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The Comparison of Different Network Address Translator (NAT) Traversal Techniques for Media Controlled by Real-time Streaming Protocol (RTSP)

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Document Type
This is an older version of an Internet-Draft that was ultimately published as RFC 7604.
Authors Magnus Westerlund , Thomas Zeng
Last updated 2015-05-14 (Latest revision 2015-04-20)
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Stream WG state Submitted to IESG for Publication
Document shepherd Flemming Andreasen
Shepherd write-up Show Last changed 2014-02-10
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Network Working Group                                      M. Westerlund
Internet-Draft                                                  Ericsson
Intended status: Informational                                   T. Zeng
Expires: October 22, 2015                                 April 20, 2015

 The Comparison of Different Network Address Translator (NAT) Traversal
 Techniques for Media Controlled by Real-time Streaming Protocol (RTSP)


   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 Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on October 22, 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
   ( in effect on the date of

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   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  . . . . . . . . . . . . . . . . . . . . . . . .   3
     1.1.  Network Address Translators . . . . . . . . . . . . . . .   5
     1.2.  Firewalls . . . . . . . . . . . . . . . . . . . . . . . .   6
     1.3.  Glossary  . . . . . . . . . . . . . . . . . . . . . . . .   6
   2.  Detecting the loss of NAT mappings  . . . . . . . . . . . . .   7
   3.  Requirements on Solutions . . . . . . . . . . . . . . . . . .   8
   4.  NAT Traversal Techniques  . . . . . . . . . . . . . . . . . .  10
     4.1.  Stand-Alone STUN  . . . . . . . . . . . . . . . . . . . .  10
       4.1.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  10
       4.1.2.  Using STUN to traverse NAT without server
               modifications . . . . . . . . . . . . . . . . . . . .  11
       4.1.3.  ALG considerations  . . . . . . . . . . . . . . . . .  13
       4.1.4.  Deployment Considerations . . . . . . . . . . . . . .  14
       4.1.5.  Security Considerations . . . . . . . . . . . . . . .  15
     4.2.  Server Embedded STUN  . . . . . . . . . . . . . . . . . .  15
       4.2.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  15
       4.2.2.  Embedding STUN in RTSP  . . . . . . . . . . . . . . .  15
       4.2.3.  Discussion On Co-located STUN Server  . . . . . . . .  17
       4.2.4.  ALG considerations  . . . . . . . . . . . . . . . . .  17
       4.2.5.  Deployment Considerations . . . . . . . . . . . . . .  17
       4.2.6.  Security Considerations . . . . . . . . . . . . . . .  18
     4.3.  ICE . . . . . . . . . . . . . . . . . . . . . . . . . . .  18
       4.3.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  18
       4.3.2.  Using ICE in RTSP . . . . . . . . . . . . . . . . . .  19
       4.3.3.  Implementation burden of ICE  . . . . . . . . . . . .  21
       4.3.4.  ALG Considerations  . . . . . . . . . . . . . . . . .  21
       4.3.5.  Deployment Considerations . . . . . . . . . . . . . .  22
       4.3.6.  Security Consideration  . . . . . . . . . . . . . . .  22
     4.4.  Latching  . . . . . . . . . . . . . . . . . . . . . . . .  22
       4.4.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  22
       4.4.2.  Necessary RTSP extensions . . . . . . . . . . . . . .  23
       4.4.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  24
       4.4.4.  Deployment Considerations . . . . . . . . . . . . . .  24
       4.4.5.  Security Consideration  . . . . . . . . . . . . . . .  25
     4.5.  A Variation to Latching . . . . . . . . . . . . . . . . .  26
       4.5.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  26
       4.5.2.  Necessary RTSP extensions . . . . . . . . . . . . . .  27
       4.5.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  27
       4.5.4.  Deployment Considerations . . . . . . . . . . . . . .  27

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       4.5.5.  Security Considerations . . . . . . . . . . . . . . .  28
     4.6.  Three Way Latching  . . . . . . . . . . . . . . . . . . .  28
       4.6.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  28
       4.6.2.  Necessary RTSP extensions . . . . . . . . . . . . . .  28
       4.6.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  29
       4.6.4.  Deployment Considerations . . . . . . . . . . . . . .  29
       4.6.5.  Security Considerations . . . . . . . . . . . . . . .  29
     4.7.  Application Level Gateways  . . . . . . . . . . . . . . .  30
       4.7.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  30
       4.7.2.  Outline On how ALGs for RTSP work . . . . . . . . . .  30
       4.7.3.  Deployment Considerations . . . . . . . . . . . . . .  31
       4.7.4.  Security Considerations . . . . . . . . . . . . . . .  32
     4.8.  TCP Tunneling . . . . . . . . . . . . . . . . . . . . . .  32
       4.8.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  33
       4.8.2.  Usage of TCP tunneling in RTSP  . . . . . . . . . . .  33
       4.8.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  33
       4.8.4.  Deployment Considerations . . . . . . . . . . . . . .  33
       4.8.5.  Security Considerations . . . . . . . . . . . . . . .  34
     4.9.  TURN (Traversal Using Relay NAT)  . . . . . . . . . . . .  34
       4.9.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  34
       4.9.2.  Usage of TURN with RTSP . . . . . . . . . . . . . . .  35
       4.9.3.  ALG Considerations  . . . . . . . . . . . . . . . . .  36
       4.9.4.  Deployment Considerations . . . . . . . . . . . . . .  36
       4.9.5.  Security Considerations . . . . . . . . . . . . . . .  37
   5.  Firewalls . . . . . . . . . . . . . . . . . . . . . . . . . .  37
   6.  Comparison of NAT traversal techniques  . . . . . . . . . . .  38
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  40
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .  40
   9.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  41
   10. Informative References  . . . . . . . . . . . . . . . . . . .  41
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  44

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 Real-time Streaming Protocol (RTSP)
   [RFC2326][I-D.ietf-mmusic-rfc2326bis], needs to deal with such
   discontinuities caused by NATs.  The problem is that, being a media
   control protocol managing one or more media streams, RTSP carries
   network address and port information within its protocol messages.
   Because of this, even if RTSP itself, when carried over Transmission
   Control Protocol (TCP) [RFC0793] for example, is not blocked by NATs,
   its media streams may be blocked by NATs.  This will occur unless
   special protocol provisions are added to support NAT-traversal.

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   Like NATs, firewalls are also middle boxes that need to be
   considered.  Firewalls help prevent unwanted traffic from getting in
   or out of the protected network.  RTSP is designed such that a
   firewall can be configured to let RTSP controlled media streams go
   through with 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 uses a filtering behavior that appear the same to
   what NATs have.  This type of firewall will be successfully traversed
   using the same solution as employed for NAT traversal, instead of
   relying on a 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 protocol extensions for each method, transition
   strategies, and security concerns.  The transition strategies are a
   discussion of how and if the method encourage a move towards not
   having any NATs on the path.

   This document is capturing the evaluation done in the process to
   recommend firewall/NAT traversal methods for RTSP streaming servers
   based on RFC 2326 [RFC2326] as well as the RTSP 2.0 core spec
   [I-D.ietf-mmusic-rfc2326bis].  The evaluation is focused on NAT
   traversal for the media streams carried over User Datagram Protocol
   (UDP) [RFC0768] with Real-time Transport Protocol (RTP) [RFC3550]
   over UDP being the main case for such usage.  The findings should be
   applicable to other protocols as long as they have similar

   At the time when the bulk of work on this document was done, a single
   level of NAT was the dominant deployment for NATs, and multiple level
   of NATs, including Carrier Grade NATs (CGNs) was not considered.
   Thus, any characterizations or findings may not be applicable in such
   scenarios, unless CGN or multiple level of NATs are explicitly noted.

   An ICE-based RTSP NAT traversal mechanism is specified in "A Network
   Address Translator (NAT) Traversal mechanism for media controlled by
   Real-Time Streaming Protocol (RTSP)" [I-D.ietf-mmusic-rtsp-nat].

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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.  In this
   specification, the term "NAT" refers to both "Basic NAT" and "Network
   Address/Port Translator (NAPT)"."

   "This document uses the term "address and port mapping" as the
   translation between an external address and port and an internal
   address and port.  Note that this is not the same as an "address
   binding" as defined in RFC 2663."

      Note: In the above it would be more correct to use external IP
      address instead of public IP address in the above text.  The
      external IP address is commonly a public one, but might be of
      other type if the NAT's external side is in a private address

   In addition to the above quote there exists a number of address and
   port mapping behaviors described in more detail in Section 4.1 of
   "Network Address Translation (NAT) Behavioral Requirements for
   Unicast UDP" [RFC4787] that are highly relevant to the discussion in
   this document.

   NATs also have a filtering behavior on traffic arriving on the
   external side.  Such behavior affects how well different methods for

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   NAT traversal works through these NATs.  See Section 5 of "Network
   Address Translation (NAT) Behavioral Requirements for Unicast UDP"
   [RFC4787] for more information on the different types of filtering
   that have been identified.

1.2.  Firewalls

   A firewall is a security gateway that enforces certain access control
   policies between two network administrative domains: a private domain
   (intranet) and a external domain, e.g.  Internet.  Many organizations
   use firewalls to prevent intrusions and an malicious attacks on
   computing resources in the private intranet [RFC2588].

   A comparison between NAT and firewall is given below:

   1.  A firewall sits at security enforcement/protection points, while
       NAT sits at borders between two address domains.

   2.  NAT does not in itself provide security, although some access
       control policies can be implemented using address translation
       schemes.  The inherent filtering behaviours are commonly mistaken
       for real security policies.

   It should be noted that many NAT devices intended for Residential or
   small office/home office (SOHO) use include both NATs and firewall

1.3.  Glossary

   Address-Dependent Mapping:  The NAT reuses the port mapping for
         subsequent packets sent from the same internal IP address and
         port to the same external IP address, regardless of the
         external port.  See [RFC4787].

   Address and Port-Dependent Mapping:  The NAT reuses the port mapping
         for subsequent packets sent from the same internal IP address
         and port to the same external IP address and port while the
         mapping is still active.  See [RFC4787].

   ALG:  Application Level Gateway, an entity that can be embedded in a
         NAT or other middlebox to perform the application layer
         functions required for a particular protocol to traverse the

   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].

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   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

   RTT:  Round Trip Times.

   SDP:  Session Description Protocol, see [RFC4566].

   SSRC: Synchronization source in RTP, see [RFC3550].

2.  Detecting the loss of NAT mappings

   Several NAT traversal techniques in the next chapter make use of the
   fact that the NAT UDP mapping's external address and port can be
   discovered.  This information is then utilized to traverse the NAT
   box.  However any such information is only good while the mapping is
   still valid.  As the IAB's UNSAF document [RFC3424] points out, the
   mapping can either timeout or change its properties.  It is therefore
   important for the NAT traversal solutions to handle the loss or
   change of NAT mappings, according to RFC3424.

   First, since NATs may also dynamically reclaim or readjust address/
   port translations, "keep-alive" and periodic re-polling may be
   required according to RFC 3424.  Secondly, it is possible to detect
   and recover from the situation where the mapping has been changed or
   removed.  The loss of a mapping can be detected when no traffic
   arrives for a while.  Below we will give some recommendation on how
   to detect loss of NAT mappings when using RTP/RTCP under RTSP

   A RTP session normally has both RTP and RTCP streams.  The loss of a
   RTP mapping can only be detected when expected traffic does not
   arrive.  If a client does not receive media data within a few seconds
   after having received the "200 OK" response to a RTSP PLAY request
   which starts the media delivery, it may be the result of a middlebox

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   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 has 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 middle box NAT/firewall, or the client is mal-

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. server is not behind a NAT.  The
   server behind a NAT is desirable, but of much lower priority.

   An important consideration for any NAT traversal technique is whether
   any protocol modification needs occur, where the implementation
   burden occur, server, client or middlebox.  If the incitement to get
   RTSP to work over a NAT is sufficient to motivate the owner of the
   server, client or middlebox to update or configure or otherwise
   perform changes to the device and its software to support the NAT
   traversal.  Thus, the question of who this burden falls on and how
   big it is is highly relevant.

   The list of feature requirements for RTSP NAT solutions are given

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   1.  Must work for all flavors of NATs, including NATs with Address
       and Port-Dependent Filtering.

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

   3.  Should have minimal impact on clients not behind NATs and which
       are not dual-hosted.  RTSP dual-hosting means that the RTSP
       signalling protocol and the media protocol (e.g.  RTP) are
       implemented on different computers with different IP addresses.

       *  For instance, no extra 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

   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 posts a DDoS threat that has
   significant amplification potentials because the RTP media streams in
   general consist of large number of IP packets.  DDoS attacks 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 in order to degrade the services
   provided by

   Note 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 is present, See 21.2.1 in
   [I-D.ietf-mmusic-rfc2326bis].  With the increased deployment of NAT
   middleboxes by operators, i.e. carrier grade NAT (CGN), the reuse of
   an IP address on the NAT's external side by many customers reduces
   the protection provided.  Also in some applications (e.g.,
   centralized conferencing), dual-hosted RTSP/RTP clients have valid

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   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
   standardization works' outcome or their failure to progress within
   IETF after the initial evaluation in this document.  For example RTP
   No-Op [I-D.ietf-avt-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 mappings
   assigned by the NAT.  Then using the knowledge of these NAT mappings
   use the external addresses to directly connect to the independent
   RTSP server.  However, this is only possible if the NAT 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

   STUN is a client-server protocol.  The STUN client sends a request to
   a STUN server and the server returns a response.  There are two types
   of STUN messages - Binding Requests and Indications.  Binding
   requests are used when determining a client's external address and
   solicits a response from the STUN server with the seen address.

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   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 RFC3489 functionality 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
   suggest using the RTP NO-OP payload format [I-D.ietf-avt-rtp-no-op]
   for key-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).


   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 (STUN server and RTSP
      server on same IP address).  Clients behind NATs that do "Address
      and Port-Dependent" Mappings cannot use this method.  See
      [RFC4787] for full definition of these terms.

   o  Based on the discontinued middlebox classification of the replaced
      STUN specification [RFC3489].  Thus brittle and unreliable.


   A RTSP client using RTP transport over UDP can use STUN to traverse a
   NAT(s) in the following way:

   1.  Use STUN to try to discover the type of NAT, and the timeout
       period for any UDP mapping on the NAT.  This is recommended to be
       performed in the background as soon as IP connectivity is
       established.  If this is performed prior to establishing a
       streaming session the delays in the session establishment will be
       reduced.  If no NAT is detected, normal SETUP should be used.

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   2.  The RTSP client determines the number of UDP ports needed by
       counting the number of needed media transport protocols sessions
       in the multi-media presentation.  This information is available
       in the media description protocol, e.g.  SDP [RFC4566].  For
       example, each RTP session will in general require two UDP ports,
       one for RTP, and one for RTCP.

   3.  For each UDP port required, establish a mapping and discover the
       public/external IP address and port number with the help of the
       STUN server.  A successful mapping looks like: client's local
       address/port <-> public address/port.

   4.  Perform the RTSP SETUP for each media.  In the transport header
       the following parameter should be included with the given values:
       "dest_addr" [I-D.ietf-mmusic-rfc2326bis] or "destination" +
       "client_port" [RFC2326] with the public/external IP address and
       port pair for both RTP and RTCP.  To be certain that this works
       servers must allow a client to setup the RTP stream on any port,
       not only even ports and with non-contiguous port numbers for RTP
       and RTCP.  This requires the new feature provided in RTSP 2.0
       [I-D.ietf-mmusic-rfc2326bis].  The server should respond with a
       transport header containing an "src_addr" or "source" +
       "server_port" parameters with the RTP and RTCP source IP address
       and port of the media stream.

   5.  To keep the mappings alive, the client should periodically send
       UDP traffic over all mappings needed for the session.  For the
       mapping carrying RTCP traffic the periodic RTCP traffic are
       likely enough.  For mappings carrying RTP traffic and for
       mappings carrying RTCP packets at too low a frequency, keep-alive
       messages should be sent.

   If a UDP mapping is lost, the above discovery process must be
   repeated.  The media stream also needs to be SETUP again to change
   the transport parameters to the new ones.  This will cause a glitch
   in media playback.

   To allow UDP packets to arrive from the server to a client behind a
   "Address Dependent" or "Address and Port Dependent" filtering NAT,
   the client must first send a UDP packet to establish filtering state
   in the NAT.  The client, before sending a RTSP PLAY request, must
   send a so called hole-punching packet 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 need to include the

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   "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.

   As hole-punching and keep-alive messages, one could have used the RTP
   No-Op packet [I-D.ietf-avt-rtp-no-op] had they been defined.  That
   would have ensured that the traffic would look like RTP and thus
   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, a
   RTP packet without any payload, or an 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 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 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 which contains addresses other than the NATs
   internal address domain.  In cases where the ALG modifies fields
   unnecessarily two alternatives exist:

   1.  Use TLS to encrypt the RTSP TCP connection to prevent the ALG
       from reading and modifying the RTSP messages.

   2.  Turn off the STUN based NAT traversal mechanism

   As it may be difficult to determine why the failure occurs, the usage
   of TLS protected RTSP message exchange at all times would avoid this

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4.1.4.  Deployment Considerations

   For the Stand-Alone usage of STUN the following applies:


   o  STUN is a solution first used by SIP [RFC3261] based applications
      (See section 1 and 2 of [RFC5389]).  As shown above, with little
      or no changes, the RTSP application can re-use STUN as a NAT
      traversal solution, avoiding the pit-fall of solving a problem

   o  Using STUN does not require RTSP server modifications, assuming it
      is a RTSP 2.0 compliant server; it only affects the client


   o  Requires a STUN server deployed in the same address domain as the

   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

   o  Does not work with port and address 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 a RTSP-aware ALG interferes with
      the use of STUN for NAT traversal unless TLS secured RTSP message
      exchange is used.

   o  Using STUN requires that RTSP servers and clients support the
      updated RTSP specification [I-D.ietf-mmusic-rfc2326bis], because
      it is no longer possible to guarantee that RTP and RTCP ports are
      adjacent to each other, as required by the "client_port" and
      "server_port" parameters in RFC2326.


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   The usage of STUN can be phased out gradually as the first step of a
   STUN capable server or client should be to check the presence of
   NATs.  The removal of STUN capability in the client implementations
   will have to wait until there is absolutely no need to use STUN.

4.1.5.  Security Considerations

   To prevent the RTSP server from being used as Denial of Service (DoS)
   attack tools the RTSP Transport header parameter "destination" and
   "dest_addr" are generally not allowed to point to any IP address
   other than the one the RTSP message originates from.  The RTSP server
   is only prepared to make an exception to this rule when the client is
   trusted (e.g., through the use of a secure authentication process, or
   through some secure method of challenging the destination to verify
   its willingness to accept the RTP traffic).  Such a restriction means
   that STUN in general does not work for use cases where RTSP and media
   transport go to different addresses.

   STUN combined with destination address restricted RTSP has the same
   security properties as the core RTSP.  It is protected from being
   used as a DoS attack tool unless the attacker has the ability to
   spoof the TCP connection carrying RTSP messages.

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

4.2.  Server Embedded STUN

4.2.1.  Introduction

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

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:

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   1.  It does not work if both RTSP client and RTSP server are behind
       separate NATs.

   2.  The RTSP server may, for security reasons, refuse to send media
       streams to an IP different from the IP in the client RTSP

   Deviations from STUN as defined in RFC 5389:

   1.  The RTSP application must provision the client with an identity
       and shared secret to use in the STUN authentication;

   2.  We require STUN server to be co-located on RTSP server's media
       source ports.

   If STUN server is co-located with RTSP server's media source port, an
   RTSP client using RTP transport over UDP can use STUN to traverse ALL
   types of NATs.  In the case of port and address dependent mapping
   NATs, the party on the inside of the NAT must initiate UDP traffic.
   The STUN Binding Request, being a UDP packet itself, can serve as the
   traffic initiating packet.  Subsequently, both the STUN Binding
   Response packets and the RTP/RTCP packets can traverse the NAT,
   regardless of whether the RTSP server or the RTSP client is behind
   NAT (however only one of the can be behind a NAT).

   Likewise, if an RTSP server is behind a NAT, then an embedded STUN
   server must be co-located on the RTSP client's RTCP port.  Also it
   will become the client that needs to disclose his destination address
   rather than the server, so the server can correctly determine its NAT
   external source address for the media streams.  In this case, we
   assume that the client has some means of establishing TCP connection
   to the RTSP server behind NAT so as to exchange RTSP messages with
   the RTSP server, potentially using a proxy or static rules.

   To minimize delay, we require that the RTSP server supporting this
   option must inform the client about the RTP and RTCP ports from where
   the server will send out RTP and RTCP packets, respectively.  This
   can be done by using the "server_port" parameter in RFC2326, and the
   "src_addr" parameter in [I-D.ietf-mmusic-rfc2326bis].  Both are in
   the RTSP Transport header.  But in general this strategy will require
   that one first do one SETUP request per media to learn the server
   ports, then perform the STUN checks, followed by a subsequent SETUP
   to change the client port and destination address to what was learned
   during the STUN checks.

   To be certain that RTCP works correctly the RTSP end-point (server or
   client) will be required to send and receive RTCP packets from the
   same port.

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4.2.3.  Discussion On Co-located STUN Server

   In order to use STUN to traverse "address and port dependent"
   filtering or mapping NATs the STUN server needs to be co-located with
   the streaming server media output ports.  This creates a de-
   multiplexing problem: we must be able to differentiate a STUN packet
   from a media packet.  This will be done based on heuristics.  The
   existing STUN heuristics is the first byte in the packet and the
   Magic Cookie field (added in RFC5389), which works fine between STUN
   and RTP or RTCP where the first byte happens to be different.  Thanks
   to the magic cookie field it is unlikely that other protocols would
   be mistaken for a STUN packet, but not assured.  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).

4.2.5.  Deployment Considerations

   For the "Embedded STUN" method the following applies:


   o  STUN is a solution first used by SIP applications.  As shown
      above, with little or no changes, RTSP application can re-use STUN
      as a NAT traversal solution, avoiding the pit-fall of solving a
      problem twice.

   o  STUN has built-in message authentication features, which makes it
      more secure against hi-jacking attacks.  See next section for an
      in-depth security discussion.

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

   o  Compared to other UDP based NAT traversal methods in this
      document, STUN requires little new protocol development (since
      STUN is already a IETF standard), and most likely less
      implementation effort, since open source STUN server and client
      implementations are available [STUN-IMPL][PJNATH].


   o  Some extensions to the RTSP core protocol, likely signaled by RTSP
      feature tags, must be introduced.

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   o  Requires an embedded STUN server to be co-located on each of the
      RTSP server's media protocol's ports (e.g.  RTP and RTCP ports),
      which means more processing is required to de-multiplex STUN
      packets from media packets.  For example, the de-multiplexer must
      be able to differentiate a RTCP RR packet from a STUN packet, and
      forward the former to the streaming server, and the latter to the
      STUN server.

   o  Does not support use cases that require the RTSP connection and
      the media reception to happen at different addresses, unless the
      server's security policy is relaxed.

   o  Interaction problems exist when a RTSP ALG is not aware of STUN
      unless TLS is used to protect the RTSP messages.

   o  Using STUN requires that RTSP servers and clients support the
      updated RTSP specification [I-D.ietf-mmusic-rfc2326bis], and they
      both agree to support the NAT traversal feature.

   o  Increases the setup delay with at least the amount of time it
      takes to perform STUN message exchanges.  Most likely an extra
      SETUP sequence will be required.


   The usage of STUN can be phased out gradually as the first step of a
   STUN capable machine can be to check the presence of NATs for the
   presently used network connection.  The removal of STUN capability in
   the client implementations will have to wait until there is
   absolutely no need to use STUN, i.e. no NATs or firewalls.

4.2.6.  Security Considerations

   See Stand-Alone STUN (Section 4.1.5).

4.3.  ICE

4.3.1.  Introduction

   ICE (Interactive Connectivity Establishment) [RFC5245] is a
   methodology for NAT traversal that has been developed for SIP using
   SDP offer/answer.  The basic idea is to try, in a staggered parallel
   fashion, all possible connection addresses that an endpoint may be
   reachable by.  This allows the endpoint to use the best available UDP
   "connection" (meaning two UDP end-points capable of reaching each
   other).  The methodology has very nice properties in that basically
   all NAT topologies are possible to traverse.

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   Here is how ICE works at a high level.  End point A collects all
   possible addresses that can be used, including local IP addresses,
   STUN derived addresses, TURN addresses, etc.  On each local port that
   any of these address and port pairs lead to, a STUN server is
   installed.  This STUN server only accepts STUN requests using the
   correct authentication through the use of a username and password.

   End-point A then sends a request to establish connectivity with end-
   point B, which includes all possible "destinations" [RFC5245] to get
   the media through to A.  Note that each of A's local address/port
   pairs (host candidates and server reflexive base) has a STUN server
   co-located.  B in turn provides A with all its possible destinations
   for the different media streams.  A and B then uses a STUN client to
   try to reach all the address and port pairs specified by A from its
   corresponding destination ports.  The destinations for which the STUN
   requests successfully complete are then indicated and one is

   If B fails to get any STUN response from A, all hope is not lost.
   Certain NAT topologies require multiple tries from both ends before
   successful connectivity is accomplished and therefore requests are
   retransmitted multiple times.  The STUN requests may also result in
   that more connectivity alternatives (destinations) are discovered and
   conveyed in the STUN responses.

4.3.2.  Using ICE in RTSP

   The usage of ICE for RTSP requires that both client and server be
   updated to include the ICE functionality.  If both parties implement
   the necessary functionality the following steps could provide ICE
   support for RTSP.

   This assumes that it is possible to establish a TCP connection for
   the RTSP messages between the client and the server.  This is not
   trivial in scenarios where the server is located behind a NAT, and
   may require some TCP ports be opened, or the deployment of proxies,

   The negotiation of ICE in RTSP of necessity will work different than
   in SIP with SDP offer/answer.  The protocol interactions are
   different and thus the possibilities for transfer of states are also
   somewhat different.  The goal is also to avoid introducing extra
   delay in the setup process at least for when the server is not behind
   a NAT in regards to the client, and the client is either having an
   address in the same address domain, or is behind NAT(s) which can
   address the address domain of the server.  This process is only
   intended to support PLAY mode, i.e. media traffic flows from server
   to client.

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   1.  The ICE usage begins in the SDP.  The SDP for the service
       indicates that ICE is supported at the server.  No candidates can
       be given here as that would not work with the on demand, DNS load
       balancing, etc., that have the SDP indicate a resource on a
       server park rather than a specific machine.

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

   3.  In each SETUP request the client includes its candidates in an
       ICE specific transport specification.  This indicates for the
       server the ICE support by the client.  One candidate is the most
       prioritized candidate and here the prioritization for this
       address should be somewhat different compared to SIP.  High
       performance candidates is recommended rather than candidates with
       the highest likellihood of success, as it is more likely that a
       server is not behind a NAT compared to a SIP user-agent.

   4.  The server responds to the SETUP (200 OK) for each media stream
       with its candidates.  A server not behind a NAT usually only
       provides a single ICE candidate.  Also here one candidate is the
       server primary address.

   5.  The connectivity checks are performed.  For the server the
       connectivity checks from the server to the clients have an
       additional usage.  They verify that there is someone willing to
       receive the media, thus preventing the server from unknowingly
       performing a DoS attack.

   6.  Connectivity checks from the client promoting a candidate pair
       were successful.  Thus no further SETUP requests are necessary
       and processing can proceed with step 7.  If another address than
       the primary has been verified by the client to work, that address
       may then be promoted for usage in a SETUP request (Go to 7).  If
       the checks for the available candidates failed and if further
       candidates have been derived during the connectivity checks, then
       those can be signalled in new candidate lines in a SETUP request
       updating the list (Go to 5).

   7.  Client issues PLAY request.  If the server also has completed its
       connectivity checks for the promoted candidate pair (based on
       username as it may be derived addresses if the client was behind
       NAT) then it can directly answer 200 OK (Go to 8).  If the
       connectivity check has not yet completed it responds with a 1xx
       code to indicate that it is verifying the connectivity.  If that
       fails within the set timeout, an error is reported back.  Client
       needs to go back to 6.

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   8.  Process completed and media can be delivered.  ICE candidates not
       used may be released.

   To keep media paths alive the client needs to periodically send data
   to the server.  This will be realized with STUN.  RTCP sent by the
   client should be able to keep RTCP open but STUN will also be used
   based on the same motivations as for ICE for SIP.

4.3.3.  Implementation burden of ICE

   The usage of ICE will require that a number of new protocols and new
   RTSP/SDP features be implemented.  This makes ICE the solution that
   has the largest impact on client and server implementations among all
   the NAT/firewall traversal methods in this document.

   RTSP server implementation requirements are:

   o  STUN server features

   o  Limited STUN client features

   o  SDP generation with more parameters.

   o  RTSP error code for ICE extension

   RTSP client implementation requirements are:

   o  Limited STUN server features

   o  Limited STUN client features

   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 manifest itself in the RTSP message primarily as
   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 signalling
   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 filtering
   state that would allow incoming STUN messages prior to any outgoing

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   hole-punching packets, and in that way speed up the connectivity
   checks and reduce the risk of failures.

4.3.5.  Deployment Considerations


   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


   o  Increases the setup delay with at least the amount of time it
      takes for the server to perform its STUN requests.

   o  Assumes that it is possible to de-multiplex between the packets of
      the media protocol and STUN packets.  This is possible for RTP as
      discussed for example in Section 5.1.2 of [RFC5764].

   o  Has fairly high implementation burden put on both RTSP server and
      client.  However, several Open Source ICE implementations do
      exist, such as [NICE][PJNATH].

4.3.6.  Security Consideration

   One should review the security consideration section of ICE and STUN
   to understand that ICE contains some potential issues.  However these
   can be avoided by correctly using ICE in RTSP.  An important factor
   is to secure the signalling, i.e. use TLS between RTSP client and
   server.  In fact ICE does help avoid the DDoS attack issue with RTSP
   substantially as it reduces the possibility for a DDoS using RTSP
   servers to attackers that are on-path between the RTSP server and the
   target and capable of intercepting the STUN connectivity check
   packets and correctly send a response to the server.

4.4.  Latching

4.4.1.  Introduction

   Latching 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

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   server to client.  Latching is similar to connection-oriented
   traffic, where one side (e.g., the RTSP client) first "connects" by
   sending a RTP packet to the other side's RTP port, the recipient then
   replies to the originating IP and port.  This method is also referred
   to as "Late binding".  It requires that all RTP/RTCP transport is
   done symmetrical, i.e. Symmetric RTP [RFC4961].  There exist a
   description for latching of SIP negotiated media streams in Session
   Border Controllers [RFC7362].

   Specifically, when the RTSP server receives the latching packet
   (a.k.a. hole-punching packet, since it is used to punch a hole in the
   firewall/NAT and to aid the server for port binding and address
   mapping) from its client, it copies the source IP and Port number and
   uses them as delivery address for media packets.  By having the
   server send media traffic back the same way as the client's packet
   are sent to the server, address mappings will be honored.  Therefore
   this technique works for all types of NATs, given that the server is
   not behind a NAT.  However, it does require server modifications.
   The format of the latching packet will have to be defined.

   Latching is very vulnerable to both hijacking and becoming a tool in
   Distributed Denial of Service (DDoS) attacks (See Security
   Considerations of [RFC7362]), because attackers can simply forge the
   source IP & Port of the latching packet.  Using the rule for
   restricting IP address to the one of the signaling connection will
   need to be applied here also.  However, that does not protect against
   hijacking from another client behind the same NAT.  This can become a
   serious issue in deployments with CGNs.

4.4.2.  Necessary RTSP extensions

   To support Latching, the RTSP signaling must be extended to allow the
   RTSP client to indicate that it will use Latching.  The client also
   needs to be able to signal its RTP SSRC to the server in its SETUP
   request.  The RTP SSRC is used to establish some basic level of
   security against hijacking attacks or simply avoid mis-association
   when multiple clients are behind the same NAT.  Care must be taken in
   choosing clients' RTP SSRC.  First, it must be unique within all the
   RTP sessions belonging to the same RTSP session.  Secondly, if the
   RTSP server is sending out media packets to multiple clients from the
   same send port, the RTP SSRC needs to be unique 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

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   The latching packet as discussed above should have field which can
   contain an 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 have been a benefit to use a well defined RTP
   payload format for this purpose as the No-Op payload format proposed
   [I-D.ietf-avt-rtp-no-op].  However, in the absence of such a
   specification an RTP packet without a payload could be used.  Using
   SSRC has the benefit that RTP and RTCP both would work as is.
   However, also other packet formats could be used that carry the
   necessary identification of the context, and such a solution is
   discussed in Section 4.5.

4.4.3.  ALG Considerations

   An RTSP ALG not supporting this method could interfer with the
   methods used to indicate that latching is to be done, as well as the
   SSRC signalling.  Thus preventing the method from working.  However,
   if the RTSP ALG instead opens the corresponding pinholes and create
   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 support this traversal method can for basic
   functionality simply pass the related signalling parameters
   transparently.  Due to the security considerations for latching it
   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


   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.


   o  Requires modifications to both RTSP server and client.

   o  Limited to work with servers that are not behind a NAT.

   o  The format of the packet for "connection setup" (a.k.a Latching
      packet) is not defined.

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   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 lack of a strong authentication mechanism and will need to use
      address restrictions.

4.4.5.  Security Consideration

   Latching's major security issue is that RTP streams can be hijacked
   and directed towards any target that the attacker desires unless
   address restrictions are used.  In the case of NATs with multiple
   clients on the inside of them, hijacking can still occur.  This
   becomes a significant threat in the context of carrier grade NATs

   The most serious security problem is the deliberate attack with the
   use of a RTSP client and Latching.  The attacker uses RTSP to setup a
   media session.  Then it uses Latching with a spoofed source address
   of the intended target of the attack.  There is no defense against
   this attack other than restricting the possible address a latching
   packet can come from to the same as the RTSP TCP connection are from.
   This prevents Latching to be used in use cases that require different
   addresses for media destination and signalling.  Even allowing only a
   limited address range containing the signalling address from where
   latching is allowed opens up a significant vulnerability as it is
   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

   One can fend off the snooping based attack by applying encryption to
   the RTSP signaling transport.  However, if the attacker is a man in
   the middle modifying latching packets, the attack is impossible to
   defend against other than through address restrictions.  As a NAT re-

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   writes the source IP and (possibly) port this cannot be
   authenticated, but authentication is required in order to protect
   against this type of DoS attack.

   Yet another issue is that these attacks also can be used to deny the
   client the service it desires from the RTSP server completely.  The
   attacker modifies or originates its own latching packets with another
   port than what the legit latching packets uses, which results in that
   the media server sends the RTP/RTCP traffic to ports the client isn't
   listening for RTP/RTCP on.

   The amount of random non-guessable material in the latching packet
   determines how well Latching can fend off stream-hijacking performed
   by parties that are off the client to server network path, i.e. lacks
   the capability to see the client's latching packets.  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/Port.  Note that this mitigation in itself opens up
   a new venue for DoS attacks against legit users trying to latch.

   To improve the security against attackers the amount of random
   material could be increased.  To achieve a longer random tag while
   still using RTP and RTCP, it will be necessary to develop RTP and
   RTCP payload formats for carrying the random material.

4.5.  A Variation to Latching

4.5.1.  Introduction

   Latching as described above requires the usage of a valid RTP format
   as the Latching packet, i.e. the first packet that the client sends
   to the server to establish a bi-directional transport flow for RTP
   streams.  There is currently no appropriate RTP packet format for
   this purpose, although the RTP No-Op format was a proposal to fix the
   problem [I-D.ietf-avt-rtp-no-op], however, that work was abandoned.
   There exists a RFC that discusses the implication of different type
   of packets as keep-alives for RTP [RFC6263] and its findings are very
   relevant to the format of the Latching packet.

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   Meanwhile, there has been NAT/firewall traversal techniques deployed
   in the wireless streaming market place that use non-RTP messages as
   Latching packets.  This section describes a variant based on a subset
   of those solutions that alters the previously described Latching

4.5.2.  Necessary RTSP extensions

   In this variation of Latching, the Latching packet is a small UDP
   packet that does not contain an RTP header.  In response to the
   client's Latching packet, the RTSP server sends back a similar
   Latching packet as a confirmation so the client can stop the so
   called "connection phase" of this NAT traversal technique.
   Afterwards, the client only has to periodically send Latching packets
   as keep-alive messages for the NAT mappings.

   The server listens on its RTP-media output port, and tries to decode
   any received UDP packet as Latching packet.  This is valid since an
   RTSP server is not expecting RTP traffic from the RTSP client.  Then,
   it can correlate the Latching packet with the RTSP client's session
   ID or the client's SSRC, and record the NAT bindings accordingly.
   The server then sends a Latching packet as the response to the

   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 re-tries and the re-try interval of the
       Latching message exchanges.

4.5.3.  ALG Considerations

   See Latching ALG consideration Section 4.4.3.

4.5.4.  Deployment Considerations

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

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

   2.  When properly defined, this kind of Latching packet exchange can
       also authenticate RTP receivers, to prevent hijacking attacks.

   This approach has the following disadvantages when compared with the
   Latching approach:

   1.  The server's sender SSRC for the RTP stream or other session
       Identity information must be signaled in RTSP's SETUP response,
       in the Transport header of the RTSP SETUP response.

4.5.5.  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 hi-jack attack.  Secondly the confirmation
   allows the client to know when Latching works and when it didn't and
   thus restart the Latching process by updating the SSRC.

   Still the main security issue remain that the RTSP server can't know
   that the source address in the latching packet was coming from a RTSP
   client wanting to receive media and not one that likes to direct the
   media traffic to an DoS target.

4.6.  Three Way Latching

4.6.1.  Introduction

   The three way latching is an attempt to try to resolve the most
   significant security issues for both previously discussed variants of
   Latching.  By adding a server request response exchange directly
   after the initial latching the server can verify that the target
   address present in the latching packet is an active listener and
   confirm its desire to establish a media flow.

4.6.2.  Necessary RTSP extensions

   Uses the same RTSP extensions as the alternative latching method
   (Section 4.5) uses.  The extensions for this variant are only in the
   format and transmission of the Latching packets.

   The client to server latching packet is similar to the Alternative
   Latching (Section 4.5), i.e. an UDP packet with some session

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   identifier and a random value.  When the server responds to the
   Latching packet with a Latching confirmation, it includes a random
   value (Nonce) of its own in addition to echoing back the one the
   client sent.  Then a third message is added to the exchange.  The
   client acknowledges the reception of the Latching confirmation
   message and echoes back the server's nonce.  Thus confirming that the
   Latched address goes to a RTSP client that initiated the latching and
   is actually present at that address.  The RTSP server will refuse to
   send any media until the Latching Acknowledgement has been received
   with a valid nonce.

4.6.3.  ALG Considerations

   See Latching ALG consideration Section 4.4.3.

4.6.4.  Deployment Considerations

   A solution with a 3-way handshake and its own Latching packets can be
   compared with the ICE-based solution (Section 4.3) and have the
   following differences:

   o  Only works for servers that are not behind a NAT.

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

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

4.6.5.  Security Considerations

   The three way latching is significantly securer than its simpler
   versions discussed above.  The client to server nonce which is
   included in signalling 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 an denial of service attack by
   diverting the media.

   The client to server nonce and its echoing back does not protect
   against on-patch attacker, 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.

   Three way latching is really only vulnerable to an on-path attacker
   that is quite capable.  First the attacker can either learn the
   client to server nonce, by intercepting the signalling, or modifying

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   the source information (target destination) of a client's latching
   packet.  Secondly, it is also on-path between the server and target
   destination and can generate a response using the server's nonce.  An
   adversary that has these capabilities are commonly capable of causing
   significantly worse damage than this using other methods.

   Three-way latching do results in that the server to client packet is
   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 do 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
   response it will generate before considering the latching to be

4.7.  Application Level Gateways

4.7.1.  Introduction

   An Application Level Gateway (ALG) reads the application level
   messages and performs necessary changes to allow the protocol to work
   through the middle box.  However this behavior has some problems in
   regards to RTSP:

   1.  It does not work when the RTSP protocol is used with end-to-end
       security.  As the ALG can't inspect and change the application
       level messages the protocol will fail due to the middle box.

   2.  ALGs need to be updated if extensions to the protocol are added.
       Due to deployment issues with changing ALGs this may also break
       the end-to-end functionality of RTSP.

   Due to the above reasons it is not recommended to use an RTSP ALG in
   NATs.  This is especially important for NATs targeted to home users
   and small office environments, since it is very hard to upgrade NATs
   deployed in home or SOHO (small office/home office) environment.

4.7.2.  Outline On how ALGs for RTSP work

   In this section, we provide a step-by-step outline on how one could
   go about writing an ALG to enable RTSP to traverse a NAT.

   1.  Detect any SETUP request.

   2.  Try to detect the usage of any of the NAT traversal methods that
       replace the address and port of the Transport header parameters
       "destination" or "dest_addr".  If any of these methods are used,

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       then the ALG should not change the address.  Ways to detect that
       these methods are used are:

       *  For embedded STUN, it would be to watch for a feature tag,
          like "nat.stun".  If any of those exists in the "supported",
          "proxy-require", or "require" headers of the RTSP exchange.

       *  For stand alone STUN and TURN based solutions: This can be
          detected by inspecting the "destination" or "dest_addr"
          parameter.  If it contains either one of the NAT's external IP
          addresses or a public IP address then such a solution is in
          use.  However if multiple NATs are used this detection may
          fail.  Remapping should only be done for addresses belonging
          to the NAT's own private address space.

       Otherwise continue to the next step.

   3.  Create UDP mappings (client given IP/port <-> external IP/port)
       where needed for all possible transport specifications in the
       transport header of the request found in (1).  Enter the external
       address and port(s) of these mappings in transport header.
       Mappings shall be created with consecutive external port numbers
       starting on an even number for RTP for each media stream.
       Mappings should also be given a long timeout period, at least 5

   4.  When the SETUP response is received from the server, the ALG may
       remove the unused UDP mappings, i.e. the ones not present in the
       transport header.  The session ID should also be bound to the UDP
       mappings part of that session.

   5.  If SETUP response settles on RTP over TCP or RTP over RTSP as
       lower transport, do nothing: let TCP tunneling take care of NAT
       traversal.  Otherwise go to next step.

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

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

4.7.3.  Deployment Considerations


   o  No impact on either client or server

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   o  Can work for any type of NATs


   o  When deployed they are hard to update to reflect protocol
      modifications and extensions.  If not updated they will break the

   o  When end-to-end security is used, the ALG functionality will fail.

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


   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 [I-D.ietf-mmusic-rfc2326bis].
   Therefore deployment of ALG may result in clients located behind NATs
   not using end-to-end security, or more likely the selection a NAT
   traversal solution that allow for security.

   The creation of an UDP mapping based on the signalling message has
   some potential security implications.  First of all if the RTSP
   client releases its ports and another application are assigned these
   instead it could receive RTP media as long as the mappings exist and
   the RTSP server has failed to be signalled or notice the lack of
   client response.

   A NAT with RTSP ALG that assigns mappings based on SETUP requests
   could potentially become victim of a resource exhaustion attack.  If
   an attacker creates a lot of RTSP sessions, even without starting
   media transmission could exhaust the pool of available UDP ports on
   the NAT.  Thus only a limited number of UDP mappings should be
   allowed to be created by the RTSP ALG.

4.8.  TCP Tunneling

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4.8.1.  Introduction

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

4.8.2.  Usage of TCP tunneling in RTSP

   The RTSP core specification [I-D.ietf-mmusic-rfc2326bis] supports
   interleaving of media data on the TCP connection that carries RTSP
   signaling.  See section 14 in [I-D.ietf-mmusic-rfc2326bis] for how to
   perform this type of TCP tunneling.  There also exists another way of
   transporting RTP over TCP defined in Appendix C.2 in
   [I-D.ietf-mmusic-rfc2326bis].  For signaling and rules on how to
   establish the TCP connection in lieu of UDP, see appendix C.2 in
   [I-D.ietf-mmusic-rfc2326bis].  This is based on the framing of RTP
   over the TCP connection as described in RFC 4571 [RFC4571].

4.8.3.  ALG Considerations

   An RTSP ALG will face a different issue with TCP tunneling, at least
   the Interleaved version.  Now the full data stream will flow 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 much resource and thus creating
   performance issues.

   The RTSP ALG can also effect the transport specifications that
   indicate that TCP tunneling can be done and its priortization,
   including removing the transport specification, thus preventing TCP

4.8.4.  Deployment Considerations


   o  Works through all types of NATs where the RTSP server in not NATed
      or at least reachable like it was not.

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   o  Functionality needs to be implemented on both server and client.

   o  Will not always meet multimedia stream's real-time requirements.


   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 denial of service
   attacks due to TCP's flow control.  The RTSP server TCP socket,
   independently if 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

4.9.  TURN (Traversal Using Relay NAT)

4.9.1.  Introduction

   Traversal Using Relay NAT (TURN) [RFC5766] is a protocol for setting
   up traffic relays that allow clients behind NATs and firewalls to
   receive incoming traffic for both UDP and TCP.  These relays are
   controlled and have limited resources.  They need to be allocated
   before usage.  TURN allows a client to temporarily bind an address/
   port pair on the relay (TURN server) to its local source address/port
   pair, which is used to contact the TURN server.  The TURN server will
   then forward packets between the two sides of the relay.

   To prevent DoS attacks on either recipient, the packets forwarded are
   restricted to the specific source address.  On the client side it is
   restricted to the source setting up the allocation.  On the external
   side this is limited to the source address/port pair that have been
   given permission by the TURN client creating the allocation.  Packets
   from any other source on this address will be discarded.

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   Using a TURN server makes it possible for a RTSP client to receive
   media streams from even an unmodified RTSP server.  However the
   problem is those RTSP servers most likely restrict media destinations
   to no other IP address than the one the RTSP message arrives from.
   This means that TURN could only be used if the server knows and
   accepts that the IP belongs to a TURN server and the TURN server
   can't be targeted at an unknown address.  Alternatively, both the
   RTSP TCP connection as well as the RTP media is relayed through the
   same TURN server.

4.9.2.  Usage of TURN with RTSP

   To use a TURN server for NAT traversal, the following steps should be

   1.  The RTSP client connects with the RTSP server.  The client
       retrieves the session description to determine the number of
       media streams.  To avoid the issue with having RTSP connection
       and media traffic from different addresses also the TCP
       connection must be done through the same TURN server as the one
       in the next step.  This will require the usage of TURN for TCP

   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 headers src_addr parameter (source and port in RTSP
       1.0).  Note that the server is required to have a mechanism to
       verify that it is allowed to send media traffic to the given
       address unless TCP relaying of the RTSP messages also is

   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

   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.

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   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

4.9.3.  ALG Considerations

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

4.9.4.  Deployment Considerations


   o  Does not require any server modifications given that the server
      includes the src_addr header in the SETUP response.

   o  Works for any type of NAT as long as the RTSP server has reachable
      IP address that is not behind a NAT.


   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

   o  The TURN server becomes a single point of failure.

   o  Since TURN forwards media packets, it necessarily introduces

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


   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.

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4.9.5.  Security Considerations

   The TURN server can become part of a denial of service attack towards
   any victim.  To perform this attack the attacker must be able to
   eavesdrop on the packets from the TURN server towards a target for
   the DoS attack.  The attacker uses the TURN server to setup a RTSP
   session with media flows going through the TURN server.  The attacker
   is in fact creating TURN mappings towards a target by spoofing the
   source address of TURN requests.  As the attacker will need the
   address of these mappings he must be able to eavesdrop or intercept
   the TURN responses going from the TURN server to the target.  Having
   these addresses, he can set up a RTSP session and start delivery of
   the media.  The attacker must be able to create these mappings.  The
   attacker in this case may be traced by the TURN username in the
   mapping requests.

   This attack requires that the attacker has access to a user account
   on the TURN server to be able set up the TURN mappings.  To prevent
   this attack the RTSP server needs to verify that the ultimate target
   destination accept this media stream.  Which would require something
   like ICE's connectivity checks being run between the RTSP server and
   the RTSP client.

5.  Firewalls

   Firewalls exist for the purpose of protecting a network from traffic
   not desired by the firewall owner.  Therefore it is a policy decision
   if a firewall will let RTSP and its media streams through or not.
   RTSP is designed to be firewall friendly in that it should be easy to
   design firewall policies to permit passage of RTSP traffic and its
   media streams.

   The firewall will need to allow the media streams associated with a
   RTSP session to pass through it.  Therefore the firewall will need an
   ALG that reads RTSP SETUP and TEARDOWN messages.  By reading the
   SETUP message the firewall can determine what type of transport and
   from where, the media stream packets will be sent.  Commonly there
   will be the need to open UDP ports for RTP/RTCP.  By looking at the
   source and destination addresses and ports the opening in the
   firewall can be minimized to the least necessary.  The opening in the
   firewall can be closed after a TEARDOWN message for that session or
   the session itself times out.

   The above possibilities for firewalls to inspect and respond to the
   signalling are prevented if end-to-end confidentiality protection is
   used for the RTSP signalling, e.g. using the specified RTSP over TLS.
   This results in that firewalls can't be actively opening pinholes for
   the media streams based on the signalling.  To enable an RTSP ALG in

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   firewall to correctly function the hop-by-hop signalling security
   (See Section 19.3) in RTSP 2.0 [I-D.ietf-mmusic-rfc2326bis] can be
   used.  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 that no
   address translation is done.  To use such a firewall a client would
   need to implement one of the above described NAT traversal methods
   that include sending packets to the server to open up the mappings.

6.  Comparison of NAT traversal techniques

   This section evaluates the techniques described above against the
   requirements listed in Section 3.

   In the following table, the columns correspond to the numbered
   requirements.  For instance, the column under R1 corresponds to the
   first requirement in Section 3: must work for all flavors of NATs.
   The rows represent the different NAT/firewall traversal techniques.
   Latch is short for Latching, "V.  Latch" is short for "variation of
   Latching" as described in Section 4.5. "3-W Latch" is short for the
   Three Way Latching described in Section 4.6.

   A Summary of the requirements are:

   R1:  Work for all flavors of NATs

   R2:  Must work with firewalls, including those with ALGs

   R3:  Should have minimal impact on clients not behind NATs, counted
      in minimal number of additional RTTs

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

   R5:  Should provide mitigation against DDoS attacks

   The following considerations are also added to requirements:

   C1:  Will solution support both Clients and Servers behind NAT

   C2:  Is the solution robust to changing NAT behaviors

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               |  R1  |  R2  |  R3  |  R4  |  R5  |  C1  |  C2  |
    STUN       | No   | Yes  |  1   | Maybe| No   | No   | No   |
    Emb. STUN  | Yes  | Yes  |  2   | Maybe| No   | No   | Yes  |
    ICE        | Yes  | Yes  | 2.5  | No   | Yes  | Yes  | Yes  |
    Latch      | Yes  | Yes  |  1   | Maybe| No   | No   | Yes  |
    V. Latch   | Yes  | Yes  |  1   | Yes  | No   | No   | Yes  |
    3-W Latch  | Yes  | Yes  | 1.5  | Maybe| Yes  | No   | Yes  |
    ALG        |(Yes) | Yes  |  0   | No   | Yes  | No   | Yes  |
    TCP Tunnel | Yes  | Yes  | 1.5  | Yes  | Yes  | No   | Yes  |
    TURN       | Yes  | Yes  |  1   | No   | Yes  |(Yes) | Yes  |

            Figure 1: Comparison of fulfillment of requirements

   Looking at Figure 1 one would draw the conclusion that using TCP
   Tunneling or Three-Way Latching is the solutions that best fulfill
   the requirements.  The different techniques were discussed in the
   MMUSIC WG.  It was established that the WG would pursue an ICE based
   solution due to its generality and capability of handling also
   servers delivering media from behind NATs.  TCP Tunneling is likely
   to be available as an alternative, due to its specification in the
   main RTSP specification.  Thus it can be used if desired and the
   potential downsides of using TCP is acceptable in particular
   deployments.  When it comes to Three-Way Latching it is a very
   competitive technique given that you don't need support for RTSP
   servers behind NATs.  There were some discussion in the WG if the
   increased implementation burden of ICE is sufficiently motivated
   compared to a the Three-Way Latching solution for this generality.
   In the end the authors believe that reuse of ICE, the greater
   flexibility and anyway need to deploy a new solution was the decisive

   The ICE based RTSP NAT traversal solution is specified in "A Network
   Address Translator (NAT) Traversal mechanism for media controlled by
   Real-Time Streaming Protocol (RTSP)" [I-D.ietf-mmusic-rtsp-nat].

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7.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an

8.  Security Considerations

   In the preceding sections we have discussed security merits of the
   different NAT/firewall traversal methods for RTSP discussed here.  In
   summary, the presence of NAT(s) is a security risk, as a client
   cannot perform source authentication of its IP address.  This
   prevents the deployment of any future RTSP extensions providing
   security against hijacking of sessions by a man-in-the-middle.

   Each of the proposed solutions has security implications.  Using STUN
   will provide the same level of security as RTSP without transport
   level security and source authentications, as long as the server does
   not allow media to be sent to a different IP-address than the RTSP
   client request was sent from.

   Using Latching will have a higher risk of session hijacking or denial
   of service than normal RTSP.  The reason is that there exists a
   probability that an attacker is able to guess the random bits that
   the client uses to prove its identity when creating the address
   bindings.  This can be solved in the variation of Latching
   (Section 4.5) with authentication features.  Still both those
   variants of Latching are vulnerable against deliberate attack from
   the RTSP client to redirect the media stream requested to any target
   assuming it can spoof the source address.  This security
   vulnerability is solved by performing a Three-way Latching procedure
   as discussed in Section 4.6.

   ICE resolves the binding vulnerability of latching by using signed
   STUN messages, as well as requiring that both sides perform
   connectivity checks to verify that the target IP address in the
   candidate pair is both reachable and willing to respond.  ICE can
   however create a significant amount of traffic if the number of
   candidate pairs are large.  Thus pacing is required and
   implementations should attempt to limit their number of candidates to
   reduce the number of packets.

   If the signalling between the ICE peers (RTSP client and Server) is
   not confidentiality and integrity protected ICE is vulnerable to
   attacks where the candidate list is manipulated.  Lack of signalling
   security will also simplify spoofing of STUN binding messages by
   revealing the secret used in signing.

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   The usage of an RTSP ALG does not in itself increase the risk for
   session hijacking.  However the deployment of ALGs as the sole
   mechanism for RTSP NAT traversal will prevent deployment of end-to-
   end encrypted RTSP signaling.

   The usage of TCP tunneling has no known security problems.  However,
   it might provide a bottleneck when it comes to end-to-end RTSP
   signaling security if TCP tunneling is used on an interleaved RTSP
   signaling connection.

   The usage of TURN has severe risk of denial of service attacks
   against a client.  The TURN server can also be used as a redirect
   point in a DDoS attack unless the server has strict enough rules for
   who may create bindings.

   The latching and variant of latching have so big security issues that
   they should not be used at all.  The three way latching as well as
   ICE mitigates these security issues and performs the important
   return-routability check that prevents spoofed source addresses, and
   should be recommended for that reason.  RTP ALG's is a security risk
   as they can create an incitement against using secure RTSP
   signalling.  That can be avoided as ALGs requires 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.
   [I-D.ietf-mmusic-rfc2326bis].  The remaining methods can be
   considered safe enough, assuming that the appropriate security
   mechanisms are used and not ignored.

9.  Acknowledgements

   The author would also like to thank all persons on the MMUSIC working
   group's mailing list that has commented on this document.  Persons
   having contributed in such way in no special order to this protocol
   are: Jonathan Rosenberg, Philippe Gentric, Tom Marshall, David Yon,
   Amir Wolf, Anders Klemets, Flemming Andreasen, Ari Keranen, Bill
   Atwood, Alissa Cooper, Colin Perkins, Sarah Banks and David Black.
   Thomas Zeng would also like to give special thanks to Greg Sherwood
   of PacketVideo for his input into this memo.

   Section 1.1 contains text originally written for RFC 4787 by Francois
   Audet and Cullen Jennings.

10.  Informative References

              Andreasen, F., "A No-Op Payload Format for RTP", draft-
              ietf-avt-rtp-no-op-04 (work in progress), May 2007.

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              Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
              and M. Stiemerling, "Real Time Streaming Protocol 2.0
              (RTSP)", draft-ietf-mmusic-rfc2326bis-40 (work in
              progress), February 2014.

              Goldberg, J., Westerlund, M., and T. Zeng, "A Network
              Address Translator (NAT) Traversal Mechanism for Media
              Controlled by Real-Time Streaming Protocol (RTSP)", draft-
              ietf-mmusic-rtsp-nat-22 (work in progress), July 2014.

   [NICE]     "Libnice - The GLib ICE implementation,
    ", May 2013.

   [PJNATH]   "PJNATH - Open Source ICE, STUN, and TURN Library,
    ", May 2013.

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

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

   [RFC2326]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
              Streaming Protocol (RTSP)", RFC 2326, April 1998.

   [RFC2588]  Finlayson, R., "IP Multicast and Firewalls", RFC 2588, May

   [RFC2663]  Srisuresh, P. and M. Holdrege, "IP Network Address
              Translator (NAT) Terminology and Considerations", RFC
              2663, August 1999.

   [RFC3022]  Srisuresh, P. and K. Egevang, "Traditional IP Network
              Address Translator (Traditional NAT)", RFC 3022, January

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC3424]  Daigle, L. and IAB, "IAB Considerations for UNilateral
              Self-Address Fixing (UNSAF) Across Network Address
              Translation", RFC 3424, November 2002.

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   [RFC3489]  Rosenberg, J., Weinberger, J., Huitema, C., and R. Mahy,
              "STUN - Simple Traversal of User Datagram Protocol (UDP)
              Through Network Address Translators (NATs)", RFC 3489,
              March 2003.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [RFC4571]  Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
              and RTP Control Protocol (RTCP) Packets over Connection-
              Oriented Transport", RFC 4571, July 2006.

   [RFC4787]  Audet, F. and C. Jennings, "Network Address Translation
              (NAT) Behavioral Requirements for Unicast UDP", BCP 127,
              RFC 4787, January 2007.

   [RFC4961]  Wing, D., "Symmetric RTP / RTP Control Protocol (RTCP)",
              BCP 131, RFC 4961, July 2007.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245, April

   [RFC5382]  Guha, S., Biswas, K., Ford, B., Sivakumar, S., and P.
              Srisuresh, "NAT Behavioral Requirements for TCP", BCP 142,
              RFC 5382, October 2008.

   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              October 2008.

   [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, May 2010.

   [RFC5766]  Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN): Relay Extensions to Session
              Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.

   [RFC6062]  Perreault, S. and J. Rosenberg, "Traversal Using Relays
              around NAT (TURN) Extensions for TCP Allocations", RFC
              6062, November 2010.

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   [RFC6263]  Marjou, X. and A. Sollaud, "Application Mechanism for
              Keeping Alive the NAT Mappings Associated with RTP / RTP
              Control Protocol (RTCP) Flows", RFC 6263, June 2011.

   [RFC7362]  Ivov, E., Kaplan, H., and D. Wing, "Latching: Hosted NAT
              Traversal (HNT) for Media in Real-Time Communication", RFC
              7362, September 2014.

              "Open Source STUN Server and Client,
    ", May 2013.

Authors' Addresses

   Magnus Westerlund
   Farogatan 6
   Stockholm  SE-164 80

   Phone: +46 8 719 0000

   Thomas Zeng


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