.
 At the time when the bulk of work on this document was done, a single
 level of NAT was the dominant deployment for NATs, and multiple
 levels of NATs, including Carrier-Grade NATs (CGNs), were not
 considered.  Thus, any characterizations or findings may not be
 applicable in such scenarios, unless CGN or multiple levels of NATs
 are explicitly noted.
 An RTSP NAT traversal mechanism based on Interactive Connectivity
 Establishment (ICE) is specified in "A Network Address Translator
 (NAT) Traversal Mechanism for Media Controlled by Real-Time Streaming
 Protocol (RTSP)" [RTSP-NAT].

1.1. Network Address Translators

 We begin by reviewing two quotes from Section 3 in "Network Address
 Translation (NAT) Behavioral Requirements for Unicast UDP" [RFC4787]
 concerning NATs and their terminology:
    Readers are urged to refer to [RFC2663] for information on NAT
    taxonomy and terminology.  Traditional NAT is the most common type
    of NAT device deployed.  Readers may refer to [RFC3022] for
    detailed information on traditional NAT.  Traditional NAT has two
    main varieties -- Basic NAT and Network Address/Port Translator
    (NAPT).
    NAPT is by far the most commonly deployed NAT device.  NAPT allows
    multiple internal hosts to share a single public IP address
    simultaneously.  When an internal host opens an outgoing TCP or
    UDP session through a NAPT, the NAPT assigns the session a public
    IP address and port number, so that subsequent response packets
    from the external endpoint can be received by the NAPT,
    translated, and forwarded to the internal host.  The effect is
    that the NAPT establishes a NAT session to translate the (private
    IP address, private port number) tuple to a (public IP address,
    public port number) tuple, and vice versa, for the duration of the
    session.  An issue of relevance to peer-to-peer applications is
    how the NAT behaves when an internal host initiates multiple
    simultaneous sessions from a single (private IP, private port)
    endpoint to multiple distinct endpoints on the external network.

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    In this specification, the term "NAT" refers to both "Basic NAT"
    and "Network Address/Port Translator (NAPT)".
    This document uses the term "Address and Port Mapping" as the
    translation between an external address and port and an internal
    address and port.  Note that this is not the same as an "address
    binding" as defined in RFC 2663.
    Note: In the above text, it would be more correct to use an
    external IP address instead of a public IP address.  The external
    IP address is commonly a public one, but it might be of another
    type if the NAT's external side is in a private address domain.
 In addition to the above quote, there exists a number of address and
 port mapping behaviors described in more detail in Section 4.1 of
 [RFC4787] that are highly relevant to the discussion in this
 document.
 NATs also have a filtering behavior on traffic arriving on the
 external side.  Such behavior affects how well different methods for
 NAT traversal works through these NATs.  See Section 5 of [RFC4787]
 for more information on the different types of filtering that have
 been identified.

1.2. Firewalls

 A firewall is a security gateway that enforces certain access control
 policies between two network administrative domains: a private domain
 (intranet) and an external domain, e.g., the Internet.  Many
 organizations use firewalls to prevent intrusions and malicious
 attacks on computing resources in the private intranet [RFC2588].
 A comparison between NAT and a firewall is given below:
 1.  A firewall sits at security enforcement/protection points, while
     NAT sits at borders between two address domains.
 2.  NAT does not in itself provide security, although some access
     control policies can be implemented using address translation
     schemes.  The inherent filtering behaviors are commonly mistaken
     for real security policies.
 It should be noted that many NAT devices intended for Residential or
 Small Office, Home Office (SOHO) use include both NATs and firewall
 functionality.

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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 is 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
 DoS:  Denial of Service
 DDoS: Distributed Denial of Service
 NAT:  Network Address Translator; see [RFC3022].
 NAPT: Network Address/Port Translator; see [RFC3022].
 RTP:  Real-Time Transport Protocol; see [RFC3550].
 RTSP: Real-Time Streaming Protocol; see [RFC2326] and [RTSP].
 RTT:  Round Trip Times
 SDP:  Session Description Protocol; see [RFC4566].
 SSRC: Synchronization source in RTP; see [RFC3550].

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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 RFC 3424.
 First, since NATs may also dynamically reclaim or readjust address/
 port translations, "keep-alive" and periodic repolling may be
 required according to RFC 3424.  Second, 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 recommendations on how
 to detect the loss of NAT mappings when using RTP/RTCP under RTSP
 control.
 An RTP session normally has both RTP and RTCP streams.  The loss of
 an RTP mapping can only be detected when expected traffic does not
 arrive.  If a client does not receive media data within a few seconds
 after having received the "200 OK" response to an RTSP PLAY request
 that starts the media delivery, it may be the result of a middlebox
 blocking the traffic.  However, for a receiver to be more certain to
 detect the case where no RTP traffic was delivered due to NAT
 trouble, one should monitor the RTCP Sender reports if they are
 received and not also blocked.  The sender report carries a field
 telling how many packets the server has sent.  If that has increased
 and no RTP packets have arrived for a few seconds, it is likely the
 mapping for the RTP stream has been removed.
 The loss of mapping for RTCP is simpler to detect.  RTCP is normally
 sent periodically in each direction, even during the RTSP ready
 state.  If RTCP packets are missing for several RTCP intervals, the
 mapping is likely lost.  Note that if neither RTCP packets nor RTSP
 messages are received by the RTSP server for a while (default 60
 seconds), 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 middlebox NAT/firewall, or the client is
 malfunctioning.

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3. Requirements on Solutions

 This section considers the set of requirements for the evaluation of
 RTSP NAT traversal solutions.
 RTSP is a client-server protocol.  Typically, service providers
 deploy RTSP servers on the Internet or otherwise reachable address
 realm.  However, there are use cases where the reverse is true: RTSP
 clients are connecting from any address realm to RTSP servers behind
 NATs, e.g., in a home.  This is the case, for instance, when home
 surveillance cameras running as RTSP servers intend to stream video
 to cell phone users in the public address realm through a home NAT.
 In terms of requirements, the primary issue to solve is the RTSP NAT
 traversal problem for RTSP servers deployed in a network where the
 server is on the external side of any NAT, i.e., the server is not
 behind a NAT.  The server behind a NAT is desirable but of much lower
 priority.
 Important considerations for any NAT traversal technique are whether
 any protocol modifications are needed and where the implementation
 burden resides (e.g., server, client, or middlebox).  If the
 incentive to get RTSP to work over a NAT is sufficient, it will
 motivate the owner of the server, client, or middlebox to update,
 configure, or otherwise perform changes to the device and its
 software in order to support NAT traversal.  Thus, the questions of
 who this burden falls on and how big it is are highly relevant.
 The list of feature requirements for RTSP NAT solutions are given
 below:
 1.  Must work for all flavors of NATs, including NATs with Address
     and Port-Dependent Filtering.
 2.  Must work for firewalls (subject to pertinent firewall
     administrative policies), including those with ALGs.
 3.  Should have minimal impact on clients not behind NATs and that
     are not dual hosted.  RTSP dual hosting means that the RTSP
     signaling protocol and the media protocol (e.g., RTP) are
     implemented on different computers with different IP addresses.
  • For instance, no extra protocol RTT before arrival of media.
 4.  Should be simple to use/implement/administer so people actually
     turn them on.
  • Discovery of the address(es) assigned by NAT should happen

automatically, if possible.

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 5.  Should authenticate dual-hosted client's media transport receiver
     to prevent usage of RTSP servers for DDoS attacks.
 The last requirement addresses the Distributed Denial-of-Service
 (DDoS) threat, which relates to NAT traversal as explained below.
 During NAT traversal, when the RTSP server determines the media
 destination (address and port) for the client, the result may be that
 the IP address of the RTP receiver host is different than the IP
 address of the RTSP client host.  This poses a DDoS threat that has
 significant amplification potentials because the RTP media streams in
 general consist of a large number of IP packets.  DDoS attacks can
 occur if the attacker can fake the messages in the NAT traversal
 mechanism to trick the RTSP server into believing that the client's
 RTP receiver is located on a host to be attacked.  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 that a simple mitigation is for the RTSP server to disallow the
 cases where the client's RTP receiver has a different IP address than
 that of the RTSP client.  This is recommended behavior in RTSP 2.0
 unless other solutions to prevent this attack are present; see
 Section 21.2.1 in [RTSP].  With the increased deployment of NAT
 middleboxes by operators, i.e., CGN, the reuse of an IP address on
 the NAT's external side by many customers reduces the protection
 provided.  Also in some applications (e.g., centralized
 conferencing), dual-hosted RTSP/RTP clients have valid use cases.
 The key is how to authenticate the messages exchanged during the NAT
 traversal process.

4. NAT-Traversal Techniques

 There exists a number of potential NAT traversal techniques that can
 be used to allow RTSP to traverse NATs.  They have different features
 and are applicable to different topologies; their costs are also
 different.  They also vary in security levels.  In the following
 sections, each technique is outlined with discussions on the
 corresponding advantages and disadvantages.
 The survey of traversal techniques was done prior to 2007 and is
 based on what was available then.  This section includes NAT
 traversal techniques that have not been formally specified anywhere
 else.  This document may be the only publicly available specification
 of some of the NAT traversal techniques.  However, that is not a real
 barrier against doing an evaluation of the NAT traversal techniques.
 Some techniques used as part of some of the traversal solutions have
 been recommended against or are no longer possible due to the outcome

Westerlund & Zeng Informational [Page 10] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

 of standardization work or their failure to progress within IETF
 after the initial evaluation in this document.  For example, RTP
 No-Op [RTP-NO-OP] was a proposed RTP payload format that failed to be
 specified; thus, it is not available for use today.  In each such
 case, the missing parts will be noted and some basic reasons will be
 given.

4.1. Stand-Alone STUN

4.1.1. Introduction

 Session Traversal Utilities for NAT (STUN) [RFC5389] is a
 standardized protocol that allows a client to use secure means to
 discover the presence of a NAT between itself and the STUN server.
 The client uses the STUN server to discover the address and port
 mappings assigned by the NAT.  Then using the knowledge of these NAT
 mappings, it uses the external addresses to directly connect to the
 independent RTSP server.  However, this is only possible if the NAT
 address and port mapping behavior is such that the STUN server and
 RTSP server will see the same external address and port for the same
 internal address and port.
 STUN is a client-server protocol.  The STUN client sends a request to
 a STUN server and the server returns a response.  There are two types
 of STUN messages -- Binding Requests and Indications.  Binding
 Requests are used when determining a client's external address and
 soliciting a response from the STUN server with the seen address.
 Indications are used by the client for keep-alive messages towards
 the server and requires no response from the server.
 The first version of STUN [RFC3489] included categorization and
 parameterization of NATs.  This was abandoned in the updated version
 [RFC5389] due to it being unreliable and brittle.  This particular
 traversal method uses the removed functionality described in RFC 3489
 to detect the NAT type to give an early failure indication when the
 NAT is showing the behavior that this method can't support.  This
 method also suggests using the RTP No-Op payload format [RTP-NO-OP]
 for keep-alives of the RTP traffic in the client-to-server direction.
 This can be replaced with another form of UDP packet as will be
 further discussed below.

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).

Westerlund & Zeng Informational [Page 11] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

 Limitations:
 o  The server must be located in either a public address realm or the
    next-hop external address realm in regards to the client.
 o  The client may only be located behind NATs that perform Endpoint-
    Independent or Address-Dependent Mappings (the STUN server and
    RTSP server on the same IP address).  Clients behind NATs that do
    Address and Port-Dependent Mappings cannot use this method.  See
    [RFC4787] for the full definition of these terms.
 o  Based on the discontinued middlebox classification of the replaced
    STUN specification [RFC3489]; thus, it is brittle and unreliable.
 Method:
 An RTSP client using RTP transport over UDP can use STUN to traverse
 a NAT(s) in the following way:
 1.  Use STUN to try to discover the type of NAT and the timeout
     period for any UDP mapping on the NAT.  This is recommended to be
     performed in the background as soon as IP connectivity is
     established.  If this is performed prior to establishing a
     streaming session, the delays in the session establishment will
     be reduced.  If no NAT is detected, normal SETUP should be used.
 2.  The RTSP client determines the number of UDP ports needed by
     counting the number of needed media transport protocols sessions
     in the multimedia presentation.  This information is available in
     the media description protocol, e.g., SDP [RFC4566].  For
     example, each RTP session will in general require two UDP ports:
     one for RTP, and one for RTCP.
 3.  For each UDP port required, establish a mapping and discover the
     public/external IP address and port number with the help of the
     STUN server.  A successful mapping looks like: client's local
     address/port <-> public address/port.
 4.  Perform the RTSP SETUP for each media.  In the Transport header,
     the following parameter should be included with the given values:
     "dest_addr" [RTSP] 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 set up the RTP stream on any port, not only even ports
     and with non-contiguous port numbers for RTP and RTCP.  This
     requires the new feature provided in RTSP 2.0 [RTSP].  The server
     should respond with a Transport header containing an "src_addr"

Westerlund & Zeng Informational [Page 12] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

     or "source" + "server_port" parameters with the RTP and RTCP
     source IP address and port of the media stream.
 5.  To keep the mappings alive, the client should periodically send
     UDP traffic over all mappings needed for the session.  For the
     mapping carrying RTCP traffic, the periodic RTCP traffic is
     likely enough.  For mappings carrying RTP traffic and for
     mappings carrying RTCP packets at too low of a frequency, keep-
     alive messages should be sent.
 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 an
 Address-Dependent or Address and Port-Dependent Filtering NAT, the
 client must first send a UDP packet to establish the filtering state
 in the NAT.  The client, before sending an RTSP PLAY request, must
 send a so-called hole-punching packet on each mapping to the IP
 address and port given as the server's source address and port.  For
 a NAT that only is Address-Dependent Filtering, the hole-punching
 packet could be sent to the server's discard port (port number 9).
 For Address and Port-Dependent Filtering NATs, the hole-punching
 packet must go to the port used for sending UDP packets to the
 client.  To be able to do that, the server needs to include the
 "src_addr" in the Transport header (which is the "source" transport
 parameter in RFC2326).  Since UDP packets are inherently unreliable,
 to ensure that at least one UDP message passes the NAT, hole-punching
 packets should be retransmitted a reasonable number of times.
 One could have used RTP No-Op packets [RTP-NO-OP] as hole-punching
 and keep-alive messages had they been defined.  That would have
 ensured that the traffic would look like RTP and thus would likely
 have the least risk of being dropped by any firewall.  The drawback
 of using RTP No-Op is that the payload type number must be
 dynamically assigned through RTSP first.  Other options are STUN, an
 RTP packet without any payload, or a UDP packet without any payload.
 For RTCP it is most suitable to use correctly generated RTCP packets.
 In general, sending unsolicited traffic to the RTSP server may
 trigger security functions resulting in the blocking of the keep-
 alive messages or termination of the RTSP session itself.
 This method is further brittle as it doesn't support Address and
 Port-Dependent Mappings.  Thus, it proposes to use the old STUN
 methods to classify the NAT behavior, thus enabling early error
 indication.  This is strictly not required but will lead to failures
 during setup when the NAT has the wrong behavior.  This failure can

Westerlund & Zeng Informational [Page 13] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

 also occur if the NAT changes the properties of the existing mapping
 and filtering state or between the classification message exchange
 and the actual RTSP session setup, for example, due to load.

4.1.3. ALG Considerations

 If a NAT supports RTSP ALG (Application Level Gateway) and is not
 aware of the STUN traversal option, service failure may happen,
 because a client discovers its NAT external IP address and port
 numbers and inserts them in its SETUP requests.  When the RTSP ALG
 processes the SETUP request, it may change the destination and port
 number, resulting in unpredictable behavior.  An ALG should not
 update address fields that contain addresses other than the NAT's
 internal address domain.  In cases where the ALG modifies fields
 unnecessarily, two alternatives exist:
 1.  Use Transport Layer Security (TLS) to encrypt the data over the
     RTSP TCP connection to prevent the ALG from reading and modifying
     the RTSP messages.
 2.  Turn off the STUN-based NAT traversal mechanism.
 As it may be difficult to determine why the failure occurs, the usage
 of TLS-protected RTSP message exchange at all times would avoid this
 issue.

4.1.4. Deployment Considerations

 For the stand-alone usage of STUN, the following applies:
 Advantages:
 o  STUN is a solution first used by applications based on SIP
    [RFC3261] (see Sections 1 and 2 of [RFC5389]).  As shown above,
    with little or no changes, the RTSP application can reuse STUN as
    a NAT traversal solution, avoiding the pitfall of solving a
    problem twice.
 o  Using STUN does not require RTSP server modifications, assuming it
    is a server that is compliant with RTSP 2.0; it only affects the
    client implementation.
 Disadvantages:
 o  Requires a STUN server deployed in the same address domain as the
    server.

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 o  Only works with NATs that perform Endpoint-Independent and
    Address-Dependent Mappings.  Address and Port-Dependent Filtering
    NATs create some issues.
 o  Brittle to NATs changing the properties of the NAT mapping and
    filtering.
 o  Does not work with Address and Port-Dependent Mapping NATs without
    server modifications.
 o  Will not work if a NAT uses multiple IP addresses, since RTSP
    servers generally require all media streams to use the same IP as
    used in the RTSP connection to prevent becoming a DDoS tool.
 o  Interaction problems exist when an RTSP-aware ALG interferes with
    the use of STUN for NAT traversal unless TLS-secured RTSP message
    exchange is used.
 o  Using STUN requires that RTSP servers and clients support the
    updated RTSP specification [RTSP], because it is no longer
    possible to guarantee that RTP and RTCP ports are adjacent to each
    other, as required by the "client_port" and "server_port"
    parameters in RFC 2326.
 Transition:
 The usage of STUN can be phased out gradually as the first step of a
 STUN-capable server or client should be to check the presence of
 NATs.  The removal of STUN capability in the client implementations
 will have to wait until there is absolutely no need to use STUN.

4.1.5. Security Considerations

 To prevent the RTSP server from being used as Denial-of-Service (DoS)
 attack tools, the RTSP Transport header parameters "destination" and
 "dest_addr" are generally not allowed to point to any IP address
 other than the one the RTSP message originates from.  The RTSP server
 is only prepared to make an exception to this rule when the client is
 trusted (e.g., through the use of a secure authentication process or
 through some secure method of challenging the destination to verify
 its willingness to accept the RTP traffic).  Such a restriction means
 that STUN in general does not work for use cases where RTSP and media
 transport go to different addresses.
 STUN combined with RTSP that is restricted by destination address has
 the same security properties as the core RTSP.  It is protected from
 being used as a DoS attack tool unless the attacker has the ability
 to spoof the TCP connection carrying RTSP messages.

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 Using STUN's support for message authentication and the secure
 transport of RTSP messages, attackers cannot modify STUN responses or
 RTSP messages (TLS) to change the media destination.  This protects
 against hijacking; however, as a client can be the initiator of an
 attack, these mechanisms cannot securely prevent RTSP servers from
 being used as DoS attack tools.

4.2. Server Embedded STUN

4.2.1. Introduction

 This section describes an alternative to the stand-alone STUN usage
 in the previous section that has quite significantly different
 behavior.

4.2.2. Embedding STUN in RTSP

 This section outlines the adaptation and embedding of STUN within
 RTSP.  This enables STUN to be used to traverse any type of NAT,
 including Address and Port-Dependent Mapping NATs.  This would
 require RTSP-level protocol changes.
 This NAT traversal solution has limitations:
 1.  It does not work if both the RTSP client and RTSP server are
     behind separate NATs.
 2.  The RTSP server may, for security reasons, refuse to send media
     streams to an IP that is 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 the STUN server to be co-located on the RTSP server's
     media source ports.
 If the STUN server is co-located with the 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 Address and Port-
 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 them can be behind a NAT).

Westerlund & Zeng Informational [Page 16] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

 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 a 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 RFC 2326 and the
 "src_addr" parameter in [RTSP].  Both are in the RTSP Transport
 header.  But in general, this strategy will require that one first
 does one SETUP request per media to learn the server ports, then
 perform the STUN checks, followed by a subsequent SETUP to change the
 client port and destination address to what was learned during the
 STUN checks.
 To be certain that RTCP works correctly, the RTSP endpoint (server or
 client) will be required to send and receive RTCP packets from the
 same port.

4.2.3. Discussion on Co-located STUN Server

 In order to use STUN to traverse Address and Port-Dependent Filtering
 or Mapping NATs, the STUN server needs to be co-located with the
 streaming server media output ports.  This creates a demultiplexing
 problem: we must be able to differentiate a STUN packet from a media
 packet.  This will be done based on heuristics.  The existing STUN
 heuristics is the first byte in the packet and the Magic Cookie field
 (added in RFC 5389), which works fine between STUN and RTP or RTCP
 where the first byte happens to be different.  Thanks to the Magic
 Cookie field, it is unlikely that other protocols would be mistaken
 for a STUN packet, but this is not assured.  For more discussion of
 this, please see Section 5.1.2 of [RFC5764].

4.2.4. ALG Considerations

 The same ALG traversal considerations as for stand-alone STUN applies
 (Section 4.1.3).

Westerlund & Zeng Informational [Page 17] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

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, the RTSP application can reuse
    STUN as a NAT traversal solution, avoiding the pitfall of solving
    a problem twice.
 o  STUN has built-in message authentication features, which makes it
    more secure against hijacking attacks.  See the next section for
    an in-depth security discussion.
 o  This solution works as long as there is only one RTSP endpoint in
    the private address realm, regardless of the NAT's type.  There
    may even be multiple NATs (see Figure 1 in [RFC5389]).
 o  Compared to other UDP-based NAT traversal methods in this
    document, STUN requires little new protocol development (since
    STUN is already an IETF standard), and most likely less
    implementation effort, since open source STUN server and client
    implementations are available [STUN-IMPL] [PJNATH].
 Disadvantages:
 o  Some extensions to the RTSP core protocol, likely signaled by RTSP
    feature tags, must be introduced.
 o  Requires an embedded STUN server to be co-located on each of the
    RTSP server's media protocol's ports (e.g., RTP and RTCP ports),
    which means more processing is required to demultiplex STUN
    packets from media packets.  For example, the demultiplexer must
    be able to differentiate an RTCP RR packet from a STUN packet and
    forward the former to the streaming server and the latter to the
    STUN server.
 o  Does not support use cases that require the RTSP connection and
    the media reception to happen at different addresses, unless the
    server's security policy is relaxed.
 o  Interaction problems exist when an 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 [RTSP], and they both agree to support
    the NAT traversal feature.

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 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 used to check the presence of NATs for
 the presently used network connection.  The removal of STUN
 capability in the client implementations will have to wait until
 there is absolutely no need to use STUN, i.e., no NATs or firewalls.

4.2.6. Security Considerations

 See Stand-Alone STUN (Section 4.1.5).

4.3. ICE

4.3.1. Introduction

 Interactive Connectivity Establishment (ICE) [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 in which an endpoint may
 be reached.  This allows the endpoint to use the best available UDP
 "connection" (meaning two UDP endpoints capable of reaching each
 other).  The methodology has very nice properties in that basically
 all NAT topologies are possible to traverse.
 Here is how ICE works at a high level.  Endpoint A collects all
 possible addresses that can be used, including local IP addresses,
 STUN-derived addresses, Traversal Using Relay NAT (TURN) addresses,
 etc.  On each local port that any of these address and port pairs
 lead to, a STUN server is installed.  This STUN server only accepts
 STUN requests using the correct authentication through the use of a
 username and password.
 Endpoint A then sends a request to establish connectivity with
 endpoint B, which includes all possible "destinations" [RFC5245] to
 get the media through to A.  Note that each of A's local address/port
 pairs (host candidates and server reflexive base) has a co-located
 STUN server.  B in turn provides A with all its possible destinations
 for the different media streams.  A and B then uses a STUN client to
 try to reach all the address and port pairs specified by A from its
 corresponding destination ports.  The destinations for which the STUN
 requests successfully complete are then indicated and one is
 selected.

Westerlund & Zeng Informational [Page 19] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

 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; therefore, requests are
 retransmitted multiple times.  The STUN requests may also result in
 more connectivity alternatives (destinations) being discovered and
 conveyed in the STUN responses.

4.3.2. Using ICE in RTSP

 The usage of ICE for RTSP requires that both client and server be
 updated to include the ICE functionality.  If both parties implement
 the necessary functionality, the following steps could provide ICE
 support for RTSP.
 This assumes that it is possible to establish a TCP connection for
 the RTSP messages between the client and the server.  This is not
 trivial in scenarios where the server is located behind a NAT, and
 may require some TCP ports be opened, or proxies are deployed, etc.
 The negotiation of ICE in RTSP of necessity will work different than
 in SIP with SDP offer/answer.  The protocol interactions are
 different, and thus the possibilities for transfer of states are also
 somewhat different.  The goal is also to avoid introducing extra
 delay in the setup process at least for when the server is not behind
 a NAT in regards to the client, and the client is either having an
 address in the same address domain or is behind the NAT(s), which can
 address the address domain of the server.  This process is only
 intended to support PLAY mode, i.e., media traffic flows from server
 to client.
 1.  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 on demand, DNS load balancing,
     etc., which have the SDP indicate a resource on a server park
     rather than a specific machine.
 2.  The client gathers addresses and puts together its candidates for
     each media stream indicated in the session description.
 3.  In each SETUP request, the client includes its candidates in an
     ICE-specific transport specification.  For the server, this
     indicates the ICE support by the client.  One candidate is the
     most prioritized candidate and here the prioritization for this
     address should be somewhat different compared to SIP.  High-
     performance candidates are recommended rather than candidates
     with the highest likelihood of success, as it is more likely that
     a server is not behind a NAT compared to a SIP user agent.

Westerlund & Zeng Informational [Page 20] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

 4.  The server responds to the SETUP (200 OK) for each media stream
     with its candidates.  A server not behind a NAT usually only
     provides a single ICE candidate.  Also, here one candidate is the
     server primary address.
 5.  The connectivity checks are performed.  For the server, the
     connectivity checks from the server to the clients have an
     additional usage.  They verify that there is someone willing to
     receive the media, thus preventing the server from unknowingly
     performing a DoS attack.
 6.  Connectivity checks from the client promoting a candidate pair
     were successful.  Thus, no further SETUP requests are necessary
     and processing can proceed with step 7.  If an address other than
     the primary has been verified by the client to work, that address
     may then be promoted for usage in a SETUP request (go to step 7).
     If the checks for the available candidates failed and if further
     candidates have been derived during the connectivity checks, then
     those can be signaled in new candidate lines in a SETUP request
     updating the list (go to step 5).
 7.  Client issues the PLAY request.  If the server also has completed
     its connectivity checks for the promoted candidate pair (based on
     the username as it may be derived addresses if the client was
     behind NAT), then it can directly answer 200 OK (go to step 8).
     If the connectivity check has not yet completed, it responds with
     a 1xx code to indicate that it is verifying the connectivity.  If
     that fails within the set timeout, an error is reported back.
     The client needs to go back to step 6.
 8.  Process completed and media can be delivered.  ICE candidates not
     used may be released.
 To keep media paths alive, the client needs to periodically send data
 to the server.  This will be realized with STUN.  RTCP sent by the
 client should be able to keep RTCP open, but STUN will also be used
 for SIP based on the same motivations as for ICE.

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 among all
 the NAT/firewall traversal methods in this document.

Westerlund & Zeng Informational [Page 21] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

 RTSP server implementation requirements are:
 o  STUN server features
 o  Limited STUN client features
 o  SDP generation with more parameters
 o  RTSP error code for ICE extension
 RTSP client implementation requirements are:
 o  Limited STUN server features
 o  Limited STUN client features
 o  RTSP error code and ICE extension

4.3.4. ALG Considerations

 If there is an RTSP ALG that doesn't support the NAT traversal
 method, it may interfere with the NAT traversal.  As the usage of ICE
 for the traversal manifests itself in the RTSP message primarily as a
 new transport specification, an ALG that passes through unknown will
 not prevent the traversal.  An ALG that discards unknown
 specifications will, however, prevent the NAT traversal.  These
 issues can be avoided by preventing the ALG to interfere with the
 signaling by using TLS for the RTSP message transport.
 An ALG that supports this traversal method can, on the most basic
 level, just pass the transport specifications through.  ALGs in NATs
 and firewalls could use the ICE candidates to establish a filtering
 state that would allow incoming STUN messages prior to any outgoing
 hole-punching packets, and in that way it could speed up the
 connectivity checks and reduce the risk of failures.

4.3.5. Deployment Considerations

 Advantages:
 o  Solves NAT connectivity discovery for basically all cases as long
    as a TCP connection between the client and server can be
    established.  This includes servers behind NATs.  (Note that a
    proxy between address domains may be required to get TCP through.)
 o  Improves defenses against DDoS attacks, since a media-receiving
    client requires authentications via STUN on its media reception
    ports.

Westerlund & Zeng Informational [Page 22] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

 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 demultiplex between the packets of
    the media protocol and STUN packets.  This is possible for RTP as
    discussed, for example, in Section 5.1.2 of [RFC5764].
 o  Has a fairly high implementation burden put on both the RTSP
    server and client.  However, several open source ICE
    implementations do exist, such as [NICE] and [PJNATH].

4.3.6. Security Considerations

 One should review the Security Considerations section of ICE and STUN
 to understand that ICE contains some potential issues.  However,
 these can be avoided by correctly using ICE in RTSP.  An important
 factor is to secure the signaling, i.e., use TLS between the RTSP
 client and server.  In fact ICE does help avoid the DDoS attack issue
 with RTSP substantially as it reduces the possibility for a DDoS
 using RTSP servers on attackers that are on path between the RTSP
 server and the target and capable of intercepting the STUN
 connectivity check packets and correctly sending a response to the
 server.  The ICE connectivity checks with their random transaction
 IDs from the server to the client serves as a return-routability
 check and prevents off-path attackers to succeed with address
 spoofing.  This is similar to Mobile IPv6's return routability
 procedure (Section 5.2.5 of [RFC6275]).

4.4. Latching

4.4.1. Introduction

 Latching is a NAT traversal solution that is based on requiring RTSP
 clients to send UDP packets to the server's media output ports.
 Conventionally, RTSP servers send RTP packets in one direction: from
 server to client.  Latching is similar to connection-oriented
 traffic, where one side (e.g., the RTSP client) first "connects" by
 sending an 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 be done symmetrically.  This in effect requires Symmetric
 RTP [RFC4961].  Refer to [RFC7362] for a description of the Latching
 of SIP-negotiated media streams in Session Border Controllers.
 Specifically, when the RTSP server receives the Latching packet
 (a.k.a. hole-punching packet, since it is used to punch a hole in the

Westerlund & Zeng Informational [Page 23] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

 firewall/NAT) from its client, it copies the source IP and Port
 number and uses them as the delivery address for media packets.  By
 having the server send media traffic back the same way as the
 client's packets are sent to the server, address and port mappings
 will be honored.  Therefore, this technique works for all types of
 NATs, given that the server is not behind a NAT.  However, it does
 require server modifications.  The format of the Latching packet will
 have to be defined.
 Latching is very vulnerable to both hijacking and becoming a tool in
 DDoS attacks (see Security Considerations in [RFC7362]) because
 attackers can simply forge the source IP and Port of the Latching
 packet.  The rule for restricting IP addresses to one of the
 signaling connections will need to be applied here also.  However,
 that does not protect against hijacking from another client behind
 the same NAT.  This can become a serious issue in deployments with
 CGNs.

4.4.2. Necessary RTSP Extensions

 To support Latching, RTSP signaling must be extended to allow the
 RTSP client to indicate that it will use Latching.  The client also
 needs to be able to signal its RTP SSRC to the server in its SETUP
 request.  The RTP SSRC is used to establish some basic level of
 security against hijacking attacks or simply to avoid mis-association
 when multiple clients are behind the same NAT.  Care must be taken in
 choosing clients' RTP SSRC.  First, it must be unique within all the
 RTP sessions belonging to the same RTSP session.  Second, if the RTSP
 server is sending out media packets to multiple clients from the same
 send port, the RTP SSRC needs to be unique among those clients' RTP
 sessions.  Recognizing that there is a potential that RTP SSRC
 collisions may occur, the RTSP server must be able to signal to a
 client that a collision has occurred and that it wants the client to
 use a different RTP SSRC carried in the SETUP response or use unique
 ports per RTSP session.  Using unique ports limits an RTSP server in
 the number of sessions it can simultaneously handle per interface IP
 addresses.
 The Latching packet as discussed above should have a field that can
 contain a client and RTP session identifier to correctly associate
 the Latching packet with the correct context.  If an RTP packet is to
 be used, there would be a benefit to using a well-defined RTP payload
 format for this purpose as the No-Op payload format proposed
 [RTP-NO-OP].  However, in the absence of such a specification, an RTP
 packet without a payload could be used.  Using SSRC is beneficial
 because RTP and RTCP both would work as is.  However, other packet
 formats could be used that carry the necessary identification of the
 context, and such a solution is discussed in Section 4.5.

Westerlund & Zeng Informational [Page 24] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

4.4.3. ALG Considerations

 An RTSP ALG not supporting this method could interfere with the
 methods used to indicate that Latching is to be done, as well as the
 SSRC signaling, thus preventing the method from working.  However, if
 the RTSP ALG instead opens the corresponding pinholes and creates the
 necessary mapping in the NAT, traversal will still work.  Securing
 the RTSP message transport using TLS will avoid this issue.
 An RTSP ALG that supports this traversal method can for basic
 functionality simply pass the related signaling parameters
 transparently.  Due to the security considerations for Latching,
 there might exist a benefit for an RTSP ALG that will enable NAT
 traversal to negotiate with the path and turn off the Latching
 procedures when the ALG handles this.  However, this opens up to
 failure modes when there are multiple levels of NAT and only one
 supports an RTSP ALG.

4.4.4. Deployment Considerations

 Advantages:
 o  Works for all types of client-facing NATs (requirement 1 in
    Section 3).
 o  Has little interaction problems with any RTSP ALG changing the
    client's information in the Transport header.
 Disadvantages:
 o  Requires modifications to both the RTSP server and client.
 o  Limited to working with servers that are not behind a NAT.
 o  The format of the packet for "connection setup" (a.k.a Latching
    packet) is not defined.
 o  SSRC management if RTP is used for Latching due to risk for mis-
    association of clients to RTSP sessions at the server if SSRC
    collision occurs.
 o  Has significant security considerations (See Section 4.4.5), due
    to the lack of a strong authentication mechanism and will need to
    use address restrictions.

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4.4.5. Security Considerations

 Latching's major security issue is that RTP streams can be hijacked
 and directed towards any target that the attacker desires unless
 address restrictions are used.  In the case of NATs with multiple
 clients on the inside of them, hijacking can still occur.  This
 becomes a significant threat in the context of CGNs.
 The most serious security problem is the deliberate attack with the
 use of an RTSP client and Latching.  The attacker uses RTSP to set up
 a media session.  Then it uses Latching with a spoofed source address
 of the intended target of the attack.  There is no defense against
 this attack other than restricting the possible address a Latching
 packet can come from to the same address as the RTSP TCP connection
 is from.  This prevents Latching to be used in use cases that require
 different addresses for media destination and signaling.  Even
 allowing only a limited address range containing the signaling
 address from where Latching is allowed opens up a significant
 vulnerability as it is difficult to determine the address usage for
 the network the client connects from.
 A hijack attack can also be performed in various ways.  The basic
 attack is based on the ability to read the RTSP signaling packets in
 order to learn the address and port the server will send from and
 also the SSRC the client will use.  Having this information, the
 attacker can send its own Latching packets containing the correct RTP
 SSRC to the correct address and port on the server.  The RTSP server
 will then use the source IP and Port from the Latching packet as the
 destination for the media packets it sends.
 Another variation of this attack is for a man in the middle to modify
 the RTP Latching packet being sent by a client to the server by
 simply changing the source IP and Port to the target one desires to
 attack.
 One can fend off the snooping-based attack by applying encryption to
 the RTSP signaling transport.  However, if the attacker is a man in
 the middle modifying Latching packets, the attack is impossible to
 defend against other than through address restrictions.  As a NAT
 rewrites the source IP and (possibly) port, this cannot be
 authenticated, but authentication is required in order to protect
 against this type of DoS attack.
 Yet another issue is that these attacks also can be used to deny the
 client the service it desires from the RTSP server completely.  The
 attacker modifies or originates its own Latching packets with a port

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 other than what the legit Latching packets use, which results in the
 media server sending the RTP/RTCP traffic to ports the client isn't
 listening for RTP/RTCP on.
 The amount of random non-guessable material in the Latching packet
 determines how well Latching can fend off stream hijacking performed
 by parties that are off the client-to-server network path, i.e., it
 lacks the capability to see the client's Latching packets.  The
 proposal above uses the 32-bit RTP SSRC field to this effect.
 Therefore, it is important that this field is derived with a non-
 predictable random number generator.  It should not be possible by
 knowing the algorithm used and a couple of basic facts to derive what
 random number a certain client will use.
 An attacker not knowing the SSRC but aware of which port numbers that
 a server sends from can deploy a brute-force attack on the server by
 testing a lot of different SSRCs until it finds a matching one.
 Therefore, a server could implement functionality that blocks packets
 to ports or from sources that receive or send multiple Latching
 packets with different invalid SSRCs, especially when they are coming
 from the same IP and Port.  Note that this mitigation in itself opens
 up a new venue for DoS attacks against legit users trying to latch.
 To improve the security against attackers, the amount of random
 material could be increased.  To achieve a longer random tag while
 still using RTP and RTCP, it will be necessary to develop RTP and
 RTCP payload formats for carrying the random material.

4.5. A Variation to Latching

4.5.1. Introduction

 Latching as described above requires the usage of a valid RTP format
 as the Latching packet, i.e., the first packet that the client sends
 to the server to establish a bidirectional transport flow for RTP
 streams.  There is currently no appropriate RTP packet format for
 this purpose, although the RTP No-Op format was a proposal to fix the
 problem [RTP-NO-OP]; however, that work was abandoned.  [RFC6263]
 discusses the implication of different types of packets as keep-
 alives for RTP, and its findings are very relevant to the format of
 the Latching packet.
 Meanwhile, there have been NAT/firewall traversal techniques deployed
 in the wireless streaming market place that use non-RTP messages as
 Latching packets.  This section describes a variant based on a subset
 of those solutions that alters the previously described Latching
 solution.

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4.5.2. Necessary RTSP Extensions

 In this variation of Latching, the Latching packet is a small UDP
 packet that does not contain an RTP header.  In response to the
 client's Latching packet, the RTSP server sends back a similar
 Latching packet as a confirmation so the client can stop the so-
 called "connection phase" of this NAT traversal technique.
 Afterwards, the client only has to periodically send Latching packets
 as keep-alive messages for the NAT mappings.
 The server listens on its RTP-media output port and tries to decode
 any received UDP packet as the Latching packet.  This is valid since
 an RTSP server is not expecting RTP traffic from the RTSP client.
 Then, it can correlate the Latching packet with the RTSP client's
 session ID or the client's SSRC and record the NAT bindings
 accordingly.  The server then sends a Latching packet as the response
 to the client.
 The Latching packet can contain the SSRC to identify the RTP stream,
 and care must be taken if the packet is bigger than 12 bytes,
 ensuring that it is distinctively different from RTP packets, whose
 header size is 12 bytes.
 RTSP signaling can be added to do the following:
 1.  Enable or disable such Latching message exchanges.  When the
     firewall/NAT has an RTSP-aware ALG, it is possible to disable
     Latching message exchange and let the ALG work out the address
     and port mappings.
 2.  Configure the number of retries and the retry interval of the
     Latching message exchanges.

4.5.3. ALG Considerations

 See Latching ALG considerations in Section 4.4.3.

4.5.4. Deployment Considerations

 This approach has the following advantages when compared with the
 Latching approach (Section 4.4):
 1.  There is no need to define an RTP payload format for firewall
     traversal; therefore, it is more simple to use, implement, and
     administer (requirement 4 in Section 3) than a Latching protocol,
     which must be defined.

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 2.  When properly defined, this kind of Latching packet exchange can
     also authenticate RTP receivers, to prevent hijacking attacks.
 This approach has the following disadvantage when compared with the
 Latching approach:
 1.  The server's sender SSRC for the RTP stream or other session
     Identity information must be signaled in the RTSP's SETUP
     response, in the Transport header of the RTSP SETUP response.

4.5.5. Security Considerations

 Compared to the security properties of Latching, this variant is
 slightly improved.  First of all it allows for a larger random field
 in the Latching packets, which makes it more unlikely for an off-path
 attacker to succeed in a hijack attack.  Second, the confirmation
 allows the client to know when Latching works and when it doesn't and
 thus when to restart the Latching process by updating the SSRC.
 Still, the main security issue remaining is that the RTSP server
 can't know that the source address in the Latching packet was coming
 from an RTSP client wanting to receive media and not from one that
 likes to direct the media traffic to a DoS target.

4.6. Three-Way Latching

4.6.1. Introduction

 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., a UDP packet with some session
 identifiers and a random value.  When the server responds to the
 Latching packet with a Latching confirmation, it includes a random
 value (nonce) of its own in addition to echoing back the one the
 client sent.  Then a third message is added to the exchange.  The
 client acknowledges the reception of the Latching confirmation

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 message and echoes back the server's nonce, thus confirming that the
 Latched address goes to an RTSP client that initiated the Latching
 and is actually present at that address.  The RTSP server will refuse
 to send any media until the Latching Acknowledgement has been
 received with a valid nonce.

4.6.3. ALG Considerations

 See Latching ALG considerations in Section 4.4.3.

4.6.4. Deployment Considerations

 A solution with a three-way handshake and its own Latching packets
 can be compared with the ICE-based solution (Section 4.3) and have
 the following differences:
 o  Only works for servers that are not behind a NAT.
 o  May be simpler to implement due to the avoidance of the ICE
    prioritization and check-board mechanisms.
 However, a Three-Way Latching protocol is very similar to using STUN
 in both directions as a Latching and verification protocol.  Using
 STUN would remove the need for implementing a new protocol.

4.6.5. Security Considerations

 Three-Way Latching is significantly more secure than its simpler
 versions discussed above.  The client-to-server nonce, which is
 included in signaling and also can be bigger than the 32 bits of
 random data that the SSRC field supports, makes it very difficult for
 an off-path attacker to perform a DoS attack by diverting the media.
 The client-to-server nonce and its echoing back does not protect
 against on-path attackers, including malicious clients.  However, the
 server-to-client nonce and its echoing back prevents malicious
 clients to divert the media stream by spoofing the source address and
 port, as it can't echo back the nonce in these cases.  This is
 similar to the Mobile IPv6 return routability procedure
 (Section 5.2.5 of [RFC6275]).
 Three-Way Latching is really only vulnerable to an on-path attacker
 that is quite capable.  First, the attacker can learn the client-
 to-server nonce either by intercepting the signaling or by modifying
 the source information (target destination) of a client's Latching
 packet.  Second, it is also on-path between the server and target
 destination and can generate a response using the server's nonce.  An

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 adversary that has these capabilities is commonly capable of causing
 significantly worse damage than this using other methods.
 Three-Way Latching results in the server-to-client packet being
 bigger than the client-to-server packet, due to the inclusion of the
 server-to-client nonce in addition to the client-to-server nonce.
 Thus, an amplification effect does exist; however, to achieve this
 amplification effect, the attacker has to create a session state on
 the RTSP server.  The RTSP server can also limit the number of
 responses it will generate before considering the Latching to be
 failed.

4.7. Application Level Gateways

4.7.1. Introduction

 An ALG reads the application level messages and performs necessary
 changes to allow the protocol to work through the middlebox.
 However, this behavior has some problems in regards to RTSP:
 1.  It does not work when RTSP 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 middlebox.
 2.  ALGs need to be updated if extensions to the protocol are added.
     Due to deployment issues with changing ALGs, this may also break
     the end-to-end functionality of RTSP.
 Due to the above reasons, it is not recommended to use an RTSP ALG in
 NATs.  This is especially important for NATs targeted to home users
 and small office environments, since it is very hard to upgrade NATs
 deployed in SOHO environments.

4.7.2. Outline on How ALGs for RTSP Work

 In this section, we provide a step-by-step outline on how one could
 go about writing an ALG to enable RTSP to traverse a NAT.
 1.  Detect any SETUP request.
 2.  Try to detect the usage of any of the NAT traversal methods that
     replace the address and port of the Transport header parameters
     "destination" or "dest_addr".  If any of these methods are used,
     then the ALG should not change the address.  Ways to detect that
     these methods are used are:

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  • For embedded STUN, it would be to watch for a feature tag,

like "nat.stun", and to see if any of those exist in the

        "supported", "proxy-require", or "require" headers of the RTSP
        exchange.
  • For stand-alone STUN and TURN-based solutions: This can be

detected by inspecting the "destination" or "dest_addr"

        parameter.  If it contains either one of the NAT's external IP
        addresses or a public IP address, then such a solution is in
        use.  However, if multiple NATs are used, this detection may
        fail.  Remapping should only be done for addresses belonging
        to the NAT's own private address space.
     Otherwise, continue to the next step.
 3.  Create UDP mappings (client given IP and Port <-> external IP and
     Port) where needed for all possible transport specifications in
     the Transport header of the request found in (step 1).  Enter the
     external address and port(s) of these mappings in the Transport
     header.  Mappings shall be created with consecutive external port
     numbers starting on an even number for RTP for each media stream.
     Mappings should also be given a long timeout period, at least 5
     minutes.
 4.  When the SETUP response is received from the server, the ALG may
     remove the unused UDP mappings, i.e., the ones not present in the
     Transport header.  The session ID should also be bound to the UDP
     mappings part of that session.
 5.  If the SETUP response settles on RTP over TCP or RTP over RTSP as
     lower transport, do nothing: let TCP tunneling take care of NAT
     traversal.  Otherwise, go to the next step.
 6.  The ALG should keep the UDP mappings belonging to the RTSP
     session as long as: an RTSP message with the session's ID has
     been sent in the last timeout interval, or a UDP message has been
     sent on any of the UDP mappings during the last timeout interval.
 7.  The ALG may remove a mapping as soon as a TEARDOWN response has
     been received for that media stream.

4.7.3. Deployment Considerations

 Advantages:
 o  No impact on either client or server.
 o  Can work for any type of NATs.

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 Disadvantages:
 o  When deployed, they are hard to update to reflect protocol
    modifications and extensions.  If not updated, they will break the
    functionality.
 o  When end-to-end security is used, the ALG functionality will fail.
 o  Can interfere with other types of traversal mechanisms, such as
    STUN.
 Transition:
 An RTSP ALG will not be phased out in any automatic way.  It must be
 removed, probably through the removal or update of the NAT it is
 associated with.

4.7.4. Security Considerations

 An ALG will not work with deployment of end-to-end RTSP signaling
 security; however, it will work with the hop-by-hop security method
 defined in Section 19.3 of RTSP 2.0 [RTSP].  Therefore, deployment of
 ALG may result in clients located behind NATs not using end-to-end
 security, or more likely the selection of a NAT traversal solution
 that allows for security.
 The creation of a UDP mapping based on the signaling message has some
 potential security implications.  First of all, if the RTSP client
 releases its ports and another application is assigned these instead,
 it could receive RTP media as long as the mappings exist and the RTSP
 server has failed to be signaled or notice the lack of client
 response.
 A NAT with RTSP ALG that assigns mappings based on SETUP requests
 could potentially become the victim of a resource exhaustion attack.
 If an attacker creates a lot of RTSP sessions, even without starting
 media transmission, this could exhaust the pool of available UDP
 ports on the NAT.  Thus, only a limited number of UDP mappings should
 be allowed to be created by the RTSP ALG.

4.8. TCP Tunneling

4.8.1. Introduction

 Using a TCP connection that is established from the client to the
 server ensures that the server can send data to the client.  The
 connection opened from the private domain ensures that the server can
 send data back to the client.  To send data originally intended to be

Westerlund & Zeng Informational [Page 33] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

 transported over UDP requires the TCP connection to support some type
 of framing of the media data packets.  Using TCP also results in the
 client having to accept that real-time performance can be impacted.
 TCP's problem of ensuring timely delivery was one of the reasons why
 RTP was developed.  Problems that arise with TCP are: head-of-line
 blocking, delay introduced by retransmissions, and a highly varying
 rate due to the congestion control algorithm.  If a sufficient amount
 of buffering (several seconds) in the receiving client can be
 tolerated, then TCP will clearly work.

4.8.2. Usage of TCP Tunneling in RTSP

 The RTSP core specification [RTSP] supports interleaving of media
 data on the TCP connection that carries RTSP signaling.  See
 Section 14 in [RTSP] for how to perform this type of TCP tunneling.
 There also exists another way of transporting RTP over TCP, which is
 defined in Appendix C.2 in [RTSP].  For signaling and rules on how to
 establish the TCP connection in lieu of UDP, see Appendix C.2 in
 [RTSP].  This is based on the framing of RTP over the TCP connection
 as described in [RFC4571].

4.8.3. ALG Considerations

 An RTSP ALG will face a different issue with TCP tunneling, at least
 the interleaved version.  Now the full data stream can end up flowing
 through the ALG implementation.  Thus, it is important that the ALG
 is efficient in dealing with the interleaved media data frames to
 avoid consuming to many resources and thus creating performance
 issues.
 The RTSP ALG can also affect the transport specifications that
 indicate that TCP tunneling can be done and its prioritization,
 including removing the transport specification, thus preventing TCP
 tunneling.

4.8.4. Deployment Considerations

 Advantage:
 o  Works through all types of NATs where the RTSP server is not NATed
    or is at least reachable like it was not.
 Disadvantages:
 o  Functionality needs to be implemented on both server and client.
 o  Will not always meet multimedia stream's real-time requirements.

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 Transition:
 The tunneling over RTSP's TCP connection is not planned to be phased
 out.  It is intended to be a fallback mechanism and for usage when
 total media reliability is desired, even at the potential price of
 loss of real-time properties.

4.8.5. Security Considerations

 The TCP tunneling of RTP has no known significant security problems
 besides those already presented in the RTSP specification.  It is
 difficult to get any amplification effect for DoS attacks due to
 TCP's flow control.  The RTSP server's TCP socket, if independently
 used for media tunneling or only RTSP messages, can be used for a
 redirected syn attack.  By spoofing the source address of any TCP
 init packets, the TCP SYNs from the server can be directed towards a
 target.
 A possible security consideration, when session media data is
 interleaved with RTSP, would be the performance bottleneck when RTSP
 encryption is applied, since all session media data also needs to be
 encrypted.

4.9. Traversal Using Relays around NAT (TURN)

4.9.1. Introduction

 TURN [RFC5766] is a protocol for setting up traffic relays that allow
 clients behind NATs and firewalls to receive incoming traffic for
 both UDP and TCP.  These relays are controlled and have limited
 resources.  They need to be allocated before usage.  TURN allows a
 client to temporarily bind an address/port pair on the relay (TURN
 server) to its local source address/port pair, which is used to
 contact the TURN server.  The TURN server will then forward packets
 between the two sides of the relay.
 To prevent DoS attacks on either recipient, the packets forwarded are
 restricted to the specific source address.  On the client side, it is
 restricted to the source setting up the allocation.  On the external
 side, it is limited to the source address/port pair that have been
 given permission by the TURN client creating the allocation.  Packets
 from any other source on this address will be discarded.
 Using a TURN server makes it possible for an RTSP client to receive
 media streams from even an unmodified RTSP server.  However, the
 problem is those RTSP servers most likely restrict media destinations
 to no other IP address than the one the RTSP message arrives from.
 This means that TURN could only be used if the server knows and

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 accepts that the IP belongs to a TURN server, and the TURN server
 can't be targeted at an unknown address.  Alternatively, both the
 RTSP TCP connection as well as the RTP media is relayed through the
 same TURN server.

4.9.2. Usage of TURN with RTSP

 To use a TURN server for NAT traversal, the following steps should be
 performed.
 1.  The RTSP client connects with the RTSP server.  The client
     retrieves the session description to determine the number of
     media streams.  To avoid the issue of having the RTSP connection
     and media traffic from different addresses, the TCP connection
     must also be done through the same TURN server as the one in the
     next step.  This will require the usage of TURN for TCP
     [RFC6062].
 2.  The client establishes the necessary bindings on the TURN server.
     It must choose the local RTP and RTCP ports that it desires to
     receive media packets.  TURN supports requesting bindings of even
     port numbers and contiguous ranges.
 3.  The RTSP client uses the acquired address and port allocations in
     the RTSP SETUP request using the destination header.
 4.  The RTSP server sends the SETUP reply, which must include the
     Transport header's "src_addr" parameter (source and port in RTSP
     1.0).  Note that the server is required to have a mechanism to
     verify that it is allowed to send media traffic to the given
     address unless TCP relaying of the RTSP messages also is
     performed.
 5.  The RTSP client uses the RTSP server's response to create TURN
     permissions for the server's media traffic.
 6.  The client requests that the server starts playing.  The server
     starts sending media packets to the given destination address and
     ports.
 7.  Media packets arrive at the TURN server on the external port; if
     the packets match an established permission, the TURN server
     forwards the media packets to the RTSP client.
 8.  If the client pauses and media is not sent for about 75% of the
     mapping timeout, the client should use TURN to refresh the
     bindings.

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4.9.3. ALG Considerations

 As the RTSP client inserts the address information of the TURN
 relay's external allocations in the SETUP messages, the ALG that
 replaces the address, without considering that the address does not
 belong to the internal address realm of the NAT, will prevent this
 mechanism from working.  This can be prevented by securing the RTSP
 signaling.

4.9.4. Deployment Considerations

 Advantages:
 o  Does not require any server modifications given that the server
    includes the "src_addr" header in the SETUP response.
 o  Works for any type of NAT as long as the RTSP server has a
    reachable IP address that is not behind a NAT.
 Disadvantages:
 o  Requires another network element, namely the TURN server.
 o  A TURN server for RTSP may not scale since the number of sessions
    it must forward is proportional to the number of client media
    sessions.
 o  The TURN server becomes a single point of failure.
 o  Since TURN forwards media packets, as a necessity it introduces
    delay.
 o  An RTSP ALG may change the necessary destinations parameter.  This
    will cause the media traffic to be sent to the wrong address.
 Transition:
 TURN is not intended to be phased out completely; see Section 19 of
 [RFC5766].  However, the usage of TURN could be reduced when the
 demand for having NAT traversal is reduced.

4.9.5. Security Considerations

 The TURN server can become part of a DoS attack towards any victim.
 To perform this attack, the attacker must be able to eavesdrop on the
 packets from the TURN server towards a target for the DoS attack.
 The attacker uses the TURN server to set up an RTSP session with
 media flows going through the TURN server.  The attacker is in fact

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 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 an RTSP session and start delivery of the
 media.  The attacker must be able to create these mappings.  The
 attacker in this case may be traced by the TURN username in the
 mapping requests.
 This attack requires that the attacker has access to a user account
 on the TURN server to be able to set up the TURN mappings.  To
 prevent this attack, the RTSP server needs to verify that the
 ultimate target destination accepts 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 an
 RTSP session to pass through it.  Therefore, the firewall will need
 an ALG that reads RTSP SETUP and TEARDOWN messages.  By reading the
 SETUP message, the firewall can determine what type of transport and
 from where the media stream packets will be sent.  Commonly, there
 will be the need to open UDP ports for RTP/RTCP.  By looking at the
 source and destination addresses and ports, the opening in the
 firewall can be minimized to the least necessary.  The opening in the
 firewall can be closed after a TEARDOWN message for that session or
 the session itself times out.
 The above possibilities for firewalls to inspect and respond to the
 signaling are prevented if end-to-end confidentiality protection is
 used for the RTSP signaling, e.g., using the specified RTSP over TLS.
 As a result, firewalls can't be actively opening pinholes for the
 media streams based on the signaling.  To enable an RTSP ALG in the
 firewall to correctly function, the hop-by-hop signaling security in
 RTSP 2.0 can be used (see Section 19.3 of [RTSP]).  If not, other
 methods have to be used to enable the transport flows for the media.
 Simpler firewalls do allow a client to receive media as long as it
 has sent packets to the target.  Depending on the security level,
 this can have the same behavior as a NAT.  The only difference is

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 that no address translation is done.  To use such a firewall, a
 client would need to implement one of the above described NAT
 traversal methods that include sending packets to the server to
 create the necessary filtering state.

6. Comparison of NAT Traversal Techniques

 This section evaluates the techniques described above against the
 requirements listed in Section 3.
 In the following table, the columns correspond to the numbered
 requirements.  For instance, the column under R1 corresponds to the
 first requirement in Section 3: must work for all flavors of NATs.
 The rows represent the different NAT/firewall traversal techniques.
 Latch is short for Latching, "V. Latch" is short for "variation of
 Latching" as described in Section 4.5, and "3-W Latch" is short for
 the Three-Way Latching described in Section 4.6.
 A summary of the requirements are:
 R1: Work for all flavors of NATs
 R2: Must work with firewalls, including those with ALGs
 R3: Should have minimal impact on clients not behind NATs, counted in
     minimal number of additional RTTs
 R4: Should be simple to use, implement, and administer
 R5: Should provide mitigation against DDoS attacks
 The following considerations are also added to the requirements:
 C1: Will the solution support both clients and servers behind NAT?
 C2: Is the solution robust as NAT behaviors change?

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  1. ———–+——+——+——+——+——+——+——+

| R1 | R2 | R3 | R4 | R5 | C1 | C2 |

  1. ———–+——+——+——+——+——+——+——+

STUN | No | Yes | 1 | Maybe| No | No | No |

  1. ———–+——+——+——+——+——+——+——+

Emb. STUN | Yes | Yes | 2 | Maybe| No | No | Yes |

  1. ———–+——+——+——+——+——+——+——+

ICE | Yes | Yes | 2.5 | No | Yes | Yes | Yes |

  1. ———–+——+——+——+——+——+——+——+

Latch | Yes | Yes | 1 | Maybe| No | No | Yes |

  1. ———–+——+——+——+——+——+——+——+

V. Latch | Yes | Yes | 1 | Yes | No | No | Yes |

  1. ———–+——+——+——+——+——+——+——+

3-W Latch | Yes | Yes | 1.5 | Maybe| Yes | No | Yes |

  1. ———–+——+——+——+——+——+——+——+

ALG |(Yes) | Yes | 0 | No | Yes | No | Yes |

  1. ———–+——+——+——+——+——+——+——+

TCP Tunnel | Yes | Yes | 1.5 | Yes | Yes | No | Yes |

  1. ———–+——+——+——+——+——+——+——+

TURN | Yes | Yes | 1 | No | Yes |(Yes) | Yes |

  1. ———–+——+——+——+——+——+——+——+
          Figure 1: Comparison of Fulfillment of Requirements
 Looking at Figure 1, one would draw the conclusion that using TCP
 Tunneling or Three-Way Latching are the solutions that best fulfill
 the requirements.  The different techniques were discussed in the
 MMUSIC WG.  It was established that the WG would pursue an ICE-based
 solution due to its generality and capability of also handling
 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 was some discussion in the WG about if
 the increased implementation burden of ICE is sufficiently motivated
 compared to a the Three-Way Latching solution for this generality.
 In the end, the authors believed that the reuse of ICE, greater
 flexibility, and any way needed to deploy a new solution were 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)" [RTSP-NAT].

Westerlund & Zeng Informational [Page 40] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

7. Security Considerations

 In the preceding sections, we have discussed security merits of the
 different NAT/firewall traversal methods for RTSP.  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
 the hijacking of sessions by a man in the middle.
 Each of the proposed solutions has security implications.  Using STUN
 will provide the same level of security as RTSP without transport-
 level security and source authentications, as long as the server does
 not allow media to be sent to a different IP address than the RTSP
 client request was sent from.
 Using Latching will have a higher risk of session hijacking or DoS
 than normal RTSP.  The reason is that there exists a probability that
 an attacker is able to guess the random bits that the client uses to
 prove its identity when creating the address bindings.  This can be
 solved in the variation of Latching (Section 4.5) with authentication
 features.  Still, both those variants of Latching are vulnerable
 against a deliberate attack from the RTSP client to redirect the
 media stream requested to any target assuming it can spoof the source
 address.  This security vulnerability is solved by performing a
 Three-way Latching procedure as discussed in Section 4.6.
 ICE resolves the binding vulnerability of Latching by using signed
 STUN messages, as well as requiring that both sides perform
 connectivity checks to verify that the target IP address in the
 candidate pair is both reachable and willing to respond.  ICE can,
 however, create a significant amount of traffic if the number of
 candidate pairs are large.  Thus, pacing is required and
 implementations should attempt to limit their number of candidates to
 reduce the number of packets.
 If the signaling between the ICE peers (RTSP client and server) is
 not confidentiality and integrity protected, ICE is vulnerable to
 attacks where the candidate list is manipulated.  The lack of
 signaling security will also simplify spoofing of STUN binding
 messages by revealing the secret used in signing.
 The usage of an RTSP ALG does not in itself increase the risk for
 session hijacking.  However, the deployment of ALGs as the sole
 mechanism for RTSP NAT traversal will prevent deployment of end-
 to-end encrypted RTSP signaling.

Westerlund & Zeng Informational [Page 41] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

 The usage of TCP tunneling has no known security problems.  However,
 it might provide a bottleneck when it comes to end-to-end RTSP
 signaling security if TCP tunneling is used on an interleaved RTSP
 signaling connection.
 The usage of TURN has severe risk of DoS attacks against a client.
 The TURN server can also be used as a redirect point in a DDoS attack
 unless the server has strict enough rules for who may create
 bindings.
 Since Latching and the variants of Latching have such big security
 issues, they should not be used at all.  Three-Way Latching as well
 as ICE mitigates these security issues and performs the important
 return-routability checks that prevent spoofed source addresses, and
 they should be recommended for that reason.  RTP ALGs are a security
 risk as they can create an incitement against using secure RTSP
 signaling.  That can be avoided as ALGs require trust in the
 middlebox, and that trust becomes explicit if one uses the hop-by-hop
 security solution as specified in Section 19.3 of RTSP 2.0.  [RTSP].
 The remaining methods can be considered safe enough, assuming that
 the appropriate security mechanisms are used and not ignored.

8. Informative References

 [NICE]      Libnice, "The GLib ICE implementation", June 2015,
             <http://nice.freedesktop.org/wiki/>.
 [PJNATH]    "PJNATH - Open Source ICE, STUN, and TURN Library", May
             2013, <http://www.pjsip.org/pjnath/docs/html/>.
 [RFC768]    Postel, J., "User Datagram Protocol", STD 6, RFC 768,
             DOI 10.17487/RFC0768, August 1980,
             <http://www.rfc-editor.org/info/rfc768>.
 [RFC793]    Postel, J., "Transmission Control Protocol", STD 7,
             RFC 793, DOI 10.17487/RFC0793, September 1981,
             <http://www.rfc-editor.org/info/rfc793>.
 [RFC2326]   Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
             Streaming Protocol (RTSP)", RFC 2326,
             DOI 10.17487/RFC2326, April 1998,
             <http://www.rfc-editor.org/info/rfc2326>.
 [RFC2588]   Finlayson, R., "IP Multicast and Firewalls", RFC 2588,
             DOI 10.17487/RFC2588, May 1999,
             <http://www.rfc-editor.org/info/rfc2588>.

Westerlund & Zeng Informational [Page 42] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

 [RFC2663]   Srisuresh, P. and M. Holdrege, "IP Network Address
             Translator (NAT) Terminology and Considerations",
             RFC 2663, DOI 10.17487/RFC2663, August 1999,
             <http://www.rfc-editor.org/info/rfc2663>.
 [RFC3022]   Srisuresh, P. and K. Egevang, "Traditional IP Network
             Address Translator (Traditional NAT)", RFC 3022,
             DOI 10.17487/RFC3022, January 2001,
             <http://www.rfc-editor.org/info/rfc3022>.
 [RFC3261]   Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
             A., Peterson, J., Sparks, R., Handley, M., and
             E. Schooler, "SIP: Session Initiation Protocol",
             RFC 3261, DOI 10.17487/RFC3261, June 2002,
             <http://www.rfc-editor.org/info/rfc3261>.
 [RFC3424]   Daigle, L., Ed. and IAB, "IAB Considerations for
             UNilateral Self-Address Fixing (UNSAF) Across Network
             Address Translation", RFC 3424, DOI 10.17487/RFC3424,
             November 2002, <http://www.rfc-editor.org/info/rfc3424>.
 [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,
             DOI 10.17487/RFC3489, March 2003,
             <http://www.rfc-editor.org/info/rfc3489>.
 [RFC3550]   Schulzrinne, H., Casner, S., Frederick, R., and V.
             Jacobson, "RTP: A Transport Protocol for Real-Time
             Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
             July 2003, <http://www.rfc-editor.org/info/rfc3550>.
 [RFC4566]   Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
             Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
             July 2006, <http://www.rfc-editor.org/info/rfc4566>.
 [RFC4571]   Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
             and RTP Control Protocol (RTCP) Packets over Connection-
             Oriented Transport", RFC 4571, DOI 10.17487/RFC4571, July
             2006, <http://www.rfc-editor.org/info/rfc4571>.
 [RFC4787]   Audet, F., Ed. and C. Jennings, "Network Address
             Translation (NAT) Behavioral Requirements for Unicast
             UDP", BCP 127, RFC 4787, DOI 10.17487/RFC4787, January
             2007, <http://www.rfc-editor.org/info/rfc4787>.

Westerlund & Zeng Informational [Page 43] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

 [RFC4961]   Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
             BCP 131, RFC 4961, DOI 10.17487/RFC4961, July 2007,
             <http://www.rfc-editor.org/info/rfc4961>.
 [RFC5245]   Rosenberg, J., "Interactive Connectivity Establishment
             (ICE): A Protocol for Network Address Translator (NAT)
             Traversal for Offer/Answer Protocols", RFC 5245,
             DOI 10.17487/RFC5245, April 2010,
             <http://www.rfc-editor.org/info/rfc5245>.
 [RFC5382]   Guha, S., Ed., Biswas, K., Ford, B., Sivakumar, S., and
             P.  Srisuresh, "NAT Behavioral Requirements for TCP",
             BCP 142, RFC 5382, DOI 10.17487/RFC5382, October 2008,
             <http://www.rfc-editor.org/info/rfc5382>.
 [RFC5389]   Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
             "Session Traversal Utilities for NAT (STUN)", RFC 5389,
             DOI 10.17487/RFC5389, October 2008,
             <http://www.rfc-editor.org/info/rfc5389>.
 [RFC5764]   McGrew, D. and E. Rescorla, "Datagram Transport Layer
             Security (DTLS) Extension to Establish Keys for the
             Secure Real-time Transport Protocol (SRTP)", RFC 5764,
             DOI 10.17487/RFC5764, May 2010,
             <http://www.rfc-editor.org/info/rfc5764>.
 [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,
             DOI 10.17487/RFC5766, April 2010,
             <http://www.rfc-editor.org/info/rfc5766>.
 [RFC6062]   Perreault, S., Ed. and J. Rosenberg, "Traversal Using
             Relays around NAT (TURN) Extensions for TCP Allocations",
             RFC 6062, DOI 10.17487/RFC6062, November 2010,
             <http://www.rfc-editor.org/info/rfc6062>.
 [RFC6263]   Marjou, X. and A. Sollaud, "Application Mechanism for
             Keeping Alive the NAT Mappings Associated with RTP / RTP
             Control Protocol (RTCP) Flows", RFC 6263,
             DOI 10.17487/RFC6263, June 2011,
             <http://www.rfc-editor.org/info/rfc6263>.
 [RFC6275]   Perkins, C., Ed., Johnson, D., and J. Arkko, "Mobility
             Support in IPv6", RFC 6275, DOI 10.17487/RFC6275, July
             2011, <http://www.rfc-editor.org/info/rfc6275>.

Westerlund & Zeng Informational [Page 44] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

 [RFC7362]   Ivov, E., Kaplan, H., and D. Wing, "Latching: Hosted NAT
             Traversal (HNT) for Media in Real-Time Communication",
             RFC 7362, DOI 10.17487/RFC7362, September 2014,
             <http://www.rfc-editor.org/info/rfc7362>.
 [RTP-NO-OP] Andreasen, F., "A No-Op Payload Format for RTP", Work in
             Progress, draft-ietf-avt-rtp-no-op-04, May 2007.
 [RTSP]      Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
             and M. Stiemerling, "Real Time Streaming Protocol 2.0
             (RTSP)", Work in Progress,
             draft-ietf-mmusic-rfc2326bis-40, February 2014.
 [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)", Work
             in Progress, draft-ietf-mmusic-rtsp-nat-22, July 2014.
 [STUN-IMPL] "Open Source STUN Client and Server", May 2013,
             <http://sourceforge.net/projects/stun/>.

Acknowledgements

 The authors would also like to thank all persons on the MMUSIC
 working group's mailing list that have commented on this document.
 Persons having contributed to this protocol, in no special order,
 are: Jonathan Rosenberg, Philippe Gentric, Tom Marshall, David Yon,
 Amir Wolf, Anders Klemets, Flemming Andreasen, Ari Keranen, Bill
 Atwood, Alissa Cooper, Colin Perkins, Sarah Banks, David Black, and
 Alvaro Retana.  Thomas Zeng would also like to give special thanks to
 Greg Sherwood of PacketVideo for his input into this memo.
 Section 1.1 contains text originally written for RFC 4787 by Francois
 Audet and Cullen Jennings.

Westerlund & Zeng Informational [Page 45] RFC 7604 Evaluation of NAT Traversal for RTSP September 2015

Authors' Addresses

 Magnus Westerlund
 Ericsson
 Farogatan 6
 Stockholm  SE-164 80
 Sweden
 Phone: +46 8 719 0000
 Email: magnus.westerlund@ericsson.com
 Thomas Zeng
 PacketVideo Corp
 Email: thomas.zeng@gmail.com

Westerlund & Zeng Informational [Page 46]

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