| Rfc | 7201 |
|---|
| Title | Options for Securing RTP Sessions |
|---|
| Author | M. Westerlund, C. Perkins |
|---|
| Date | April
2014 |
|---|
| Format: | TXT, HTML |
|---|
| Status: | INFORMATIONAL |
|---|
|
Internet Engineering Task Force (IETF) M. Westerlund
Request for Comments: 7201 Ericsson
Category: Informational C. Perkins
ISSN: 2070-1721 University of Glasgow
April 2014
Options for Securing RTP Sessions
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 Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/rfc7201.
Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Background . . . . . . . . . . . . . . . . . . . . . . . . . 5
2.1. Point-to-Point Sessions . . . . . . . . . . . . . . . . . 5
2.2. Sessions Using an RTP Mixer . . . . . . . . . . . . . . . 5
2.3. Sessions Using an RTP Translator . . . . . . . . . . . . 6
2.3.1. Transport Translator (Relay) . . . . . . . . . . . . 6
2.3.2. Gateway . . . . . . . . . . . . . . . . . . . . . . . 7
2.3.3. Media Transcoder . . . . . . . . . . . . . . . . . . 8
2.4. Any Source Multicast . . . . . . . . . . . . . . . . . . 8
2.5. Source-Specific Multicast . . . . . . . . . . . . . . . . 8
3. Security Options . . . . . . . . . . . . . . . . . . . . . . 10
3.1. Secure RTP . . . . . . . . . . . . . . . . . . . . . . . 10
3.1.1. Key Management for SRTP: DTLS-SRTP . . . . . . . . . 12
3.1.2. Key Management for SRTP: MIKEY . . . . . . . . . . . 14
3.1.3. Key Management for SRTP: Security Descriptions . . . 15
3.1.4. Key Management for SRTP: Encrypted Key Transport . . 16
3.1.5. Key Management for SRTP: ZRTP and Other Solutions . . 17
3.2. RTP Legacy Confidentiality . . . . . . . . . . . . . . . 17
3.3. IPsec . . . . . . . . . . . . . . . . . . . . . . . . . . 17
3.4. RTP over TLS over TCP . . . . . . . . . . . . . . . . . . 18
3.5. RTP over Datagram TLS (DTLS) . . . . . . . . . . . . . . 18
3.6. Media Content Security/Digital Rights Management . . . . 19
3.6.1. ISMA Encryption and Authentication . . . . . . . . . 19
4. Securing RTP Applications . . . . . . . . . . . . . . . . . . 20
4.1. Application Requirements . . . . . . . . . . . . . . . . 20
4.1.1. Confidentiality . . . . . . . . . . . . . . . . . . . 20
4.1.2. Integrity . . . . . . . . . . . . . . . . . . . . . . 21
4.1.3. Source Authentication . . . . . . . . . . . . . . . . 22
4.1.4. Identifiers and Identity . . . . . . . . . . . . . . 23
4.1.5. Privacy . . . . . . . . . . . . . . . . . . . . . . . 24
4.2. Application Structure . . . . . . . . . . . . . . . . . . 25
4.3. Automatic Key Management . . . . . . . . . . . . . . . . 25
4.4. End-to-End Security vs. Tunnels . . . . . . . . . . . . . 25
4.5. Plaintext Keys . . . . . . . . . . . . . . . . . . . . . 26
4.6. Interoperability . . . . . . . . . . . . . . . . . . . . 26
5. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 26
5.1. Media Security for SIP-Established Sessions Using
DTLS-SRTP . . . . . . . . . . . . . . . . . . . . . . . . 27
5.2. Media Security for WebRTC Sessions . . . . . . . . . . . 27
5.3. IP Multimedia Subsystem (IMS) Media Security . . . . . . 28
5.4. 3GPP Packet-Switched Streaming Service (PSS) . . . . . . 29
5.5. RTSP 2.0 . . . . . . . . . . . . . . . . . . . . . . . . 30
6. Security Considerations . . . . . . . . . . . . . . . . . . . 31
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 31
8. Informative References . . . . . . . . . . . . . . . . . . . 31
1. Introduction
The 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 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 [RFC3365]. Because of the
heterogeneity of RTP applications and use cases, however, a single
security solution cannot be mandated [RFC7202]. 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. 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. Section 4 provides guidelines
and important aspects to consider when securing an RTP application.
Finally, in Section 5, we give some examples of application domains
where guidelines for security exist.
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 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 |<------->| B |
+---+ +---+
Figure 1: Point-to-Point Topology
2.2. Sessions Using an RTP Mixer
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.
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 |<---->| |<---->| 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 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.
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 and discuss three different
categories of RTP translators: transport translators, gateways, and
media transcoders.
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 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
participants unchanged. The resulting topology is very similar to an
Any Source Multicast (ASM) session (as discussed in Section 2.4) but
is 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,
thus, 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 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.
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 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.
2.5. Source-Specific Multicast
Source-Specific Multicast (SSM) [RFC4607] 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 [RFC5760].
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
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 |<----+ +----->| 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
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.
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 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 [RFC4949].
3.1. Secure RTP
The Secure Real-time Transport Protocol (SRTP) [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 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 Section 3.1.3)
and Datagram Transport Layer Security Extension for SRTP (DTLS-SRTP)
(see Section 3.1.1) 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 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 an 80-bit
authentication tag. This is the default cryptographic transform
that needs to be supported. The transforms are defined in SRTP
[RFC3711], with the corresponding SRTP crypto suite defined in
[RFC4568] and SRTP protection profile defined 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 defined 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 Cipher Block Chaining Message Authentication Code (CBC-MAC),
and one using Galois Counter Mode.
ARIA: A Korean block cipher [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-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 [RFC3711]
and [RFC6188], and the SRTP crypto suites are defined in
[RFC6188].
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 [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. Timed Efficient Stream Loss-Tolerant Authentication (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 the need for buffering the packets for a while before
authenticity can be verified.
[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 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 [RFC6904] 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) [RTSP] 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.
3.1.1. Key Management for SRTP: DTLS-SRTP
A Datagram Transport Layer Security (DTLS) extension exists for
establishing SRTP keys [RFC5763][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
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 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., [RFC4474], 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 [RFC5763] 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 [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 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" [RFC5479].
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 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: [RFC4650] 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 [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 specify the session keying
material depending on the use case. Usage of this mode requires
one round-trip time.
TICKET: Ticket Payload (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: Identity-Based Authenticated Key Exchange (IBAKE) [RFC6267]
uses a KMS infrastructure but with lower demand on the KMS. It
claims to provide both perfect forward and backwards secrecy.
SAKKE: [RFC6509] 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. [RFC4567] 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 [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 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 [RFC5197].
MIKEY with pre-shared keys is 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 [RTSP] specifies use of
the RSA-R mode. There are some SIP endpoints 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 plaintext 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 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 [T3GPP.33.328].
3.1.4. Key Management for SRTP: Encrypted Key Transport
Encrypted Key Transport (EKT) [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 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.
3.1.5. Key Management for SRTP: ZRTP and Other Solutions
The ZRTP [RFC6189] 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.
3.2. RTP Legacy Confidentiality
Section 9 of the RTP standard [RFC3550] defines a Data Encryption
Standard (DES) or 3DES-based encryption of RTP and RTCP packets.
This mechanism is keyed using plaintext 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 used.
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 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.
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 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 [RFC2326] and 2.0 [RTSP] 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.
3.5. RTP over Datagram TLS (DTLS)
DTLS [RFC6347] is based on TLS [RFC5246] 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.
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 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
Section 3.6.1) and the deprecated 3GPP Packet-switched Streaming
Service solution; see Annex K of [T3GPP.26.234R8].
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 an 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
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.
4.1.1. Confidentiality
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 [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 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).
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 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.
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 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 [RFC4568] 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.
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 [RFC4474] 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.
4.1.4. Identifiers and Identity
There exist many different types of systems providing identifiers
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 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
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.
4.1.5. Privacy
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 [RFC5766]) 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" [RFC7022].
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.
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
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.
4.3. Automatic Key Management
The guidelines for Cryptographic Key Management [RFC4107] 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.
4.4. End-to-End Security vs. Tunnels
If the security mechanism only provides a secured tunnel, for
example, like some common uses of IPsec (Section 3.3), 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
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.
4.5. Plaintext Keys
Key management solutions that use plaintext keys, like SDP security
descriptions (Section 3.1.3), 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.
4.6. 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 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.
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
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
[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
"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 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 [RFC4474]. 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.
5.2. Media Security for WebRTC Sessions
Web Real-Time Communication (WebRTC) [WebRTC] 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 [RFC3711], with keying done using DTLS-
SRTP [RFC5764]. The security configuration is further defined in
"WebRTC Security Architecture" [WebRTC-SEC].
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
[WebRTC-SEC] 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 Interactive Connectivity Establishment (ICE)
negotiation [RFC5245] 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.
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 (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 [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, 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 [RFC5763]
instead of security descriptions.
5.4. 3GPP Packet-Switched Streaming Service (PSS)
The 3GPP Release 11 PSS specification of the Packet-switched
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 (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 [RFC2326] 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 protected by
ISMACryp 2.0 [ISMACryp2] 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
[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" part [OMASCP] 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 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.
5.5. RTSP 2.0
The Real-time Streaming Protocol 2.0 [RTSP] 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 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
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.
6. Security Considerations
This entire document is about security. Please read it.
7. Acknowledgements
We thank the IESG for their careful review of [RFC7202], which led to
the writing of this memo. John Mattsson has contributed the IMS
Media Security example (Section 5.3).
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.
8. Informative References
[AES-GCM] McGrew, D. and K. Igoe, "AES-GCM and AES-CCM
Authenticated Encryption in Secure RTP (SRTP)", Work in
Progress, September 2013.
[ARIA-SRTP] 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.
[EKT] McGrew, D. and D. Wing, "Encrypted Key Transport for
Secure RTP", Work in Progress, February 2014.
[ISMACryp2] Internet Streaming Media Alliance (ISMA), "ISMA
Encryption and Authentication Version 2.0", November
2007, <http://www.oipf.tv/images/site/DOCS/mpegif/ISMA/
isma_easpec2.0.pdf>.
[OMABCAST] Open Mobile Alliance, "Mobile Broadcast Services Version
1.0", February 2009,
<http://technical.openmobilealliance.org/Technical/
release_program/bcast_v1_0.aspx>.
[OMADRMv2] Open Mobile Alliance, "OMA Digital Rights Management
V2.0", July 2008,
<http://technical.openmobilealliance.org/
Technical/release_program/drm_v2_0.aspx>.
[OMASCP] Open Mobile Alliance, "Service and Content Protection for
Mobile Broadcast Services", January 2013,
<http://technical.openmobilealliance.org/Technical/
release_program/docs/BCAST/V1_0_1-20130109-A/
OMA-TS-BCAST_SvcCntProtection-V1_0_1-20130109-A.pdf>.
[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.
[RFC4107] Bellovin, S. and R. Housley, "Guidelines for
Cryptographic Key Management", BCP 107, RFC 4107, June
2005.
[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
2006.
[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.
[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
2006.
[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
2006.
[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
2010.
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security
(TLS) Protocol Version 1.2", RFC 5246, August 2008.
[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 2012.
[RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of
Variable Bit Rate Audio with Secure RTP", RFC 6562, March
2012.
[RFC6904] Lennox, J., "Encryption of Header Extensions in the
Secure Real-time Transport Protocol (SRTP)", RFC 6904,
April 2013.
[RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla,
"Guidelines for Choosing RTP Control Protocol (RTCP)
Canonical Names (CNAMEs)", RFC 7022, September 2013.
[RFC7202] Perkins, C. and M. Westerlund, "Securing the RTP Protocol
Framework: Why RTP Does Not Mandate a Single Media
Security Solution", RFC 7202, April 2014.
[RTSP] Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M.,
and M. Stiemerling, "Real Time Streaming Protocol 2.0
(RTSP)", Work in Progress, February 2014.
[T3GPP.26.234R11]
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,
<http://www.3gpp.org/DynaReport/26234.htm>.
[T3GPP.26.234R8]
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,
<http://www.3gpp.org/DynaReport/26234.htm>.
[T3GPP.26.346]
3GPP, "Multimedia Broadcast/Multicast Service (MBMS);
Protocols and codecs", 3GPP TS 26.346 10.7.0, March 2013,
<http://www.3gpp.org/DynaReport/26346.htm>.
[T3GPP.33.246]
3GPP, "3G Security; Security of Multimedia Broadcast/
Multicast Service (MBMS)", 3GPP TS 33.246 11.1.0,
December 2012,
<http://www.3gpp.org/DynaReport/33246.htm>.
[T3GPP.33.328]
3GPP, "IP Multimedia Subsystem (IMS) media plane
security", 3GPP TS 33.328 12.1.0, December 2012,
<http://www.3gpp.org/DynaReport/33328.htm>.
[WebRTC-SEC]
Rescorla, E., "WebRTC Security Architecture", Work in
Progress, February 2014.
[WebRTC] Alvestrand, H., "Overview: Real Time Protocols for
Browser-based Applications", Work in Progress, February
2014.
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: http://csperkins.org/