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Network Working Group                                      M. Westerlund
Internet-Draft                                                  Ericsson
Intended status: Informational                                   T. Zeng
Expires: July 27, 2014
                                                        January 23, 2014


 The Evaluation of Different Network Address Translator (NAT) Traversal
 Techniques for Media Controlled by Real-time Streaming Protocol (RTSP)
                draft-ietf-mmusic-rtsp-nat-evaluation-12

Abstract

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

Status of This Memo

   This 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
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   Internet-Drafts are draft documents valid for a maximum of six months
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   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 July 27, 2014.

Copyright Notice

   Copyright (c) 2014 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents



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   (http://trustee.ietf.org/license-info) in effect on the date of
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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
     1.1.  Network Address Translators . . . . . . . . . . . . . . .   4
     1.2.  Firewalls . . . . . . . . . . . . . . . . . . . . . . . .   5
     1.3.  Glossary  . . . . . . . . . . . . . . . . . . . . . . . .   6
   2.  Detecting the loss of NAT mappings  . . . . . . . . . . . . .   7
   3.  Requirements on Solutions . . . . . . . . . . . . . . . . . .   8
   4.  NAT Traversal Techniques  . . . . . . . . . . . . . . . . . .   9
     4.1.  Stand-Alone STUN  . . . . . . . . . . . . . . . . . . . .  10
       4.1.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  10
       4.1.2.  Using STUN to traverse NAT without server
               modifications . . . . . . . . . . . . . . . . . . . .  10
       4.1.3.  ALG considerations  . . . . . . . . . . . . . . . . .  12
       4.1.4.  Deployment Considerations . . . . . . . . . . . . . .  13
       4.1.5.  Security Considerations . . . . . . . . . . . . . . .  14
     4.2.  Server Embedded STUN  . . . . . . . . . . . . . . . . . .  14
       4.2.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  14
       4.2.2.  Embedding STUN in RTSP  . . . . . . . . . . . . . . .  14
       4.2.3.  Discussion On Co-located STUN Server  . . . . . . . .  16
       4.2.4.  ALG considerations  . . . . . . . . . . . . . . . . .  16
       4.2.5.  Deployment Considerations . . . . . . . . . . . . . .  16
       4.2.6.  Security Considerations . . . . . . . . . . . . . . .  17
     4.3.  ICE . . . . . . . . . . . . . . . . . . . . . . . . . . .  17
       4.3.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  17
       4.3.2.  Using ICE in RTSP . . . . . . . . . . . . . . . . . .  18
       4.3.3.  Implementation burden of ICE  . . . . . . . . . . . .  20
       4.3.4.  Deployment Considerations . . . . . . . . . . . . . .  20
       4.3.5.  Security Consideration  . . . . . . . . . . . . . . .  21
     4.4.  Latching  . . . . . . . . . . . . . . . . . . . . . . . .  21
       4.4.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  21
       4.4.2.  Necessary RTSP extensions . . . . . . . . . . . . . .  22
       4.4.3.  Deployment Considerations . . . . . . . . . . . . . .  22
       4.4.4.  Security Consideration  . . . . . . . . . . . . . . .  23
     4.5.  A Variation to Latching . . . . . . . . . . . . . . . . .  24
       4.5.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  24
       4.5.2.  Necessary RTSP extensions . . . . . . . . . . . . . .  25
       4.5.3.  Deployment Considerations . . . . . . . . . . . . . .  25
       4.5.4.  Security Considerations . . . . . . . . . . . . . . .  26
     4.6.  Three Way Latching  . . . . . . . . . . . . . . . . . . .  26



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       4.6.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  26
       4.6.2.  Necessary RTSP extensions . . . . . . . . . . . . . .  26
       4.6.3.  Deployment Considerations . . . . . . . . . . . . . .  27
     4.7.  Application Level Gateways  . . . . . . . . . . . . . . .  27
       4.7.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  27
       4.7.2.  Outline On how ALGs for RTSP work . . . . . . . . . .  28
       4.7.3.  Deployment Considerations . . . . . . . . . . . . . .  29
       4.7.4.  Security Considerations . . . . . . . . . . . . . . .  29
     4.8.  TCP Tunneling . . . . . . . . . . . . . . . . . . . . . .  30
       4.8.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  30
       4.8.2.  Usage of TCP tunneling in RTSP  . . . . . . . . . . .  30
       4.8.3.  Deployment Considerations . . . . . . . . . . . . . .  30
       4.8.4.  Security Considerations . . . . . . . . . . . . . . .  31
     4.9.  TURN (Traversal Using Relay NAT)  . . . . . . . . . . . .  31
       4.9.1.  Introduction  . . . . . . . . . . . . . . . . . . . .  31
       4.9.2.  Usage of TURN with RTSP . . . . . . . . . . . . . . .  32
       4.9.3.  Deployment Considerations . . . . . . . . . . . . . .  32
       4.9.4.  Security Considerations . . . . . . . . . . . . . . .  33
   5.  Firewalls . . . . . . . . . . . . . . . . . . . . . . . . . .  34
   6.  Comparison of NAT traversal techniques  . . . . . . . . . . .  34
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  36
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .  36
   9.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  37
   10. Informative References  . . . . . . . . . . . . . . . . . . .  37
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  39

1.  Introduction

   Today there is a proliferate deployment of different flavors of
   Network Address Translator (NAT) boxes that in many cases only
   loosely follow standards
   [RFC3022][RFC2663][RFC3424][RFC4787][RFC5382].  NATs cause
   discontinuity in address realms [RFC3424], therefore an application
   protocol, such as Real-time Streaming Protocol (RTSP)
   [RFC2326][I-D.ietf-mmusic-rfc2326bis], needs to deal with such
   discontinuities caused by NATs.  The problem is that, being a media
   control protocol managing one or more media streams, RTSP carries
   network address and port information within its protocol messages.
   Because of this, even if RTSP itself, when carried over Transmission
   Control Protocol (TCP) [RFC0793] for example, is not blocked by NATs,
   its media streams may be blocked by NATs.  This will occur unless
   special protocol provisions are added to support NAT-traversal.

   Like NATs, Firewalls are also middle boxes that need to be
   considered.  Firewalls help prevent unwanted traffic from getting in
   or out of the protected network.  RTSP is designed such that a
   firewall can be configured to let RTSP controlled media streams go
   through with minimal implementation effort.  The minimal effort is to



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   implement an Application Level Gateway (ALG) to interpret RTSP
   parameters.  There is also a large class of firewalls, commonly home
   firewalls, that uses a similar filtering behavior to what NAT has.
   This type of firewalls can be handled using the same solution as
   employed for NAT traversal instead of relying on ALGs.

   This document describes several NAT-traversal mechanisms for RTSP
   controlled media streaming.  Many of these NAT solutions fall into
   the category of "UNilateral Self-Address Fixing (UNSAF)" as defined
   in [RFC3424] and quoted below:

   "UNSAF is a process whereby some originating process attempts to
   determine or fix the address (and port) by which it is known - e.g.
   to be able to use address data in the protocol exchange, or to
   advertise a public address from which it will receive connections."

   Following the guidelines spelled out in RFC 3424, we describe the
   required RTSP protocol extensions for each method, transition
   strategies, and security concerns.

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

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

1.1.  Network Address Translators

   We begin by reviewing two quotes from Section 3 in "Network Address
   Translation (NAT) Behavioral Requirements for Unicast UDP" [RFC4787]
   concering NATs and their terminology:

   "Readers are urged to refer to [RFC2663] for information on NAT
   taxonomy and terminology.  Traditional NAT is the most common type of
   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).





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   NAPT is by far the most commonly deployed NAT device.  NAPT allows
   multiple internal hosts to share a single public IP address
   simultaneously.  When an internal host opens an outgoing TCP or UDP
   session through a NAPT, the NAPT assigns the session a public IP
   address and port number, so that subsequent response packets from the
   external endpoint can be received by the NAPT, translated, and
   forwarded to the internal host.  The effect is that the NAPT
   establishes a NAT session to translate the (private IP address,
   private port number) tuple to a (public IP address, public port
   number) tuple, and vice versa, for the duration of the session.  An
   issue of relevance to peer-to-peer applications is how the NAT
   behaves when an internal host initiates multiple simultaneous
   sessions from a single (private IP, private port) endpoint to
   multiple distinct endpoints on the external network.  In this
   specification, the term "NAT" refers to both "Basic NAT" and "Network
   Address/Port Translator (NAPT)"."

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

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

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

   NATs also have a filtering behavior on traffic arriving on the
   external side.  Such behavior affects how well different methods for
   NAT traversal works through these NATs.  See Section 5 of "Network
   Address Translation (NAT) Behavioral Requirements for Unicast UDP"
   [RFC4787] for more information on the different types of filtering
   that have been identified.

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. public Internet.  Many
   organizations use firewalls to prevent privacy intrusions and
   malicious attacks to corporate computing resources in the private
   intranet [RFC2588].



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   A comparison between NAT and Firewall is given below:

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

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

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

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

1.3.  Glossary

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

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

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

   Endpoint-Independent Mapping:  The NAT reuses the port mapping for
         subsequent packets sent from the same internal IP address and
         port to any external IP address and port.  See [RFC4787].

   ICE:  Interactive Connectivity Establishment, see [RFC5245].

   DNS:  Domain Name Service

   DoS:  Denial of Service

   DDoS: Distributed Denial of Service

   NAT:  Network Address Translator, see [RFC3022].



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   NAPT: Network Address/Port Translator, see [RFC3022].

   RTP:  Real-time Transport Protocol, see [RFC3550].

   RTSP: Real-Time Streaming Protocol, see [RFC2326] and
         [I-D.ietf-mmusic-rfc2326bis].

   RTT:  Round Trip Times.

   SDP:  Session Description Protocol, see [RFC4566].

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

2.  Detecting the loss of NAT mappings

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

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

   A RTP session normally has both RTP and RTCP streams.  The loss of a
   RTP mapping can only be detected when expected traffic does not
   arrive.  If a client does not receive data within a few seconds after
   having received the "200 OK" response to a PLAY request, there are
   likely some middleboxes blocking the traffic.  However, for a
   receiver to be more certain to detect the case where no RTP traffic
   was delivered due to NAT trouble, one should monitor the RTCP Sender
   reports if they are received and not also blocked.  The sender report
   carries a field telling how many packets the server has sent.  If
   that has increased and no RTP packets has arrived for a few seconds
   it is likely the RTP mapping has been removed.

   The loss of mapping for RTCP is simpler to detect.  RTCP is normally
   sent periodically in each direction, even during the RTSP ready
   state.  If RTCP packets are missing for several RTCP intervals, the



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   mapping is likely lost.  Note that if neither RTCP packets nor RTSP
   messages are received by the RTSP server for a while, the RTSP server
   has the option to delete the corresponding RTP session, SSRC and RTSP
   session ID, because either the client can not get through a middle
   box NAT/Firewall, or the client is mal-functioning.

3.  Requirements on Solutions

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

   RTSP is a client-server protocol.  Typically service providers deploy
   RTSP servers in the public address realm.  However, there are use
   cases where the reverse is true: RTSP clients are connecting from
   public address realm to RTSP servers behind home NATs.  This is the
   case for instance when home surveillance cameras running as RTSP
   servers intend to stream video to cell phone users in the public
   address realm through a home NAT.  In terms of requirements, the
   first requirement should be to solve the RTSP NAT traversal problem
   for RTSP servers deployed in a public network, i.e. no NAT at the
   server side.

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

   1.  Must 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 in the open and not dual-
       hosted.  RTSP dual-hosting means that the RTSP signalling
       protocol and the media protocol (e.g. RTP) are implemented on
       different computers with different IP addresses.

       *  For instance, no extra delay from RTSP connection till arrival
          of media

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

       *  Otherwise people will resort to TCP tunneling through NATs

       *  Discovery of the address(es) assigned by NAT should happen
          automatically, if possible





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

   The last requirement addresses the Distributed Denial-of-Service
   (DDoS) threat, which relates to NAT traversal as explained below.

   During NAT traversal, when the RTSP server determines the media
   destination (address and port) for the client, the result may be that
   the public IP address of the RTP receiver host is different than the
   public IP address of the RTSP client host.  This posts a DDoS threat
   that has significant amplification potentials because the RTP media
   streams in general consist of large number of IP packets.  DDoS
   attacks occur if the attacker fakes the messages in the NAT traversal
   mechanism to trick the RTSP server into believing that the client's
   RTP receiver is located on a separate host.  For example, user A may
   use his RTSP client to direct the RTSP server to send video RTP
   streams to target.example.com in order to degrade the services
   provided by target.example.com.  Note a simple preventative measure
   commonly deployed is for the RTSP server to disallow the cases where
   the client's RTP receiver has a different public IP address than that
   of the RTSP client.  With the increased deployment of NAT middleboxes
   by operators, i.e. carrier grade NAT (CGN), the reusing of a public
   IP address for many customers reduces the protection provided.  Also
   in some applications (e.g., centralized conferencing), dual-hosted
   RTSP/RTP clients have valid use cases.  The key is how to
   authenticate the messages exchanged during the NAT traversal process.

4.  NAT Traversal Techniques

   There exists a number of potential NAT traversal techniques that can
   be used to allow RTSP to traverse NATs.  They have different features
   and are applicable to different topologies; their costs are also
   different.  They also vary in security levels.  In the following
   sections, each technique is outlined with discussions on the
   corresponding advantages and disadvantages.

   The main evaluation was done prior to 2007 and is based on what was
   available then.  This section includes NAT traversal techniques that
   have not been formally specified anywhere else.  The overview section
   of this document may be the only publicly available specification of
   some of the NAT traversal techniques.  However that is not a real
   barrier against doing an evaluation of the NAT traversal techniques.
   Some other techniques have been recommended against or are no longer
   possible due to standardization works' outcome or their failure to
   progress within IETF after the initial evaluation in this document,
   e.g. RTP No-Op [I-D.ietf-avt-rtp-no-op].





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4.1.  Stand-Alone STUN

4.1.1.  Introduction

   Session Traversal Utilities for NAT (STUN) [RFC5389] is a
   standardized protocol that allows a client to use secure means to
   discover the presence of a NAT between itself and the STUN server.
   The client uses the STUN server to discover the address mappings
   assigned by the NAT.  STUN is a client-server protocol.  The STUN
   client sends a request to a STUN server and the server returns a
   response.  There are two types of STUN messages - Binding Requests
   and Indications.  Binding requests are used when determining a
   client's external address and solicits a response from the STUN
   server with the seen address.

   The first version of STUN [RFC3489] included categorization and
   parameterization of NATs.  This was abandoned in the updated version
   [RFC5389] due to it being unreliable and brittle.  Some of the below
   discussed methods are based on RFC3489 functionality which will be
   called out and the downside of that will be part of the
   characterization.

4.1.2.  Using STUN to traverse NAT without server modifications

   This section describes how a client can use STUN to traverse NATs to
   RTSP servers without requiring server modifications.  Note that this
   method has limited applicability and requires the server to be
   available in the external/public address realm in regards to the
   client located behind a NAT(s).

   Limitations:

   o  The server must be located in either a public address realm or the
      next hop external address realm in regards to the client.

   o  The client may only be located behind NATs that perform "Endpoint-
      Independent" or "Address-Dependent" Mappings.  Clients behind NATs
      that do "Address and Port-Dependent" Mappings cannot use this
      method.  See [RFC4787] for full definition of these terms.

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

   Method:

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




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

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

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

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

   5.  To keep the mappings alive, the client should periodically send
       UDP traffic over all mappings needed for the session.  For the
       mapping carrying RTCP traffic the periodic RTCP traffic are
       likely enough.  For mappings carrying RTP traffic and for
       mappings carrying RTCP packets at too low a frequency, keep-alive
       messages should be sent.  As keep alive messages, one could use
       the RTP No-Op packet [I-D.ietf-avt-rtp-no-op] to the streaming
       server's discard port (port number 9).  The drawback of using RTP
       No-Op is that the payload type number must be dynamically
       assigned through RTSP first.  Otherwise STUN could be used for
       the keep-alive as well as empty UDP packets.

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



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   To allow UDP packets to arrive from the server to a client behind a
   "Address Dependent" filtering NAT, the client must first send a UDP
   packet to establish filtering state in the NAT.  The client, before
   sending a RTSP PLAY request, must send a so called hole-punching
   packet (such as a RTP No-Op packet) on each mapping, to the IP
   address given as the servers source address.  To create minimum
   problems for the server these UDP packets should be sent to the
   server's discard port (port number 9).  Since UDP packets are
   inherently unreliable, to ensure that at least one UDP message passes
   the NAT, hole-punching packets should be retransmitted a reasonable
   number of times.

   For an "Address and Port Dependent" filtering NAT the client must
   send messages to the exact ports used by the server to send UDP
   packets before sending a RTSP PLAY request.  This makes it possible
   to use the above described process with the following additional
   restrictions: for each port mapping, hole-punching packets need to be
   sent first to the server's source address/port.  To minimize
   potential effects on the server from these messages the following
   type of hole punching packets must be sent.  RTP: an empty or less
   than 12 bytes UDP packet.  RTCP: A correctly formatted RTCP RR or SR
   message.  The above described adaptations for restricted NATs will
   not work unless the server includes the "src_addr" in the "Transport"
   header (which is the "source" transport parameter in RFC2326).

   This method is brittle because it assumes one can use STUN to
   classify the NAT behavior, which was found to be problematic
   [RFC5389].  If the NAT changes the properties of the existing mapping
   and filtering state for example due to load, then the methods will
   fail.

4.1.3.  ALG considerations

   If a NAT supports RTSP ALG (Application Level Gateway) and is not
   aware of the STUN traversal option, service failure may happen,
   because a client discovers its public IP address and port numbers,
   and inserts them in its SETUP requests.  When the RTSP ALG processes
   the SETUP request it may change the destination and port number,
   resulting in unpredictable behavior.  An ALG should not update
   address fields which contains addresses other than the NATs internal
   address domain.  In cases where the ALG modifies fields 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




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

4.1.4.  Deployment Considerations

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

   Advantages:

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

   o  Using STUN does not require RTSP server modifications; it only
      affects the client implementation.

   Disadvantages:

   o  Requires a STUN server deployed in the public address space.

   o  Only works with NATs that perform endpoint independent and address
      dependent mappings.  Address and Port-Dependent filtering NATs
      create some issues.

   o  Brittle to NATs changing the properties of the NAT mapping and
      filtering.

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

   o  Will mostly not work if a NAT uses multiple IP addresses, since
      RTSP servers generally require all media streams to use the same
      IP as used in the RTSP connection to prevent becoming a DDoS tool.

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

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

   Transition:




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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
   behavior.

4.2.2.  Embedding STUN in RTSP

   This section outlines the adaptation and embedding of STUN within
   RTSP.  This enables STUN to be used to traverse any type of NAT,
   including address and Port-Dependent mapping NATs.  This would
   require RTSP level protocol changes.

   This NAT traversal solution has limitations:





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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
       requests.

   Deviations from STUN as defined in RFC 5389:

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

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

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

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

   To minimize delay, we require that the RTSP server supporting this
   option must inform the client about the RTP and RTCP ports from where
   the server will send out RTP and RTCP packets, respectively.  This
   can be done by using the "server_port" parameter in RFC2326, and the
   "src_addr" parameter in [I-D.ietf-mmusic-rfc2326bis].  Both are in
   the RTSP Transport header.  But in general this strategy will require
   that one first do one SETUP request per media 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.

4.2.4.  ALG considerations

   The same ALG traversal considerations as for Stand-Alone STUN applies
   (Section 4.1.3).

4.2.5.  Deployment Considerations

   For the "Embedded STUN" method the following applies:

   Advantages:

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

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

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

   o  Compared to other UDP based NAT traversal methods in this
      document, STUN requires little new protocol development (since
      STUN is already a IETF standard), and most likely less
      implementation effort, since open source STUN server and client
      implementations are available [STUN-IMPL][PJNATH].  There is the
      need to embed STUN in RTSP server and client, which require a de-
      multiplexer between STUN packets and RTP/RTCP packets.  There is
      also a need to register the proper feature tags.

   Disadvantages:




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

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

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

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

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

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

   Transition:

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

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



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   other).  The methodology has very nice properties in that basically
   all NAT topologies are possible to traverse.

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

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

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

4.3.2.  Using ICE in RTSP

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

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

   The negotiation of ICE in RTSP of necessity will work different than
   in SIP with SDP offer/answer.  The protocol interactions are
   different and thus the possibilities for transfer of states are also
   somewhat different.  The goal is also to avoid introducing extra
   delay in the setup process at least for when the server is using a
   public address and the client is either having a public address or is




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   behind NAT(s).  This process is only intended to support PLAY mode,
   i.e. media traffic flows from server to client.

   1.  The ICE usage begins in the SDP.  The SDP for the service
       indicates that ICE is supported at the server.  No candidates can
       be given here as that would not work with the on demand, DNS load
       balancing, etc., that 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 rather than always successful is recommended, as it
       is most likely to be a server in the public.

   4.  The server responds to the SETUP (200 OK) for each media stream
       with its candidates.  A server with a public address usually only
       provides a single ICE candidate.  Also here one candidate is the
       server primary address.

   5.  The connectivity checks are performed.  For the server the
       connectivity checks from the server to the clients have an
       additional usage.  They verify that there is someone willingly to
       receive the media, thus 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




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       fails within the set timeout, an error is reported back.  Client
       needs to go back to 6.

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

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

4.3.3.  Implementation burden of ICE

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

   RTSP server implementation requirements are:

   o  STUN server features

   o  Limited STUN client features

   o  SDP generation with more parameters.

   o  RTSP error code for ICE extension

   RTSP client implementation requirements are:

   o  Limited STUN server features

   o  Limited STUN client features

   o  RTSP error code and ICE extension

4.3.4.  Deployment Considerations

   Advantages:

   o  Solves NAT connectivity discovery for basically all cases as long
      as a TCP connection between the client and server can be
      established.  This includes servers behind NATs.  (Note that a
      proxy between address domains may be required to get TCP through).

   o  Improves defenses against DDoS attacks, since a media receiving
      client requires authentications, via STUN on its media reception
      ports.



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   Disadvantages:

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

   o  Assumes that it is possible to de-multiplex between the packets of
      the media protocol and STUN packets.

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

4.3.5.  Security Consideration

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

4.4.  Latching

4.4.1.  Introduction

   Latching [I-D.ietf-mmusic-latching] is a NAT traversal solution that
   is based on requiring RTSP clients to send UDP packets to the
   server's media output ports.  Conventionally, RTSP servers send RTP
   packets in one direction: from server to client.  Latching is similar
   to connection-oriented traffic, where one side (e.g., the RTSP
   client) first "connects" by sending a RTP packet to the other side's
   RTP port, the recipient then replies to the originating IP and port.
   This method is also referred to as "Late binding".  It requires that
   all RTP/RTCP transport is done symmetrical, i.e. Symmetric RTP
   [RFC4961].

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



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   vulnerable to DDoS attacks (See Security Considerations of
   [I-D.ietf-mmusic-latching]), because attackers can simply forge the
   source IP & Port of the latching packet.  Using the rule for
   restricting IP address to the one of the signaling connection will
   need to be applied here also.  However, that does not protect against
   hijacking from another client behind the same NAT.  This can become a
   serious issue in deployments with CGNs.

4.4.2.  Necessary RTSP extensions

   To support Latching, the RTSP signaling must be extended to allow the
   RTSP client to indicate that it will use Latching.  The client also
   needs to be able to signal its RTP SSRC to the server in its SETUP
   request.  The RTP SSRC is used to establish some basic level of
   security against hijacking attacks or simply avoid mis-association
   when multiple clients are behind the same NAT.  Care must be taken in
   choosing clients' RTP SSRC.  First, it must be unique within all the
   RTP sessions belonging to the same RTSP session.  Secondly, if the
   RTSP server is sending out media packets to multiple clients from the
   same send port, the RTP SSRC needs to be unique amongst those
   clients' RTP sessions.  Recognizing that there is a potential that
   RTP SSRC collisions may occur, the RTSP server must be able to signal
   to a client that a collision has occurred and that it wants the
   client to use a different RTP SSRC carried in the SETUP response or
   use unique ports per RTSP session.  Using unique ports limits an RTSP
   server in the number of sessions it can simultaneously handle per
   interface IP addresses.

4.4.3.  Deployment Considerations

   Advantages:

   o  Works for all types of client-facing NATs.  (Requirement 1 in
      Section 3).

   o  Has no interaction problems with any RTSP ALG changing the
      client's information in the transport header.

   Disadvantages:

   o  Requires modifications to both RTSP server and client.

   o  Limited to work with servers that have an public IP address.

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




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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 (Section 4.4.4) due to
      lack of strong authentication mechanism, compare with STUN message
      authentication, and will need to use address restrictions.

4.4.4.  Security Consideration

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

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

   A hijack attack can also be performed in various ways.  The basic
   attack is based on the ability to read the RTSP signaling packets in
   order to learn the address and port the server will send from and
   also the SSRC the client will use.  Having this information the
   attacker can send its own Latching packets containing the correct RTP
   SSRC to the correct address and port on the server.  The RTSP server
   will then use the source IP and port from the Latching packet as the
   destination for the media packets it sends.

   Another variation of this attack is for a man in the middle to modify
   the RTP latching packet being sent by a client to the server by
   simply changing the source IP to the target one desires to attack.

   One can fend off the snooping based attack by applying encryption to
   the RTSP signaling transport.  However, if the attacker is an man in
   the middle modifying latching packets, the attack is impossible to
   defend against other than through address restrictions.  As a NAT re-
   writes the source IP and (possibly) port this cannot be



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   authenticated, but authentication is required in order to protect
   against this type of DoS attack.

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

   The amount of random non-guessable material in the latching packet
   determines how well Latching can fend off stream-hijacking performed
   by parties that are off the client to server network path, i.e. lacks
   the capability to see the client's latching packets.  This proposal
   uses the 32-bit RTP SSRC field to this effect.  Therefore it is
   important that this field is derived with a non-predictable random
   number generator.  It should not be possible by knowing the algorithm
   used and a couple of basic facts, to derive what random number a
   certain client will use.

   An attacker not knowing the SSRC but aware of which port numbers that
   a server sends from can deploy a brute force attack on the server by
   testing a lot of different SSRCs until it finds a matching one.
   Therefore a server could implement functionality that blocks packets
   to ports or from sources that receive or send multiple Latching
   packets with different invalid SSRCs, especially when they are coming
   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 set up virtual RTP connection.  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
   solution.

4.5.2.  Necessary RTSP extensions

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

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

   The Latching packet can contain the SSRC to identify the RTP stream,
   and care must be taken if the packet is bigger than 12 bytes,
   ensuring that it is distinctively different from RTP packets, whose
   header size is 12 bytes.

   RTSP signaling can be added to do the following:

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

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

4.5.3.  Deployment Considerations

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

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



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

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

   1.  RTP traffic is normally accompanied by RTCP traffic.  This
       approach needs to rely on RTCP RRs and SRs to enable NAT
       traversal for RTCP endpoints, use RTP/RTCP Multiplexing
       [RFC5761], or use the same type of Latching packets also for RTCP
       endpoints.

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

4.5.4.  Security Considerations

   Compared to the security properties of Latching this variant is
   slightly improved.  First of all it allows for a larger random field
   in the Latching packets which makes it more unlikely for an off-path
   attacker to succeed in a hi-jack attack.  Secondly the confirmation
   allows the client to know when Latching works and when it didn't and
   thus restart the Latching process by updating the SSRC.  Thirdly if
   an authentication mechanism is included in the latching packet
   hijacking attacks can be prevented.

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

4.6.  Three Way Latching

4.6.1.  Introduction

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

4.6.2.  Necessary RTSP extensions

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



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   The client to server latching packet is similar to the Alternative
   Latching (Section 4.5), i.e. an UDP packet with some session
   identifier and a random value.  When the server responds to the
   Latching packet with a Latching confirmation, it includes a random
   value (Nonce) of its own in addition to echoing back the one the
   client sent.  Then a third message is added to the exchange.  The
   client acknowledges the reception of the Latching confirmation
   message and echoes back the server's nonce.  Thus confirming that the
   Latched address goes to a RTSP client that initiated the latching and
   is actually present at that address.  The RTSP server will refuse to
   send any media until the Latching Acknowledgement has been received
   with a valid nonce.

4.6.3.  Deployment Considerations

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

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

   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.

4.7.  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



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   and small office environments, since it is very hard to upgrade NATs
   deployed in home or SOHO (small office/home office) environment.

4.7.2.  Outline On how ALGs for RTSP work

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

   1.  Detect any SETUP request.

   2.  Try to detect the usage of any of the NAT traversal methods that
       replace the address and port of the Transport header parameters
       "destination" or "dest_addr".  If any of these methods are used,
       then the ALG should not change the address.  Ways to detect that
       these methods are used are:

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

       *  For stand alone STUN and TURN based solutions: This can be
          detected by inspecting the "destination" or "dest_addr"
          parameter.  If it contains either one of the NAT's external IP
          addresses or a public IP address then such a solution is in
          use.  However if multiple NATs are used this detection may
          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 public port numbers
       starting on an even number for RTP for each media stream.
       Mappings should also be given a long timeout period, at least 5
       minutes.

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

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




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

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

4.7.3.  Deployment Considerations

   Advantage:

   o  No impact on either client or server

   o  Can work for any type of NATs

   Disadvantage:

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

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

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

   Transition:

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

4.7.4.  Security Considerations

   An ALG will not work with deployment of end-to-end RTSP signaling
   security.  Therefore deployment of ALG will likely result in clients
   located behind NATs not using end-to-end security.

   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



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   an attacker creates a lot of RTSP sessions, even without starting
   media transmission could exhaust the pool of available UDP ports on
   the NAT.  Thus only a limited number of UDP mappings should be
   allowed to be created by the RTSP ALG.

4.8.  TCP Tunneling

4.8.1.  Introduction

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

4.8.2.  Usage of TCP tunneling in RTSP

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

4.8.3.  Deployment Considerations

   Advantage:

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

   Disadvantage:

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

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




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   Transition:

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

4.8.4.  Security Considerations

   The TCP tunneling of RTP has no known security problems besides those
   already presented in the RTSP specification.  It is not possible to
   get any amplification effect for denial of service attacks due to
   TCP's flow control.  A possible security consideration, when session
   media data is interleaved with RTSP, would be the performance
   bottleneck when RTSP encryption is applied, since all session media
   data also needs to be encrypted.

4.9.  TURN (Traversal Using Relay NAT)

4.9.1.  Introduction

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

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

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






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4.9.2.  Usage of TURN with RTSP

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

   1.  The RTSP client connects with the RTSP server.  The client
       retrieves the session description to determine the number of
       media streams.  To avoid the issue with having RTSP connection
       and media traffic from different addresses also the TCP
       connection must be done through the same TURN server as the one
       in the next step.  This will require the usage of TURN for TCP
       [RFC6062].

   2.  The client establishes the necessary bindings on the TURN server.
       It must choose the local RTP and RTCP ports that it desires to
       receive media packets.  TURN supports requesting bindings of even
       port numbers and contiguous ranges.

   3.  The RTSP client uses the acquired address and port allocations in
       the RTSP SETUP request using the destination header.

   4.  The RTSP Server sends the SETUP reply, which must include the
       transport headers src_addr parameter (source and port in RTSP
       1.0).  Note that the server is required to have a mechanism to
       verify that it is allowed to send media traffic to the given
       address.

   5.  The RTSP Client uses the RTSP Server's response to create TURN
       permissions for the server's media traffic.

   6.  The client requests that the server starts playing.  The server
       starts sending media packets to the given destination address and
       ports.

   7.  Media packets arrive at the TURN server on the external port; If
       the packets match an established permission, the TURN server
       forwards the media packets to the RTSP client.

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

4.9.3.  Deployment Considerations

   Advantages:

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



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

   Disadvantage:

   o  Requires another network element, namely the TURN server.

   o  A TURN server for RTSP may not scale since the number of sessions
      it must forward is proportional to the number of client media
      sessions.

   o  TURN server becomes a single point of failure.

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

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

   Transition:

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

4.9.4.  Security Considerations

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

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




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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.

   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



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   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

   ------------+------+------+------+------+------+------+------+
               |  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.



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   In the end the authors believe that reuse of ICE, the greater
   flexibility and anyway need to deploy a new solution was the decisive
   factors.

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

7.  IANA Considerations

   This document makes no request of IANA.

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

8.  Security Considerations

   In the preceding 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.  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




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   redirect point in a DDoS attack unless the server has strict enough
   rules for who may create bindings.

9.  Acknowledgements

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

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

10.  Informative References

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

   [I-D.ietf-mmusic-latching]
              Ivov, E., Kaplan, H., and D. Wing, "Latching: Hosted NAT
              Traversal (HNT) for Media in Real-Time Communication",
              draft-ietf-mmusic-latching-04 (work in progress), October
              2013.

   [I-D.ietf-mmusic-rfc2326bis]
              Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
              and M. Stiemerling, "Real Time Streaming Protocol 2.0
              (RTSP)", draft-ietf-mmusic-rfc2326bis-39 (work in
              progress), January 2014.

   [I-D.ietf-mmusic-rtsp-nat]
              Goldberg, J., Westerlund, M., and T. Zeng, "A Network
              Address Translator (NAT) Traversal mechanism for media
              controlled by Real-Time Streaming Protocol (RTSP)", draft-
              ietf-mmusic-rtsp-nat-18 (work in progress), January 2014.

   [NICE]     "Libnice - The GLib ICE implementation,
              http://nice.freedesktop.org/wiki/", May 2013.

   [PJNATH]   "PJNATH - Open Source ICE, STUN, and TURN Library,
              http://www.pjsip.org/pjnath/docs/html/", May 2013.





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

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

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

   [RFC2588]  Finlayson, R., "IP Multicast and Firewalls", RFC 2588, May
              1999.

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

   [RFC3022]  Srisuresh, P. and K. Egevang, "Traditional IP Network
              Address Translator (Traditional NAT)", RFC 3022, January
              2001.

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

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

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

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

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

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

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





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Internet-Draft    Evaluation of NAT Traversal for RTSP      January 2014


   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245, April
              2010.

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

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

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761, April 2010.

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

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

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

   [STUN-IMPL]
              "Open Source STUN Server and Client,
              http://sourceforge.net/projects/stun/", May 2013.

Authors' Addresses

   Magnus Westerlund
   Ericsson
   Farogatan 6
   Stockholm  SE-164 80
   Sweden

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


   Thomas Zeng

   Email: thomas.zeng@gmail.com




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