rfc5479
Network Working Group D. Wing, Ed.
Request for Comments: 5479 Cisco
Category: Informational S. Fries
Siemens AG
H. Tschofenig
Nokia Siemens Networks
F. Audet
Nortel
April 2009
Requirements and Analysis of Media Security Management Protocols
Status of This Memo
This memo provides information for the Internet community. It does
not specify an Internet standard of any kind. Distribution of this
memo is unlimited.
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Abstract
This document describes requirements for a protocol to negotiate a
security context for SIP-signaled Secure RTP (SRTP) media. In
addition to the natural security requirements, this negotiation
protocol must interoperate well with SIP in certain ways. A number
of proposals have been published and a summary of these proposals is
in the appendix of this document.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Attack Scenarios . . . . . . . . . . . . . . . . . . . . . . . 5
4. Call Scenarios and Requirements Considerations . . . . . . . . 7
4.1. Clipping Media before Signaling Answer . . . . . . . . . . 7
4.2. Retargeting and Forking . . . . . . . . . . . . . . . . . 8
4.3. Recording . . . . . . . . . . . . . . . . . . . . . . . . 11
4.4. PSTN Gateway . . . . . . . . . . . . . . . . . . . . . . . 12
4.5. Call Setup Performance . . . . . . . . . . . . . . . . . . 12
4.6. Transcoding . . . . . . . . . . . . . . . . . . . . . . . 13
4.7. Upgrading to SRTP . . . . . . . . . . . . . . . . . . . . 13
4.8. Interworking with Other Signaling Protocols . . . . . . . 14
4.9. Certificates . . . . . . . . . . . . . . . . . . . . . . . 14
5. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 14
5.1. Key Management Protocol Requirements . . . . . . . . . . . 15
5.2. Security Requirements . . . . . . . . . . . . . . . . . . 16
5.3. Requirements outside of the Key Management Protocol . . . 19
6. Security Considerations . . . . . . . . . . . . . . . . . . . 20
7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 20
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
8.1. Normative References . . . . . . . . . . . . . . . . . . . 20
8.2. Informative References . . . . . . . . . . . . . . . . . . 21
Appendix A. Overview and Evaluation of Existing Keying
Mechanisms . . . . . . . . . . . . . . . . . . . . . 24
A.1. Signaling Path Keying Techniques . . . . . . . . . . . . . 25
A.1.1. MIKEY-NULL . . . . . . . . . . . . . . . . . . . . . . 25
A.1.2. MIKEY-PSK . . . . . . . . . . . . . . . . . . . . . . 25
A.1.3. MIKEY-RSA . . . . . . . . . . . . . . . . . . . . . . 25
A.1.4. MIKEY-RSA-R . . . . . . . . . . . . . . . . . . . . . 25
A.1.5. MIKEY-DHSIGN . . . . . . . . . . . . . . . . . . . . . 26
A.1.6. MIKEY-DHHMAC . . . . . . . . . . . . . . . . . . . . . 26
A.1.7. MIKEY-ECIES and MIKEY-ECMQV (MIKEY-ECC) . . . . . . . 26
A.1.8. SDP Security Descriptions with SIPS . . . . . . . . . 26
A.1.9. SDP Security Descriptions with S/MIME . . . . . . . . 27
A.1.10. SDP-DH (Expired) . . . . . . . . . . . . . . . . . . . 27
A.1.11. MIKEYv2 in SDP (Expired) . . . . . . . . . . . . . . . 27
A.2. Media Path Keying Technique . . . . . . . . . . . . . . . 27
A.2.1. ZRTP . . . . . . . . . . . . . . . . . . . . . . . . . 27
A.3. Signaling and Media Path Keying Techniques . . . . . . . . 28
A.3.1. EKT . . . . . . . . . . . . . . . . . . . . . . . . . 28
A.3.2. DTLS-SRTP . . . . . . . . . . . . . . . . . . . . . . 28
A.3.3. MIKEYv2 Inband (Expired) . . . . . . . . . . . . . . . 29
A.4. Evaluation Criteria - SIP . . . . . . . . . . . . . . . . 29
A.4.1. Secure Retargeting and Secure Forking . . . . . . . . 29
A.4.2. Clipping Media before SDP Answer . . . . . . . . . . . 31
A.4.3. SSRC and ROC . . . . . . . . . . . . . . . . . . . . . 33
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A.5. Evaluation Criteria - Security . . . . . . . . . . . . . . 35
A.5.1. Distribution and Validation of Persistent Public
Keys and Certificates . . . . . . . . . . . . . . . . 35
A.5.2. Perfect Forward Secrecy . . . . . . . . . . . . . . . 38
A.5.3. Best Effort Encryption . . . . . . . . . . . . . . . . 39
A.5.4. Upgrading Algorithms . . . . . . . . . . . . . . . . . 40
Appendix B. Out-of-Scope . . . . . . . . . . . . . . . . . . . . 42
B.1. Shared Key Conferencing . . . . . . . . . . . . . . . . . 42
1. Introduction
The work on media security started when the Session Initiation
Protocol (SIP) was still in its infancy. With the increased SIP
deployment and the availability of new SIP extensions and related
protocols, the need for end-to-end security was re-evaluated. The
procedure of re-evaluating prior protocol work and design decisions
is not an uncommon strategy and, to some extent, considered necessary
to ensure that the developed protocols indeed meet the previously
envisioned needs for the users on the Internet.
This document summarizes media security requirements, i.e.,
requirements for mechanisms that negotiate security context such as
cryptographic keys and parameters for SRTP.
The organization of this document is as follows: Section 2 introduces
terminology, Section 3 describes various attack scenarios against the
signaling path and media path, Section 4 provides an overview about
possible call scenarios, and Section 5 lists requirements for media
security. The main part of the document concludes with the security
considerations Section 6, and acknowledgements in Section 7.
Appendix A lists and compares available solution proposals. The
following Appendix A.4 compares the different approaches regarding
their suitability for the SIP signaling scenarios described in
Appendix A, while Appendix A.5 provides a comparison regarding
security aspects. Appendix B lists non-goals for this document.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119], with the
important qualification that, unless otherwise stated, these terms
apply to the design of the media security key management protocol,
not its implementation or application.
Furthermore, the terminology described in SIP [RFC3261] regarding
functions and components are used throughout the document.
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Additionally, the following items are used in this document:
AOR (Address-of-Record): A SIP or SIPS URI that points to a domain
with a location service that can map the URI to another URI where
the user might be available. Typically, the location service is
populated through registrations. An AOR is frequently thought of
as the "public address" of the user.
SSRC: The 32-bit value that defines the synchronization source, used
in RTP. These are generally unique, but collisions can occur.
two-time pad: The use of the same key and the same keystream to
encrypt different data. For SRTP, a two-time pad occurs if two
senders are using the same key and the same RTP SSRC value.
Perfect Forward Secrecy (PFS): The property that disclosure of the
long-term secret keying material that is used to derive an agreed
ephemeral key does not compromise the secrecy of agreed keys from
earlier runs.
active adversary: An active adversary is able to alter data
communication to affect its operation (see also [RFC4949]).
passive adversary: A passive adversary is able to learn information
from data communication, but not alter that data communication
(see also [RFC4949]).
signaling path: The signaling path is the route taken by SIP
signaling messages transmitted between the calling and called user
agents. This can be either direct signaling between the calling
and called user agents or, more commonly, involves the SIP proxy
servers that were involved in the call setup.
media path: The media path is the route taken by media packets
exchanged by the endpoints. In the simplest case, the endpoints
exchange media directly, and the "media path" is defined by a
quartet of IP addresses and TCP/UDP ports, along with an IP route.
In other cases, this path may include RTP relays, mixers,
transcoders, session border controllers, NATs, or media gateways.
Moreover, as this document discusses requirements for media security,
the nomenclature R-XXX is used to mark requirements, where XXX is the
requirement, which needs to be met.
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3. Attack Scenarios
The discussion in this section relates to requirements R-ASSOC
(specified in Section 5.1) R-PASS-MEDIA, R-PASS-SIG, R-SIG-MEDIA,
R-ACT-ACT, and R-ID-BINDING (specified in Section 5.2).
This document classifies adversaries according to their access and
their capabilities. An adversary might have access:
1. only to the media path,
2. only to the signaling path,
3. to the media path and to the signaling path.
An attacker that can solely be located along the signaling path, and
does not have access to media (item 2), is not considered in this
document.
There are two different types of adversaries: active and passive. An
active adversary may need to be active with regard to the key
exchange relevant information traveling along the media path or
traveling along the signaling path.
Based on their robustness against the adversary capabilities
described above, we can group security mechanisms using the following
labels. This list is generally ordered from easiest to compromise
(at the top) to more difficult to compromise:
+---------------+---------+--------------------------------------+
| SIP signaling | media | abbreviation |
+---------------+---------+--------------------------------------+
| none | passive | no-signaling-passive-media |
| none | active | no-signaling-active-media |
| passive | passive | passive-signaling-passive-media |
| passive | active | passive-signaling-active-media |
| active | passive | active-signaling-passive-media |
| active | active | active-signaling-active-media |
| active | active | active-signaling-active-media-detect |
+---------------+---------+--------------------------------------+
no-signaling-passive-media:
Access only to the media path is sufficient to reveal the content
of the media traffic.
passive-signaling-passive-media:
Passive attack on the signaling and passive attack on the media
path is necessary to reveal the content of the media traffic.
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passive-signaling-active-media:
Passive attack on the signaling and active attack on the media
path is necessary to reveal the content of the media traffic.
active-signaling-passive-media:
Active attack on the signaling path and passive attack on the
media path is necessary to reveal the content of the media
traffic.
no-signaling-active-media:
Active attack on the media path is sufficient to reveal the
content of the media traffic.
active-signaling-active-media:
Active attack on both the signaling path and the media path is
necessary to reveal the content of the media traffic.
active-signaling-active-media-detect:
Active attack on both signaling and media path is necessary to
reveal the content of the media traffic (as with active-signaling-
active-media), and the attack is detectable by protocol messages
exchanged between the endpoints.
For example, unencrypted RTP is vulnerable to no-signaling-passive-
media.
As another example, SDP Security Descriptions [RFC4568], when
protected by TLS (as it is commonly implemented and deployed), belong
in the passive-signaling-passive-media category since the adversary
needs to learn the SDP Security Descriptions key by seeing the SIP
signaling message at a SIP proxy (assuming that the adversary is in
control of the SIP proxy). The media traffic can be decrypted using
that learned key.
As another example, DTLS-SRTP (Datagram Transport Layer Security
Extension for SRTP) falls into active-signaling-active-media category
when DTLS-SRTP is used with a public-key-based ciphersuite with self-
signed certificates and without SIP Identity [RFC4474]. An adversary
would have to modify the fingerprint that is sent along the signaling
path and subsequently to modify the certificates carried in the DTLS
handshake that travel along the media path. If DTLS-SRTP is used
with both SIP Identity [RFC4474] and SIP Connected Identity
[RFC4916], the RFC-4474 signature protects both the offer and the
answer, and such a system would then belong to the active-signaling-
active-media-detect category (provided, of course, the signaling path
to the RFC-4474 authenticator and verifier is secured as per RFC
4474, and the RFC-4474 authenticator and verifier are behaving as per
RFC 4474).
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The above discussion of DTLS-SRTP demonstrates how a single security
protocol can be in different classes depending on the mode in which
it is operated. Other protocols can achieve a similar effect by
adding functions outside of the on-the-wire key management protocol
itself. Although it may be appropriate to deploy lower-classed
mechanisms in some cases, the ultimate security requirement for a
media security negotiation protocol is that it have a mode of
operation available in which is detect-attack, which provides
protection against the passive and active attacks and provides
detection of such attacks. That is, there must be a way to use the
protocol so that an active attack is required against both the
signaling and media paths, and so that such attacks are detectable by
the endpoints.
4. Call Scenarios and Requirements Considerations
The following subsections describe call scenarios that pose the most
challenge to the key management system for media data in cooperation
with SIP signaling.
Throughout the subsections, requirements are stated by using the
nomenclature R- to state an explicit requirement. All of the stated
requirements are explained in detail in Section 5. They are listed
according to their association to the key management protocol, to
attack scenarios, and requirements that can be met inside the key
management protocol or outside of the key management protocol.
4.1. Clipping Media before Signaling Answer
The discussion in this section relates to requirements R-AVOID-
CLIPPING and R-ALLOW-RTP.
Per the Session Description Protocol (SDP) Offer/Answer Model
[RFC3264]:
Once the offerer has sent the offer, it MUST be prepared to
receive media for any recvonly streams described by that offer.
It MUST be prepared to send and receive media for any sendrecv
streams in the offer, and send media for any sendonly streams in
the offer (of course, it cannot actually send until the peer
provides an answer with the needed address and port information).
To meet this requirement with SRTP, the offerer needs to know the
SRTP key for arriving media. If either endpoint receives encrypted
media before it has access to the associated SRTP key, it cannot play
the media -- causing clipping.
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For key exchange mechanisms that send the answerer's key in SDP, a
SIP provisional response [RFC3261], such as 183 (session progress),
is useful. However, the 183 messages are not reliable unless both
the calling and called endpoint support Provisional Response
ACKnowledgement (PRACK) [RFC3262], use TCP across all SIP proxies,
implement Security Preconditions [RFC5027], or both ends implement
Interactive Connectivity Establishment [ICE] and the answerer
implements the reliable provisional response mechanism described in
ICE. Unfortunately, there is not wide deployment of any of these
techniques and there is industry reluctance to require these
techniques to avoid the problems described in this section.
Note that the receipt of an SDP answer is not always sufficient to
allow media to be played to the offerer. Sometimes, the offerer must
send media in order to open up firewall holes or NAT bindings before
media can be received (for details, see [MIDDLEBOX]). In this case,
even a solution that makes the key available before the SDP answer
arrives will not help.
Preventing the arrival of early media (i.e., media that arrives at
the SDP offerer before the SDP answer arrives) might obsolete the
R-AVOID-CLIPPING requirement, but at the time of writing such early
media exists in many normal call scenarios.
4.2. Retargeting and Forking
The discussion in this section relates to requirements R-FORK-
RETARGET, R-DISTINCT, R-HERFP, and R-BEST-SECURE.
In SIP, a request sent to a specific AOR but delivered to a different
AOR is called a "retarget". A typical scenario is a "call
forwarding" feature. In Figure 1, Alice sends an INVITE in step 1
that is sent to Bob in step 2. Bob responds with a redirect (SIP
response code 3xx) pointing to Carol in step 3. This redirect
typically does not propagate back to Alice but only goes to a proxy
(i.e., the retargeting proxy) that sends the original INVITE to Carol
in step 4.
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+-----+
|Alice|
+--+--+
|
| INVITE (1)
V
+----+----+
| proxy |
++-+-----++
| ^ |
INVITE (2) | | | INVITE (4)
& redirect (3) | | |
V | V
++-++ ++----+
|Bob| |Carol|
+---+ +-----+
Figure 1: Retargeting
Using retargeting might lead to situations where the User Agent
Client (UAC) does not know where its request will be going. This
might not immediately seem like a serious problem; after all, when
one places a telephone call on the Public Switched Telephone Network
(PSTN), one never really knows if it will be forwarded to a different
number, who will pick up the line when it rings, and so on. However,
when considering SIP mechanisms for authenticating the called party,
this function can also make it difficult to differentiate an
intermediary that is behaving legitimately from an attacker. From
this perspective, the main problems with retargeting are:
Not detectable by the caller: The originating user agent has no
means of anticipating that the condition will arise, nor any means
of determining that it has occurred until the call has already
been set up.
Not preventable by the caller: There is no existing mechanism that
might be employed by the originating user agent in order to
guarantee that the call will not be retargeted.
The mechanism used by SIP for identifying the calling party is SIP
Identity [RFC4474]. However, due to the nature of retargeting, SIP
Identity can only identify the calling party (that is, the party that
initiated the SIP request). Some key exchange mechanisms predate SIP
Identity and include their own identity mechanism (e.g., Multimedia
Internet KEYing (MIKEY)). However, those built-in identity mechanism
also suffer from the SIP retargeting problem. While Connected
Identity [RFC4916] allows positive identification of the called
party, the primary difficulty still remains that the calling party
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does not know if a mismatched called party is legitimate (i.e., due
to authorized retargeting) or illegitimate (i.e., due to unauthorized
retargeting by an attacker above to modify SIP signaling).
In SIP, 'forking' is the delivery of a request to multiple locations.
This happens when a single AOR is registered more than once. An
example of forking is when a user has a desk phone, PC client, and
mobile handset all registered with the same AOR.
+-----+
|Alice|
+--+--+
|
| INVITE
V
+-----+-----+
| proxy |
++---------++
| |
INVITE | | INVITE
V V
+--+--+ +--+--+
|Bob-1| |Bob-2|
+-----+ +-----+
Figure 2: Forking
With forking, both Bob-1 and Bob-2 might send back SDP answers in SIP
responses. Alice will see those intermediate (18x) and final (200)
responses. It is useful for Alice to be able to associate the SIP
response with the incoming media stream. Although this association
can be done with ICE [ICE], and ICE is useful to make this
association with RTP, it is not desirable to require ICE to
accomplish this association.
Forking and retargeting are often used together. For example, a boss
and secretary might have both phones ring (forking) and rollover to
voice mail if neither phone is answered (retargeting).
To maintain the security of the media traffic, only the endpoint that
answers the call should know the SRTP keys for the session. Forked
and retargeted calls only reveal sensitive information to non-
responders when the signaling messages contain sensitive information
(e.g., SRTP keys) that is accessible by parties that receive the
offer, but may not respond (i.e., the original recipients in a
retargeted call, or non-answering endpoints in a forked call). For
key exchange mechanisms that do not provide secure forking or secure
retargeting, one workaround is to rekey immediately after forking or
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retargeting. However, because the originator may not be aware that
the call forked this mechanism requires rekeying immediately after
every session is established. This doubles the number of messages
processed by the network.
Further compounding this problem is a unique feature of SIP that,
when forking is used, there is always only one final error response
delivered to the sender of the request: the forking proxy is
responsible for choosing which final response to choose in the event
where forking results in multiple final error responses being
received by the forking proxy. This means that if a request is
rejected, say with information that the keying information was
rejected and providing the far end's credentials, it is very possible
that the rejection will never reach the sender. This problem, called
the Heterogeneous Error Response Forking Problem (HERFP) [RFC3326],
is difficult to solve in SIP. Because we expect the HERFP to
continue to be a problem in SIP for the foreseeable future, a media
security system should function even in the presence of HERFP
behavior.
4.3. Recording
The discussion in this section relates to requirement R-RECORDING.
Some business environments, such as stock brokerages, banks, and
catalog call centers, require recording calls with customers. This
is the familiar "this call is being recorded for quality purposes"
heard during calls to these sorts of businesses. In these
environments, media recording is typically performed by an
intermediate device (with RTP, this is typically implemented in a
'sniffer').
When performing such call recording with SRTP, the end-to-end
security is compromised. This is unavoidable, but necessary because
the operation of the business requires such recording. It is
desirable that the media security is not unduly compromised by the
media recording. The endpoint within the organization needs to be
informed that there is an intermediate device and needs to cooperate
with that intermediate device.
This scenario does not place a requirement directly on the key
management protocol. The requirement could be met directly by the
key management protocol (e.g., MIKEY-NULL or [RFC4568]) or through an
external out-of-band mechanism (e.g., [SRTP-KEY]).
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4.4. PSTN Gateway
The discussion in this section relates to requirement R-PSTN.
It is desirable, even when one leg of a call is on the PSTN, that the
IP leg of the call be protected with SRTP.
A typical case of using media security where two entities are having
a Voice over IP (VoIP) conversation over IP-capable networks.
However, there are cases where the other end of the communication is
not connected to an IP-capable network. In this kind of setting,
there needs to be some kind of gateway at the edge of the IP network
that converts the VoIP conversation to a format understood by the
other network. An example of such a gateway is a PSTN gateway
sitting at the edge of IP and PSTN networks (such as the architecture
described in [RFC3372]).
If media security (e.g., SRTP protection) is employed in this kind of
gateway-setting, then media security and the related key management
is terminated at the PSTN gateway. The other network (e.g., PSTN)
may have its own measures to protect the communication, but this
means that from media security point of view the media security is
not employed truly end-to-end between the communicating entities.
4.5. Call Setup Performance
The discussion in this section relates to requirement R-REUSE.
Some devices lack sufficient processing power to perform public key
operations or Diffie-Hellman operations for each call, or prefer to
avoid performing those operations on every call. The ability to
reuse previous public key or Diffie-Hellman operations can vastly
decrease the call setup delay and processing requirements for such
devices.
In certain devices, it can take a second or two to perform a Diffie-
Hellman operation. Examples of these devices include handsets, IP
Multimedia Services Identity Modules (ISIMs), and PSTN gateways.
PSTN gateways typically utilize a Digital Signal Processor (DSP) that
is not yet involved with typical DSP operations at the beginning of a
call; thus, the DSP could be used to perform the calculation, so as
to avoid having the central host processor perform the calculation.
However, not all PSTN gateways use DSPs (some have only central
processors or their DSPs are incapable of performing the necessary
public key or Diffie-Hellman operation), and handsets lack a
separate, unused processor to perform these operations.
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Two scenarios where R-REUSE is useful are calls between an endpoint
and its voicemail server or its PSTN gateway. In those scenarios,
calls are made relatively often and it can be useful for the
voicemail server or PSTN gateway to avoid public key operations for
subsequent calls.
Storing keys across sessions often interferes with perfect forward
secrecy (R-PFS).
4.6. Transcoding
The discussion in this section relates to requirement R-TRANSCODER.
In some environments, it is necessary for network equipment to
transcode from one codec (e.g., a highly compressed codec that makes
efficient use of wireless bandwidth) to another codec (e.g., a
standardized codec to a SIP peering interface). With RTP, a
transcoding function can be performed with the combination of a SIP
back-to-back user agent (B2BUA) to modify the SDP and a processor to
perform the transcoding between the codecs. However, with end-to-end
secured SRTP, a transcoding function implemented the same way is a
man-in-the-middle attack, and the key management system prevents its
use.
However, such a network-based transcoder can still be realized with
the cooperation and approval of the endpoint, and can provide end-to-
transcoder and transcoder-to-end security.
4.7. Upgrading to SRTP
The discussion in this section relates to the requirement R-ALLOW-
RTP.
Legitimate RTP media can be sent to an endpoint for announcements,
colorful ringback tones (e.g., music), advertising, or normal call
progress tones. The RTP may be received before an associated SDP
answer. For details on various scenarios, see [EARLY-MEDIA].
While receiving such RTP exposes the calling party to a risk of
receiving malicious RTP from an attacker, SRTP endpoints will need to
receive and play out RTP media in order to be compatible with
deployed systems that send RTP to calling parties.
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4.8. Interworking with Other Signaling Protocols
The discussion in this section relates to the requirement R-OTHER-
SIGNALING.
In many environments, some devices are signaled with protocols other
than SIP that do not share SIP's offer/answer model (e.g., [H.248.1]
or do not utilize SDP (e.g., H.323). In other environments, both
endpoints may be SIP, but may use different key management systems
(e.g., one uses MIKEY-RSA, the other MIKEY-RSA-R).
In these environments, it is desirable to have SRTP -- rather than
RTP -- between the two endpoints. It is always possible, although
undesirable, to interwork those disparate signaling systems or
disparate key management systems by decrypting and re-encrypting each
SRTP packet in a device in the middle of the network (often the same
device performing the signaling interworking). This is undesirable
due to the cost and increased attack area, as such an SRTP/SRTP
interworking device is a valuable attack target.
At the time of this writing, interworking is considered important.
Interworking without decryption/encryption of the SRTP, while useful,
is not yet deemed critical because the scale of such SRTP deployments
is, to date, relatively small.
4.9. Certificates
The discussion in this section relates to R-CERTS.
On the Internet and on some private networks, validating another
peer's certificate is often done through a trust anchor -- a list of
Certificate Authorities that are trusted. It can be difficult or
expensive for a peer to obtain these certificates. In all cases,
both parties to the call would need to trust the same trust anchor
(i.e., "certificate authority"). For these reasons, it is important
that the media plane key management protocol offer a mechanism that
allows end-users who have no prior association to authenticate to
each other without acquiring credentials from a third-party trust
point. Note that this does not rule out mechanisms in which servers
have certificates and attest to the identities of end-users.
5. Requirements
This section is divided into several parts: requirements specific to
the key management protocol (Section 5.1), attack scenarios
(Section 5.2), and requirements that can be met inside the key
management protocol or outside of the key management protocol
(Section 5.3).
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5.1. Key Management Protocol Requirements
SIP Forking and Retargeting, from Section 4.2:
R-FORK-RETARGET:
The media security key management protocol MUST
securely support forking and retargeting when all
endpoints are willing to use SRTP without causing
the call setup to fail. This requirement means the
endpoints that did not answer the call MUST NOT
learn the SRTP keys (in either direction) used by
the answering endpoint.
R-DISTINCT:
The media security key management protocol MUST be
capable of creating distinct, independent cryptographic
contexts for each endpoint in a forked session.
R-HERFP:
The media security key management protocol MUST function
securely even in the presence of HERFP behavior, i.e., the
rejection of key information does not reach the sender.
Performance considerations:
R-REUSE:
The media security key management protocol MAY support the
reuse of a previously established security context.
Note: reuse of the security context does not imply reuse of RTP
parameters (e.g., payload type or SSRC).
Media considerations:
R-AVOID-CLIPPING:
The media security key management protocol SHOULD
avoid clipping media before SDP answer without
requiring Security Preconditions [RFC5027]. This
requirement comes from Section 4.1.
R-RTP-CHECK:
If SRTP key negotiation is performed over the media
path (i.e., using the same UDP/TCP ports as media
packets), the key negotiation packets MUST NOT pass the
RTP validity check defined in Appendix A.1 of
[RFC3550], so that SRTP negotiation packets can be
differentiated from RTP packets.
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R-ASSOC:
The media security key management protocol SHOULD include a
mechanism for associating key management messages with both
the signaling traffic that initiated the session and with
protected media traffic. It is useful to associate key
management messages with call signaling messages, as this
allows the SDP offerer to avoid performing CPU-consuming
operations (e.g., Diffie-Hellman or public key operations)
with attackers that have not seen the signaling messages.
For example, if using a Diffie-Hellman keying technique
with security preconditions that forks to 20 endpoints, the
call initiator would get 20 provisional responses
containing 20 signed Diffie-Hellman key pairs. Calculating
20 Diffie-Hellman secrets and validating signatures can be
a difficult task for some devices. Hence, in the case of
forking, it is not desirable to perform a Diffie-Hellman
operation with every party, but rather only with the party
that answers the call (and incur some media clipping). To
do this, the signaling and media need to be associated so
the calling party knows which key management exchange needs
to be completed. This might be done by using the transport
address indicated in the SDP, although NATs can complicate
this association.
Note: due to RTP's design requirements, it is expected that
SRTP receivers will have to perform authentication of any
received SRTP packets.
R-NEGOTIATE:
The media security key management protocol MUST allow a
SIP User Agent to negotiate media security parameters
for each individual session. Such negotiation MUST NOT
cause a two-time pad (Section 9.1 of [RFC3711]).
R-PSTN:
The media security key management protocol MUST support
termination of media security in a PSTN gateway. This
requirement is from Section 4.4.
5.2. Security Requirements
This section describes overall security requirements and specific
requirements from the attack scenarios (Section 3).
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Overall security requirements:
R-PFS:
The media security key management protocol MUST be able to
support perfect forward secrecy.
R-COMPUTE:
The media security key management protocol MUST support
offering additional SRTP cipher suites without incurring
significant computational expense.
R-CERTS:
The key management protocol MUST NOT require that end-users
obtain credentials (certificates or private keys) from a
third- party trust anchor.
R-FIPS:
The media security key management protocol SHOULD use
algorithms that allow FIPS 140-2 [FIPS-140-2] certification
or similar country-specific certification (e.g.,
[AISITSEC]).
The United States Government can only purchase and use
crypto implementations that have been validated by the
FIPS-140 [FIPS-140-2] process:
The FIPS-140 standard is applicable to all Federal agencies
that use cryptographic-based security systems to protect
sensitive information in computer and telecommunication
systems, including voice systems. The adoption and use
of this standard is available to private and commercial
organizations.
Some commercial organizations, such as banks and defense
contractors, require or prefer equipment that has received the
same validation.
R-DOS:
The media security key management protocol MUST NOT introduce
any new significant denial-of-service vulnerabilities (e.g.,
the protocol should not request the endpoint to perform CPU-
intensive operations without the client being able to
validate or authorize the request).
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R-EXISTING:
The media security key management protocol SHOULD allow
endpoints to authenticate using pre-existing
cryptographic credentials, e.g., certificates or
pre-shared keys.
R-AGILITY:
The media security key management protocol MUST provide
crypto- agility, i.e., the ability to adapt to evolving
cryptography and security requirements (update of
cryptographic algorithms without substantial disruption
to deployed implementations).
R-DOWNGRADE:
The media security key management protocol MUST protect
cipher suite negotiation against downgrading attacks.
R-PASS-MEDIA:
The media security key management protocol MUST have a
mode that prevents a passive adversary with access to
the media path from gaining access to keying material
used to protect SRTP media packets.
R-PASS-SIG:
The media security key management protocol MUST have a
mode in which it prevents a passive adversary with
access to the signaling path from gaining access to
keying material used to protect SRTP media packets.
R-SIG-MEDIA:
The media security key management protocol MUST have a
mode in which it defends itself from an attacker that
is solely on the media path and from an attacker that
is solely on the signaling path. A successful attack
refers to the ability for the adversary to obtain
keying material to decrypt the SRTP encrypted media
traffic.
R-ID-BINDING:
The media security key management protocol MUST enable
the media security keys to be cryptographically bound
to an identity of the endpoint.
Note: This allows domains to deploy SIP Identity [RFC4474].
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R-ACT-ACT:
The media security key management protocol MUST support a
mode of operation that provides
active-signaling-active-media-detect robustness, and MAY
support modes of operation that provide lower levels of
robustness (as described in Section 3).
Note: Failing to meet R-ACT-ACT indicates the protocol cannot
provide secure end-to-end media.
5.3. Requirements outside of the Key Management Protocol
The requirements in this section are for an overall VoIP security
system. These requirements can be met within the key management
protocol itself, or can be solved outside of the key management
protocol itself (e.g., solved in SIP or in SDP).
R-BEST-SECURE:
Even when some endpoints of a forked or retargeted
call are incapable of using SRTP, a solution MUST be
described that allows the establishment of SRTP
associations with SRTP-capable endpoints and/or RTP
associations with non-SRTP-capable endpoints.
R-OTHER-SIGNALING:
A solution SHOULD be able to negotiate keys for
SRTP sessions created via different call
signaling protocols (e.g., between Jabber, SIP,
H.323, Media Gateway Control Protocol (MGCP).
R-RECORDING:
A solution SHOULD be described that supports recording
of decrypted media. This requirement comes from
Section 4.3.
R-TRANSCODER:
A solution SHOULD be described that supports
intermediate nodes (e.g., transcoders), terminating or
processing media, between the endpoints.
R-ALLOW-RTP: A solution SHOULD be described that allows RTP media to
be received by the calling party until SRTP has been
negotiated with the answerer, after which SRTP is
preferred over RTP.
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6. Security Considerations
This document lists requirements for securing media traffic. As
such, it addresses security throughout the document.
7. Acknowledgements
For contributions to the requirements portion of this document, the
authors would like to thank the active participants of the RTPSEC BoF
and on the RTPSEC mailing list, and a special thanks to Steffen Fries
and Dragan Ignjatic for their excellent MIKEY comparison [RFC5197]
document.
The authors would furthermore like to thank the following people for
their review, suggestions, and comments: Flemming Andreasen, Richard
Barnes, Mark Baugher, Wolfgang Buecker, Werner Dittmann, Lakshminath
Dondeti, John Elwell, Martin Euchner, Hans-Heinrich Grusdt, Christer
Holmberg, Guenther Horn, Peter Howard, Leo Huang, Dragan Ignjatic,
Cullen Jennings, Alan Johnston, Vesa Lehtovirta, Matt Lepinski, David
McGrew, David Oran, Colin Perkins, Eric Raymond, Eric Rescorla, Peter
Schneider, Frank Shearar, Srinath Thiruvengadam, Dave Ward, Dan York,
and Phil Zimmermann.
8. References
8.1. Normative References
[FIPS-140-2] NIST, "Security Requirements for Cryptographic
Modules", June 2005, <http://csrc.nist.gov/
publications/fips/fips140-2/fips1402.pdf>.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G.,
Johnston, A., Peterson, J., Sparks, R., Handley, M.,
and E. Schooler, "SIP: Session Initiation Protocol",
RFC 3261, June 2002.
[RFC3262] Rosenberg, J. and H. Schulzrinne, "Reliability of
Provisional Responses in Session Initiation Protocol
(SIP)", RFC 3262, June 2002.
[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer
Model with Session Description Protocol (SDP)",
RFC 3264, June 2002.
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RFC 5479 Media Security Requirements April 2009
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and
K. Norrman, "The Secure Real-time Transport Protocol
(SRTP)", RFC 3711, March 2004.
8.2. Informative References
[AISITSEC] Bundesamt fuer Sicherheit in der Informationstechnik
[Federal Office of Information Security - Germany],
"Anwendungshinweise und Interpretationen (AIS) zu
ITSEC", January 2002,
<http://www.bsi.de/zertifiz/zert/interpr/
aisitsec.htm>.
[DTLS-SRTP] McGrew, D. and E. Rescorla, "Datagram Transport Layer
Security (DTLS) Extension to Establish Keys for Secure
Real-time Transport Protocol (SRTP)", Work
in Progress, October 2008.
[EARLY-MEDIA] Stucker, B., "Coping with Early Media in the Session
Initiation Protocol (SIP)", Work in Progress,
October 2006.
[EKT] McGrew, D., "Encrypted Key Transport for Secure RTP",
Work in Progress, July 2007.
[H.248.1] ITU, "Gateway control protocol", Recommendation H.248,
June 2000, <http://www.itu.int/rec/T-REC-H.248/e>.
[ICE] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator
(NAT) Traversal for Offer/Answer Protocols", Work
in Progress, October 2007.
[MIDDLEBOX] Stucker, B. and H. Tschofenig, "Analysis of Middlebox
Interactions for Signaling Protocol Communication
along the Media Path", Work in Progress, July 2008.
[MIKEY-ECC] Milne, A., "ECC Algorithms for MIKEY", Work
in Progress, June 2007.
[MIKEYv2] Dondeti, L., "MIKEYv2: SRTP Key Management using
MIKEY, revisited", Work in Progress, March 2007.
[MULTIPART] Wing, D. and C. Jennings, "Session Initiation Protocol
(SIP) Offer/Answer with Multipart Alternative", Work
in Progress, March 2006.
Wing, et al. Informational [Page 21]
RFC 5479 Media Security Requirements April 2009
[RFC3326] Schulzrinne, H., Oran, D., and G. Camarillo, "The
Reason Header Field for the Session Initiation
Protocol (SIP)", RFC 3326, December 2002.
[RFC3372] Vemuri, A. and J. Peterson, "Session Initiation
Protocol for Telephones (SIP-T): Context and
Architectures", BCP 63, RFC 3372, September 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.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and
K. Norrman, "MIKEY: Multimedia Internet KEYing",
RFC 3830, August 2004.
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for
Authenticated Identity Management in the Session
Initiation Protocol (SIP)", RFC 4474, August 2006.
[RFC4492] Blake-Wilson, S., Bolyard, N., Gupta, V., Hawk, C.,
and B. Moeller, "Elliptic Curve Cryptography (ECC)
Cipher Suites for Transport Layer Security (TLS)",
RFC 4492, May 2006.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for
Media Streams", RFC 4568, July 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.
[RFC4916] Elwell, J., "Connected Identity in the Session
Initiation Protocol (SIP)", RFC 4916, June 2007.
Wing, et al. Informational [Page 22]
RFC 5479 Media Security Requirements April 2009
[RFC4949] Shirey, R., "Internet Security Glossary, Version 2",
FYI 36, RFC 4949, August 2007.
[RFC5027] Andreasen, F. and D. Wing, "Security Preconditions for
Session Description Protocol (SDP) Media Streams",
RFC 5027, October 2007.
[RFC5197] Fries, S. and D. Ignjatic, "On the Applicability of
Various Multimedia Internet KEYing (MIKEY) Modes and
Extensions", RFC 5197, June 2008.
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer
Security (TLS) Protocol Version 1.2", RFC 5246,
August 2008.
[SDP-CAP] Andreasen, F., "SDP Capability Negotiation", Work
in Progress, July 2008.
[SDP-DH] Baugher, M. and D. McGrew, "Diffie-Hellman Exchanges
for Multimedia Sessions", Work in Progress,
February 2006.
[SIP-CERTS] Jennings, C. and J. Fischl, "Certificate Management
Service for The Session Initiation Protocol (SIP)",
Work in Progress, November 2008.
[SIP-DTLS] Fischl, J., "Datagram Transport Layer Security (DTLS)
Protocol for Protection of Media Traffic Established
with the Session Initiation Protocol", Work
in Progress, July 2007.
[SRTP-KEY] Wing, D., Audet, F., Fries, S., Tschofenig, H., and A.
Johnston, "Secure Media Recording and Transcoding with
the Session Initiation Protocol", Work in Progress,
October 2008.
[ZRTP] Zimmermann, P., Johnston, A., and J. Callas, "ZRTP:
Media Path Key Agreement for Secure RTP", Work
in Progress, February 2009.
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RFC 5479 Media Security Requirements April 2009
Appendix A. Overview and Evaluation of Existing Keying Mechanisms
Based on how the SRTP keys are exchanged, each SRTP key exchange
mechanism belongs to one general category:
signaling path:
All the keying is carried in the call signaling (SIP
or SDP) path.
media path:
All the keying is carried in the SRTP/SRTCP media path,
and no signaling whatsoever is carried in the call
signaling path.
signaling and media path:
Parts of the keying are carried in the
SRTP/SRTCP media path, and parts are
carried in the call signaling (SIP or SDP)
path.
One of the significant benefits of SRTP over other end-to-end
encryption mechanisms, such as for example IPsec, is that SRTP is
bandwidth efficient and SRTP retains the header of RTP packets.
Bandwidth efficiency is vital for VoIP in many scenarios where access
bandwidth is limited or expensive, and retaining the RTP header is
important for troubleshooting packet loss, delay, and jitter.
Related to SRTP's characteristics is a goal that any SRTP keying
mechanism to also be efficient and not cause additional call setup
delay. Contributors to additional call setup delay include network
or database operations: retrieval of certificates and additional SIP
or media path messages, and computational overhead of establishing
keys or validating certificates.
When examining the choice between keying in the signaling path,
keying in the media path, or keying in both paths, it is important to
realize the media path is generally "faster" than the SIP signaling
path. The SIP signaling path has computational elements involved
that parse and route SIP messages. The media path, on the other
hand, does not normally have computational elements involved, and
even when computational elements such as firewalls are involved, they
cause very little additional delay. Thus, the media path can be
useful for exchanging several messages to establish SRTP keys. A
disadvantage of keying over the media path is that interworking
different key exchange requires the interworking function be in the
media path, rather than just in the signaling path; in practice, this
involvement is probably unavoidable anyway.
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A.1. Signaling Path Keying Techniques
A.1.1. MIKEY-NULL
MIKEY-NULL [RFC3830] has the offerer indicate the SRTP keys for both
directions. The key is sent unencrypted in SDP, which means the SDP
must be encrypted hop-by-hop (e.g., by using TLS (SIPS)) or end-to-
end (e.g., by using Secure/Multipurpose Internet Mail Extensions
(S/MIME)).
MIKEY-NULL requires one message from offerer to answerer (half a
round trip), and does not add additional media path messages.
A.1.2. MIKEY-PSK
MIKEY-PSK (pre-shared key) [RFC3830] requires that all endpoints
share one common key. MIKEY-PSK has the offerer encrypt the SRTP
keys for both directions using this pre-shared key.
MIKEY-PSK requires one message from offerer to answerer (half a round
trip), and does not add additional media path messages.
A.1.3. MIKEY-RSA
MIKEY-RSA [RFC3830] has the offerer encrypt the keys for both
directions using the intended answerer's public key, which is
obtained from a mechanism outside of MIKEY.
MIKEY-RSA requires one message from offerer to answerer (half a round
trip), and does not add additional media path messages. MIKEY-RSA
requires the offerer to obtain the intended answerer's certificate.
A.1.4. MIKEY-RSA-R
MIKEY-RSA-R [RFC4738] is essentially the same as MIKEY-RSA but
reverses the role of the offerer and the answerer with regards to
providing the keys. That is, the answerer encrypts the keys for both
directions using the offerer's public key. Both the offerer and
answerer validate each other's public keys using a standard X.509
validation techniques. MIKEY-RSA-R also enables sending certificates
in the MIKEY message.
MIKEY-RSA-R requires one message from offerer to answer, and one
message from answerer to offerer (full round trip), and does not add
additional media path messages. MIKEY-RSA-R requires the offerer
validate the answerer's certificate.
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A.1.5. MIKEY-DHSIGN
In MIKEY-DHSIGN [RFC3830], the offerer and answerer derive the key
from a Diffie-Hellman (DH) exchange. In order to prevent an active
man-in-the-middle, the DH exchange itself is signed using each
endpoint's private key and the associated public keys are validated
using standard X.509 validation techniques.
MIKEY-DHSIGN requires one message from offerer to answerer, and one
message from answerer to offerer (full round trip), and does not add
additional media path messages. MIKEY-DHSIGN requires the offerer
and answerer to validate each other's certificates. MIKEY-DHSIGN
also enables sending the answerer's certificate in the MIKEY message.
A.1.6. MIKEY-DHHMAC
MIKEY-DHHMAC [RFC4650] uses a pre-shared secret to HMAC the Diffie-
Hellman exchange, essentially combining aspects of MIKEY-PSK with
MIKEY-DHSIGN, but without MIKEY-DHSIGN's need for certificate
authentication.
MIKEY-DHHMAC requires one message from offerer to answerer, and one
message from answerer to offerer (full round trip), and does not add
additional media path messages.
A.1.7. MIKEY-ECIES and MIKEY-ECMQV (MIKEY-ECC)
ECC Algorithms For MIKEY [MIKEY-ECC] describes how ECC can be used
with MIKEY-RSA (using Elliptic Curve Digital Signature Algorithm
(ECDSA) signature) and with MIKEY-DHSIGN (using a new DH-Group code),
and also defines two new ECC-based algorithms, Elliptic Curve
Integrated Encryption Scheme (ECIES) and Elliptic Curve Menezes-Qu-
Vanstone (ECMQV) .
With this proposal, the ECDSA signature, MIKEY-ECIES, and MIKEY-ECMQV
function exactly like MIKEY-RSA, and the new DH-Group code function
exactly like MIKEY-DHSIGN. Therefore, these ECC mechanisms are not
discussed separately in this document.
A.1.8. SDP Security Descriptions with SIPS
SDP Security Descriptions [RFC4568] have each side indicate the key
they will use for transmitting SRTP media, and the keys are sent in
the clear in SDP. SDP Security Descriptions rely on hop-by-hop (TLS
via "SIPS:") encryption to protect the keys exchanged in signaling.
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SDP Security Descriptions requires one message from offerer to
answerer, and one message from answerer to offerer (full round trip),
and does not add additional media path messages.
A.1.9. SDP Security Descriptions with S/MIME
This keying mechanism is identical to Appendix A.1.8 except that,
rather than protecting the signaling with TLS, the entire SDP is
encrypted with S/MIME.
A.1.10. SDP-DH (Expired)
SDP Diffie-Hellman [SDP-DH] exchanges Diffie-Hellman messages in the
signaling path to establish session keys. To protect against active
man-in-the-middle attacks, the Diffie-Hellman exchange needs to be
protected with S/MIME, SIPS, or SIP Identity [RFC4474] and SIP
Connected Identity [RFC4916].
SDP-DH requires one message from offerer to answerer, and one message
from answerer to offerer (full round trip), and does not add
additional media path messages.
A.1.11. MIKEYv2 in SDP (Expired)
MIKEYv2 [MIKEYv2] adds mode negotiation to MIKEYv1 and removes the
time synchronization requirement. It therefore now takes 2 round
trips to complete. In the first round trip, the communicating
parties learn each other's identities, agree on a MIKEY mode, crypto
algorithm, SRTP policy, and exchanges nonces for replay protection.
In the second round trip, they negotiate unicast and/or group SRTP
context for SRTP and/or SRTCP.
Furthermore, MIKEYv2 also defines an in-band negotiation mode as an
alternative to SDP (see Appendix A.3.3).
A.2. Media Path Keying Technique
A.2.1. ZRTP
ZRTP [ZRTP] does not exchange information in the signaling path
(although it's possible for endpoints to exchange a hash of the ZRTP
Hello message with "a=zrtp-hash" in the initial offer if sent over an
integrity-protected signaling channel. This provides some useful
correlation between the signaling and media layers). In ZRTP, the
keys are exchanged entirely in the media path using a Diffie-Hellman
exchange. The advantage to this mechanism is that the signaling
channel is used only for call setup and the media channel is used to
establish an encrypted channel -- much like encryption devices on the
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PSTN. ZRTP uses voice authentication of its Diffie-Hellman exchange
by having each person read digits or words to the other person.
Subsequent sessions with the same ZRTP endpoint can be authenticated
using the stored hash of the previously negotiated key rather than
voice authentication. ZRTP uses four media path messages (Hello,
Commit, DHPart1, and DHPart2) to establish the SRTP key, and three
media path confirmation messages. These initial messages are all
sent as non-RTP packets.
Note: that when ZRTP probing is used, unencrypted RTP can be
exchanged until the SRTP keys are established.
A.3. Signaling and Media Path Keying Techniques
A.3.1. EKT
EKT [EKT] relies on another SRTP key exchange protocol, such as SDP
Security Descriptions or MIKEY, for bootstrapping. In the initial
phase, each member of a conference uses an SRTP key exchange protocol
to establish a common key encryption key (KEK). Each member may use
the KEK to securely transport its SRTP master key and current SRTP
rollover counter (ROC), via RTCP, to the other participants in the
session.
EKT requires the offerer to send some parameters (EKT_Cipher, KEK,
and security parameter index (SPI)) via the bootstrapping protocol
such as SDP Security Descriptions or MIKEY. Each answerer sends an
SRTCP message that contains the answerer's SRTP Master Key, rollover
counter, and the SRTP sequence number. Rekeying is done by sending a
new SRTCP message. For reliable transport, multiple RTCP messages
need to be sent.
A.3.2. DTLS-SRTP
DTLS-SRTP [DTLS-SRTP] exchanges public key fingerprints in SDP
[SIP-DTLS] and then establishes a DTLS session over the media
channel. The endpoints use the DTLS handshake to agree on crypto
suites and establish SRTP session keys. SRTP packets are then
exchanged between the endpoints.
DTLS-SRTP requires one message from offerer to answerer (half round
trip), and one message from the answerer to offerer (full round trip)
so the offerer can correlate the SDP answer with the answering
endpoint. DTLS-SRTP uses four media path messages to establish the
SRTP key.
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This document assumes DTLS will use TLS_RSA_WITH_AES_128_CBC_SHA as
its cipher suite, which is the mandatory-to-implement cipher suite in
TLS [RFC5246].
A.3.3. MIKEYv2 Inband (Expired)
As defined in Appendix A.1.11, MIKEYv2 also defines an in-band
negotiation mode as an alternative to SDP (see Appendix A.3.3). The
details are not sorted out in the document yet on what in-band
actually means (i.e., UDP, RTP, RTCP, etc.).
A.4. Evaluation Criteria - SIP
This section considers how each keying mechanism interacts with SIP
features.
A.4.1. Secure Retargeting and Secure Forking
Retargeting and forking of signaling requests is described within
Section 4.2. The following builds upon this description.
The following list compares the behavior of secure forking, answering
association, two-time pads, and secure retargeting for each keying
mechanism.
MIKEY-NULL
Secure Forking: No, all AORs see offerer's and answerer's keys.
Answer is associated with media by the SSRC in MIKEY.
Additionally, a two-time pad occurs if two branches choose the
same 32-bit SSRC and transmit SRTP packets.
Secure Retargeting: No, all targets see offerer's and
answerer's keys. Suffers from retargeting identity problem.
MIKEY-PSK
Secure Forking: No, all AORs see offerer's and answerer's keys.
Answer is associated with media by the SSRC in MIKEY. Note
that all AORs must share the same pre-shared key in order for
forking to work at all with MIKEY-PSK. Additionally, a two-
time pad occurs if two branches choose the same 32-bit SSRC and
transmit SRTP packets.
Secure Retargeting: Not secure. For retargeting to work, the
final target must possess the correct PSK. As this is likely
in scenarios where the call is targeted to another device
belonging to the same user (forking), it is very unlikely that
other users will possess that PSK and be able to successfully
answer that call.
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MIKEY-RSA
Secure Forking: No, all AORs see offerer's and answerer's keys.
Answer is associated with media by the SSRC in MIKEY. Note
that all AORs must share the same private key in order for
forking to work at all with MIKEY-RSA. Additionally, a two-
time pad occurs if two branches choose the same 32-bit SSRC and
transmit SRTP packets.
Secure Retargeting: No.
MIKEY-RSA-R
Secure Forking: Yes, answer is associated with media by the
SSRC in MIKEY.
Secure Retargeting: Yes.
MIKEY-DHSIGN
Secure Forking: Yes, each forked endpoint negotiates unique
keys with the offerer for both directions. Answer is
associated with media by the SSRC in MIKEY.
Secure Retargeting: Yes, each target negotiates unique keys
with the offerer for both directions.
MIKEYv2 in SDP
The behavior will depend on which mode is picked.
MIKEY-DHHMAC
Secure Forking: Yes, each forked endpoint negotiates unique
keys with the offerer for both directions. Answer is
associated with media by the SSRC in MIKEY.
Secure Retargeting: Yes, each target negotiates unique keys
with the offerer for both directions. Note that for the keys
to be meaningful, it would require the PSK to be the same for
all the potential intermediaries, which would only happen
within a single domain.
SDP Security Descriptions with SIPS
Secure Forking: No, each forked endpoint sees the offerer's
key. Answer is not associated with media.
Secure Retargeting: No, each target sees the offerer's key.
SDP Security Descriptions with S/MIME
Secure Forking: No, each forked endpoint sees the offerer's
key. Answer is not associated with media.
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Secure Retargeting: No, each target sees the offerer's key.
Suffers from retargeting identity problem.
SDP-DH
Secure Forking: Yes, each forked endpoint calculates a unique
SRTP key. Answer is not associated with media.
Secure Retargeting: Yes, the final target calculates a unique
SRTP key.
ZRTP
Secure Forking: Yes, each forked endpoint calculates a unique
SRTP key. With the "a=zrtp-hash" attribute, the media can be
associated with an answer.
Secure Retargeting: Yes, the final target calculates a unique
SRTP key.
EKT
Secure Forking: Inherited from the bootstrapping mechanism (the
specific MIKEY mode or SDP Security Descriptions). Answer is
associated with media by the SPI in the EKT protocol. Answer
is associated with media by the SPI in the EKT protocol.
Secure Retargeting: Inherited from the bootstrapping mechanism
(the specific MIKEY mode or SDP Security Descriptions).
DTLS-SRTP
Secure Forking: Yes, each forked endpoint calculates a unique
SRTP key. Answer is associated with media by the certificate
fingerprint in signaling and certificate in the media path.
Secure Retargeting: Yes, the final target calculates a unique
SRTP key.
MIKEYv2 Inband
The behavior will depend on which mode is picked.
A.4.2. Clipping Media before SDP Answer
Clipping media before receiving the signaling answer is described
within Section 4.1. The following builds upon this description.
Furthermore, the problem of clipping gets compounded when forking is
used. For example, if using a Diffie-Hellman keying technique with
security preconditions that forks to 20 endpoints, the call initiator
would get 20 provisional responses containing 20 signed Diffie-
Hellman half keys. Calculating 20 DH secrets and validating
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signatures can be a difficult task depending on the device
capabilities.
The following list compares the behavior of clipping before SDP
answer for each keying mechanism.
MIKEY-NULL
Not clipped. The offerer provides the answerer's keys.
MIKEY-PSK
Not clipped. The offerer provides the answerer's keys.
MIKEY-RSA
Not clipped. The offerer provides the answerer's keys.
MIKEY-RSA-R
Clipped. The answer contains the answerer's encryption key.
MIKEY-DHSIGN
Clipped. The answer contains the answerer's Diffie-Hellman
response.
MIKEY-DHHMAC
Clipped. The answer contains the answerer's Diffie-Hellman
response.
MIKEYv2 in SDP
The behavior will depend on which mode is picked.
SDP Security Descriptions with SIPS
Clipped. The answer contains the answerer's encryption key.
SDP Security Descriptions with S/MIME
Clipped. The answer contains the answerer's encryption key.
SDP-DH
Clipped. The answer contains the answerer's Diffie-Hellman
response.
ZRTP
Not clipped because the session initially uses RTP. While RTP
is flowing, both ends negotiate SRTP keys in the media path and
then switch to using SRTP.
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EKT
Not clipped, as long as the first RTCP packet (containing the
answerer's key) is not lost in transit. The answerer sends its
encryption key in RTCP, which arrives at the same time (or
before) the first SRTP packet encrypted with that key.
Note: RTCP needs to work, in the answerer-to-offerer
direction, before the offerer can decrypt SRTP media.
DTLS-SRTP
No clipping after the DTLS-SRTP handshake has completed. SRTP
keys are exchanged in the media path. Need to wait for SDP
answer to ensure DTLS-SRTP handshake was done with an
authorized party.
If a middlebox interferes with the media path, there can be
clipping [MIDDLEBOX].
MIKEYv2 Inband
Not clipped. Keys are exchanged in the media path without
relying on the signaling path.
A.4.3. SSRC and ROC
In SRTP, a cryptographic context is defined as the SSRC, destination
network address, and destination transport port number. Whereas RTP,
a flow is defined as the destination network address and destination
transport port number. This results in a problem -- how to
communicate the SSRC so that the SSRC can be used for the
cryptographic context.
Two approaches have emerged for this communication. One, used by all
MIKEY modes, is to communicate the SSRCs to the peer in the MIKEY
exchange. Another, used by SDP Security Descriptions, is to apply
"late binding" -- that is, any new packet containing a previously
unseen SSRC (which arrives at the same destination network address
and destination transport port number) will create a new
cryptographic context. Another approach, common amongst techniques
with media-path SRTP key establishment, is to require a handshake
over that media path before SRTP packets are sent. MIKEY's approach
changes RTP's SSRC collision detection behavior by requiring RTP to
pre-establish the SSRC values for each session.
Another related issue is that SRTP introduces a rollover counter
(ROC), which records how many times the SRTP sequence number has
rolled over. As the sequence number is used for SRTP's default
ciphers, it is important that all endpoints know the value of the
ROC. The ROC starts at 0 at the beginning of a session.
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Some keying mechanisms cause a two-time pad to occur if two endpoints
of a forked call have an SSRC collision.
Note: A proposal has been made to send the ROC value on every Nth
SRTP packet[RFC4771]. This proposal has not yet been incorporated
into this document.
The following list examines handling of SSRC and ROC:
MIKEY-NULL
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
MIKEY-PSK
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
MIKEY-RSA
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
MIKEY-RSA-R
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
MIKEY-DHSIGN
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
MIKEY-DHHMAC
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
MIKEYv2 in SDP
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
SDP Security Descriptions with SIPS
Neither SSRC nor ROC are signaled. SSRC "late binding" is
used.
SDP Security Descriptions with S/MIME
Neither SSRC nor ROC are signaled. SSRC "late binding" is
used.
SDP-DH
Neither SSRC nor ROC are signaled. SSRC "late binding" is
used.
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ZRTP
Neither SSRC nor ROC are signaled. SSRC "late binding" is
used.
EKT
The SSRC of the SRTCP packet containing an EKT update
corresponds to the SRTP master key and other parameters within
that packet.
DTLS-SRTP
Neither SSRC nor ROC are signaled. SSRC "late binding" is
used.
MIKEYv2 Inband
Each endpoint indicates a set of SSRCs and the ROC for SRTP
packets it transmits.
A.5. Evaluation Criteria - Security
This section evaluates each keying mechanism on the basis of their
security properties.
A.5.1. Distribution and Validation of Persistent Public Keys and
Certificates
Using persistent public keys for confidentiality and authentication
can introduce requirements for two types of systems, often
implemented using certificates: (1) a system to distribute those
persistent public keys certificates, and (2) a system for validating
those persistent public keys. We refer to the former as a key
distribution system and the latter as an authentication
infrastructure. In many cases, a monolithic public key
infrastructure (PKI) is used to fulfill both of these roles.
However, these functions can be provided by many other systems. For
instance, key distribution may be accomplished by any public
repository of keys. Any system in which the two endpoints have
access to trust anchors and intermediate CA certificates that can be
used to validate other endpoints' certificates (including a system of
self-signed certificates) can be used to support certificate
validation in the below schemes.
With real-time communications, it is desirable to avoid fetching or
validating certificates that delay call setup. Rather, it is
preferable to fetch or validate certificates in such a way that call
setup is not delayed. For example, a certificate can be validated
while the phone is ringing or can be validated while ring-back tones
are being played or even while the called party is answering the
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phone and saying "hello". Even better is to avoid fetching or
validating persistent public keys at all.
SRTP key exchange mechanisms that require a particular authentication
infrastructure to operate (whether for distribution or validation)
are gated on the deployment of a such an infrastructure available to
both endpoints. This means that no media security is achievable
until such an infrastructure exists. For SIP, something like sip-
certs [SIP-CERTS] might be used to obtain the certificate of a peer.
Note: Even if sip-certs [SIP-CERTS] were deployed, the retargeting
problem (Appendix A.4.1) would still prevent successful deployment
of keying techniques which require the offerer to obtain the
actual target's public key.
The following list compares the requirements introduced by the use of
public-key cryptography in each keying mechanism, both for public key
distribution and for certificate validation.
MIKEY-NULL
Public-key cryptography is not used.
MIKEY-PSK
Public-key cryptography is not used. Rather, all endpoints
must have some way to exchange per-endpoint or per-system
pre-shared keys.
MIKEY-RSA
The offerer obtains the intended answerer's public key before
initiating the call. This public key is used to encrypt the
SRTP keys. There is no defined mechanism for the offerer to
obtain the answerer's public key, although [SIP-CERTS] might be
viable in the future.
The offer may also contain a certificate for the offerer, which
would require an authentication infrastructure in order to be
validated by the receiver.
MIKEY-RSA-R
The offer contains the offerer's certificate, and the answer
contains the answerer's certificate. The answerer uses the
public key in the certificate to encrypt the SRTP keys that
will be used by the offerer and the answerer. An
authentication infrastructure is necessary to validate the
certificates.
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MIKEY-DHSIGN
An authentication infrastructure is used to authenticate the
public key that is included in the MIKEY message.
MIKEY-DHHMAC
Public-key cryptography is not used. Rather, all endpoints
must have some way to exchange per-endpoint or per-system
pre-shared keys.
MIKEYv2 in SDP
The behavior will depend on which mode is picked.
SDP Security Descriptions with SIPS
Public-key cryptography is not used.
SDP Security Descriptions with S/MIME
Use of S/MIME requires that the endpoints be able to fetch and
validate certificates for each other. The offerer must obtain
the intended target's certificate and encrypts the SDP offer
with the public key contained in target's certificate. The
answerer must obtain the offerer's certificate and encrypt the
SDP answer with the public key contained in the offerer's
certificate.
SDP-DH
Public-key cryptography is not used.
ZRTP
Public-key cryptography is used (Diffie-Hellman), but without
dependence on persistent public keys. Thus, certificates are
not fetched or validated.
EKT
Public-key cryptography is not used by itself, but might be
used by the EKT bootstrapping keying mechanism (such as certain
MIKEY modes).
DTLS-SRTP
Remote party's certificate is sent in media path, and a
fingerprint of the same certificate is sent in the signaling
path.
MIKEYv2 Inband
The behavior will depend on which mode is picked.
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A.5.2. Perfect Forward Secrecy
In the context of SRTP, Perfect Forward Secrecy is the property that
SRTP session keys that protected a previous session are not
compromised if the static keys belonging to the endpoints are
compromised. That is, if someone were to record your encrypted
session content and later acquires either party's private key, that
encrypted session content would be safe from decryption if your key
exchange mechanism had perfect forward secrecy.
The following list describes how each key exchange mechanism provides
PFS.
MIKEY-NULL
Not applicable; MIKEY-NULL does not have a long-term secret.
MIKEY-PSK
No PFS.
MIKEY-RSA
No PFS.
MIKEY-RSA-R
No PFS.
MIKEY-DHSIGN
PFS is provided with the Diffie-Hellman exchange.
MIKEY-DHHMAC
PFS is provided with the Diffie-Hellman exchange.
MIKEYv2 in SDP
The behavior will depend on which mode is picked.
SDP Security Descriptions with SIPS
Not applicable; SDP Security Descriptions does not have a long-
term secret.
SDP Security Descriptions with S/MIME
Not applicable; SDP Security Descriptions does not have a long-
term secret.
SDP-DH
PFS is provided with the Diffie-Hellman exchange.
ZRTP
PFS is provided with the Diffie-Hellman exchange.
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EKT
No PFS.
DTLS-SRTP
PFS is provided if the negotiated cipher suite uses ephemeral
keys (e.g., Diffie-Hellman (DHE_RSA [RFC5246]) or Elliptic
Curve Diffie-Hellman [RFC4492]).
MIKEYv2 Inband
The behavior will depend on which mode is picked.
A.5.3. Best Effort Encryption
With best effort encryption, SRTP is used with endpoints that support
SRTP, otherwise RTP is used.
SIP needs a backwards-compatible best effort encryption in order for
SRTP to work successfully with SIP retargeting and forking when there
is a mix of forked or retargeted devices that support SRTP and don't
support SRTP.
Consider the case of Bob, with a phone that only does RTP and a
voice mail system that supports SRTP and RTP. If Alice calls Bob
with an SRTP offer, Bob's RTP-only phone will reject the media
stream (with an empty "m=" line) because Bob's phone doesn't
understand SRTP (RTP/SAVP). Alice's phone will see this rejected
media stream and may terminate the entire call (BYE) and
re-initiate the call as RTP-only, or Alice's phone may decide to
continue with call setup with the SRTP-capable leg (the voice mail
system). If Alice's phone decided to re-initiate the call as RTP-
only, and Bob doesn't answer his phone, Alice will then leave
voice mail using only RTP, rather than SRTP as expected.
Currently, several techniques are commonly considered as candidates
to provide opportunistic encryption:
multipart/alternative
[MULTIPART] describes how to form a multipart/alternative body
part in SIP. The significant issues with this technique are (1)
that multipart MIME is incompatible with existing SIP proxies,
firewalls, Session Border Controllers, and endpoints and (2) when
forking, the Heterogeneous Error Response Forking Problem (HERFP)
[RFC3326] causes problems if such non-multipart-capable endpoints
were involved in the forking.
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session attribute
With this technique, the endpoints signal their desire to do SRTP
by signaling RTP (RTP/AVP), and using an attribute ("a=") in the
SDP. This technique is entirely backwards compatible with
non-SRT-aware endpoints, but doesn't use the RTP/SAVP protocol
registered by SRTP [RFC3711].
SDP Capability Negotiation
SDP Capability Negotiation [SDP-CAP] provides a backwards-
compatible mechanism to allow offering both SRTP and RTP in a
single offer. This is the preferred technique.
Probing
With this technique, the endpoints first establish an RTP session
using RTP (RTP/AVP). The endpoints send probe messages, over the
media path, to determine if the remote endpoint supports their
keying technique. A disadvantage of probing is an active attacker
can interfere with probes, and until probing completes (and SRTP
is established) the media is in the clear.
The preferred technique, SDP Capability Negotiation [SDP-CAP], can be
used with all key exchange mechanisms. What remains unique is ZRTP,
which can also accomplish its best effort encryption by probing
(sending ZRTP messages over the media path) or by session attribute
(see "a=zrtp-hash" in [ZRTP]). Current implementations of ZRTP use
probing.
A.5.4. Upgrading Algorithms
It is necessary to allow upgrading SRTP encryption and hash
algorithms, as well as upgrading the cryptographic functions used for
the key exchange mechanism. With SIP's offer/answer model, this can
be computationally expensive because the offer needs to contain all
combinations of the key exchange mechanisms (all MIKEY modes, SDP
Security Descriptions), all SRTP cryptographic suites (AES-128,
AES-256) and all SRTP cryptographic hash functions (SHA-1, SHA-256)
that the offerer supports. In order to do this, the offerer has to
expend CPU resources to build an offer containing all of this
information that becomes computationally prohibitive.
Thus, it is important to keep the offerer's CPU impact fixed so that
offering multiple new SRTP encryption and hash functions incurs no
additional expense.
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The following list describes the CPU effort involved in using each
key exchange technique.
MIKEY-NULL
No significant computational expense.
MIKEY-PSK
No significant computational expense.
MIKEY-RSA
For each offered SRTP crypto suite, the offerer has to perform
RSA operation to encrypt the TGK (TEK Generation Key).
MIKEY-RSA-R
For each offered SRTP crypto suite, the offerer has to perform
public key operation to sign the MIKEY message.
MIKEY-DHSIGN
For each offered SRTP crypto suite, the offerer has to perform
Diffie-Hellman operation, and a public key operation to sign
the Diffie-Hellman output.
MIKEY-DHHMAC
For each offered SRTP crypto suite, the offerer has to perform
Diffie-Hellman operation.
MIKEYv2 in SDP
The behavior will depend on which mode is picked.
SDP Security Descriptions with SIPS
No significant computational expense.
SDP Security Descriptions with S/MIME
S/MIME requires the offerer and the answerer to encrypt the SDP
with the other's public key, and to decrypt the received SDP
with their own private key.
SDP-DH
For each offered SRTP crypto suite, the offerer has to perform
a Diffie-Hellman operation.
ZRTP
The offerer has no additional computational expense at all, as
the offer contains no information about ZRTP or might contain
"a=zrtp-hash".
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EKT
The offerer's computational expense depends entirely on the EKT
bootstrapping mechanism selected (one or more MIKEY modes or
SDP Security Descriptions).
DTLS-SRTP
The offerer has no additional computational expense at all, as
the offer contains only a fingerprint of the certificate that
will be presented in the DTLS exchange.
MIKEYv2 Inband
The behavior will depend on which mode is picked.
Appendix B. Out-of-Scope
The compromise of an endpoint that has access to decrypted media
(e.g., SIP user agent, transcoder, recorder) is out of scope of this
document. Such a compromise might be via privilege escalation,
installation of a virus or trojan horse, or similar attacks.
B.1. Shared Key Conferencing
The consensus on the RTPSEC mailing list was to concentrate on
unicast, point-to-point sessions. Thus, there are no requirements
related to shared key conferencing. This section is retained for
informational purposes.
For efficient scaling, large audio and video conference bridges
operate most efficiently by encrypting the current speaker once and
distributing that stream to the conference attendees. Typically,
inactive participants receive the same streams -- they hear (or see)
the active speaker(s), and the active speakers receive distinct
streams that don't include themselves. In order to maintain the
confidentiality of such conferences where listeners share a common
key, all listeners must rekeyed when a listener joins or leaves a
conference.
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An important use case for mixers/translators is a conference bridge:
+----+
A --- 1 --->| |
<-- 2 ----| M |
| I |
B --- 3 --->| X |
<-- 4 ----| E |
| R |
C --- 5 --->| |
<-- 6 ----| |
+----+
Figure 3: Centralized Keying
In the figure above, 1, 3, and 5 are RTP media contributions from
Alice, Bob, and Carol, and 2, 4, and 6 are the RTP flows to those
devices carrying the "mixed" media.
Several scenarios are possible:
a. Multiple inbound sessions: 1, 3, and 5 are distinct RTP sessions,
b. Multiple outbound sessions: 2, 4, and 6 are distinct RTP
sessions,
c. Single inbound session: 1, 3, and 5 are just different sources
within the same RTP session,
d. Single outbound session: 2, 4, and 6 are different flows of the
same (multi-unicast) RTP session.
If there are multiple inbound sessions and multiple outbound sessions
(scenarios a and b), then every keying mechanism behaves as if the
mixer were an endpoint and can set up a point-to-point secure session
between the participant and the mixer. This is the simplest
situation, but is computationally wasteful, since SRTP processing has
to be done independently for each participant. The use of multiple
inbound sessions (scenario a) doesn't waste computational resources,
though it does consume additional cryptographic context on the mixer
for each participant and has the advantage of data origin
authentication.
To support a single outbound session (scenario d), the mixer has to
dictate its encryption key to the participants. Some keying
mechanisms allow the transmitter to determine its own key, and others
allow the offerer to determine the key for the offerer and answerer.
Depending on how the call is established, the offerer might be a
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participant (such as a participant dialing into a conference bridge)
or the offerer might be the mixer (such as a conference bridge
calling a participant). The use of offerless INVITEs may help some
keying mechanisms reverse the role of offerer/answerer. A
difficulty, however, is knowing a priori if the role should be
reversed for a particular call. The significant advantage of a
single outbound session is the number of SRTP encryption operations
remains constant even as the number of participants increases.
However, a disadvantage is that data origin authentication is lost,
allowing any participant to spoof the sender (because all
participants know the sender's SRTP key).
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Authors' Addresses
Dan Wing (editor)
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134
USA
EMail: dwing@cisco.com
Steffen Fries
Siemens AG
Otto-Hahn-Ring 6
Munich, Bavaria 81739
Germany
EMail: steffen.fries@siemens.com
Hannes Tschofenig
Nokia Siemens Networks
Linnoitustie 6
Espoo, 02600
Finland
Phone: +358 (50) 4871445
EMail: Hannes.Tschofenig@nsn.com
URI: http://www.tschofenig.priv.at
Francois Audet
Nortel
4655 Great America Parkway
Santa Clara, CA 95054
USA
EMail: audet@nortel.com
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ERRATA
