Network Working Group                                      M. Westerlund
Internet-Draft                                                  Ericsson
Intended status: Informational                             C. S. Perkins
Expires: April 24, 2014                            University of Glasgow
                                                        October 21, 2013

                   Options for Securing RTP Sessions


   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/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

Status of This Memo

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

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   This Internet-Draft will expire on April 24, 2014.

Copyright Notice

   Copyright (c) 2013 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   ( in effect on the date of

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   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Background  . . . . . . . . . . . . . . . . . . . . . . . . .   4
     2.1.  Point-to-Point Sessions . . . . . . . . . . . . . . . . .   4
     2.2.  Sessions Using an RTP Mixer . . . . . . . . . . . . . . .   4
     2.3.  Sessions Using an RTP Translator  . . . . . . . . . . . .   5
       2.3.1.  Transport Translator (Relay)  . . . . . . . . . . . .   5
       2.3.2.  Gateway . . . . . . . . . . . . . . . . . . . . . . .   6
       2.3.3.  Media Transcoder  . . . . . . . . . . . . . . . . . .   7
     2.4.  Any Source Multicast  . . . . . . . . . . . . . . . . . .   7
     2.5.  Source-Specific Multicast . . . . . . . . . . . . . . . .   7
   3.  Security Options  . . . . . . . . . . . . . . . . . . . . . .   9
     3.1.  Secure RTP  . . . . . . . . . . . . . . . . . . . . . . .   9
       3.1.1.  Key Management for SRTP: DTLS-SRTP  . . . . . . . . .  11
       3.1.2.  Key Management for SRTP: MIKEY  . . . . . . . . . . .  12
       3.1.3.  Key Management for SRTP: Security Descriptions  . . .  14
       3.1.4.  Key Management for SRTP: Encrypted Key Transport  . .  15
       3.1.5.  Key Management for SRTP: Other systems  . . . . . . .  15
     3.2.  RTP Legacy Confidentiality  . . . . . . . . . . . . . . .  15
     3.3.  IPsec . . . . . . . . . . . . . . . . . . . . . . . . . .  16
     3.4.  RTP over TLS over TCP . . . . . . . . . . . . . . . . . .  16
     3.5.  RTP over Datagram TLS (DTLS)  . . . . . . . . . . . . . .  16
     3.6.  Media Content Security/Digital Rights Management  . . . .  17
       3.6.1.  ISMA Encryption and Authentication  . . . . . . . . .  18
   4.  Securing RTP Applications . . . . . . . . . . . . . . . . . .  18
     4.1.  Application Requirements  . . . . . . . . . . . . . . . .  18
       4.1.1.  Confidentiality . . . . . . . . . . . . . . . . . . .  18
       4.1.2.  Integrity . . . . . . . . . . . . . . . . . . . . . .  20
       4.1.3.  Source Authentication . . . . . . . . . . . . . . . .  20
       4.1.4.  Identity  . . . . . . . . . . . . . . . . . . . . . .  22
       4.1.5.  Privacy . . . . . . . . . . . . . . . . . . . . . . .  22
     4.2.  Application Structure . . . . . . . . . . . . . . . . . .  23
     4.3.  Interoperability  . . . . . . . . . . . . . . . . . . . .  23
   5.  Examples  . . . . . . . . . . . . . . . . . . . . . . . . . .  24
     5.1.  Media Security for SIP-established Sessions using DTLS-
           SRTP  . . . . . . . . . . . . . . . . . . . . . . . . . .  24
     5.2.  Media Security for WebRTC Sessions  . . . . . . . . . . .  25
     5.3.  IP Multimedia Subsystem (IMS) Media Security  . . . . . .  26
     5.4.  3GPP Packet Based Streaming Service (PSS) . . . . . . . .  26
     5.5.  RTSP 2.0  . . . . . . . . . . . . . . . . . . . . . . . .  27

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   6.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  28
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  28
   8.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  28
   9.  Informative References  . . . . . . . . . . . . . . . . . . .  28
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  33

1.  Introduction

   Real-time Transport Protocol (RTP) [RFC3550] 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 centralised group
   teleconferences, to large-scale television distribution services.
   The types of media can vary significantly, as can the signalling
   methods used to establish the RTP sessions.

   This multi-dimensional heterogeneity has so far prevented development
   of a single security solution that meets the needs of the different
   applications.  Instead 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 indication of actual and intended usage at 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

   It is important that application developers consider the security
   goals and requirements for their application.  The IETF considers it
   important that protocols implement, and makes available to the user,
   secure modes of operation [RFC3365].  Because of the heterogeneity of
   RTP applications and use cases, however, a single security solution
   cannot be mandated [I-D.ietf-avt-srtp-not-mandatory].  Instead,
   application developers need to select mechanisms that provide
   appropriate security for their environment.  It is strongly
   encouraged that common mechanisms are used by related applications in
   common environments.  The IETF publishes guidelines for specific
   classes of applications, so it is worth searching for such

   The remainder of this document is structured as follows.  Section 2
   provides additional background.  Section 3 outlines the available
   security mechanisms at the time of this writing, and lists their key
   security properties and constraints.  That is followed by guidelines

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   and important aspects to consider when securing an RTP application in
   Section 4.  Finally, we give some examples of application domains
   where guidelines for security exist in Section 5.

2.  Background

   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.

2.1.  Point-to-Point Sessions

   The most basic use case is two directly connected end-points, 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 can't 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
   attack is taking place.

   The source authentication can also be tied to a user or an end-
   point's verifiable identity to ensure that the peer knows who they
   are communicating with.  Here the combination of the security
   protocol protecting the RTP session and its RTP and RTCP traffic and
   the key-management protocol becomes important in which security
   statements one can do.

                            +---+         +---+
                            | A |<------->| B |
                            +---+         +---+

                     Figure 1: Point-to-point topology

2.2.  Sessions Using an RTP Mixer

   An RTP mixer is an RTP session-level middlebox that one can build a
   multi-party RTP based conference around.  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 is that it generates a new
   media stream, and 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

                   +---+      +------------+      +---+
                   | A |<---->|            |<---->| B |
                   +---+      |            |      +---+
                              |    Mixer   |
                   +---+      |            |      +---+
                   | C |<---->|            |<---->| 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 end-points 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.

2.3.  Sessions Using an RTP Translator

   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 three different categories of RTP translator:
   transport translators, gateways, and media transcoders.  We discuss
   each in turn.

2.3.1.  Transport Translator (Relay)

   A transport translator [RFC5117] operates on a level below RTP and
   RTCP.  It relays the RTP/RTCP traffic from one end-point to one or
   more other addresses.  This can be done based only on IP addresses
   and transport protocol ports, with 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
   Any Source Multicast (ASM) session (as discussed in Section 2.4), but
   implemented at the application layer.

                   +---+      +------------+      +---+
                   | A |<---->|            |<---->| B |
                   +---+      |    Relay   |      +---+
                              | Translator |
                   +---+      |            |      +---+
                   | C |<---->|            |<---->| 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
   so can complicate keying and source authentication mechanisms.  This
   is further discussed in Section 2.4.

2.3.2.  Gateway

   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 |<---->|  Gateway  |<---->| 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 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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2.3.3.  Media Transcoder

   A Media transcoder is a special type of gateway device that changes
   the encoding of the media being transported by RTP.  The discussion
   in Section 2.3.2 applies.  A media transcoder alters the media data,
   and thus needs to be trusted with access to the security context.

2.4.  Any Source Multicast

   Any Source Multicast [RFC1112] 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 group security context
   will have the necessary secrets to decrypt and verify integrity of
   the traffic.  Thus use of any group security context fails if the
   goal is to separate individual sources; alternate solutions are

                       +---+     /       \    +---+
                       | A |----/         \---| B |
                       +---+   /   Multi-  \  +---+
                              +    Cast     +
                       +---+   \  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.

2.5.  Source-Specific Multicast

   Source-Specific Multicast [RFC4607] allows only a specific end-point
   to send traffic to the multicast group, irrespective of the number of
   RTP media sources.  The end-point is known as the media Distribution
   Source.  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
   of the Feedback Targets.  The feedback targets aggregate reception
   quality feedback and forward it upstream towards the distribution

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   source.  The distribution source forwards (possibly aggregated and
   summarised) 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  |<----+     +----->| 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, and 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
   authenticate the media sources to prevent unwanted traffic being
   distributed via the SSM group.  Similarly, the receivers need to

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   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

3.  Security Options

   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 designer by
   reviewing the types of solution that are available.  This section
   will use a number of different security related terms, described in
   the Internet Security Glossary, Version 2 [RFC4949].

3.1.  Secure RTP

   The Secure RTP (SRTP) protocol [RFC3711] 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 SSRC fields, but protects
   the sensitive content.

   SRTP works with transforms where different combinations of encryption
   algorithm, authentication algorithm, and pseudo-random 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, SDP
   Security Descriptions (see Section 3.1.3) and DTLS-SRTP (see
   Section 3.1.1) use pre-defined 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 MIKEY protocol (see Section 3.1.2)
   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:

   AES-CM and HMAC-SHA-1:  AES Counter Mode encryption with 128-bit keys
      combined with 160-bit keyed HMAC-SHA-1 with 80-bit authentication
      tag.  This is the default cryptographic transform that needs to be

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      supported.  The transforms are defined in SRTP [RFC3711], with the
      corresponding SRTP crypto suite in [RFC4568] and SRTP protection
      profile in [RFC5764].

   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 [RFC3711], with the corresponding SRTP
      crypto suite in [RFC4568].  The corresponding SRTP protection
      profile is not defined.

   SEED:  A Korean national standard cryptographic transform that is
      defined to be used with SRTP in [RFC5669].  Three options are
      defined, one using SHA-1 authentication, one using Counter mode
      with CBC-MAC, and finally one using Galois Counter mode.

   ARIA:  A Korean block cipher [I-D.ietf-avtcore-aria-srtp], 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
      bits 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 [RFC3711]
      and [RFC6188], with the SRTP crypto suites in [RFC6188].

   AES-GCM and AES-CCM:  AES Galois Counter Mode and AES Counter 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 [I-D.ietf-avtcore-srtp-aes-gcm].

   NULL:  SRTP [RFC3711] 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 [RFC5764].

   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
   sources.  TESLA [RFC4383] 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 need for buffering the packets for a
   while before authenticity can be verified.

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   [RFC4771] 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 secure RTP group, for example a broadcast session.

   RTP header extensions are normally carried in the clear and only
   integrity protected in SRTP.  This can be problematic in some cases,
   so [RFC6904] defines an extension to also encrypt selected header

   SRTP is specified and deployed in a number of RTP usage contexts;
   Significant support in SIP-established VoIP clients including IMS;
   RTSP [I-D.ietf-mmusic-rfc2326bis] 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, and
   instead relies on an external key management protocol.  There are
   several protocols that can be used.  The following sections outline
   some popular schemes.

3.1.1.  Key Management for SRTP: DTLS-SRTP

   A Datagram Transport Layer Security extension exists for establishing
   SRTP keys [RFC5763][RFC5764].  This extension provides secure key-
   exchange between two peers, enabling perfect forward secrecy and
   binding strong identity verification to an end-point.  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.  TLS resumption can be used when
   establishing additional media streams with the same peer, and reduces
   the set-up time to one RTT for these streams (see [RFC5764] 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 end-points of the hand-shake.  For
   example some cipher suits provide perfect forward secrecy (PFS),
   while other 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 end-point.  One choice can be to
   rely on the certificates and use a PKI to verify them to make an

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   identity assertion.  However, this is not the most common way,
   instead self-signed certificate are common to use, and instead
   establish trust through signalling or other third party solutions.

   DTLS-SRTP key management can use the signalling protocol in four
   ways.  First, to agree on using DTLS-SRTP for media security.
   Secondly, 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.
   Thirdly, to exchange hashes of each side's certificates to bind these
   to the signalling, and ensure there is no man-in-the-middle attack.
   This assumes that one can trust the signalling solution to be
   resistant to modification, and not be in collaboration with an
   attacker.  Finally to provide an assertable identity, e.g.  [RFC4474]
   that can be used to prevent modification of the signalling and the
   exchange of certificate hashes.  That way enabling binding between
   the key-exchange and the signalling.

   This usage is well defined for SIP/SDP in [RFC5763], and in most
   cases can be adopted for use with other bi-directional signalling
   solutions.  It is to be noted that there is work underway to revisit
   the SIP Identity mechanism [RFC4474] 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 man-in-the-middle
   attacks.  This do requires trust in some DTLS-SRTP external party,
   either a PKI, a signalling system or some identity provider.

   DTLS-SRTP usage is clearly on the rise.  It is mandatory to support
   in WebRTC.  It has growing support among SIP end-points.  DTLS-SRTP
   was developed in IETF primarily to meet security requirements for

3.1.2.  Key Management for SRTP: MIKEY

   Multimedia Internet Keying (MIKEY) [RFC3830] 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 useable 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.

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   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:  [RFC4650] defines a variant of
      the Diffie-Hellman exchange that uses a pre-shared key in a keyed
      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 [RFC4738] 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 to specify the session keying
      material depending on use case.  Usage of this mode requires one
      round-trip time.

   TICKET:  [RFC6043] 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:  [RFC6267] uses a key management services (KMS) infrastructure
      but with lower demand on the KMS.  Claims to provides both perfect
      forward and backwards secrecy, the exact meaning is unclear (See
      Perfect Forward Secrecy in [RFC4949]).

   SAKKE:  [RFC6509] provides Sakai-Kasahara Key Encryption in MIKEY.
      Based on Identity based Public Key Cryptography and a KMS
      infrastructure to establish a shared secret value and certificate

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      less signatures to provide source authentication.  Its features
      include simplex transmission, scalability, low-latency call set-
      up, and support for secure deferred delivery.

   MIKEY messages have several different transports.  [RFC4567] defines
   how MIKEY messages can be embedded in general SDP for usage with the
   signalling protocols SIP, SAP and RTSP.  There also exist a 3GPP
   defined usage of MIKEY that sends MIKEY messages directly over UDP
   [T3GPP.33.246] to key the receivers of Multimedia Broadcast and
   Multicast Service (MBMS) [T3GPP.26.346].  [RFC3830] 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 ones solution and based on that evaluate which modes that
   are candidates for ones usage.  More information on the applicability
   of the different MIKEY modes can be found in [RFC5197].

   MIKEY with pre-shared keys are used by 3GPP MBMS [T3GPP.33.246] and
   IMS media security [T3GPP.33.328] specifies the use of the TICKET
   mode transported over SIP and HTTP.  RTSP 2.0
   [I-D.ietf-mmusic-rfc2326bis] specifies use of the RSA-R mode.  There
   are some SIP end-points that support MIKEY.  The modes they use are
   unknown to the authors.

3.1.3.  Key Management for SRTP: Security Descriptions

   [RFC4568] provides a keying solution based on sending plain text keys
   in SDP [RFC4566].  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
   issues since it's difficult to guarantee unique SSRCs (as needed to
   avoid a "two-time pad" attack - see Section 9 of [RFC3711]).

   Since keys are transported in plain text 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, where instead hop-by-hop security is provided
   between signalling nodes using TLS.  This leaves the keying material
   sensitive to capture by the traversed signalling nodes.  Thus, in
   most cases, the security properties of security descriptions are
   weak.  The usage of security descriptions usually requires additional
   security measures, e.g.  the signalling nodes be 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.

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   Security Descriptions is the most commonly deployed keying solution
   for SIP-based end-points, where almost all end-points that support
   SRTP also support Security Descriptions.  It is also used for access
   protection in IMS Media Security [T3GPP.33.328].

3.1.4.  Key Management for SRTP: Encrypted Key Transport

   Encrypted Key Transport (EKT) [I-D.ietf-avtcore-srtp-ekt] 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 can also be used in sessions where end-
   points 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 on the bridge or
   gateway.  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 a 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 are in some Cisco video
   conferencing products.

3.1.5.  Key Management for SRTP: Other systems

   The ZRTP [RFC6189] key-management system for SRTP was proposed as an
   alternative to DTLS-SRTP.  ZRTP provides best effort encryption
   independent of the signalling 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.  Commercial implementations exist.

   Additional proprietary solutions are also known to exist.

3.2.  RTP Legacy Confidentiality

   Section 9 of the RTP standard [RFC3550] defines a DES or 3DES based
   encryption of RTP and RTCP packets.  This mechanism is keyed using
   plain text keys in SDP [RFC4566] using the "k=" SDP field.  This
   method can provide confidentiality but, as discussed in Section 9 of
   [RFC3550], it has extremely weak security properties and is not to be

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3.3.  IPsec

   IPsec [RFC4301] 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 end-point leaves the
   traffic vulnerable to attack between the VPN termination node and the
   end-point.  Thus usage of IPsec requires careful thought and design
   of its usage so that it meets the security goals.  A important
   question is how one ensures the IPsec terminating peer and the
   ultimate destination are the same.  Applications can have issues
   using existing APIs with determining if IPsec is being used or not,
   and when used who the authenticated peer entity is.

   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.

3.4.  RTP over TLS over TCP

   Just as RTP can be sent over TCP [RFC4571], it can also be sent over
   TLS over TCP [RFC4572], 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 multi-party scenarios using a central node for media
   distribution, the security provide 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 [RFC2326] and 2.0 [I-D.ietf-mmusic-rfc2326bis] specifies 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 uses TLS to bypass firewalls.

3.5.  RTP over Datagram TLS (DTLS)

   Datagram Transport Layer Security (DTLS) [RFC6347] is a based on TLS
   [RFC5246], but designed to work over a 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 an

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   datagram transport such as UDP.  The two peers establish an 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 to 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 signalling-side agreement to communicate to the certificates
   used by the end-point 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.

3.6.  Media Content Security/Digital Rights Management

   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 end-point (e.g., some types of centralised 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 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

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   second layer with additional transport security ensuring source
   authentication and integrity of the RTP headers associated with the
   encrypted payloads.  This can also results in the need to have two
   different key-management systems as the entity protecting the packets
   and payloads are different with different set of keys.

   The aspect of two tiers of security are present in ISMACryp (see
   Section 3.6.1) and the deprecated 3GPP Packet Based Streaming Service
   Annex.K [T3GPP.26.234R8] solution.

3.6.1.  ISMA Encryption and Authentication

   The Internet Streaming Media Alliance (ISMA) has defined ISMA
   Encryption and Authentication 2.0 [ISMACryp2].  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 [RFC3640].  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.

   Key-management for a ISMACryp based system can be achieved through
   Open Mobile Alliance (OMA) Digital Rights Management 2.0 [OMADRMv2],
   for example.

4.  Securing RTP Applications

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

4.1.  Application Requirements

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

4.1.1.  Confidentiality

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

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   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 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 bit-rate speech codecs that produce
      different sized packets based on the speech input [RFC6562].
      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.

   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, 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).

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4.1.2.  Integrity

   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 considers 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, and 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.

4.1.3.  Source Authentication

   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 to not be circumvented in the time frame when you would

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      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 multi-party 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 signalling:  A step up in the assertion
      that can be done in base-level systems is to tie the signalling 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 that the
      receiver established the session with.  How feasible this is
      depends on the properties of the key-management system, the
      ability to tie the signalling to a particular source, and the
      degree of trust the receiver places on the different nodes

      For example, systems where the key-exchange is done using the
      signalling systems, such as Security Descriptions [RFC4568],
      enable a direct binding between signalling and key-exchange.  In
      such systems, the actual security depends on the trust one can
      place in the signalling system to correctly associate the peer's
      identity with the key-exchange.

   Using Identities:  If the applications have access to a system that
      can provide verifiable identities, then the source authentication
      can be bound to that identity.  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
      identity, can provide a strong assertion about the source of the
      traffic.  SIP identity [RFC4474] provides one example of how this
      can be done, and could be used to bind DTLS-SRTP certificates 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 identity.  If the signalling 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 identity
      of the media sender.  However, using an signalling authentication
      mechanism built on a group key can limit the media layer to
      asserting only group membership.

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4.1.4.  Identity

   There exist many different types of identity systems with different
   properties (e.g., SIP identity [RFC4474]).  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 identities.  What an identity really is can also vary
   but, in the context of communication, one of the most obvious is the
   identity of the human user one communicates with.  However, the human
   user can also have additional identities in a particular role.  For
   example, the human Alice, can also be a police officer and in some
   cases her identity as police officer will be more relevant then that
   she is Alice.  This is common in contact with organizations, where it
   is important to prove the persons right to represent the
   organization.  Some examples of identity mechanisms that can be used:

   Certificate based:  A certificate is used to prove the identity, by
      having access to the private part of the certificate one can
      perform signing to assert ones 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 identity certificate.
      However, the trust is based on all steps in the certificate chain
      being verifiable and trusted.  Thus 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 identity, then perform
      assertions on their behalf or provision the requester with short-
      term credentials to assert their identity.  The verifier can then
      contact the IdP to request verification of a particular identity.
      Here the trust is highly dependent on how much one trusts the IdP.
      The system also becomes dependent on having access to the relevant

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

4.1.5.  Privacy

   RTP applications need to consider what privacy goals they have.  As
   RTP applications communicate directly between peers in many cases,

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   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 end-point.  The
   main way of avoid such concerns is the introduction of relay (e.g., a
   TURN server [RFC5766]) or centralized media mixers or forwarders that
   hides the address of a peer from any other peer.  The security and
   trust placed in these relays obviously needs to be carefully

   RTP itself can contribute to enabling a particular user to be tracked
   between communication sessions if the 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 clearly has privacy implications.  Instead
   cryptographically random ones could be used as defined by Guidelines
   for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)

   If there exist privacy goals, these 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.

4.2.  Application Structure

   When it comes to RTP security, the most appropriate solution is often
   highly dependent on the topology of the communication session.  The
   signalling 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 have different requirements on underlying
   infrastructure it is important to take that aspect into consideration
   early in the design.

4.3.  Interoperability

   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 of determining
   what security level is actually negotiated in a particular session.

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   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

5.  Examples

   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.

5.1.  Media Security for SIP-established Sessions using DTLS-SRTP

   The IETF evaluated media security for RTP sessions established using
   point-to-point SIP sessions in 2009.  A number of requirements were
   determined, and based on those, the existing solutions for media
   security and especially the keying methods were analysed.  The
   resulting requirements and analysis were published in [RFC5479].
   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 the
   Framework for Establishing a Secure Real-time Transport Protocol
   (SRTP) Security Context Using Datagram Transport Layer Security
   (DTLS) [RFC5763].  On a high level the framework uses SIP with SDP
   offer/answer procedures to exchange the network addresses where the
   server end-point will have a DTLS-SRTP enable server running.  The
   SIP signalling is also used to exchange the fingerprints of the
   certificate each end-point will use in the DTLS establishment
   process.  When the signalling 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

   DTLS has a number of good security properties.  For example, to
   enable a man in the middle someone in the signalling path needs to
   perform an active action and modify both the signalling message and
   the DTLS handshake.  There also exists solutions that enables the
   fingerprints to be bound to identities.  SIP Identity provides an
   identity established by the first proxy for each user [RFC4474].
   This reduces the number of nodes the connecting user User Agent has
   to trust to include just the first hop proxy, rather than the full

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   signalling path.  The biggest security weakness of this system is its
   dependency on the signalling.  SIP signalling passes multiple nodes
   and there is usually no message security deployed, only hop-by-hop
   transport security, if any, between the nodes.

5.2.  Media Security for WebRTC Sessions

   Web Real-Time Communication (WebRTC) [I-D.ietf-rtcweb-overview] is a
   solution providing JavaScript web applications with real-time media
   directly between browsers.  Media is transported using RTP protected
   using a mandatory application of SRTP [RFC3711], with keying done
   using DTLS-SRTP [RFC5764].  The security configuration is further
   defined in the WebRTC Security Architecture

   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
   verified.  An approach identified in the WebRTC security architecture
   [I-D.ietf-rtcweb-security-arch] 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 [RFC6189]), 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 ICE negotiation [RFC5245] provides IP
      addresses and ports for the browser, this leaks location
      information in the signalling 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 re-used between session
      instances.  Thus to prevent tracking, such information is ought

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      not be re-used between sessions, or the information ought not 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

   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.

5.3.  IP Multimedia Subsystem (IMS) Media Security

   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
   services such as voice mail, interworking with legacy systems (PSTN,
   GSM, and 3G), and transcoding.  End-points are authenticated during
   the SIP registration using either IMS-AKA (using SIM credentials) or
   SIP Digest (using password).

   In IMS media security [T3GPP.33.328], 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 [RFC4568] is considered to give good protection against
   eavesdropping over the accesses that are not already encrypted (GSM,
   3G, 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 end-point.

   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 [RFC5763]
   instead of security descriptions.

5.4.  3GPP Packet Based Streaming Service (PSS)

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   The 3GPP Release 11 PSS specification of the Packet Based Streaming
   Service (PSS) [T3GPP.26.234R11] 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.  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 RTSP 1.0 [RFC2326] controlled media streaming over RTP.  Thus
   an RTSP client whose user wants to access a protected content will
   request a session description (SDP [RFC4566]) for the protected
   content.  This SDP will indicate that the media is ISMACryp 2.0
   [ISMACryp2] protected media encoding application units (AUs).  The
   key(s) used to protect the media are provided in either 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
   [OMADRMv2] will be used to retrieve the key.  If multiple keys are to
   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" section of the OMA
   Mobile Broadcast Services [OMABCAST].

   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 has no access to unprotected content.  Instead all
   that want to access the media is 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 signalling, 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.

5.5.  RTSP 2.0

   Real-time Streaming Protocol 2.0 [I-D.ietf-mmusic-rfc2326bis] offers
   an interesting comparison to the PSS service (Section 5.4) 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 mandatory to

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   implement security mechanism.  As it specifies how one transport
   media over RTP it is also defining security mechanisms for the RTP
   transported media streams.

   The security goals 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
   signalling also has to be protected, requiring the use of TLS between
   the client and server.

   Using TLS-protected signalling 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
   enables the reuse of the RTSP servers 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.

6.  IANA Considerations

   This document makes no request of IANA.

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

7.  Security Considerations

   This entire document is about security.  Please read it.

8.  Acknowledgements

   We thank the IESG for their careful review of
   [I-D.ietf-avt-srtp-not-mandatory] which led to the writing of this
   memo.  John Mattsson has contributed the IMS Media Security example
   (Section 5.3).

   The authors wished to thank Christian Correll, Dan Wing, Kevin Gross,
   Alan Johnston, Michael Peck, Ole Jacobsen, and John Mattsson for
   review and proposals for improvements of the text.

9.  Informative References

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              Perkins, C. and M. Westerlund, "Securing the RTP Protocol
              Framework: Why RTP Does Not Mandate a Single Media
              Security Solution", draft-ietf-avt-srtp-not-mandatory-14
              (work in progress), October 2013.

              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)", draft-ietf-avtcore-aria-srtp-05
              (work in progress), September 2013.

              McGrew, D. and K. Igoe, "AES-GCM and AES-CCM Authenticated
              Encryption in Secure RTP (SRTP)", draft-ietf-avtcore-srtp-
              aes-gcm-10 (work in progress), September 2013.

              McGrew, D., Wing, D., and K. Fischer, "Encrypted Key
              Transport for Secure RTP", draft-ietf-avtcore-srtp-ekt-00
              (work in progress), July 2012.

              Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
              and M. Stiemerling, "Real Time Streaming Protocol 2.0
              (RTSP)", draft-ietf-mmusic-rfc2326bis-38 (work in
              progress), October 2013.

              Alvestrand, H., "Overview: Real Time Protocols for Brower-
              based Applications", draft-ietf-rtcweb-overview-08 (work
              in progress), September 2013.

              Rescorla, E., "WebRTC Security Architecture", draft-ietf-
              rtcweb-security-arch-07 (work in progress), July 2013.

              Internet Streaming Media Alliance (ISMA), "ISMA Encryption
              and Authentication, Version 2.0 release version", November

              Open Mobile Alliance, "OMA Mobile Broadcast Services
              V1.0", February 2009.


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              Open Mobile Alliance, "OMA Digital Rights Management
              V2.0", July 2008.

   [RFC1112]  Deering, S., "Host extensions for IP multicasting", STD 5,
              RFC 1112, August 1989.

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

   [RFC3365]  Schiller, J., "Strong Security Requirements for Internet
              Engineering Task Force Standard Protocols", BCP 61, RFC
              3365, August 2002.

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

   [RFC3640]  van der Meer, J., Mackie, D., Swaminathan, V., Singer, D.,
              and P. Gentric, "RTP Payload Format for Transport of
              MPEG-4 Elementary Streams", RFC 3640, November 2003.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [RFC3830]  Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
              Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
              August 2004.

   [RFC4301]  Kent, S. and K. Seo, "Security Architecture for the
              Internet Protocol", RFC 4301, December 2005.

   [RFC4383]  Baugher, M. and E. Carrara, "The Use of Timed Efficient
              Stream Loss-Tolerant Authentication (TESLA) in the Secure
              Real-time Transport Protocol (SRTP)", RFC 4383, February

   [RFC4474]  Peterson, J. and C. Jennings, "Enhancements for
              Authenticated Identity Management in the Session
              Initiation Protocol (SIP)", RFC 4474, August 2006.

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

   [RFC4567]  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)", RFC 4567, July 2006.

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   [RFC4568]  Andreasen, F., Baugher, M., and D. Wing, "Session
              Description Protocol (SDP) Security Descriptions for Media
              Streams", RFC 4568, July 2006.

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

   [RFC4572]  Lennox, J., "Connection-Oriented Media Transport over the
              Transport Layer Security (TLS) Protocol in the Session
              Description Protocol (SDP)", RFC 4572, July 2006.

   [RFC4607]  Holbrook, H. and B. Cain, "Source-Specific Multicast for
              IP", RFC 4607, August 2006.

   [RFC4650]  Euchner, M., "HMAC-Authenticated Diffie-Hellman for
              Multimedia Internet KEYing (MIKEY)", RFC 4650, September

   [RFC4738]  Ignjatic, D., Dondeti, L., Audet, F., and P. Lin, "MIKEY-
              RSA-R: An Additional Mode of Key Distribution in
              Multimedia Internet KEYing (MIKEY)", RFC 4738, November

   [RFC4771]  Lehtovirta, V., Naslund, M., and K. Norrman, "Integrity
              Transform Carrying Roll-Over Counter for the Secure Real-
              time Transport Protocol (SRTP)", RFC 4771, January 2007.

   [RFC4949]  Shirey, R., "Internet Security Glossary, Version 2", RFC
              4949, August 2007.

   [RFC5117]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
              January 2008.

   [RFC5197]  Fries, S. and D. Ignjatic, "On the Applicability of
              Various Multimedia Internet KEYing (MIKEY) Modes and
              Extensions", RFC 5197, June 2008.

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

   [RFC5246]  Dierks, T. and E. Rescorla, "The Transport Layer Security
              (TLS) Protocol Version 1.2", RFC 5246, August 2008.

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   [RFC5479]  Wing, D., Fries, S., Tschofenig, H., and F. Audet,
              "Requirements and Analysis of Media Security Management
              Protocols", RFC 5479, April 2009.

   [RFC5669]  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)", RFC 5669, August 2010.

   [RFC5760]  Ott, J., Chesterfield, J., and E. Schooler, "RTP Control
              Protocol (RTCP) Extensions for Single-Source Multicast
              Sessions with Unicast Feedback", RFC 5760, February 2010.

   [RFC5763]  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)", RFC 5763, May 2010.

   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for the Secure
              Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.

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

   [RFC6043]  Mattsson, J. and T. Tian, "MIKEY-TICKET: Ticket-Based
              Modes of Key Distribution in Multimedia Internet KEYing
              (MIKEY)", RFC 6043, March 2011.

   [RFC6188]  McGrew, D., "The Use of AES-192 and AES-256 in Secure
              RTP", RFC 6188, March 2011.

   [RFC6189]  Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media
              Path Key Agreement for Unicast Secure RTP", RFC 6189,
              April 2011.

   [RFC6267]  Cakulev, V. and G. Sundaram, "MIKEY-IBAKE: Identity-Based
              Authenticated Key Exchange (IBAKE) Mode of Key
              Distribution in Multimedia Internet KEYing (MIKEY)", RFC
              6267, June 2011.

   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6347, January 2012.

   [RFC6509]  Groves, M., "MIKEY-SAKKE: Sakai-Kasahara Key Encryption in
              Multimedia Internet KEYing (MIKEY)", RFC 6509, February

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   [RFC6562]  Perkins, C. and JM. Valin, "Guidelines for the Use of
              Variable Bit Rate Audio with Secure RTP", RFC 6562, March

   [RFC6904]  Lennox, J., "Encryption of Header Extensions in the Secure
              Real-time Transport Protocol (SRTP)", RFC 6904, April

   [RFC7022]  Begen, A., Perkins, C., Wing, D., and E. Rescorla,
              "Guidelines for Choosing RTP Control Protocol (RTCP)
              Canonical Names (CNAMEs)", RFC 7022, September 2013.

              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.

              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.

              3GPP, "Multimedia Broadcast/Multicast Service (MBMS);
              Protocols and codecs", 3GPP TS 26.346 10.7.0, March 2013.

              3GPP, "3G Security; Security of Multimedia Broadcast/
              Multicast Service (MBMS)", 3GPP TS 33.246 12.1.0, December

              3GPP, "IP Multimedia Subsystem (IMS) media plane
              security", 3GPP TS 33.328 12.1.0, December 2012.

Authors' Addresses

   Magnus Westerlund
   Farogatan 6
   SE-164 80 Kista

   Phone: +46 10 714 82 87

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


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