<pre>Internet Engineering Task Force (IETF)                     M. Westerlund
Request for Comments: 7201                                      Ericsson
Category: Informational                                       C. Perkins
ISSN: 2070-1721                                    University of Glasgow
                                                              April 2014


                   <span class="h1">Options for Securing RTP Sessions</span>

Abstract

   The Real-time Transport Protocol (RTP) is used in a large number of
   different application domains and environments.  This heterogeneity
   implies that different security mechanisms are needed to provide
   services such as confidentiality, integrity, and source
   authentication of RTP and RTP Control Protocol (RTCP) packets
   suitable for the various environments.  The range of solutions makes
   it difficult for RTP-based application developers to pick the most
   suitable mechanism.  This document provides an overview of a number
   of security solutions for RTP and gives guidance for developers on
   how to choose the appropriate security mechanism.

Status of This Memo

   This document is not an Internet Standards Track specification; it is
   published for informational purposes.

   This document is a product of the Internet Engineering Task Force
   (IETF).  It represents the consensus of the IETF community.  It has
   received public review and has been approved for publication by the
   Internet Engineering Steering Group (IESG).  Not all documents
   approved by the IESG are a candidate for any level of Internet
   Standard; see <a href="./rfc5741#section-2">Section&nbsp;2 of RFC 5741</a>.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   <a href="http://www.rfc-editor.org/info/rfc7201">http://www.rfc-editor.org/info/rfc7201</a>.














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Copyright Notice

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

   This document is subject to <a href="https://www.rfc-editor.org/bcp/bcp78">BCP 78</a> and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (<a href="http://trustee.ietf.org/license-info">http://trustee.ietf.org/license-info</a>) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.





































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Table of Contents

   <a href="#section-1">1</a>.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   <a href="#page-4">4</a>
   <a href="#section-2">2</a>.  Background  . . . . . . . . . . . . . . . . . . . . . . . . .   <a href="#page-5">5</a>
     <a href="#section-2.1">2.1</a>.  Point-to-Point Sessions . . . . . . . . . . . . . . . . .   <a href="#page-5">5</a>
     <a href="#section-2.2">2.2</a>.  Sessions Using an RTP Mixer . . . . . . . . . . . . . . .   <a href="#page-5">5</a>
     <a href="#section-2.3">2.3</a>.  Sessions Using an RTP Translator  . . . . . . . . . . . .   <a href="#page-6">6</a>
       <a href="#section-2.3.1">2.3.1</a>.  Transport Translator (Relay)  . . . . . . . . . . . .   <a href="#page-6">6</a>
       <a href="#section-2.3.2">2.3.2</a>.  Gateway . . . . . . . . . . . . . . . . . . . . . . .   <a href="#page-7">7</a>
       <a href="#section-2.3.3">2.3.3</a>.  Media Transcoder  . . . . . . . . . . . . . . . . . .   <a href="#page-8">8</a>
     <a href="#section-2.4">2.4</a>.  Any Source Multicast  . . . . . . . . . . . . . . . . . .   <a href="#page-8">8</a>
     <a href="#section-2.5">2.5</a>.  Source-Specific Multicast . . . . . . . . . . . . . . . .   <a href="#page-8">8</a>
   <a href="#section-3">3</a>.  Security Options  . . . . . . . . . . . . . . . . . . . . . .  <a href="#page-10">10</a>
     <a href="#section-3.1">3.1</a>.  Secure RTP  . . . . . . . . . . . . . . . . . . . . . . .  <a href="#page-10">10</a>
       <a href="#section-3.1.1">3.1.1</a>.  Key Management for SRTP: DTLS-SRTP  . . . . . . . . .  <a href="#page-12">12</a>
       <a href="#section-3.1.2">3.1.2</a>.  Key Management for SRTP: MIKEY  . . . . . . . . . . .  <a href="#page-14">14</a>
       <a href="#section-3.1.3">3.1.3</a>.  Key Management for SRTP: Security Descriptions  . . .  <a href="#page-15">15</a>
       <a href="#section-3.1.4">3.1.4</a>.  Key Management for SRTP: Encrypted Key Transport  . .  <a href="#page-16">16</a>
       <a href="#section-3.1.5">3.1.5</a>.  Key Management for SRTP: ZRTP and Other Solutions . .  <a href="#page-17">17</a>
     <a href="#section-3.2">3.2</a>.  RTP Legacy Confidentiality  . . . . . . . . . . . . . . .  <a href="#page-17">17</a>
     <a href="#section-3.3">3.3</a>.  IPsec . . . . . . . . . . . . . . . . . . . . . . . . . .  <a href="#page-17">17</a>
     <a href="#section-3.4">3.4</a>.  RTP over TLS over TCP . . . . . . . . . . . . . . . . . .  <a href="#page-18">18</a>
     <a href="#section-3.5">3.5</a>.  RTP over Datagram TLS (DTLS)  . . . . . . . . . . . . . .  <a href="#page-18">18</a>
     <a href="#section-3.6">3.6</a>.  Media Content Security/Digital Rights Management  . . . .  <a href="#page-19">19</a>
       <a href="#section-3.6.1">3.6.1</a>.  ISMA Encryption and Authentication  . . . . . . . . .  <a href="#page-19">19</a>
   <a href="#section-4">4</a>.  Securing RTP Applications . . . . . . . . . . . . . . . . . .  <a href="#page-20">20</a>
     <a href="#section-4.1">4.1</a>.  Application Requirements  . . . . . . . . . . . . . . . .  <a href="#page-20">20</a>
       <a href="#section-4.1.1">4.1.1</a>.  Confidentiality . . . . . . . . . . . . . . . . . . .  <a href="#page-20">20</a>
       <a href="#section-4.1.2">4.1.2</a>.  Integrity . . . . . . . . . . . . . . . . . . . . . .  <a href="#page-21">21</a>
       <a href="#section-4.1.3">4.1.3</a>.  Source Authentication . . . . . . . . . . . . . . . .  <a href="#page-22">22</a>
       <a href="#section-4.1.4">4.1.4</a>.  Identifiers and Identity  . . . . . . . . . . . . . .  <a href="#page-23">23</a>
       <a href="#section-4.1.5">4.1.5</a>.  Privacy . . . . . . . . . . . . . . . . . . . . . . .  <a href="#page-24">24</a>
     <a href="#section-4.2">4.2</a>.  Application Structure . . . . . . . . . . . . . . . . . .  <a href="#page-25">25</a>
     <a href="#section-4.3">4.3</a>.  Automatic Key Management  . . . . . . . . . . . . . . . .  <a href="#page-25">25</a>
     <a href="#section-4.4">4.4</a>.  End-to-End Security vs. Tunnels . . . . . . . . . . . . .  <a href="#page-25">25</a>
     <a href="#section-4.5">4.5</a>.  Plaintext Keys  . . . . . . . . . . . . . . . . . . . . .  <a href="#page-26">26</a>
     <a href="#section-4.6">4.6</a>.  Interoperability  . . . . . . . . . . . . . . . . . . . .  <a href="#page-26">26</a>
   <a href="#section-5">5</a>.  Examples  . . . . . . . . . . . . . . . . . . . . . . . . . .  <a href="#page-26">26</a>
     5.1.  Media Security for SIP-Established Sessions Using
           DTLS-SRTP . . . . . . . . . . . . . . . . . . . . . . . .  <a href="#page-27">27</a>
     <a href="#section-5.2">5.2</a>.  Media Security for WebRTC Sessions  . . . . . . . . . . .  <a href="#page-27">27</a>
     <a href="#section-5.3">5.3</a>.  IP Multimedia Subsystem (IMS) Media Security  . . . . . .  <a href="#page-28">28</a>
     <a href="#section-5.4">5.4</a>.  3GPP Packet-Switched Streaming Service (PSS)  . . . . . .  <a href="#page-29">29</a>
     <a href="#section-5.5">5.5</a>.  RTSP 2.0  . . . . . . . . . . . . . . . . . . . . . . . .  <a href="#page-30">30</a>
   <a href="#section-6">6</a>.  Security Considerations . . . . . . . . . . . . . . . . . . .  <a href="#page-31">31</a>
   <a href="#section-7">7</a>.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  <a href="#page-31">31</a>
   <a href="#section-8">8</a>.  Informative References  . . . . . . . . . . . . . . . . . . .  <a href="#page-31">31</a>




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<span class="h2"><a class="selflink" id="section-1" href="#section-1">1</a>.  Introduction</span>

   The Real-time Transport Protocol (RTP) [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>] is widely used in a
   large variety of multimedia applications, including Voice over IP
   (VoIP), centralized multimedia conferencing, sensor data transport,
   and Internet television (IPTV) services.  These applications can
   range from point-to-point phone calls, through centralized group
   teleconferences, to large-scale television distribution services.
   The types of media can vary significantly, as can the signaling
   methods used to establish the RTP sessions.

   So far, this multidimensional heterogeneity has prevented development
   of a single security solution that meets the needs of the different
   applications.  Instead, a significant number of different solutions
   have been developed to meet different sets of security goals.  This
   makes it difficult for application developers to know what solutions
   exist and whether their properties are appropriate.  This memo gives
   an overview of the available RTP solutions and provides guidance on
   their applicability for different application domains.  It also
   attempts to provide an indication of actual and intended usage at the
   time of writing as additional input to help with considerations such
   as interoperability, availability of implementations, etc.  The
   guidance provided is not exhaustive, and this memo does not provide
   normative recommendations.

   It is important that application developers consider the security
   goals and requirements for their application.  The IETF considers it
   important that protocols implement secure modes of operation and
   makes them available to users [<a href="./rfc3365" title="&quot;Strong Security Requirements for Internet Engineering Task Force Standard Protocols&quot;">RFC3365</a>].  Because of the
   heterogeneity of RTP applications and use cases, however, a single
   security solution cannot be mandated [<a href="./rfc7202" title="&quot;Securing the RTP Protocol Framework: Why RTP Does Not Mandate a Single Media Security Solution&quot;">RFC7202</a>].  Instead, application
   developers need to select mechanisms that provide appropriate
   security for their environment.  It is strongly encouraged that
   common mechanisms be used by related applications in common
   environments.  The IETF publishes guidelines for specific classes of
   applications, so it is worth searching for such guidelines.

   The remainder of this document is structured as follows.  <a href="#section-2">Section 2</a>
   provides additional background.  <a href="#section-3">Section 3</a> outlines the available
   security mechanisms at the time of this writing and lists their key
   security properties and constraints.  <a href="#section-4">Section 4</a> provides guidelines
   and important aspects to consider when securing an RTP application.
   Finally, in <a href="#section-5">Section 5</a>, we give some examples of application domains
   where guidelines for security exist.







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<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a>.  Background</span>

   RTP can be used in a wide variety of topologies due to its support
   for point-to-point sessions, multicast groups, and other topologies
   built around different types of RTP middleboxes.  In the following,
   we review the different topologies supported by RTP to understand
   their implications for the security properties and trust relations
   that can exist in RTP sessions.

<span class="h3"><a class="selflink" id="section-2.1" href="#section-2.1">2.1</a>.  Point-to-Point Sessions</span>

   The most basic use case is two directly connected endpoints, shown in
   Figure 1, where A has established an RTP session with B.  In this
   case, the RTP security is primarily about ensuring that any third
   party be unable to compromise the confidentiality and integrity of
   the media communication.  This requires confidentiality protection of
   the RTP session, integrity protection of the RTP/RTCP packets, and
   source authentication of all the packets to ensure no man-in-the-
   middle (MITM) attack is taking place.

   The source authentication can also be tied to a user or an endpoint's
   verifiable identity to ensure that the peer knows with whom they are
   communicating.  Here, the combination of the security protocol
   protecting the RTP session (and, hence, the RTP and RTCP traffic) and
   the key management protocol becomes important to determine what
   security claims can be made.

   +---+         +---+
   | A |&lt;-------&gt;| B |
   +---+         +---+

                     Figure 1: Point-to-Point Topology

<span class="h3"><a class="selflink" id="section-2.2" href="#section-2.2">2.2</a>.  Sessions Using an RTP Mixer</span>

   An RTP mixer is an RTP session-level middlebox around which one can
   build a multiparty RTP-based conference.  The RTP mixer might
   actually perform media mixing, like mixing audio or compositing video
   images into a new media stream being sent from the mixer to a given
   participant, or it might provide a conceptual stream; for example,
   the video of the current active speaker.  From a security point of
   view, the important features of an RTP mixer are that it generates a
   new media stream, has its own source identifier, and does not simply
   forward the original media.







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   An RTP session using a mixer might have a topology like that in
   Figure 2.  In this example, participants A through D each send
   unicast RTP traffic to the RTP mixer, and receive an RTP stream from
   the mixer, comprising a mixture of the streams from the other
   participants.

   +---+      +------------+      +---+
   | A |&lt;----&gt;|            |&lt;----&gt;| B |
   +---+      |            |      +---+
              |    Mixer   |
   +---+      |            |      +---+
   | C |&lt;----&gt;|            |&lt;----&gt;| D |
   +---+      +------------+      +---+

                   Figure 2: Example RTP Mixer Topology

   A consequence of an RTP mixer having its own source identifier and
   acting as an active participant towards the other endpoints is that
   the RTP mixer needs to be a trusted device that has access to the
   security context(s) established.  The RTP mixer can also become a
   security-enforcing entity.  For example, a common approach to secure
   the topology in Figure 2 is to establish a security context between
   the mixer and each participant independently and have the mixer
   source authenticate each peer.  The mixer then ensures that one
   participant cannot impersonate another.

<span class="h3"><a class="selflink" id="section-2.3" href="#section-2.3">2.3</a>.  Sessions Using an RTP Translator</span>

   RTP translators are middleboxes that provide various levels of
   in-network media translation and transcoding.  Their security
   properties vary widely, depending on which type of operations they
   attempt to perform.  We identify and discuss three different
   categories of RTP translators: transport translators, gateways, and
   media transcoders.

<span class="h4"><a class="selflink" id="section-2.3.1" href="#section-2.3.1">2.3.1</a>.  Transport Translator (Relay)</span>

   A transport translator [<a href="./rfc5117" title="&quot;RTP Topologies&quot;">RFC5117</a>] operates on a level below RTP and
   RTCP.  It relays the RTP/RTCP traffic from one endpoint to one or
   more other addresses.  This can be done based only on IP addresses
   and transport protocol ports, and each receive port on the translator
   can have a very basic list of where to forward traffic.  Transport
   translators also need to implement ingress filtering to prevent
   random traffic from being forwarded that isn't coming from a
   participant in the conference.

   Figure 3 shows an example transport translator, where traffic from
   any one of the four participants will be forwarded to the other three



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   participants unchanged.  The resulting topology is very similar to an
   Any Source Multicast (ASM) session (as discussed in <a href="#section-2.4">Section 2.4</a>) but
   is implemented at the application layer.

   +---+      +------------+      +---+
   | A |&lt;----&gt;|            |&lt;----&gt;| B |
   +---+      |    Relay   |      +---+
              | Translator |
   +---+      |            |      +---+
   | C |&lt;----&gt;|            |&lt;----&gt;| D |
   +---+      +------------+      +---+

                  Figure 3: RTP Relay Translator Topology

   A transport translator can often operate without needing access to
   the security context, as long as the security mechanism does not
   provide protection over the transport-layer information.  A transport
   translator does, however, make the group communication visible and,
   thus, can complicate keying and source authentication mechanisms.
   This is further discussed in <a href="#section-2.4">Section 2.4</a>.

<span class="h4"><a class="selflink" id="section-2.3.2" href="#section-2.3.2">2.3.2</a>.  Gateway</span>

   Gateways are deployed when the endpoints are not fully compatible.
   Figure 4 shows an example topology.  The functions a gateway provides
   can be diverse and range from transport-layer relaying between two
   domains not allowing direct communication, via transport or media
   protocol function initiation or termination, to protocol- or media-
   encoding translation.  The supported security protocol might even be
   one of the reasons a gateway is needed.

   +---+      +-----------+      +---+
   | A |&lt;----&gt;|  Gateway  |&lt;----&gt;| B |
   +---+      +-----------+      +---+

                      Figure 4: RTP Gateway Topology

   The choice of security protocol, and the details of the gateway
   function, will determine if the gateway needs to be trusted with
   access to the application security context.  Many gateways need to be
   trusted by all peers to perform the translation; in other cases, some
   or all peers might not be aware of the presence of the gateway.  The
   security protocols have different properties depending on the degree
   of trust and visibility needed.  Ensuring communication is possible
   without trusting the gateway can be a strong incentive for accepting
   different security properties.  Some security solutions will be able
   to detect the gateways as manipulating the media stream, unless the
   gateway is a trusted device.



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<span class="h4"><a class="selflink" id="section-2.3.3" href="#section-2.3.3">2.3.3</a>.  Media Transcoder</span>

   A media transcoder is a special type of gateway device that changes
   the encoding of the media being transported by RTP.  The discussion
   in <a href="#section-2.3.2">Section 2.3.2</a> applies.  A media transcoder alters the media data
   and, thus, needs to be trusted with access to the security context.

<span class="h3"><a class="selflink" id="section-2.4" href="#section-2.4">2.4</a>.  Any Source Multicast</span>

   Any Source Multicast [<a href="./rfc1112" title="&quot;Host extensions for IP multicasting&quot;">RFC1112</a>] is the original multicast model where
   any multicast group participant can send to the multicast group and
   get their packets delivered to all group members (see Figure 5).
   This form of communication has interesting security properties due to
   the many-to-many nature of the group.  Source authentication is
   important, but all participants with access to the group security
   context will have the necessary secrets to decrypt and verify the
   integrity of the traffic.  Thus, use of any group security context
   fails if the goal is to separate individual sources; alternate
   solutions are needed.

              +-----+
   +---+     /       \    +---+
   | A |----/         \---| B |
   +---+   /           \  +---+
          +  Multicast  +
   +---+   \  Network  /  +---+
   | C |----\         /---| D |
   +---+     \       /    +---+
              +-----+

                Figure 5: Any Source Multicast (ASM) Group

   In addition, the potential large size of multicast groups creates
   some considerations for the scalability of the solution and how the
   key management is handled.

<span class="h3"><a class="selflink" id="section-2.5" href="#section-2.5">2.5</a>.  Source-Specific Multicast</span>

   Source-Specific Multicast (SSM) [<a href="./rfc4607" title="&quot;Source-Specific Multicast for IP&quot;">RFC4607</a>] allows only a specific
   endpoint to send traffic to the multicast group, irrespective of the
   number of RTP media sources.  The endpoint is known as the media
   distribution source.  For the RTP session to function correctly with
   RTCP over an SSM session, extensions have been defined in [<a href="./rfc5760" title="&quot;RTP Control Protocol (RTCP) Extensions for Single-Source Multicast Sessions with Unicast Feedback&quot;">RFC5760</a>].
   Figure 6 shows a sample SSM-based RTP session where several media
   sources, MS1...MSm, all send media to a distribution source, which
   then forwards the media data to the SSM group for delivery to the
   receivers, R1...Rn, and the feedback targets, FT1...FTn.  RTCP
   reception quality feedback is sent unicast from each receiver to one



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   of the feedback targets.  The feedback targets aggregate reception
   quality feedback and forward it upstream towards the distribution
   source.  The distribution source forwards (possibly aggregated and
   summarized) reception feedback to the SSM group and back to the
   original media sources.  The feedback targets are also members of the
   SSM group and receive the media data, so they can send unicast repair
   data to the receivers in response to feedback if appropriate.

    +-----+  +-----+          +-----+
    | MS1 |  | MS2 |   ....   | MSm |
    +-----+  +-----+          +-----+
       ^        ^                ^
       |        |                |
       V        V                V
   +---------------------------------+
   |       Distribution Source       |
   +--------+                        |
   | FT Agg |                        |
   +--------+------------------------+
     ^ ^           |
     :  .          |
     :   +...................+
     :             |          .
     :            / \          .
   +------+      /   \       +-----+
   | FT1  |&lt;----+     +-----&gt;| FT2 |
   +------+    /       \     +-----+
     ^  ^     /         \     ^  ^
     :  :    /           \    :  :
     :  :   /             \   :  :
     :  :  /               \  :  :
     :   ./\               /\.   :
     :   /. \             / .\   :
     :  V  . V           V .  V  :
    +----+ +----+     +----+ +----+
    | R1 | | R2 | ... |Rn-1| | Rn |
    +----+ +----+     +----+ +----+

     Figure 6: Example SSM-Based RTP Session with Two Feedback Targets

   The use of SSM makes it more difficult to inject traffic into the
   multicast group, but not impossible.  Source authentication
   requirements apply for SSM sessions, too; an individual verification
   of who sent the RTP and RTCP packets is needed.  An RTP session using
   SSM will have a group security context that includes the media
   sources, distribution source, feedback targets, and the receivers.
   Each has a different role and will be trusted to perform different
   actions.  For example, the distribution source will need to



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   authenticate the media sources to prevent unwanted traffic from being
   distributed via the SSM group.  Similarly, the receivers need to
   authenticate both the distribution source and their feedback target
   to prevent injection attacks from malicious devices claiming to be
   feedback targets.  An understanding of the trust relationships and
   group security context is needed between all components of the
   system.

<span class="h2"><a class="selflink" id="section-3" href="#section-3">3</a>.  Security Options</span>

   This section provides an overview of security requirements and the
   current RTP security mechanisms that implement those requirements.
   This cannot be a complete survey, since new security mechanisms are
   defined regularly.  The goal is to help applications designers by
   reviewing the types of solutions that are available.  This section
   will use a number of different security-related terms, as described
   in the Internet Security Glossary, Version 2 [<a href="./rfc4949" title="&quot;Internet Security Glossary, Version 2&quot;">RFC4949</a>].

<span class="h3"><a class="selflink" id="section-3.1" href="#section-3.1">3.1</a>.  Secure RTP</span>

   The Secure Real-time Transport Protocol (SRTP) [<a href="./rfc3711" title="&quot;The Secure Real-time Transport Protocol (SRTP)&quot;">RFC3711</a>] is one of
   the most commonly used mechanisms to provide confidentiality,
   integrity protection, source authentication, and replay protection
   for RTP.  SRTP was developed with RTP header compression and third-
   party monitors in mind.  Thus, the RTP header is not encrypted in RTP
   data packets, and the first 8 bytes of the first RTCP packet header
   in each compound RTCP packet are not encrypted.  The entirety of RTP
   packets and compound RTCP packets are integrity protected.  This
   allows RTP header compression to work and lets third-party monitors
   determine what RTP traffic flows exist based on the synchronization
   source (SSRC) fields, but it protects the sensitive content.

   SRTP works with transforms where different combinations of encryption
   algorithm, authentication algorithm, and pseudorandom function can be
   used, and the authentication tag length can be set to any value.
   SRTP can also be easily extended with additional cryptographic
   transforms.  This gives flexibility but requires more security
   knowledge by the application developer.  To simplify things, Session
   Description Protocol (SDP) security descriptions (see <a href="#section-3.1.3">Section 3.1.3</a>)
   and Datagram Transport Layer Security Extension for SRTP (DTLS-SRTP)
   (see <a href="#section-3.1.1">Section 3.1.1</a>) use predefined combinations of transforms, known
   as SRTP crypto suites and SRTP protection profiles, that bundle
   together transforms and other parameters, making them easier to use
   but reducing flexibility.  The Multimedia Internet Keying (MIKEY)
   protocol (see <a href="#section-3.1.2">Section 3.1.2</a>) provides flexibility to negotiate the
   full selection of transforms.  At the time of this writing, the
   following transforms, SRTP crypto suites, and SRTP protection
   profiles are defined or under definition:



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   AES-CM and HMAC-SHA-1:  AES Counter Mode encryption with 128-bit keys
      combined with 160-bit keyed HMAC-SHA-1 with an 80-bit
      authentication tag.  This is the default cryptographic transform
      that needs to be supported.  The transforms are defined in SRTP
      [<a href="./rfc3711" title="&quot;The Secure Real-time Transport Protocol (SRTP)&quot;">RFC3711</a>], with the corresponding SRTP crypto suite defined in
      [<a href="./rfc4568" title="&quot;Session Description Protocol (SDP) Security Descriptions for Media Streams&quot;">RFC4568</a>] and SRTP protection profile defined in [<a href="./rfc5764" title="&quot;Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)&quot;">RFC5764</a>].

   AES-f8 and HMAC-SHA-1:  AES f8-mode encryption using 128-bit keys
      combined with keyed HMAC-SHA-1 using 80-bit authentication.  The
      transforms are defined in [<a href="./rfc3711" title="&quot;The Secure Real-time Transport Protocol (SRTP)&quot;">RFC3711</a>], with the corresponding SRTP
      crypto suite defined in [<a href="./rfc4568" title="&quot;Session Description Protocol (SDP) Security Descriptions for Media Streams&quot;">RFC4568</a>].  The corresponding SRTP
      protection profile is not defined.

   SEED:  A Korean national standard cryptographic transform that is
      defined to be used with SRTP in [<a href="./rfc5669" title="&quot;The SEED Cipher Algorithm and Its Use with the Secure Real- Time Transport Protocol (SRTP)&quot;">RFC5669</a>].  Three options are
      defined: one using SHA-1 authentication, one using Counter Mode
      with Cipher Block Chaining Message Authentication Code (CBC-MAC),
      and one using Galois Counter Mode.

   ARIA:  A Korean block cipher [<a href="#ref-ARIA-SRTP" title="&quot;The ARIA Algorithm and Its Use with the Secure Real-time Transport Protocol(SRTP)&quot;">ARIA-SRTP</a>] that supports 128-, 192-,
      and 256-bit keys.  It also defines three options: Counter Mode
      where combined with HMAC-SHA-1 with 80- or 32-bit authentication
      tags, Counter Mode with CBC-MAC, and Galois Counter Mode.  It also
      defines a different key derivation function than the AES-based
      systems.

   AES-192-CM and AES-256-CM:  Cryptographic transforms for SRTP based
      on AES-192 and AES-256 Counter Mode encryption and 160-bit keyed
      HMAC-SHA-1 with 80- and 32-bit authentication tags.  These provide
      192- and 256-bit encryption keys, but otherwise match the default
      128-bit AES-CM transform.  The transforms are defined in [<a href="./rfc3711" title="&quot;The Secure Real-time Transport Protocol (SRTP)&quot;">RFC3711</a>]
      and [<a href="./rfc6188" title="&quot;The Use of AES-192 and AES-256 in Secure RTP&quot;">RFC6188</a>], and the SRTP crypto suites are defined in
      [<a href="./rfc6188" title="&quot;The Use of AES-192 and AES-256 in Secure RTP&quot;">RFC6188</a>].

   AES-GCM and AES-CCM:  AES Galois Counter Mode and AES Counter Mode
      with CBC-MAC for AES-128 and AES-256.  This authentication is
      included in the cipher text, which becomes expanded with the
      length of the authentication tag instead of using the SRTP
      authentication tag.  This is defined in [<a href="#ref-AES-GCM" title="&quot;AES-GCM and AES-CCM Authenticated Encryption in Secure RTP (SRTP)&quot;">AES-GCM</a>].

   NULL:  SRTP [<a href="./rfc3711" title="&quot;The Secure Real-time Transport Protocol (SRTP)&quot;">RFC3711</a>] also provides a NULL cipher that can be used
      when no confidentiality for RTP/RTCP is requested.  The
      corresponding SRTP protection profile is defined in [<a href="./rfc5764" title="&quot;Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)&quot;">RFC5764</a>].

   The source authentication guarantees provided by SRTP depend on the
   cryptographic transform and key management used.  Some transforms
   give strong source authentication even in multiparty sessions; others
   give weaker guarantees and can authenticate group membership but not



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   sources.  Timed Efficient Stream Loss-Tolerant Authentication (TESLA)
   [<a href="./rfc4383" title="&quot;The Use of Timed Efficient Stream Loss-Tolerant Authentication (TESLA) in the Secure Real-time Transport Protocol (SRTP)&quot;">RFC4383</a>] offers a complement to the regular symmetric keyed
   authentication transforms, like HMAC-SHA-1, and can provide
   per-source authentication in some group communication scenarios.  The
   downside is the need for buffering the packets for a while before
   authenticity can be verified.

   [<a id="ref-RFC4771">RFC4771</a>] defines a variant of the authentication tag that enables a
   receiver to obtain the Roll over Counter for the RTP sequence number
   that is part of the Initialization Vector (IV) for many cryptographic
   transforms.  This enables quicker and easier options for joining a
   long-lived RTP group; for example, a broadcast session.

   RTP header extensions are normally carried in the clear and are only
   integrity protected in SRTP.  This can be problematic in some cases,
   so [<a href="./rfc6904" title="&quot;Encryption of Header Extensions in the Secure Real-time Transport Protocol (SRTP)&quot;">RFC6904</a>] defines an extension to also encrypt selected header
   extensions.

   SRTP is specified and deployed in a number of RTP usage contexts;
   significant support is provided in SIP-established VoIP clients,
   including IP Multimedia Subsystems (IMS), and in the Real Time
   Streaming Protocol (RTSP) [<a href="#ref-RTSP" title="&quot;Real Time Streaming Protocol 2.0 (RTSP)&quot;">RTSP</a>] and RTP-based media streaming.
   Thus, SRTP in general is widely deployed.  When it comes to
   cryptographic transforms, the default (AES-CM and HMAC-SHA-1) is the
   most commonly used, but it might be expected that AES-GCM,
   AES-192-CM, and AES-256-CM will gain usage in future, especially due
   to the AES- and GCM-specific instructions in new CPUs.

   SRTP does not contain an integrated key management solution; instead,
   it relies on an external key management protocol.  There are several
   protocols that can be used.  The following sections outline some
   popular schemes.

<span class="h4"><a class="selflink" id="section-3.1.1" href="#section-3.1.1">3.1.1</a>.  Key Management for SRTP: DTLS-SRTP</span>

   A Datagram Transport Layer Security (DTLS) extension exists for
   establishing SRTP keys [<a href="./rfc5763" title="&quot;Framework for Establishing a Secure Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS)&quot;">RFC5763</a>][RFC5764].  This extension provides
   secure key exchange between two peers, enabling Perfect Forward
   Secrecy (PFS) and binding strong identity verification to an
   endpoint.  PFS is a property of the key agreement protocol that
   ensures that a session key derived from a set of long-term keys will
   not be compromised if one of the long-term keys is compromised in the
   future.  The default key generation will generate a key that contains
   material contributed by both peers.  The key exchange happens in the
   media plane directly between the peers.  The common key exchange
   procedures will take two round trips assuming no losses.  Transport
   Layer Security (TLS) resumption can be used when establishing
   additional media streams with the same peer, and it reduces the setup



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   time to one RTT for these streams (see [<a href="./rfc5764" title="&quot;Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)&quot;">RFC5764</a>] for a discussion of
   TLS resumption in this context).

   The actual security properties of an established SRTP session using
   DTLS will depend on the cipher suites offered and used, as well as
   the mechanism for identifying the endpoints of the handshake.  For
   example, some cipher suites provide PFS, while others do not.  When
   using DTLS, the application designer needs to select which cipher
   suites DTLS-SRTP can offer and accept so that the desired security
   properties are achieved.  The next choice is how to verify the
   identity of the peer endpoint.  One choice can be to rely on the
   certificates and use a PKI to verify them to make an identity
   assertion.  However, this is not the most common way; instead, self-
   signed certificates are common to use to establish trust through
   signaling or other third-party solutions.

   DTLS-SRTP key management can use the signaling protocol in four ways:
   First, to agree on using DTLS-SRTP for media security.  Second, to
   determine the network location (address and port) where each side is
   running a DTLS listener to let the parts perform the key management
   handshakes that generate the keys used by SRTP.  Third, to exchange
   hashes of each side's certificates to bind these to the signaling and
   ensure there is no MITM attack.  This assumes that one can trust the
   signaling solution to be resistant to modification and not be in
   collaboration with an attacker.  Finally, to provide an asserted
   identity, e.g., [<a href="./rfc4474" title="&quot;Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)&quot;">RFC4474</a>], that can be used to prevent modification
   of the signaling and the exchange of certificate hashes.  That way,
   it enables binding between the key exchange and the signaling.

   This usage is well defined for SIP/SDP in [<a href="./rfc5763" title="&quot;Framework for Establishing a Secure Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS)&quot;">RFC5763</a>] and, in most
   cases, can be adopted for use with other bidirectional signaling
   solutions.  It is to be noted that there is work underway to revisit
   the SIP Identity mechanism [<a href="./rfc4474" title="&quot;Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)&quot;">RFC4474</a>] in the IETF STIR working group.

   The main question regarding DTLS-SRTP's security properties is how
   one verifies any peer identity or at least prevents MITM attacks.
   This does require trust in some DTLS-SRTP external parties: either a
   PKI, a signaling system, or some identity provider.

   DTLS-SRTP usage is clearly on the rise.  It is mandatory to support
   in Web Real-Time Communication (WebRTC).  It has growing support
   among SIP endpoints.  DTLS-SRTP was developed in IETF primarily to
   meet security requirements for RTP-based media established using SIP.
   The requirements considered can be reviewed in "Requirements and
   Analysis of Media Security Management Protocols" [<a href="./rfc5479" title="&quot;Requirements and Analysis of Media Security Management Protocols&quot;">RFC5479</a>].






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<span class="h4"><a class="selflink" id="section-3.1.2" href="#section-3.1.2">3.1.2</a>.  Key Management for SRTP: MIKEY</span>

   Multimedia Internet Keying (MIKEY) [<a href="./rfc3830" title="&quot;MIKEY: Multimedia Internet KEYing&quot;">RFC3830</a>] is a keying protocol
   that has several modes with different properties.  MIKEY can be used
   in point-to-point applications using SIP and RTSP (e.g., VoIP calls)
   but is also suitable for use in broadcast and multicast applications
   and centralized group communications.

   MIKEY can establish multiple security contexts or cryptographic
   sessions with a single message.  It is usable in scenarios where one
   entity generates the key and needs to distribute the key to a number
   of participants.  The different modes and the resulting properties
   are highly dependent on the cryptographic method used to establish
   the session keys actually used by the security protocol, like SRTP.

   MIKEY has the following modes of operation:

   Pre-Shared Key:  Uses a pre-shared secret for symmetric key crypto
      used to secure a keying message carrying the already-generated
      session key.  This system is the most efficient from the
      perspective of having small messages and processing demands.  The
      downside is scalability, where usually the effort for the
      provisioning of pre-shared keys is only manageable if the number
      of endpoints is small.

   Public Key Encryption:  Uses a public key crypto to secure a keying
      message carrying the already-generated session key.  This is more
      resource intensive but enables scalable systems.  It does require
      a public key infrastructure to enable verification.

   Diffie-Hellman:  Uses Diffie-Hellman key agreement to generate the
      session key, thus providing perfect forward secrecy.  The downside
      is high resource consumption in bandwidth and processing during
      the MIKEY exchange.  This method can't be used to establish group
      keys as each pair of peers performing the MIKEY exchange will
      establish different keys.

   HMAC-Authenticated Diffie-Hellman:  [<a href="./rfc4650" title="&quot;HMAC-Authenticated Diffie-Hellman for Multimedia Internet KEYing (MIKEY)&quot;">RFC4650</a>] defines a variant of
      the Diffie-Hellman exchange that uses a pre-shared key in a keyed
      Hashed Message Authentication Code (HMAC) to verify authenticity
      of the keying material instead of a digital signature as in the
      previous method.  This method is still restricted to
      point-to-point usage.

   RSA-R:  MIKEY-RSA in Reverse mode [<a href="./rfc4738" title="&quot;MIKEY- RSA-R: An Additional Mode of Key Distribution in Multimedia Internet KEYing (MIKEY)&quot;">RFC4738</a>] is a variant of the
      public key method, which doesn't rely on the initiator of the key
      exchange knowing the responder's certificate.  This method lets
      both the initiator and the responder specify the session keying



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      material depending on the use case.  Usage of this mode requires
      one round-trip time.

   TICKET:  Ticket Payload (TICKET) [<a href="./rfc6043" title="&quot;MIKEY-TICKET: Ticket-Based Modes of Key Distribution in Multimedia Internet KEYing (MIKEY)&quot;">RFC6043</a>] is a MIKEY extension using
      a trusted centralized key management service (KMS).  The initiator
      and responder do not share any credentials; instead, they trust a
      third party, the KMS, with which they both have or can establish
      shared credentials.

   IBAKE:  Identity-Based Authenticated Key Exchange (IBAKE) [<a href="./rfc6267" title="&quot;MIKEY-IBAKE: Identity-Based Authenticated Key Exchange (IBAKE) Mode of Key Distribution in Multimedia Internet KEYing (MIKEY)&quot;">RFC6267</a>]
      uses a KMS infrastructure but with lower demand on the KMS.  It
      claims to provide both perfect forward and backwards secrecy.

   SAKKE:  [<a href="./rfc6509" title="&quot;MIKEY-SAKKE: Sakai-Kasahara Key Encryption in Multimedia Internet KEYing (MIKEY)&quot;">RFC6509</a>] provides Sakai-Kasahara Key Encryption (SAKKE) in
      MIKEY.  It is based on Identity-based Public Key Cryptography and
      a KMS infrastructure to establish a shared secret value and
      certificateless signatures to provide source authentication.  Its
      features include simplex transmission, scalability, low-latency
      call setup, and support for secure deferred delivery.

   MIKEY messages have several different transports.  [<a href="./rfc4567" title="&quot;Key Management Extensions for Session Description Protocol (SDP) and Real Time Streaming Protocol (RTSP)&quot;">RFC4567</a>] defines
   how MIKEY messages can be embedded in general SDP for usage with the
   signaling protocols SIP, Session Announcement Protocol (SAP), and
   RTSP.  There also exists a usage of MIKEY defined by the Third
   Generation Partnership Project (3GPP) that sends MIKEY messages
   directly over UDP [<a href="#ref-T3GPP.33.246">T3GPP.33.246</a>] to key the receivers of Multimedia
   Broadcast and Multicast Service (MBMS) [<a href="#ref-T3GPP.26.346">T3GPP.26.346</a>].  [<a href="./rfc3830" title="&quot;MIKEY: Multimedia Internet KEYing&quot;">RFC3830</a>]
   defines the application/mikey media type, allowing MIKEY to be used
   in, e.g., email and HTTP.

   Based on the many choices, it is important to consider the properties
   needed in one's solution and based on that evaluate which modes are
   candidates for use.  More information on the applicability of the
   different MIKEY modes can be found in [<a href="./rfc5197" title="&quot;On the Applicability of Various Multimedia Internet KEYing (MIKEY) Modes and Extensions&quot;">RFC5197</a>].

   MIKEY with pre-shared keys is used by 3GPP MBMS [<a href="#ref-T3GPP.33.246">T3GPP.33.246</a>], and
   IMS media security [<a href="#ref-T3GPP.33.328">T3GPP.33.328</a>] specifies the use of the TICKET
   mode transported over SIP and HTTP.  RTSP 2.0 [<a href="#ref-RTSP" title="&quot;Real Time Streaming Protocol 2.0 (RTSP)&quot;">RTSP</a>] specifies use of
   the RSA-R mode.  There are some SIP endpoints that support MIKEY.
   The modes they use are unknown to the authors.

<span class="h4"><a class="selflink" id="section-3.1.3" href="#section-3.1.3">3.1.3</a>.  Key Management for SRTP: Security Descriptions</span>

   [<a id="ref-RFC4568">RFC4568</a>] provides a keying solution based on sending plaintext keys
   in SDP [<a href="./rfc4566" title="&quot;SDP: Session Description Protocol&quot;">RFC4566</a>].  It is primarily used with SIP and the SDP Offer/
   Answer model and is well defined in point-to-point sessions where
   each side declares its own unique key.  Using security descriptions
   to establish group keys is less well defined and can have security



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   issues since it's difficult to guarantee unique SSRCs (as needed to
   avoid a "two-time pad" attack -- see <a href="./rfc3711#section-9">Section&nbsp;9 of [RFC3711]</a>).

   Since keys are transported in plaintext in SDP, they can easily be
   intercepted unless the SDP carrying protocol provides strong
   end-to-end confidentiality and authentication guarantees.  This is
   not normally the case; instead, hop-by-hop security is provided
   between signaling nodes using TLS.  This leaves the keying material
   sensitive to capture by the traversed signaling nodes.  Thus, in most
   cases, the security properties of security descriptions are weak.
   The usage of security descriptions usually requires additional
   security measures; for example, the signaling nodes are trusted and
   protected by strict access control.  Usage of security descriptions
   requires careful design in order to ensure that the security goals
   can be met.

   Security descriptions are the most commonly deployed keying solution
   for SIP-based endpoints, where almost all endpoints that support SRTP
   also support security descriptions.  It is also used for access
   protection in IMS Media Security [<a href="#ref-T3GPP.33.328">T3GPP.33.328</a>].

<span class="h4"><a class="selflink" id="section-3.1.4" href="#section-3.1.4">3.1.4</a>.  Key Management for SRTP: Encrypted Key Transport</span>

   Encrypted Key Transport (EKT) [<a href="#ref-EKT" title="&quot;Encrypted Key Transport for Secure RTP&quot;">EKT</a>] is an SRTP extension that enables
   group keying despite using a keying mechanism like DTLS-SRTP that
   doesn't support group keys.  It is designed for centralized
   conferencing, but it can also be used in sessions where endpoints
   connect to a conference bridge or a gateway and need to be
   provisioned with the keys each participant on the bridge or gateway
   uses to avoid decryption and encryption cycles.  This can enable
   interworking between DTLS-SRTP and other keying systems where either
   party can set the key (e.g., interworking with security
   descriptions).

   The mechanism is based on establishing an additional EKT key, which
   everyone uses to protect their actual session key.  The actual
   session key is sent in an expanded authentication tag to the other
   session participants.  This key is only sent occasionally or
   periodically depending on use cases and depending on what
   requirements exist for timely delivery or notification.

   The only known deployment of EKT so far is in some Cisco video
   conferencing products.








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<span class="h4"><a class="selflink" id="section-3.1.5" href="#section-3.1.5">3.1.5</a>.  Key Management for SRTP: ZRTP and Other Solutions</span>

   The ZRTP [<a href="./rfc6189" title="&quot;ZRTP: Media Path Key Agreement for Unicast Secure RTP&quot;">RFC6189</a>] key management system for SRTP was proposed as an
   alternative to DTLS-SRTP.  ZRTP provides best effort encryption
   independent of the signaling protocol and utilizes key continuity,
   Short Authentication Strings, or a PKI for authentication.  ZRTP
   wasn't adopted as an IETF Standards Track protocol, but was instead
   published as an Informational RFC in the IETF stream.  Commercial
   implementations exist.

   Additional proprietary solutions are also known to exist.

<span class="h3"><a class="selflink" id="section-3.2" href="#section-3.2">3.2</a>.  RTP Legacy Confidentiality</span>

   <a href="#section-9">Section 9</a> of the RTP standard [<a href="./rfc3550" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>] defines a Data Encryption
   Standard (DES) or 3DES-based encryption of RTP and RTCP packets.
   This mechanism is keyed using plaintext keys in SDP [<a href="./rfc4566" title="&quot;SDP: Session Description Protocol&quot;">RFC4566</a>] using
   the "k=" SDP field.  This method can provide confidentiality but, as
   discussed in <a href="./rfc3550#section-9">Section&nbsp;9 of [RFC3550]</a>, it has extremely weak security
   properties and is not to be used.

<span class="h3"><a class="selflink" id="section-3.3" href="#section-3.3">3.3</a>.  IPsec</span>

   IPsec [<a href="./rfc4301" title="&quot;Security Architecture for the Internet Protocol&quot;">RFC4301</a>] can be used in either tunnel or transport mode to
   protect RTP and RTCP packets in transit from one network interface to
   another.  This can be sufficient when the network interfaces have a
   direct relation or in a secured environment where it can be
   controlled who can read the packets from those interfaces.

   The main concern with using IPsec to protect RTP traffic is that in
   most cases, using a VPN approach that terminates the security
   association at some node prior to the RTP endpoint leaves the traffic
   vulnerable to attack between the VPN termination node and the
   endpoint.  Thus, usage of IPsec requires careful thought and design
   of its usage so that it meets the security goals.  An important
   question is how one ensures the IPsec terminating peer and the
   ultimate destination are the same.  Applications can have issues
   using existing APIs when determining if IPsec is being used or not
   and when determining who the authenticated peer entity is when IPsec
   is used.

   IPsec with RTP is more commonly used as a security solution between
   infrastructure nodes that exchange many RTP sessions and media
   streams.  The establishment of a secure tunnel between such nodes
   minimizes the key management overhead.






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<span class="h3"><a class="selflink" id="section-3.4" href="#section-3.4">3.4</a>.  RTP over TLS over TCP</span>

   Just as RTP can be sent over TCP [<a href="./rfc4571" title="&quot;Framing Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) Packets over Connection- Oriented Transport&quot;">RFC4571</a>], it can also be sent over
   TLS over TCP [<a href="./rfc4572" title="&quot;Connection-Oriented Media Transport over the Transport Layer Security (TLS) Protocol in the Session Description Protocol (SDP)&quot;">RFC4572</a>], using TLS to provide point-to-point security
   services.  The security properties TLS provides are confidentiality,
   integrity protection, and possible source authentication if the
   client or server certificates are verified and provide a usable
   identity.  When used in multiparty scenarios using a central node for
   media distribution, the security provided is only between the central
   node and the peers, so the security properties for the whole session
   are dependent on what trust one can place in the central node.

   RTSP 1.0 [<a href="./rfc2326" title="&quot;Real Time Streaming Protocol (RTSP)&quot;">RFC2326</a>] and 2.0 [<a href="#ref-RTSP" title="&quot;Real Time Streaming Protocol 2.0 (RTSP)&quot;">RTSP</a>] specify the usage of RTP over the
   same TLS/TCP connection that the RTSP messages are sent over.  It
   appears that RTP over TLS/TCP is also used in some proprietary
   solutions that use TLS to bypass firewalls.

<span class="h3"><a class="selflink" id="section-3.5" href="#section-3.5">3.5</a>.  RTP over Datagram TLS (DTLS)</span>

   DTLS [<a href="./rfc6347" title="&quot;Datagram Transport Layer Security Version 1.2&quot;">RFC6347</a>] is based on TLS [<a href="./rfc5246" title="&quot;The Transport Layer Security (TLS) Protocol Version 1.2&quot;">RFC5246</a>] but designed to work over an
   unreliable datagram-oriented transport rather than requiring reliable
   byte stream semantics from the transport protocol.  Accordingly, DTLS
   can provide point-to-point security for RTP flows analogous to that
   provided by TLS but over a datagram transport such as UDP.  The two
   peers establish a DTLS association between each other, including the
   possibility to do certificate-based source authentication when
   establishing the association.  All RTP and RTCP packets flowing will
   be protected by this DTLS association.

   Note that using DTLS for RTP flows is different from using DTLS-SRTP
   key management.  DTLS-SRTP uses the same key management steps as
   DTLS, but uses SRTP for the per-packet security operations.  Using
   DTLS for RTP flows uses the normal datagram TLS data protection,
   wrapping complete RTP packets.  When using DTLS for RTP flows, the
   RTP and RTCP packets are completely encrypted with no headers in the
   clear; when using DTLS-SRTP, the RTP headers are in the clear and
   only the payload data is encrypted.

   DTLS can use similar techniques to those available for DTLS-SRTP to
   bind a signaling-side agreement to communicate to the certificates
   used by the endpoint when doing the DTLS handshake.  This enables use
   without having a certificate-based trust chain to a trusted
   certificate root.

   There does not appear to be significant usage of DTLS for RTP.






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<span class="h3"><a class="selflink" id="section-3.6" href="#section-3.6">3.6</a>.  Media Content Security/Digital Rights Management</span>

   Mechanisms have been defined that encrypt only the media content
   operating within the RTP payload data and leaving the RTP headers and
   RTCP unaffected.  There are several reasons why this might be
   appropriate, but a common rationale is to ensure that the content
   stored by RTSP streaming servers has the media content in a protected
   format that cannot be read by the streaming server (this is mostly
   done in the context of Digital Rights Management).  These approaches
   then use a key management solution between the rights provider and
   the consuming client to deliver the key used to protect the content
   and do not give the media server access to the security context.
   Such methods have several security weaknesses such as the fact that
   the same key is handed out to a potentially large group of receiving
   clients, increasing the risk of a leak.

   Use of this type of solution can be of interest in environments that
   allow middleboxes to rewrite the RTP headers and select which streams
   are delivered to an endpoint (e.g., some types of centralized video
   conference systems).  The advantage of encrypting and possibly
   integrity protecting the payload but not the headers is that the
   middlebox can't eavesdrop on the media content, but it can still
   provide stream switching functionality.  The downside of such a
   system is that it likely needs two levels of security: the payload-
   level solution, to provide confidentiality and source authentication,
   and a second layer with additional transport security ensuring source
   authentication and integrity of the RTP headers associated with the
   encrypted payloads.  This can also result in the need to have two
   different key management systems as the entity protecting the packets
   and payloads are different with a different set of keys.

   The aspect of two tiers of security are present in ISMACryp (see
   <a href="#section-3.6.1">Section 3.6.1</a>) and the deprecated 3GPP Packet-switched Streaming
   Service solution; see Annex K of [<a href="#ref-T3GPP.26.234R8">T3GPP.26.234R8</a>].

<span class="h4"><a class="selflink" id="section-3.6.1" href="#section-3.6.1">3.6.1</a>.  ISMA Encryption and Authentication</span>

   The Internet Streaming Media Alliance (ISMA) has defined ISMA
   Encryption and Authentication 2.0 [<a href="#ref-ISMACryp2" title="&quot;ISMA Encryption and Authentication Version 2.0&quot;">ISMACryp2</a>].  This specification
   defines how one encrypts and packetizes the encrypted application
   data units (ADUs) in an RTP payload using the MPEG-4 generic payload
   format [<a href="./rfc3640" title="&quot;RTP Payload Format for Transport of MPEG-4 Elementary Streams&quot;">RFC3640</a>].  The ADU types that are allowed are those that can
   be stored as elementary streams in an ISO Media File format-based
   file.  ISMACryp uses SRTP for packet-level integrity and source
   authentication from a streaming server to the receiver.






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   Key management for an ISMACryp-based system can be achieved through
   Open Mobile Alliance (OMA) Digital Rights Management 2.0 [<a href="#ref-OMADRMv2" title="&quot;OMA Digital Rights Management V2.0&quot;">OMADRMv2</a>],
   for example.

<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a>.  Securing RTP Applications</span>

   In the following, we provide guidelines for how to choose appropriate
   security mechanisms for RTP applications.

<span class="h3"><a class="selflink" id="section-4.1" href="#section-4.1">4.1</a>.  Application Requirements</span>

   This section discusses a number of application requirements that need
   to be considered.  An application designer choosing security
   solutions requires a good understanding of what level of security is
   needed and what behavior they strive to achieve.

<span class="h4"><a class="selflink" id="section-4.1.1" href="#section-4.1.1">4.1.1</a>.  Confidentiality</span>

   When it comes to confidentiality of an RTP session, there are several
   aspects to consider:

   Probability of compromise:  When using encryption to provide media
      confidentiality, it is necessary to have some rough understanding
      of the security goal and how long one can expect the protected
      content to remain confidential.  National or other regulations
      might provide additional requirements on a particular usage of an
      RTP.  From that, one can determine which encryption algorithms are
      to be used from the set of available transforms.

   Potential for other leakage:  RTP-based security in most of its forms
      simply wraps RTP and RTCP packets into cryptographic containers.
      This commonly means that the size of the original RTP payload is
      visible to observers of the protected packet flow.  This can
      provide information to those observers.  A well-documented case is
      the risk with variable bitrate speech codecs that produce
      different sized packets based on the speech input [<a href="./rfc6562" title="&quot;Guidelines for the Use of Variable Bit Rate Audio with Secure RTP&quot;">RFC6562</a>].
      Potential threats such as these need to be considered and, if they
      are significant, then restrictions will be needed on mode choices
      in the codec, or additional padding will need to be added to make
      all packets equal size and remove the informational leakage.

      Another case is RTP header extensions.  If SRTP is used, header
      extensions are normally not protected by the security mechanism
      protecting the RTP payload.  If the header extension carries
      information that is considered sensitive, then the application
      needs to be modified to ensure that mechanisms used to protect
      against such information leakage are employed.




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   Who has access:  When considering the confidentiality properties of a
      system, it is important to consider where the media handled in the
      clear.  For example, if the system is based on an RTP mixer that
      needs the keys to decrypt the media, process it, and repacketize
      it, then is the mixer providing the security guarantees expected
      by the other parts of the system?  Furthermore, it is important to
      consider who has access to the keys.  The policies for the
      handling of the keys, and who can access the keys, need to be
      considered along with the confidentiality goals.

   As can be seen, the actual confidentiality level has likely more to
   do with the application's usage of centralized nodes, and the details
   of the key management solution chosen, than with the actual choice of
   encryption algorithm (although, of course, the encryption algorithm
   needs to be chosen appropriately for the desired security level).

<span class="h4"><a class="selflink" id="section-4.1.2" href="#section-4.1.2">4.1.2</a>.  Integrity</span>

   Protection against modification of content by a third party, or due
   to errors in the network, is another factor to consider.  The first
   aspect that one assesses is what resilience one has against
   modifications to the content.  Some media types are extremely
   sensitive to network bit errors, whereas others might be able to
   tolerate some degree of data corruption.  Equally important is to
   consider the sensitivity of the content, who is providing the
   integrity assertion, what is the source of the integrity tag, and
   what are the risks of modifications happening prior to that point
   where protection is applied.  These issues affect what cryptographic
   algorithm is used, the length of the integrity tags, and whether the
   entire payload is protected.

   RTP applications that rely on central nodes need to consider if
   hop-by-hop integrity is acceptable or if true end-to-end integrity
   protection is needed.  Is it important to be able to tell if a
   middlebox has modified the data?  There are some uses of RTP that
   require trusted middleboxes that can modify the data in a way that
   doesn't break integrity protection as seen by the receiver, for
   example, local advertisement insertion in IPTV systems.  There are
   also uses where it is essential that such in-network modification be
   detectable.  RTP can support both with appropriate choices of
   security mechanisms.

   Integrity of the data is commonly closely tied to the question of
   source authentication.  That is, it becomes important to know who
   makes an integrity assertion for the data.






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<span class="h4"><a class="selflink" id="section-4.1.3" href="#section-4.1.3">4.1.3</a>.  Source Authentication</span>

   Source authentication is about determining who sent a particular RTP
   or RTCP packet.  It is normally closely tied with integrity, since a
   receiver generally also wants to ensure that the data received is
   what the source really sent, so source authentication without
   integrity is not particularly useful.  Similarly, integrity
   protection without source authentication is also not particularly
   useful; a claim that a packet is unchanged that cannot itself be
   validated as from the source (or some from other known and trusted
   party) is meaningless.

   Source authentication can be asserted in several different ways:

   Base level:  Using cryptographic mechanisms that give authentication
      with some type of key management provide an implicit method for
      source authentication.  Assuming that the mechanism has sufficient
      strength not to be circumvented in the time frame when you would
      accept the packet as valid, it is possible to assert a source-
      authenticated statement; this message is likely from a source that
      has the cryptographic key(s) to this communication.

      What that assertion actually means is highly dependent on the
      application and how it handles the keys.  If only the two peers
      have access to the keys, this can form a basis for a strong trust
      relationship that traffic is authenticated coming from one of the
      peers.  However, in a multiparty scenario where security contexts
      are shared among participants, most base-level authentication
      solutions can't even assert that this packet is from the same
      source as the previous packet.

   Binding the source and the signaling:  A step up in the assertion
      that can be done in base-level systems is to tie the signaling to
      the key exchange.  Here, the goal is to at least be able to assert
      that the source of the packets is the same entity with which the
      receiver established the session.  How feasible this is depends on
      the properties of the key management system, the ability to tie
      the signaling to a particular source, and the degree of trust the
      receiver places on the different nodes involved.

      For example, systems where the key exchange is done using the
      signaling systems, such as security descriptions [<a href="./rfc4568" title="&quot;Session Description Protocol (SDP) Security Descriptions for Media Streams&quot;">RFC4568</a>] enable
      a direct binding between signaling and key exchange.  In such
      systems, the actual security depends on the trust one can place in
      the signaling system to correctly associate the peer's identifier
      with the key exchange.





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   Using identifiers:  If the applications have access to a system that
      can provide verifiable identifiers, then the source authentication
      can be bound to that identifier.  For example, in a point-to-point
      communication, even symmetric key crypto, where the key management
      can assert that the key has only been exchanged with a particular
      identifier, can provide a strong assertion about the source of the
      traffic.  SIP Identity [<a href="./rfc4474" title="&quot;Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)&quot;">RFC4474</a>] provides one example of how this
      can be done and could be used to bind DTLS-SRTP certificates used
      by an endpoint to the identity provider's public key to
      authenticate the source of a DTLS-SRTP flow.

      Note that all levels of the system need to have matching
      capability to assert identifiers.  If the signaling can assert
      that only a given entity in a multiparty session has a key, then
      the media layer might be able to provide guarantees about the
      identifier used by the media sender.  However, using a signaling
      authentication mechanism built on a group key can limit the media
      layer to asserting only group membership.

<span class="h4"><a class="selflink" id="section-4.1.4" href="#section-4.1.4">4.1.4</a>.  Identifiers and Identity</span>

   There exist many different types of systems providing identifiers
   with different properties (e.g., SIP Identity [<a href="./rfc4474" title="&quot;Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)&quot;">RFC4474</a>]).  In the
   context of RTP applications, the most important property is the
   possibility to perform source authentication and verify such
   assertions in relation to any claimed identifiers.  What an
   identifier really represents can also vary but, in the context of
   communication, one of the most obvious is the identifiers
   representing the identity of the human user with which one
   communicates.  However, the human user can also have additional
   identifiers in a particular role.  For example, the human (Alice) can
   also be a police officer, and in some cases, an identifier for her
   role as police officer will be more relevant than one that asserts
   that she is Alice.  This is common in contact with organizations,
   where it is important to prove the person's right to represent the
   organization.  Some examples of identifier/identity mechanisms that
   can be used:

   Certificate based:  A certificate is used to assert the identifiers
      used to claim an identity; by having access to the private part of
      the certificate, one can perform signing to assert one's identity.
      Any entity interested in verifying the assertion then needs the
      public part of the certificate.  By having the certificate, one
      can verify the signature against the certificate.  The next step
      is to determine if one trusts the certificate's trust chain.
      Commonly, by provisioning the verifier with the public part of a
      root certificate, this enables the verifier to verify a trust
      chain from the root certificate down to the identifier in the



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      certificate.  However, the trust is based on all steps in the
      certificate chain being verifiable and trusted.  Thus, the
      provisioning of root certificates and the ability to revoke
      compromised certificates are aspects that will require
      infrastructure.

   Online identity providers:  An online identity provider (IdP) can
      authenticate a user's right to use an identifier and then perform
      assertions on their behalf or provision the requester with short-
      term credentials to assert the identifiers.  The verifier can then
      contact the IdP to request verification of a particular
      identifier.  Here, the trust is highly dependent on how much one
      trusts the IdP.  The system also becomes dependent on having
      access to the relevant IdP.

   In all of the above examples, an important part of the security
   properties is related to the method for authenticating the access to
   the identity.

<span class="h4"><a class="selflink" id="section-4.1.5" href="#section-4.1.5">4.1.5</a>.  Privacy</span>

   RTP applications need to consider what privacy goals they have.  As
   RTP applications communicate directly between peers in many cases,
   the IP addresses of any communication peer will be available.  The
   main privacy concern with IP addresses is related to geographical
   location and the possibility to track a user of an endpoint.  The
   main way to avoid such concerns is the introduction of relay (e.g., a
   Traversal Using Relay NAT (TURN) server [<a href="./rfc5766" title="&quot;Traversal Using Relays around NAT (TURN): Relay Extensions to Session Traversal Utilities for NAT (STUN)&quot;">RFC5766</a>]) or centralized
   media mixers or forwarders that hide the address of a peer from any
   other peer.  The security and trust placed in these relays obviously
   needs to be carefully considered.

   RTP itself can contribute to enabling a particular user to be tracked
   between communication sessions if the Canonical Name (CNAME) is
   generated according to the RTP specification in the form of
   user@host.  Such RTCP CNAMEs are likely long-term stable over
   multiple sessions, allowing tracking of users.  This can be desirable
   for long-term fault tracking and diagnosis, but it clearly has
   privacy implications.  Instead, cryptographically random ones could
   be used as defined by "Guidelines for Choosing RTP Control Protocol
   (RTCP) CNAMEs" [<a href="./rfc7022" title="&quot;Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)&quot;">RFC7022</a>].

   If privacy goals exist, they need to be considered and the system
   designed with them in mind.  In addition, certain RTP features might
   have to be configured to safeguard privacy or have requirements on
   how the implementation is done.





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<span class="h3"><a class="selflink" id="section-4.2" href="#section-4.2">4.2</a>.  Application Structure</span>

   When it comes to RTP security, the most appropriate solution is often
   highly dependent on the topology of the communication session.  The
   signaling also impacts what information can be provided and if this
   can be instance specific or common for a group.  In the end, the key
   management system will highly affect the security properties achieved
   by the application.  At the same time, the communication structure of
   the application limits what key management methods are applicable.
   As different key management methods have different requirements on
   underlying infrastructure, it is important to take that aspect into
   consideration early in the design.

<span class="h3"><a class="selflink" id="section-4.3" href="#section-4.3">4.3</a>.  Automatic Key Management</span>

   The guidelines for Cryptographic Key Management [<a href="./rfc4107" title="&quot;Guidelines for Cryptographic Key Management&quot;">RFC4107</a>] provide an
   overview of why automatic key management is important.  They also
   provide a strong recommendation on using automatic key management.
   Most of the security solutions reviewed in this document provide or
   support automatic key management, at least to establish session keys.
   In some more long-term use cases, credentials might need to be
   manually deployed in certain cases.

   For SRTP, an important aspect of automatic key management is to
   ensure that two-time pads do not occur, in particular by preventing
   multiple endpoints using the same session key and SSRC.  In these
   cases, automatic key management methods can have strong dependencies
   on signaling features to function correctly.  If those dependencies
   can't be fulfilled, additional constrains on usage, e.g., per-
   endpoint session keys, might be needed to avoid the issue.

   When selecting security mechanisms for an RTP application, it is
   important to consider the properties of the key management.  Using
   key management that is both automatic and integrated will provide
   minimal interruption for the user and is important to ensure that
   security can, and will remain, to be on by default.

<span class="h3"><a class="selflink" id="section-4.4" href="#section-4.4">4.4</a>.  End-to-End Security vs. Tunnels</span>

   If the security mechanism only provides a secured tunnel, for
   example, like some common uses of IPsec (<a href="#section-3.3">Section 3.3</a>), it is
   important to consider the full end-to-end properties of the system.
   How does one ensure that the path from the endpoint to the local
   tunnel ingress/egress is secure and can be trusted (and similarly for
   the other end of the tunnel)?  How does one handle the source
   authentication of the peer, as the security protocol identifies the
   other end of the tunnel?  These are some of the issues that arise
   when one considers a tunnel-based security protocol rather than an



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   end-to-end one.  Even with clear requirements and knowledge that one
   still can achieve the security properties using a tunnel-based
   solution, one ought to prefer to use end-to-end mechanisms, as they
   are much less likely to violate any assumptions made about
   deployment.  These assumptions can also be difficult to automatically
   verify.

<span class="h3"><a class="selflink" id="section-4.5" href="#section-4.5">4.5</a>.  Plaintext Keys</span>

   Key management solutions that use plaintext keys, like SDP security
   descriptions (<a href="#section-3.1.3">Section 3.1.3</a>), require care to ensure a secure
   transport of the signaling messages that contain the plaintext keys.
   For plaintext keys, the security properties of the system depend on
   how securely the plaintext keys are protected end-to-end between the
   sender and receiver(s).  Not only does one need to consider what
   transport protection is provided for the signaling message, including
   the keys, but also the degree to which any intermediaries in the
   signaling are trusted.  Untrusted intermediaries can perform MITM
   attacks on the communication or can log the keys, resulting in the
   encryption being compromised significantly after the actual
   communication occurred.

<span class="h3"><a class="selflink" id="section-4.6" href="#section-4.6">4.6</a>.  Interoperability</span>

   Few RTP applications exist as independent applications that never
   interoperate with anything else.  Rather, they enable communication
   with a potentially large number of other systems.  To minimize the
   number of security mechanisms that need to be implemented, it is
   important to consider if one can use the same security mechanisms as
   other applications.  This can also reduce problems with determining
   what security level is actually negotiated in a particular session.

   The desire to be interoperable can, in some cases, be in conflict
   with the security requirements of an application.  To meet the
   security goals, it might be necessary to sacrifice interoperability.
   Alternatively, one can implement multiple security mechanisms; this,
   however, introduces the complication of ensuring that the user
   understands what it means to use a particular security system.  In
   addition, the application can then become vulnerable to bid-down
   attacks.

<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a>.  Examples</span>

   In the following, we describe a number of example security solutions
   for applications using RTP services or frameworks.  These examples
   are provided to illustrate the choices available.  They are not
   normative recommendations for security.




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<span class="h3"><a class="selflink" id="section-5.1" href="#section-5.1">5.1</a>.  Media Security for SIP-Established Sessions Using DTLS-SRTP</span>

   In 2009, the IETF evaluated media security for RTP sessions
   established using point-to-point SIP sessions.  A number of
   requirements were determined, and based on those, the existing
   solutions for media security and especially the keying methods were
   analyzed.  The resulting requirements and analysis were published in
   [<a href="./rfc5479" title="&quot;Requirements and Analysis of Media Security Management Protocols&quot;">RFC5479</a>].  Based on this analysis and working group discussion,
   DTLS-SRTP was determined to be the best solution.

   The security solution for SIP using DTLS-SRTP is defined in
   "Framework for Establishing a Secure Real-time Transport Protocol
   (SRTP) Security Context Using Datagram Transport Layer Security
   (DTLS)" [<a href="./rfc5763" title="&quot;Framework for Establishing a Secure Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS)&quot;">RFC5763</a>].  On a high level, the framework uses SIP with SDP
   offer/answer procedures to exchange the network addresses where the
   server endpoint will have a DTLS-SRTP-enabled server running.  The
   SIP signaling is also used to exchange the fingerprints of the
   certificate each endpoint will use in the DTLS establishment process.
   When the signaling is sufficiently completed, the DTLS-SRTP client
   performs DTLS handshakes and establishes SRTP session keys.  The
   clients also verify the fingerprints of the certificates to verify
   that no man in the middle has inserted themselves into the exchange.

   DTLS has a number of good security properties.  For example, to
   enable a MITM, someone in the signaling path needs to perform an
   active action and modify both the signaling message and the DTLS
   handshake.  Solutions also exist that enable the fingerprints to be
   bound to identities.  SIP Identity provides an identity established
   by the first proxy for each user [<a href="./rfc4474" title="&quot;Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)&quot;">RFC4474</a>].  This reduces the number
   of nodes the connecting User Agent has to trust to include just the
   first-hop proxy rather than the full signaling path.  The biggest
   security weakness of this system is its dependency on the signaling.
   SIP signaling passes multiple nodes and there is usually no message
   security deployed, only hop-by-hop transport security, if any,
   between the nodes.

<span class="h3"><a class="selflink" id="section-5.2" href="#section-5.2">5.2</a>.  Media Security for WebRTC Sessions</span>

   Web Real-Time Communication (WebRTC) [<a href="#ref-WebRTC" title="&quot;Overview: Real Time Protocols for Browser-based Applications&quot;">WebRTC</a>] is a solution providing
   JavaScript web applications with real-time media directly between
   browsers.  Media is transported using RTP and protected using a
   mandatory application of SRTP [<a href="./rfc3711" title="&quot;The Secure Real-time Transport Protocol (SRTP)&quot;">RFC3711</a>], with keying done using DTLS-
   SRTP [<a href="./rfc5764" title="&quot;Datagram Transport Layer Security (DTLS) Extension to Establish Keys for the Secure Real-time Transport Protocol (SRTP)&quot;">RFC5764</a>].  The security configuration is further defined in
   "WebRTC Security Architecture" [<a href="#ref-WebRTC-SEC">WebRTC-SEC</a>].

   A hash of the peer's certificate is provided to the JavaScript web
   application, allowing that web application to verify identity of the
   peer.  There are several ways in which the certificate hashes can be



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   verified.  An approach identified in the WebRTC security architecture
   [<a href="#ref-WebRTC-SEC">WebRTC-SEC</a>] is to use an identity provider.  In this solution, the
   identity provider, which is a third party to the web application,
   signs the DTLS-SRTP hash combined with a statement on the validity of
   the user identity that has been used to sign the hash.  The receiver
   of such an identity assertion can then independently verify the user
   identity to ensure that it is the identity that the receiver intended
   to communicate with, and that the cryptographic assertion holds; this
   way, a user can be certain that the application also can't perform a
   MITM and acquire the keys to the media communication.  Other ways of
   verifying the certificate hashes exist; for example, they could be
   verified against a hash carried in some out-of-band channel (e.g.,
   compare with a hash printed on a business card) or using a verbal
   short authentication string (e.g., as in ZRTP [<a href="./rfc6189" title="&quot;ZRTP: Media Path Key Agreement for Unicast Secure RTP&quot;">RFC6189</a>]) or using
   hash continuity.

   In the development of WebRTC, there has also been attention given to
   privacy considerations.  The main RTP-related concerns that have been
   raised are:

   Location disclosure:  As Interactive Connectivity Establishment (ICE)
      negotiation [<a href="./rfc5245" title="&quot;Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols&quot;">RFC5245</a>] provides IP addresses and ports for the
      browser, this leaks location information in the signaling to the
      peer.  To prevent this, one can block the usage of any ICE
      candidate that isn't a relay candidate, i.e., where the IP and
      port provided belong to the service providers media traffic relay.

   Prevent tracking between sessions:  Static RTP CNAMEs and DTLS-SRTP
      certificates provide information that is reused between session
      instances.  Thus, to prevent tracking, such information ought not
      be reused between sessions, or the information ought not be sent
      in the clear.  Note that generating new certificates each time
      prevents continuity in authentication, however, as WebRTC users
      are expected to use multiple devices to access the same
      communication service, such continuity can't be expected anyway;
      instead, the above-described identity mechanism has to be relied
      on.

   Note: The above cases are focused on providing privacy from other
   parties, not on providing privacy from the web server that provides
   the WebRTC JavaScript application.

<span class="h3"><a class="selflink" id="section-5.3" href="#section-5.3">5.3</a>.  IP Multimedia Subsystem (IMS) Media Security</span>

   In IMS, the core network is controlled by a single operator or by
   several operators with high trust in each other.  Except for some
   types of accesses, the operator is in full control, and no packages
   are routed over the Internet.  Nodes in the core network offer



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   services such as voice mail, interworking with legacy systems (Public
   Switched Telephone Network (PSTN), Global System for Mobile
   Communications (GSM), and 3G), and transcoding.  Endpoints are
   authenticated during the SIP registration using either IMS and
   Authentication and Key Agreement (AKA) (using Subscriber Identity
   Module (SIM) credentials) or SIP Digest (using a password).

   In IMS media security [<a href="#ref-T3GPP.33.328">T3GPP.33.328</a>], end-to-end encryption is,
   therefore, not seen as needed or desired as it would hinder, for
   example, interworking and transcoding, making calls between
   incompatible terminals impossible.  Because of this, IMS media
   security mostly uses end-to-access-edge security where SRTP is
   terminated in the first node in the core network.  As the SIP
   signaling is trusted and encrypted (with TLS or IPsec), security
   descriptions [<a href="./rfc4568" title="&quot;Session Description Protocol (SDP) Security Descriptions for Media Streams&quot;">RFC4568</a>] is considered to give good protection against
   eavesdropping over the accesses that are not already encrypted (GSM,
   3G, and Long Term Evolution (LTE)).  Media source authentication is
   based on knowledge of the SRTP session key and trust in that the IMS
   network will only forward media from the correct endpoint.

   For enterprises and government agencies, which might have weaker
   trust in the IMS core network and can be assumed to have compatible
   terminals, end-to-end security can be achieved by deploying their own
   key management server.

   Work on interworking with WebRTC is currently ongoing; the security
   will still be end-to-access-edge but using DTLS-SRTP [<a href="./rfc5763" title="&quot;Framework for Establishing a Secure Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS)&quot;">RFC5763</a>]
   instead of security descriptions.

<span class="h3"><a class="selflink" id="section-5.4" href="#section-5.4">5.4</a>.  3GPP Packet-Switched Streaming Service (PSS)</span>

   The 3GPP Release 11 PSS specification of the Packet-switched
   Streaming Service (PSS) [<a href="#ref-T3GPP.26.234R11">T3GPP.26.234R11</a>] defines, in Annex R, a set
   of security mechanisms.  These security mechanisms are concerned with
   protecting the content from being copied, i.e., Digital Rights
   Management (DRM).  To meet these goals with the specified solution,
   the client implementation and the application platform are trusted to
   protect against access and modification by an attacker.

   PSS is media controlled by RTSP 1.0 [<a href="./rfc2326" title="&quot;Real Time Streaming Protocol (RTSP)&quot;">RFC2326</a>] streaming over RTP.
   Thus, an RTSP client whose user wants to access a protected content
   will request a session description (SDP [<a href="./rfc4566" title="&quot;SDP: Session Description Protocol&quot;">RFC4566</a>]) for the protected
   content.  This SDP will indicate that the media is protected by
   ISMACryp 2.0 [<a href="#ref-ISMACryp2" title="&quot;ISMA Encryption and Authentication Version 2.0&quot;">ISMACryp2</a>] encoding application units (AUs).  The
   key(s) used to protect the media is provided in one of two ways.  If
   a single key is used, then the client uses some DRM system to
   retrieve the key as indicated in the SDP.  Commonly, OMA DRM v2
   [<a href="#ref-OMADRMv2" title="&quot;OMA Digital Rights Management V2.0&quot;">OMADRMv2</a>] will be used to retrieve the key.  If multiple keys are to



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   be used, then an additional RTSP stream for key updates in parallel
   with the media streams is established, where key updates are sent to
   the client using Short Term Key Messages defined in the "Service and
   Content Protection for Mobile Broadcast Services" part [<a href="#ref-OMASCP" title="&quot;Service and Content Protection for Mobile Broadcast Services&quot;">OMASCP</a>] of
   the OMA Mobile Broadcast Services [<a href="#ref-OMABCAST" title="&quot;Mobile Broadcast Services Version 1.0&quot;">OMABCAST</a>].

   Worth noting is that this solution doesn't provide any integrity
   verification method for the RTP header and payload header
   information; only the encoded media AU is protected. 3GPP has not
   defined any requirement for supporting any solution that could
   provide that service.  Thus, replay or insertion attacks are
   possible.  Another property is that the media content can be
   protected by the ones providing the media, so that the operators of
   the RTSP server have no access to unprotected content.  Instead, all
   that want to access the media are supposed to contact the DRM keying
   server, and if the device is acceptable, they will be given the key
   to decrypt the media.

   To protect the signaling, RTSP 1.0 supports the usage of TLS.  This
   is, however, not explicitly discussed in the PSS specification.
   Usage of TLS can prevent both modification of the session description
   information and help maintain some privacy of what content the user
   is watching as all URLs would then be confidentiality protected.

<span class="h3"><a class="selflink" id="section-5.5" href="#section-5.5">5.5</a>.  RTSP 2.0</span>

   The Real-time Streaming Protocol 2.0 [<a href="#ref-RTSP" title="&quot;Real Time Streaming Protocol 2.0 (RTSP)&quot;">RTSP</a>] offers an interesting
   comparison to the PSS service (<a href="#section-5.4">Section 5.4</a>) that is based on RTSP 1.0
   and service requirements perceived by mobile operators.  A major
   difference between RTSP 1.0 and RTSP 2.0 is that 2.0 is fully defined
   under the requirement to have a mandatory-to-implement security
   mechanism.  As it specifies one transport media over RTP, it is also
   defining security mechanisms for the RTP-transported media streams.

   The security goal for RTP in RTSP 2.0 is to ensure that there is
   confidentiality, integrity, and source authentication between the
   RTSP server and the client.  This to prevent eavesdropping on what
   the user is watching for privacy reasons and to prevent replay or
   injection attacks on the media stream.  To reach these goals, the
   signaling also has to be protected, requiring the use of TLS between
   the client and server.

   Using TLS-protected signaling, the client and server agree on the
   media transport method when doing the SETUP request and response.
   The secured media transport is SRTP (SAVP/RTP) normally over UDP.
   The key management for SRTP is MIKEY using RSA-R mode.  The RSA-R
   mode is selected as it allows the RTSP server to select the key
   despite having the RTSP client initiate the MIKEY exchange.  It also



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   enables the reuse of the RTSP server's TLS certificate when creating
   the MIKEY messages, thus ensuring a binding between the RTSP server
   and the key exchange.  Assuming the SETUP process works, this will
   establish a SRTP crypto context to be used between the RTSP server
   and the client for the RTP-transported media streams.

<span class="h2"><a class="selflink" id="section-6" href="#section-6">6</a>.  Security Considerations</span>

   This entire document is about security.  Please read it.

<span class="h2"><a class="selflink" id="section-7" href="#section-7">7</a>.  Acknowledgements</span>

   We thank the IESG for their careful review of [<a href="./rfc7202" title="&quot;Securing the RTP Protocol Framework: Why RTP Does Not Mandate a Single Media Security Solution&quot;">RFC7202</a>], which led to
   the writing of this memo.  John Mattsson has contributed the IMS
   Media Security example (<a href="#section-5.3">Section 5.3</a>).

   The authors wish to thank Christian Correll, Dan Wing, Kevin Gross,
   Alan Johnston, Michael Peck, Ole Jacobsen, Spencer Dawkins, Stephen
   Farrell, John Mattsson, and Suresh Krishnan for their reviews and
   proposals for improvements to the text.

<span class="h2"><a class="selflink" id="section-8" href="#section-8">8</a>.  Informative References</span>

   [<a id="ref-AES-GCM">AES-GCM</a>]   McGrew, D. and K. Igoe, "AES-GCM and AES-CCM
               Authenticated Encryption in Secure RTP (SRTP)", Work in
               Progress, September 2013.

   [<a id="ref-ARIA-SRTP">ARIA-SRTP</a>] Kim, W., Lee, J., Kim, D., Park, J., and D. Kwon, "The
               ARIA Algorithm and Its Use with the Secure Real-time
               Transport Protocol(SRTP)", Work in Progress, November
               2013.

   [<a id="ref-EKT">EKT</a>]       McGrew, D. and D. Wing, "Encrypted Key Transport for
               Secure RTP", Work in Progress, February 2014.

   [<a id="ref-ISMACryp2">ISMACryp2</a>] Internet Streaming Media Alliance (ISMA), "ISMA
               Encryption and Authentication Version 2.0", November
               2007, &lt;<a href="http://www.oipf.tv/images/site/DOCS/mpegif/ISMA/isma_easpec2.0.pdf">http://www.oipf.tv/images/site/DOCS/mpegif/ISMA/</a>
               <a href="http://www.oipf.tv/images/site/DOCS/mpegif/ISMA/isma_easpec2.0.pdf">isma_easpec2.0.pdf</a>&gt;.

   [<a id="ref-OMABCAST">OMABCAST</a>]  Open Mobile Alliance, "Mobile Broadcast Services Version
               1.0", February 2009,
               &lt;<a href="http://technical.openmobilealliance.org/Technical/release_program/bcast_v1_0.aspx">http://technical.openmobilealliance.org/Technical/</a>
               <a href="http://technical.openmobilealliance.org/Technical/release_program/bcast_v1_0.aspx">release_program/bcast_v1_0.aspx</a>&gt;.







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   [<a id="ref-OMADRMv2">OMADRMv2</a>]  Open Mobile Alliance, "OMA Digital Rights Management
               V2.0", July 2008,
               &lt;<a href="http://technical.openmobilealliance.org/Technical/release_program/drm_v2_0.aspx">http://technical.openmobilealliance.org/</a>
               <a href="http://technical.openmobilealliance.org/Technical/release_program/drm_v2_0.aspx">Technical/release_program/drm_v2_0.aspx</a>&gt;.

   [<a id="ref-OMASCP">OMASCP</a>]    Open Mobile Alliance, "Service and Content Protection for
               Mobile Broadcast Services", January 2013,
               &lt;<a href="http://technical.openmobilealliance.org/Technical/release_program/docs/BCAST/V1_0_1-20130109-A/OMA-TS-BCAST_SvcCntProtection-V1_0_1-20130109-A.pdf">http://technical.openmobilealliance.org/Technical/</a>
               <a href="http://technical.openmobilealliance.org/Technical/release_program/docs/BCAST/V1_0_1-20130109-A/OMA-TS-BCAST_SvcCntProtection-V1_0_1-20130109-A.pdf">release_program/docs/BCAST/V1_0_1-20130109-A/</a>
               <a href="http://technical.openmobilealliance.org/Technical/release_program/docs/BCAST/V1_0_1-20130109-A/OMA-TS-BCAST_SvcCntProtection-V1_0_1-20130109-A.pdf">OMA-TS-BCAST_SvcCntProtection-V1_0_1-20130109-A.pdf</a>&gt;.

   [<a id="ref-RFC1112">RFC1112</a>]   Deering, S., "Host extensions for IP multicasting", STD
               5, <a href="./rfc1112">RFC 1112</a>, August 1989.

   [<a id="ref-RFC2326">RFC2326</a>]   Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time
               Streaming Protocol (RTSP)", <a href="./rfc2326">RFC 2326</a>, April 1998.

   [<a id="ref-RFC3365">RFC3365</a>]   Schiller, J., "Strong Security Requirements for Internet
               Engineering Task Force Standard Protocols", <a href="https://www.rfc-editor.org/bcp/bcp61">BCP 61</a>, <a href="./rfc3365">RFC</a>
               <a href="./rfc3365">3365</a>, August 2002.

   [<a id="ref-RFC3550">RFC3550</a>]   Schulzrinne, H., Casner, S., Frederick, R., and V.
               Jacobson, "RTP: A Transport Protocol for Real-Time
               Applications", STD 64, <a href="./rfc3550">RFC 3550</a>, July 2003.

   [<a id="ref-RFC3640">RFC3640</a>]   van der Meer, J., Mackie, D., Swaminathan, V., Singer,
               D., and P. Gentric, "RTP Payload Format for Transport of
               MPEG-4 Elementary Streams", <a href="./rfc3640">RFC 3640</a>, November 2003.

   [<a id="ref-RFC3711">RFC3711</a>]   Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
               Norrman, "The Secure Real-time Transport Protocol
               (SRTP)", <a href="./rfc3711">RFC 3711</a>, March 2004.

   [<a id="ref-RFC3830">RFC3830</a>]   Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
               Norrman, "MIKEY: Multimedia Internet KEYing", <a href="./rfc3830">RFC 3830</a>,
               August 2004.

   [<a id="ref-RFC4107">RFC4107</a>]   Bellovin, S. and R. Housley, "Guidelines for
               Cryptographic Key Management", <a href="https://www.rfc-editor.org/bcp/bcp107">BCP 107</a>, <a href="./rfc4107">RFC 4107</a>, June
               2005.

   [<a id="ref-RFC4301">RFC4301</a>]   Kent, S. and K. Seo, "Security Architecture for the
               Internet Protocol", <a href="./rfc4301">RFC 4301</a>, December 2005.

   [<a id="ref-RFC4383">RFC4383</a>]   Baugher, M. and E. Carrara, "The Use of Timed Efficient
               Stream Loss-Tolerant Authentication (TESLA) in the Secure
               Real-time Transport Protocol (SRTP)", <a href="./rfc4383">RFC 4383</a>, February
               2006.



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   [<a id="ref-RFC4474">RFC4474</a>]   Peterson, J. and C. Jennings, "Enhancements for
               Authenticated Identity Management in the Session
               Initiation Protocol (SIP)", <a href="./rfc4474">RFC 4474</a>, August 2006.

   [<a id="ref-RFC4566">RFC4566</a>]   Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
               Description Protocol", <a href="./rfc4566">RFC 4566</a>, July 2006.

   [<a id="ref-RFC4567">RFC4567</a>]   Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E.
               Carrara, "Key Management Extensions for Session
               Description Protocol (SDP) and Real Time Streaming
               Protocol (RTSP)", <a href="./rfc4567">RFC 4567</a>, July 2006.

   [<a id="ref-RFC4568">RFC4568</a>]   Andreasen, F., Baugher, M., and D. Wing, "Session
               Description Protocol (SDP) Security Descriptions for
               Media Streams", <a href="./rfc4568">RFC 4568</a>, July 2006.

   [<a id="ref-RFC4571">RFC4571</a>]   Lazzaro, J., "Framing Real-time Transport Protocol (RTP)
               and RTP Control Protocol (RTCP) Packets over Connection-
               Oriented Transport", <a href="./rfc4571">RFC 4571</a>, July 2006.

   [<a id="ref-RFC4572">RFC4572</a>]   Lennox, J., "Connection-Oriented Media Transport over the
               Transport Layer Security (TLS) Protocol in the Session
               Description Protocol (SDP)", <a href="./rfc4572">RFC 4572</a>, July 2006.

   [<a id="ref-RFC4607">RFC4607</a>]   Holbrook, H. and B. Cain, "Source-Specific Multicast for
               IP", <a href="./rfc4607">RFC 4607</a>, August 2006.

   [<a id="ref-RFC4650">RFC4650</a>]   Euchner, M., "HMAC-Authenticated Diffie-Hellman for
               Multimedia Internet KEYing (MIKEY)", <a href="./rfc4650">RFC 4650</a>, September
               2006.

   [<a id="ref-RFC4738">RFC4738</a>]   Ignjatic, D., Dondeti, L., Audet, F., and P. Lin, "MIKEY-
               RSA-R: An Additional Mode of Key Distribution in
               Multimedia Internet KEYing (MIKEY)", <a href="./rfc4738">RFC 4738</a>, November
               2006.

   [<a id="ref-RFC4771">RFC4771</a>]   Lehtovirta, V., Naslund, M., and K. Norrman, "Integrity
               Transform Carrying Roll-Over Counter for the Secure Real-
               time Transport Protocol (SRTP)", <a href="./rfc4771">RFC 4771</a>, January 2007.

   [<a id="ref-RFC4949">RFC4949</a>]   Shirey, R., "Internet Security Glossary, Version 2", <a href="./rfc4949">RFC</a>
               <a href="./rfc4949">4949</a>, August 2007.

   [<a id="ref-RFC5117">RFC5117</a>]   Westerlund, M. and S. Wenger, "RTP Topologies", <a href="./rfc5117">RFC 5117</a>,
               January 2008.






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   [<a id="ref-RFC5197">RFC5197</a>]   Fries, S. and D. Ignjatic, "On the Applicability of
               Various Multimedia Internet KEYing (MIKEY) Modes and
               Extensions", <a href="./rfc5197">RFC 5197</a>, June 2008.

   [<a id="ref-RFC5245">RFC5245</a>]   Rosenberg, J., "Interactive Connectivity Establishment
               (ICE): A Protocol for Network Address Translator (NAT)
               Traversal for Offer/Answer Protocols", <a href="./rfc5245">RFC 5245</a>, April
               2010.

   [<a id="ref-RFC5246">RFC5246</a>]   Dierks, T. and E. Rescorla, "The Transport Layer Security
               (TLS) Protocol Version 1.2", <a href="./rfc5246">RFC 5246</a>, August 2008.

   [<a id="ref-RFC5479">RFC5479</a>]   Wing, D., Fries, S., Tschofenig, H., and F. Audet,
               "Requirements and Analysis of Media Security Management
               Protocols", <a href="./rfc5479">RFC 5479</a>, April 2009.

   [<a id="ref-RFC5669">RFC5669</a>]   Yoon, S., Kim, J., Park, H., Jeong, H., and Y. Won, "The
               SEED Cipher Algorithm and Its Use with the Secure Real-
               Time Transport Protocol (SRTP)", <a href="./rfc5669">RFC 5669</a>, August 2010.

   [<a id="ref-RFC5760">RFC5760</a>]   Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
               Protocol (RTCP) Extensions for Single-Source Multicast
               Sessions with Unicast Feedback", <a href="./rfc5760">RFC 5760</a>, February 2010.

   [<a id="ref-RFC5763">RFC5763</a>]   Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
               for Establishing a Secure Real-time Transport Protocol
               (SRTP) Security Context Using Datagram Transport Layer
               Security (DTLS)", <a href="./rfc5763">RFC 5763</a>, May 2010.

   [<a id="ref-RFC5764">RFC5764</a>]   McGrew, D. and E. Rescorla, "Datagram Transport Layer
               Security (DTLS) Extension to Establish Keys for the
               Secure Real-time Transport Protocol (SRTP)", <a href="./rfc5764">RFC 5764</a>,
               May 2010.

   [<a id="ref-RFC5766">RFC5766</a>]   Mahy, R., Matthews, P., and J. Rosenberg, "Traversal
               Using Relays around NAT (TURN): Relay Extensions to
               Session Traversal Utilities for NAT (STUN)", <a href="./rfc5766">RFC 5766</a>,
               April 2010.

   [<a id="ref-RFC6043">RFC6043</a>]   Mattsson, J. and T. Tian, "MIKEY-TICKET: Ticket-Based
               Modes of Key Distribution in Multimedia Internet KEYing
               (MIKEY)", <a href="./rfc6043">RFC 6043</a>, March 2011.

   [<a id="ref-RFC6188">RFC6188</a>]   McGrew, D., "The Use of AES-192 and AES-256 in Secure
               RTP", <a href="./rfc6188">RFC 6188</a>, March 2011.






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   [<a id="ref-RFC6189">RFC6189</a>]   Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media
               Path Key Agreement for Unicast Secure RTP", <a href="./rfc6189">RFC 6189</a>,
               April 2011.

   [<a id="ref-RFC6267">RFC6267</a>]   Cakulev, V. and G. Sundaram, "MIKEY-IBAKE: Identity-Based
               Authenticated Key Exchange (IBAKE) Mode of Key
               Distribution in Multimedia Internet KEYing (MIKEY)", <a href="./rfc6267">RFC</a>
               <a href="./rfc6267">6267</a>, June 2011.

   [<a id="ref-RFC6347">RFC6347</a>]   Rescorla, E. and N. Modadugu, "Datagram Transport Layer
               Security Version 1.2", <a href="./rfc6347">RFC 6347</a>, January 2012.

   [<a id="ref-RFC6509">RFC6509</a>]   Groves, M., "MIKEY-SAKKE: Sakai-Kasahara Key Encryption
               in Multimedia Internet KEYing (MIKEY)", <a href="./rfc6509">RFC 6509</a>,
               February 2012.

   [<a id="ref-RFC6562">RFC6562</a>]   Perkins, C. and JM. Valin, "Guidelines for the Use of
               Variable Bit Rate Audio with Secure RTP", <a href="./rfc6562">RFC 6562</a>, March
               2012.

   [<a id="ref-RFC6904">RFC6904</a>]   Lennox, J., "Encryption of Header Extensions in the
               Secure Real-time Transport Protocol (SRTP)", <a href="./rfc6904">RFC 6904</a>,
               April 2013.

   [<a id="ref-RFC7022">RFC7022</a>]   Begen, A., Perkins, C., Wing, D., and E. Rescorla,
               "Guidelines for Choosing RTP Control Protocol (RTCP)
               Canonical Names (CNAMEs)", <a href="./rfc7022">RFC 7022</a>, September 2013.

   [<a id="ref-RFC7202">RFC7202</a>]   Perkins, C. and M. Westerlund, "Securing the RTP Protocol
               Framework: Why RTP Does Not Mandate a Single Media
               Security Solution", <a href="./rfc7202">RFC 7202</a>, April 2014.

   [<a id="ref-RTSP">RTSP</a>]      Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
               and M. Stiemerling, "Real Time Streaming Protocol 2.0
               (RTSP)", Work in Progress, February 2014.

   [<a id="ref-T3GPP.26.234R11">T3GPP.26.234R11</a>]
               3GPP, "Technical Specification Group Services and System
               Aspects; Transparent end-to-end Packet-switched Streaming
               Service (PSS); Protocols and codecs", 3GPP TS 26.234
               11.1.0, September 2012,
               &lt;<a href="http://www.3gpp.org/DynaReport/26234.htm">http://www.3gpp.org/DynaReport/26234.htm</a>&gt;.









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   [<a id="ref-T3GPP.26.234R8">T3GPP.26.234R8</a>]
               3GPP, "Technical Specification Group Services and System
               Aspects; Transparent end-to-end Packet-switched Streaming
               Service (PSS); Protocols and codecs", 3GPP TS 26.234
               8.4.0, September 2009,
               &lt;<a href="http://www.3gpp.org/DynaReport/26234.htm">http://www.3gpp.org/DynaReport/26234.htm</a>&gt;.

   [<a id="ref-T3GPP.26.346">T3GPP.26.346</a>]
               3GPP, "Multimedia Broadcast/Multicast Service (MBMS);
               Protocols and codecs", 3GPP TS 26.346 10.7.0, March 2013,
               &lt;<a href="http://www.3gpp.org/DynaReport/26346.htm">http://www.3gpp.org/DynaReport/26346.htm</a>&gt;.

   [<a id="ref-T3GPP.33.246">T3GPP.33.246</a>]
               3GPP, "3G Security; Security of Multimedia Broadcast/
               Multicast Service (MBMS)", 3GPP TS 33.246 11.1.0,
               December 2012,
               &lt;<a href="http://www.3gpp.org/DynaReport/33246.htm">http://www.3gpp.org/DynaReport/33246.htm</a>&gt;.

   [<a id="ref-T3GPP.33.328">T3GPP.33.328</a>]
               3GPP, "IP Multimedia Subsystem (IMS) media plane
               security", 3GPP TS 33.328 12.1.0, December 2012,
               &lt;<a href="http://www.3gpp.org/DynaReport/33328.htm">http://www.3gpp.org/DynaReport/33328.htm</a>&gt;.

   [<a id="ref-WebRTC-SEC">WebRTC-SEC</a>]
               Rescorla, E., <a style="text-decoration: none" href='https://www.google.com/search?sitesearch=datatracker.ietf.org%2Fdoc%2Fhtml%2F&amp;q=inurl:draft-+%22WebRTC+Security+Architecture%22'>"WebRTC Security Architecture"</a>, Work in
               Progress, February 2014.

   [<a id="ref-WebRTC">WebRTC</a>]   Alvestrand, H., "Overview: Real Time Protocols for
               Browser-based Applications", Work in Progress, February
               2014.





















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Authors' Addresses

   Magnus Westerlund
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

   Phone: +46 10 714 82 87
   EMail: magnus.westerlund@ericsson.com


   Colin Perkins
   University of Glasgow
   School of Computing Science
   Glasgow  G12 8QQ
   United Kingdom

   EMail: csp@csperkins.org
   URI:   <a href="http://csperkins.org/">http://csperkins.org/</a>































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