This document describes the Secure Real-time Transport Protocol
(SRTP), a profile of the Real-time Transport Protocol (RTP), which
can provide confidentiality, message authentication, and replay
protection to the RTP traffic and to the control traffic for RTP,
RTCP (the Real-time Transport Control Protocol) [RFC3350].
SRTP provides a framework for encryption and message authentication
of RTP and RTCP streams (Section 3). SRTP defines a set of default
cryptographic transforms (Sections 4 and 5), and it allows new
transforms to be introduced in the future (Section 6). With
appropriate key management (Sections 7 and 8), SRTP is secure
(Sections 9) for unicast and multicast RTP applications (Section 11).
SRTP can achieve high throughput and low packet expansion. SRTP
proves to be a suitable protection for heterogeneous environments
(mix of wired and wireless networks). To get such features, default
transforms are described, based on an additive stream cipher for
encryption, a keyed-hash based function for message authentication,
and an "implicit" index for sequencing/synchronization based on the
RTP sequence number for SRTP and an index number for Secure RTCP
(SRTCP).
The keywords "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]. The
terminology conforms to [RFC2828] with the following exception. For
simplicity we use the term "random" throughout the document to denote
randomly or pseudo-randomly generated values. Large amounts of
random bits may be difficult to obtain, and for the security of SRTP,
pseudo-randomness is sufficient [RFC1750].
By convention, the adopted representation is the network byte order,
i.e., the left most bit (octet) is the most significant one. By XOR
we mean bitwise addition modulo 2 of binary strings, and || denotes
concatenation. In other words, if C = A || B, then the most
significant bits of C are the bits of A, and the least significant
bits of C equal the bits of B. Hexadecimal numbers are prefixed by
0x.
Baugher, et al. Standards Track [Page 3]
RFC 3711 SRTP March 2004
The word "encryption" includes also use of the NULL algorithm (which
in practice does leave the data in the clear).
With slight abuse of notation, we use the terms "message
authentication" and "authentication tag" as is common practice, even
though in some circumstances, e.g., group communication, the service
provided is actually only integrity protection and not data origin
authentication.
The security goals for SRTP are to ensure:
* the confidentiality of the RTP and RTCP payloads, and
* the integrity of the entire RTP and RTCP packets, together with
protection against replayed packets.
These security services are optional and independent from each other,
except that SRTCP integrity protection is mandatory (malicious or
erroneous alteration of RTCP messages could otherwise disrupt the
processing of the RTP stream).
Other, functional, goals for the protocol are:
* a framework that permits upgrading with new cryptographic
transforms,
* low bandwidth cost, i.e., a framework preserving RTP header
compression efficiency,
and, asserted by the pre-defined transforms:
* a low computational cost,
* a small footprint (i.e., small code size and data memory for
keying information and replay lists),
* limited packet expansion to support the bandwidth economy goal,
* independence from the underlying transport, network, and physical
layers used by RTP, in particular high tolerance to packet loss
and re-ordering.
These properties ensure that SRTP is a suitable protection scheme for
RTP/RTCP in both wired and wireless scenarios.
Baugher, et al. Standards Track [Page 4]
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Besides the above mentioned direct goals, SRTP provides for some
additional features. They have been introduced to lighten the burden
on key management and to further increase security. They include:
* A single "master key" can provide keying material for
confidentiality and integrity protection, both for the SRTP stream
and the corresponding SRTCP stream. This is achieved with a key
derivation function (see Section 4.3), providing "session keys"
for the respective security primitive, securely derived from the
master key.
* In addition, the key derivation can be configured to periodically
refresh the session keys, which limits the amount of ciphertext
produced by a fixed key, available for an adversary to
cryptanalyze.
* "Salting keys" are used to protect against pre-computation and
time-memory tradeoff attacks [MF00] [BS00].
Detailed rationale for these features can be found in Section 7.
RTP is the Real-time Transport Protocol [RFC3550]. We define SRTP as
a profile of RTP. This profile is an extension to the RTP
Audio/Video Profile [RFC3551]. Except where explicitly noted, all
aspects of that profile apply, with the addition of the SRTP security
features. Conceptually, we consider SRTP to be a "bump in the stack"
implementation which resides between the RTP application and the
transport layer. SRTP intercepts RTP packets and then forwards an
equivalent SRTP packet on the sending side, and intercepts SRTP
packets and passes an equivalent RTP packet up the stack on the
receiving side.
Secure RTCP (SRTCP) provides the same security services to RTCP as
SRTP does to RTP. SRTCP message authentication is MANDATORY and
thereby protects the RTCP fields to keep track of membership, provide
feedback to RTP senders, or maintain packet sequence counters. SRTCP
is described in Section 3.4.
Baugher, et al. Standards Track [Page 5]
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The format of an SRTP packet is illustrated in Figure 1.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
|V=2|P|X| CC |M| PT | sequence number | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| timestamp | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| synchronization source (SSRC) identifier | |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
| contributing source (CSRC) identifiers | |
| .... | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| RTP extension (OPTIONAL) | |
+>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| | payload ... | |
| | +-------------------------------+ |
| | | RTP padding | RTP pad count | |
+>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
| ~ SRTP MKI (OPTIONAL) ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| : authentication tag (RECOMMENDED) : |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| |
+- Encrypted Portion* Authenticated Portion ---+
Figure 1. The format of an SRTP packet. *Encrypted Portion is the
same size as the plaintext for the Section 4 pre-defined transforms.
The "Encrypted Portion" of an SRTP packet consists of the encryption
of the RTP payload (including RTP padding when present) of the
equivalent RTP packet. The Encrypted Portion MAY be the exact size
of the plaintext or MAY be larger. Figure 1 shows the RTP payload
including any possible padding for RTP [RFC3550].
None of the pre-defined encryption transforms uses any padding; for
these, the RTP and SRTP payload sizes match exactly. New transforms
added to SRTP (following Section 6) may require padding, and may
hence produce larger payloads. RTP provides its own padding format
(as seen in Fig. 1), which due to the padding indicator in the RTP
header has merits in terms of compactness relative to paddings using
prefix-free codes. This RTP padding SHALL be the default method for
transforms requiring padding. Transforms MAY specify other padding
methods, and MUST then specify the amount, format, and processing of
their padding. It is important to note that encryption transforms
Baugher, et al. Standards Track [Page 6]
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that use padding are vulnerable to subtle attacks, especially when
message authentication is not used [V02]. Each specification for a
new encryption transform needs to carefully consider and describe the
security implications of the padding that it uses. Message
authentication codes define their own padding, so this default does
not apply to authentication transforms.
The OPTIONAL MKI and the RECOMMENDED authentication tag are the only
fields defined by SRTP that are not in RTP. Only 8-bit alignment is
assumed.
MKI (Master Key Identifier): configurable length, OPTIONAL. The
MKI is defined, signaled, and used by key management. The
MKI identifies the master key from which the session
key(s) were derived that authenticate and/or encrypt the
particular packet. Note that the MKI SHALL NOT identify
the SRTP cryptographic context, which is identified
according to Section 3.2.3. The MKI MAY be used by key
management for the purposes of re-keying, identifying a
particular master key within the cryptographic context
(Section 3.2.1).
Authentication tag: configurable length, RECOMMENDED. The
authentication tag is used to carry message authentication
data. The Authenticated Portion of an SRTP packet
consists of the RTP header followed by the Encrypted
Portion of the SRTP packet. Thus, if both encryption and
authentication are applied, encryption SHALL be applied
before authentication on the sender side and conversely on
the receiver side. The authentication tag provides
authentication of the RTP header and payload, and it
indirectly provides replay protection by authenticating
the sequence number. Note that the MKI is not integrity
protected as this does not provide any extra protection.
Each SRTP stream requires the sender and receiver to maintain
cryptographic state information. This information is called the
"cryptographic context".
SRTP uses two types of keys: session keys and master keys. By a
"session key", we mean a key which is used directly in a
cryptographic transform (e.g., encryption or message authentication),
and by a "master key", we mean a random bit string (given by the key
management protocol) from which session keys are derived in a
Baugher, et al. Standards Track [Page 7]
RFC 3711 SRTP March 2004
cryptographically secure way. The master key(s) and other parameters
in the cryptographic context are provided by key management
mechanisms external to SRTP, see Section 8.
Transform-independent parameters are present in the cryptographic
context independently of the particular encryption or authentication
transforms that are used. The transform-independent parameters of
the cryptographic context for SRTP consist of:
* a 32-bit unsigned rollover counter (ROC), which records how many
times the 16-bit RTP sequence number has been reset to zero after
passing through 65,535. Unlike the sequence number (SEQ), which
SRTP extracts from the RTP packet header, the ROC is maintained by
SRTP as described in Section 3.3.1.
We define the index of the SRTP packet corresponding to a given
ROC and RTP sequence number to be the 48-bit quantity
i = 2^16 * ROC + SEQ.
* for the receiver only, a 16-bit sequence number s_l, which can be
thought of as the highest received RTP sequence number (see
Section 3.3.1 for its handling), which SHOULD be authenticated
since message authentication is RECOMMENDED,
* an identifier for the encryption algorithm, i.e., the cipher and
its mode of operation,
* an identifier for the message authentication algorithm,
* a replay list, maintained by the receiver only (when
authentication and replay protection are provided), containing
indices of recently received and authenticated SRTP packets,
* an MKI indicator (0/1) as to whether an MKI is present in SRTP and
SRTCP packets,
* if the MKI indicator is set to one, the length (in octets) of the
MKI field, and (for the sender) the actual value of the currently
active MKI (the value of the MKI indicator and length MUST be kept
fixed for the lifetime of the context),
* the master key(s), which MUST be random and kept secret,
Baugher, et al. Standards Track [Page 8]
RFC 3711 SRTP March 2004
* for each master key, there is a counter of the number of SRTP
packets that have been processed (sent) with that master key
(essential for security, see Sections 3.3.1 and 9),
* non-negative integers n_e, and n_a, determining the length of the
session keys for encryption, and message authentication.
In addition, for each master key, an SRTP stream MAY use the
following associated values:
* a master salt, to be used in the key derivation of session keys.
This value, when used, MUST be random, but MAY be public. Use of
master salt is strongly RECOMMENDED, see Section 9.2. A "NULL"
salt is treated as 00...0.
* an integer in the set {1,2,4,...,2^24}, the "key_derivation_rate",
where an unspecified value is treated as zero. The constraint to
be a power of 2 simplifies the session-key derivation
implementation, see Section 4.3.
* an MKI value,
* <From, To> values, specifying the lifetime for a master key,
expressed in terms of the two 48-bit index values inside whose
range (including the range end-points) the master key is valid.
For the use of <From, To>, see Section 8.1.1. <From, To> is an
alternative to the MKI and assumes that a master key is in one-
to-one correspondence with the SRTP session key on which the
<From, To> range is defined.
SRTCP SHALL by default share the crypto context with SRTP, except:
* no rollover counter and s_l-value need to be maintained as the
RTCP index is explicitly carried in each SRTCP packet,
* a separate replay list is maintained (when replay protection is
provided),
* SRTCP maintains a separate counter for its master key (even if the
master key is the same as that for SRTP, see below), as a means to
maintain a count of the number of SRTCP packets that have been
processed with that key.
Note in particular that the master key(s) MAY be shared between SRTP
and the corresponding SRTCP, if the pre-defined transforms (including
the key derivation) are used but the session key(s) MUST NOT be so
shared.
Baugher, et al. Standards Track [Page 9]
RFC 3711 SRTP March 2004
In addition, there can be cases (see Sections 8 and 9.1) where
several SRTP streams within a given RTP session, identified by their
synchronization source (SSRCs, which is part of the RTP header),
share most of the crypto context parameters (including possibly
master and session keys). In such cases, just as in the normal
SRTP/SRTCP parameter sharing above, separate replay lists and packet
counters for each stream (SSRC) MUST still be maintained. Also,
separate SRTP indices MUST then be maintained.
A summary of parameters, pre-defined transforms, and default values
for the above parameters (and other SRTP parameters) can be found in
Sections 5 and 8.2.
All encryption, authentication/integrity, and key derivation
parameters are defined in the transforms section (Section 4).
Typical examples of such parameters are block size of ciphers,
session keys, data for the Initialization Vector (IV) formation, etc.
Future SRTP transform specifications MUST include a section to list
the additional cryptographic context's parameters for that transform,
if any.
Recall that an RTP session for each participant is defined [RFC3550]
by a pair of destination transport addresses (one network address
plus a port pair for RTP and RTCP), and that a multimedia session is
defined as a collection of RTP sessions. For example, a particular
multimedia session could include an audio RTP session, a video RTP
session, and a text RTP session.
A cryptographic context SHALL be uniquely identified by the triplet
context identifier:
context id = <SSRC, destination network address, destination
transport port number>
where the destination network address and the destination transport
port are the ones in the SRTP packet. It is assumed that, when
presented with this information, the key management returns a context
with the information as described in Section 3.2.
As noted above, SRTP and SRTCP by default share the bulk of the
parameters in the cryptographic context. Thus, retrieving the crypto
context parameters for an SRTCP stream in practice may imply a
binding to the correspondent SRTP crypto context. It is up to the
implementation to assure such binding, since the RTCP port may not be
Baugher, et al. Standards Track [Page 10]
RFC 3711 SRTP March 2004
directly deducible from the RTP port only. Alternatively, the key
management may choose to provide separate SRTP- and SRTCP- contexts,
duplicating the common parameters (such as master key(s)). The
latter approach then also enables SRTP and SRTCP to use, e.g.,
distinct transforms, if so desired. Similar considerations arise
when multiple SRTP streams, forming part of one single RTP session,
share keys and other parameters.
If no valid context can be found for a packet corresponding to a
certain context identifier, that packet MUST be discarded.
The following applies to SRTP. SRTCP is described in Section 3.4.
Assuming initialization of the cryptographic context(s) has taken
place via key management, the sender SHALL do the following to
construct an SRTP packet:
1. Determine which cryptographic context to use as described in
Section 3.2.3.
2. Determine the index of the SRTP packet using the rollover counter,
the highest sequence number in the cryptographic context, and the
sequence number in the RTP packet, as described in Section 3.3.1.
3. Determine the master key and master salt. This is done using the
index determined in the previous step or the current MKI in the
cryptographic context, according to Section 8.1.
4. Determine the session keys and session salt (if they are used by
the transform) as described in Section 4.3, using master key,
master salt, key_derivation_rate, and session key-lengths in the
cryptographic context with the index, determined in Steps 2 and 3.
5. Encrypt the RTP payload to produce the Encrypted Portion of the
packet (see Section 4.1, for the defined ciphers). This step uses
the encryption algorithm indicated in the cryptographic context,
the session encryption key and the session salt (if used) found in
Step 4 together with the index found in Step 2.
6. If the MKI indicator is set to one, append the MKI to the packet.
7. For message authentication, compute the authentication tag for the
Authenticated Portion of the packet, as described in Section 4.2.
This step uses the current rollover counter, the authentication
Baugher, et al. Standards Track [Page 11]
RFC 3711 SRTP March 2004
algorithm indicated in the cryptographic context, and the session
authentication key found in Step 4. Append the authentication tag
to the packet.
8. If necessary, update the ROC as in Section 3.3.1, using the packet
index determined in Step 2.
To authenticate and decrypt an SRTP packet, the receiver SHALL do the
following:
1. Determine which cryptographic context to use as described in
Section 3.2.3.
2. Run the algorithm in Section 3.3.1 to get the index of the SRTP
packet. The algorithm uses the rollover counter and highest
sequence number in the cryptographic context with the sequence
number in the SRTP packet, as described in Section 3.3.1.
3. Determine the master key and master salt. If the MKI indicator in
the context is set to one, use the MKI in the SRTP packet,
otherwise use the index from the previous step, according to
Section 8.1.
4. Determine the session keys, and session salt (if used by the
transform) as described in Section 4.3, using master key, master
salt, key_derivation_rate and session key-lengths in the
cryptographic context with the index, determined in Steps 2 and 3.
5. For message authentication and replay protection, first check if
the packet has been replayed (Section 3.3.2), using the Replay
List and the index as determined in Step 2. If the packet is
judged to be replayed, then the packet MUST be discarded, and the
event SHOULD be logged.
Next, perform verification of the authentication tag, using the
rollover counter from Step 2, the authentication algorithm
indicated in the cryptographic context, and the session
authentication key from Step 4. If the result is "AUTHENTICATION
FAILURE" (see Section 4.2), the packet MUST be discarded from
further processing and the event SHOULD be logged.
6. Decrypt the Encrypted Portion of the packet (see Section 4.1, for
the defined ciphers), using the decryption algorithm indicated in
the cryptographic context, the session encryption key and salt (if
used) found in Step 4 with the index from Step 2.
Baugher, et al. Standards Track [Page 12]
RFC 3711 SRTP March 2004
7. Update the rollover counter and highest sequence number, s_l, in
the cryptographic context as in Section 3.3.1, using the packet
index estimated in Step 2. If replay protection is provided, also
update the Replay List as described in Section 3.3.2.
8. When present, remove the MKI and authentication tag fields from
the packet.
SRTP implementations use an "implicit" packet index for sequencing,
i.e., not all of the index is explicitly carried in the SRTP packet.
For the pre-defined transforms, the index i is used in replay
protection (Section 3.3.2), encryption (Section 4.1), message
authentication (Section 4.2), and for the key derivation (Section
4.3).
When the session starts, the sender side MUST set the rollover
counter, ROC, to zero. Each time the RTP sequence number, SEQ, wraps
modulo 2^16, the sender side MUST increment ROC by one, modulo 2^32
(see security aspects below). The sender's packet index is then
defined as
i = 2^16 * ROC + SEQ.
Receiver-side implementations use the RTP sequence number to
determine the correct index of a packet, which is the location of the
packet in the sequence of all SRTP packets. A robust approach for
the proper use of a rollover counter requires its handling and use to
be well defined. In particular, out-of-order RTP packets with
sequence numbers close to 2^16 or zero must be properly handled.
The index estimate is based on the receiver's locally maintained ROC
and s_l values. At the setup of the session, the ROC MUST be set to
zero. Receivers joining an on-going session MUST be given the
current ROC value using out-of-band signaling such as key-management
signaling. Furthermore, the receiver SHALL initialize s_l to the RTP
sequence number (SEQ) of the first observed SRTP packet (unless the
initial value is provided by out of band signaling such as key
management).
On consecutive SRTP packets, the receiver SHOULD estimate the index
as
i = 2^16 * v + SEQ,
where v is chosen from the set { ROC-1, ROC, ROC+1 } (modulo 2^32)
such that i is closest (in modulo 2^48 sense) to the value 2^16 * ROC
+ s_l (see Appendix A for pseudocode).
Baugher, et al. Standards Track [Page 13]
RFC 3711 SRTP March 2004
After the packet has been processed and authenticated (when enabled
for SRTP packets for the session), the receiver MUST use v to
conditionally update its s_l and ROC variables as follows. If
v=(ROC-1) mod 2^32, then there is no update to s_l or ROC. If v=ROC,
then s_l is set to SEQ if and only if SEQ is larger than the current
s_l; there is no change to ROC. If v=(ROC+1) mod 2^32, then s_l is
set to SEQ and ROC is set to v.
After a re-keying occurs (changing to a new master key), the rollover
counter always maintains its sequence of values, i.e., it MUST NOT be
reset to zero.
As the rollover counter is 32 bits long and the sequence number is 16
bits long, the maximum number of packets belonging to a given SRTP
stream that can be secured with the same key is 2^48 using the pre-
defined transforms. After that number of SRTP packets have been sent
with a given (master or session) key, the sender MUST NOT send any
more packets with that key. (There exists a similar limit for SRTCP,
which in practice may be more restrictive, see Section 9.2.) This
limitation enforces a security benefit by providing an upper bound on
the amount of traffic that can pass before cryptographic keys are
changed. Re-keying (see Section 8.1) MUST be triggered, before this
amount of traffic, and MAY be triggered earlier, e.g., for increased
security and access control to media. Recurring key derivation by
means of a non-zero key_derivation_rate (see Section 4.3), also gives
stronger security but does not change the above absolute maximum
value.
On the receiver side, there is a caveat to updating s_l and ROC: if
message authentication is not present, neither the initialization of
s_l, nor the ROC update can be made completely robust. The
receiver's "implicit index" approach works for the pre-defined
transforms as long as the reorder and loss of the packets are not too
great and bit-errors do not occur in unfortunate ways. In
particular, 2^15 packets would need to be lost, or a packet would
need to be 2^15 packets out of sequence before synchronization is
lost. Such drastic loss or reorder is likely to disrupt the RTP
application itself.
The algorithm for the index estimate and ROC update is a matter of
implementation, and should take into consideration the environment
(e.g., packet loss rate) and the cases when synchronization is likely
to be lost, e.g., when the initial sequence number (randomly chosen
by RTP) is not known in advance (not sent in the key management
protocol) but may be near to wrap modulo 2^16.
Baugher, et al. Standards Track [Page 14]
RFC 3711 SRTP March 2004
A more elaborate and more robust scheme than the one given above is
the handling of RTP's own "rollover counter", see Appendix A.1 of
[RFC3550].
Secure replay protection is only possible when integrity protection
is present. It is RECOMMENDED to use replay protection, both for RTP
and RTCP, as integrity protection alone cannot assure security
against replay attacks.
A packet is "replayed" when it is stored by an adversary, and then
re-injected into the network. When message authentication is
provided, SRTP protects against such attacks through a Replay List.
Each SRTP receiver maintains a Replay List, which conceptually
contains the indices of all of the packets which have been received
and authenticated. In practice, the list can use a "sliding window"
approach, so that a fixed amount of storage suffices for replay
protection. Packet indices which lag behind the packet index in the
context by more than SRTP-WINDOW-SIZE can be assumed to have been
received, where SRTP-WINDOW-SIZE is a receiver-side, implementation-
dependent parameter and MUST be at least 64, but which MAY be set to
a higher value.
The receiver checks the index of an incoming packet against the
replay list and the window. Only packets with index ahead of the
window, or, inside the window but not already received, SHALL be
accepted.
After the packet has been authenticated (if necessary the window is
first moved ahead), the replay list SHALL be updated with the new
index.
The Replay List can be efficiently implemented by using a bitmap to
represent which packets have been received, as described in the
Security Architecture for IP [RFC2401].
Secure RTCP follows the definition of Secure RTP. SRTCP adds three
mandatory new fields (the SRTCP index, an "encrypt-flag", and the
authentication tag) and one optional field (the MKI) to the RTCP
packet definition. The three mandatory fields MUST be appended to an
RTCP packet in order to form an equivalent SRTCP packet. The added
fields follow any other profile-specific extensions.
Baugher, et al. Standards Track [Page 15]
RFC 3711 SRTP March 2004
According to Section 6.1 of [RFC3550], there is a REQUIRED packet
format for compound packets. SRTCP MUST be given packets according
to that requirement in the sense that the first part MUST be a sender
report or a receiver report. However, the RTCP encryption prefix (a
random 32-bit quantity) specified in that Section MUST NOT be used
since, as is stated there, it is only applicable to the encryption
method specified in [RFC3550] and is not needed by the cryptographic
mechanisms used in SRTP.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
|V=2|P| RC | PT=SR or RR | length | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| SSRC of sender | |
+>+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
| ~ sender info ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| ~ report block 1 ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| ~ report block 2 ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| ~ ... ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| |V=2|P| SC | PT=SDES=202 | length | |
| +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
| | SSRC/CSRC_1 | |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| ~ SDES items ~ |
| +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
| ~ ... ~ |
+>+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
| |E| SRTCP index | |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
| ~ SRTCP MKI (OPTIONAL) ~ |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| : authentication tag : |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| |
+-- Encrypted Portion Authenticated Portion -----+
Figure 2. An example of the format of a Secure RTCP packet,
consisting of an underlying RTCP compound packet with a Sender Report
and SDES packet.
Baugher, et al. Standards Track [Page 16]
RFC 3711 SRTP March 2004
The Encrypted Portion of an SRTCP packet consists of the encryption
(Section 4.1) of the RTCP payload of the equivalent compound RTCP
packet, from the first RTCP packet, i.e., from the ninth (9) octet to
the end of the compound packet. The Authenticated Portion of an
SRTCP packet consists of the entire equivalent (eventually compound)
RTCP packet, the E flag, and the SRTCP index (after any encryption
has been applied to the payload).
The added fields are:
E-flag: 1 bit, REQUIRED
The E-flag indicates if the current SRTCP packet is
encrypted or unencrypted. Section 9.1 of [RFC3550] allows
the split of a compound RTCP packet into two lower-layer
packets, one to be encrypted and one to be sent in the
clear. The E bit set to "1" indicates encrypted packet, and
"0" indicates non-encrypted packet.
SRTCP index: 31 bits, REQUIRED
The SRTCP index is a 31-bit counter for the SRTCP packet.
The index is explicitly included in each packet, in contrast
to the "implicit" index approach used for SRTP. The SRTCP
index MUST be set to zero before the first SRTCP packet is
sent, and MUST be incremented by one, modulo 2^31, after
each SRTCP packet is sent. In particular, after a re-key,
the SRTCP index MUST NOT be reset to zero again.
Authentication Tag: configurable length, REQUIRED
The authentication tag is used to carry message
authentication data.
MKI: configurable length, OPTIONAL
The MKI is the Master Key Indicator, and functions according
to the MKI definition in Section 3.
SRTCP uses the cryptographic context parameters and packet processing
of SRTP by default, with the following changes:
* The receiver does not need to "estimate" the index, as it is
explicitly signaled in the packet.
* Pre-defined SRTCP encryption is as specified in Section 4.1, but
using the definition of the SRTCP Encrypted Portion given in this
section, and using the SRTCP index as the index i. The encryption
transform and related parameters SHALL by default be the same
selected for the protection of the associated SRTP stream(s),
while the NULL algorithm SHALL be applied to the RTCP packets not
to be encrypted. SRTCP may have a different encryption transform
Baugher, et al. Standards Track [Page 17]
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than the one used by the corresponding SRTP. The expected use for
this feature is when the former has NULL-encryption and the latter
has a non NULL-encryption.
The E-flag is assigned a value by the sender depending on whether the
packet was encrypted or not.
* SRTCP decryption is performed as in Section 4, but only if the E
flag is equal to 1. If so, the Encrypted Portion is decrypted,
using the SRTCP index as the index i. In case the E-flag is 0,
the payload is simply left unmodified.
* SRTCP replay protection is as defined in Section 3.3.2, but using
the SRTCP index as the index i and a separate Replay List that is
specific to SRTCP.
* The pre-defined SRTCP authentication tag is specified as in
Section 4.2, but with the Authenticated Portion of the SRTCP
packet given in this section (which includes the index). The
authentication transform and related parameters (e.g., key size)
SHALL by default be the same as selected for the protection of the
associated SRTP stream(s).
* In the last step of the processing, only the sender needs to
update the value of the SRTCP index by incrementing it modulo 2^31
and for security reasons the sender MUST also check the number of
SRTCP packets processed, see Section 9.2.
Message authentication for RTCP is REQUIRED, as it is the control
protocol (e.g., it has a BYE packet) for RTP.
Precautions must be taken so that the packet expansion in SRTCP (due
to the added fields) does not cause SRTCP messages to use more than
their share of RTCP bandwidth. To avoid this, the following two
measures MUST be taken:
1. When initializing the RTCP variable "avg_rtcp_size" defined in
chapter 6.3 of [RFC3550], it MUST include the size of the fields
that will be added by SRTCP (index, E-bit, authentication tag, and
when present, the MKI).
2. When updating the "avg_rtcp_size" using the variable "packet_size"
(section 6.3.3 of [RFC3550]), the value of "packet_size" MUST
include the size of the additional fields added by SRTCP.
Baugher, et al. Standards Track [Page 18]
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With these measures in place the SRTCP messages will not use more
than the allotted bandwidth. The effect of the size of the added
fields on the SRTCP traffic will be that messages will be sent with
longer packet intervals. The increase in the intervals will be
directly proportional to size of the added fields. For the pre-
defined transforms, the size of the added fields will be at least 14
octets, and upper bounded depending on MKI and the authentication tag
sizes.
While there are numerous encryption and message authentication
algorithms that can be used in SRTP, below we define default
algorithms in order to avoid the complexity of specifying the
encodings for the signaling of algorithm and parameter identifiers.
The defined algorithms have been chosen as they fulfill the goals
listed in Section 2. Recommendations on how to extend SRTP with new
transforms are given in Section 6.
The following parameters are common to both pre-defined, non-NULL,
encryption transforms specified in this section.
* BLOCK_CIPHER-MODE indicates the block cipher used and its mode of
operation
* n_b is the bit-size of the block for the block cipher
* k_e is the session encryption key
* n_e is the bit-length of k_e
* k_s is the session salting key
* n_s is the bit-length of k_s
* SRTP_PREFIX_LENGTH is the octet length of the keystream prefix, a
non-negative integer, specified by the message authentication code
in use.
The distinct session keys and salts for SRTP/SRTCP are by default
derived as specified in Section 4.3.
The encryption transforms defined in SRTP map the SRTP packet index
and secret key into a pseudo-random keystream segment. Each
keystream segment encrypts a single RTP packet. The process of
encrypting a packet consists of generating the keystream segment
corresponding to the packet, and then bitwise exclusive-oring that
keystream segment onto the payload of the RTP packet to produce the
Encrypted Portion of the SRTP packet. In case the payload size is
not an integer multiple of n_b bits, the excess (least significant)
bits of the keystream are simply discarded. Decryption is done the
same way, but swapping the roles of the plaintext and ciphertext.
Baugher, et al. Standards Track [Page 19]
RFC 3711 SRTP March 2004
+----+ +------------------+---------------------------------+
| KG |-->| Keystream Prefix | Keystream Suffix |---+
+----+ +------------------+---------------------------------+ |
|
+---------------------------------+ v
| Payload of RTP Packet |->(*)
+---------------------------------+ |
|
+---------------------------------+ |
| Encrypted Portion of SRTP Packet|<--+
+---------------------------------+
Figure 3: Default SRTP Encryption Processing. Here KG denotes the
keystream generator, and (*) denotes bitwise exclusive-or.
The definition of how the keystream is generated, given the index,
depends on the cipher and its mode of operation. Below, two such
keystream generators are defined. The NULL cipher is also defined,
to be used when encryption of RTP is not required.
The SRTP definition of the keystream is illustrated in Figure 3. The
initial octets of each keystream segment MAY be reserved for use in a
message authentication code, in which case the keystream used for
encryption starts immediately after the last reserved octet. The
initial reserved octets are called the "keystream prefix" (not to be
confused with the "encryption prefix" of [RFC3550, Section 6.1]), and
the remaining octets are called the "keystream suffix". The
keystream prefix MUST NOT be used for encryption. The process is
illustrated in Figure 3.
The number of octets in the keystream prefix is denoted as
SRTP_PREFIX_LENGTH. The keystream prefix is indicated by a positive,
non-zero value of SRTP_PREFIX_LENGTH. This means that, even if
confidentiality is not to be provided, the keystream generator output
may still need to be computed for packet authentication, in which
case the default keystream generator (mode) SHALL be used.
The default cipher is the Advanced Encryption Standard (AES) [AES],
and we define two modes of running AES, (1) Segmented Integer Counter
Mode AES and (2) AES in f8-mode. In the remainder of this section,
let E(k,x) be AES applied to key k and input block x.
Baugher, et al. Standards Track [Page 20]
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Conceptually, counter mode [AES-CTR] consists of encrypting
successive integers. The actual definition is somewhat more
complicated, in order to randomize the starting point of the integer
sequence. Each packet is encrypted with a distinct keystream
segment, which SHALL be computed as follows.
A keystream segment SHALL be the concatenation of the 128-bit output
blocks of the AES cipher in the encrypt direction, using key k = k_e,
in which the block indices are in increasing order. Symbolically,
each keystream segment looks like
E(k, IV) || E(k, IV + 1 mod 2^128) || E(k, IV + 2 mod 2^128) ...
where the 128-bit integer value IV SHALL be defined by the SSRC, the
SRTP packet index i, and the SRTP session salting key k_s, as below.
IV = (k_s * 2^16) XOR (SSRC * 2^64) XOR (i * 2^16)
Each of the three terms in the XOR-sum above is padded with as many
leading zeros as needed to make the operation well-defined,
considered as a 128-bit value.
The inclusion of the SSRC allows the use of the same key to protect
distinct SRTP streams within the same RTP session, see the security
caveats in Section 9.1.
In the case of SRTCP, the SSRC of the first header of the compound
packet MUST be used, i SHALL be the 31-bit SRTCP index and k_e, k_s
SHALL be replaced by the SRTCP encryption session key and salt.
Note that the initial value, IV, is fixed for each packet and is
formed by "reserving" 16 zeros in the least significant bits for the
purpose of the counter. The number of blocks of keystream generated
for any fixed value of IV MUST NOT exceed 2^16 to avoid keystream
re-use, see below. The AES has a block size of 128 bits, so 2^16
output blocks are sufficient to generate the 2^23 bits of keystream
needed to encrypt the largest possible RTP packet (except for IPv6
"jumbograms" [RFC2675], which are not likely to be used for RTP-based
multimedia traffic). This restriction on the maximum bit-size of the
packet that can be encrypted ensures the security of the encryption
method by limiting the effectiveness of probabilistic attacks [BDJR].
For a particular Counter Mode key, each IV value used as an input
MUST be distinct, in order to avoid the security exposure of a two-
time pad situation (Section 9.1). To satisfy this constraint, an
implementation MUST ensure that the combination of the SRTP packet
Baugher, et al. Standards Track [Page 21]
RFC 3711 SRTP March 2004
index of ROC || SEQ, and the SSRC used in the construction of the IV
are distinct for any particular key. The failure to ensure this
uniqueness could be catastrophic for Secure RTP. This is in contrast
to the situation for RTP itself, which may be able to tolerate such
failures. It is RECOMMENDED that, if a dedicated security module is
present, the RTP sequence numbers and SSRC either be generated or
checked by that module (i.e., sequence-number and SSRC processing in
an SRTP system needs to be protected as well as the key).
To encrypt UMTS (Universal Mobile Telecommunications System, as 3G
networks) data, a solution (see [f8-a] [f8-b]) known as the f8-
algorithm has been developed. On a high level, the proposed scheme
is a variant of Output Feedback Mode (OFB) [HAC], with a more
elaborate initialization and feedback function. As in normal OFB,
the core consists of a block cipher. We also define here the use of
AES as a block cipher to be used in what we shall call "f8-mode of
operation" RTP encryption. The AES f8-mode SHALL use the same
default sizes for session key and salt as AES counter mode.
Figure 4 shows the structure of block cipher, E, running in f8-mode.
Baugher, et al. Standards Track [Page 22]
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IV
|
v
+------+
| |
+--->| E |
| +------+
| |
m -> (*) +-----------+-------------+-- ... ------+
| IV' | | | |
| | j=1 -> (*) j=2 -> (*) ... j=L-1 ->(*)
| | | | |
| | +-> (*) +-> (*) ... +-> (*)
| | | | | | | |
| v | v | v | v
| +------+ | +------+ | +------+ | +------+
k_e ---+--->| E | | | E | | | E | | | E |
| | | | | | | | | | |
+------+ | +------+ | +------+ | +------+
| | | | | | |
+------+ +--------+ +-- ... ----+ |
| | | |
v v v v
S(0) S(1) S(2) . . . S(L-1)
Figure 4. f8-mode of operation (asterisk, (*), denotes bitwise XOR).
The figure represents the KG in Figure 3, when AES-f8 is used.
The Initialization Vector (IV) SHALL be determined as described in
Section 4.1.2.2 (and in Section 4.1.2.3 for SRTCP).
Let IV', S(j), and m denote n_b-bit blocks. The keystream,
S(0) ||... || S(L-1), for an N-bit message SHALL be defined by
setting IV' = E(k_e XOR m, IV), and S(-1) = 00..0. For
j = 0,1,..,L-1 where L = N/n_b (rounded up to nearest integer if it
is not already an integer) compute
S(j) = E(k_e, IV' XOR j XOR S(j-1))
Notice that the IV is not used directly. Instead it is fed through E
under another key to produce an internal, "masked" value (denoted
IV') to prevent an attacker from gaining known input/output pairs.
Baugher, et al. Standards Track [Page 23]
RFC 3711 SRTP March 2004
The role of the internal counter, j, is to prevent short keystream
cycles. The value of the key mask m SHALL be
m = k_s || 0x555..5,
i.e., the session salting key, appended by the binary pattern 0101..
to fill out the entire desired key size, n_e.
The sender SHOULD NOT generate more than 2^32 blocks, which is
sufficient to generate 2^39 bits of keystream. Unlike counter mode,
there is no absolute threshold above (below) which f8 is guaranteed
to be insecure (secure). The above bound has been chosen to limit,
with sufficient security margin, the probability of degenerative
behavior in the f8 keystream generation.
The purpose of the following IV formation is to provide a feature
which we call implicit header authentication (IHA), see Section 9.5.
The SRTP IV for 128-bit block AES-f8 SHALL be formed in the following
way:
IV = 0x00 || M || PT || SEQ || TS || SSRC || ROC
M, PT, SEQ, TS, SSRC SHALL be taken from the RTP header; ROC is from
the cryptographic context.
The presence of the SSRC as part of the IV allows AES-f8 to be used
when a master key is shared between multiple streams within the same
RTP session, see Section 9.1.
The SRTCP IV for 128-bit block AES-f8 SHALL be formed in the
following way:
IV= 0..0 || E || SRTCP index || V || P || RC || PT || length || SSRC
where V, P, RC, PT, length, SSRC SHALL be taken from the first header
in the RTCP compound packet. E and SRTCP index are the 1-bit and
31-bit fields added to the packet.
Baugher, et al. Standards Track [Page 24]
RFC 3711 SRTP March 2004
The NULL cipher is used when no confidentiality for RTP/RTCP is
requested. The keystream can be thought of as "000..0", i.e., the
encryption SHALL simply copy the plaintext input into the ciphertext
output.
Throughout this section, M will denote data to be integrity
protected. In the case of SRTP, M SHALL consist of the Authenticated
Portion of the packet (as specified in Figure 1) concatenated with
the ROC, M = Authenticated Portion || ROC; in the case of SRTCP, M
SHALL consist of the Authenticated Portion (as specified in Figure 2)
only.
Common parameters:
* AUTH_ALG is the authentication algorithm
* k_a is the session message authentication key
* n_a is the bit-length of the authentication key
* n_tag is the bit-length of the output authentication tag
* SRTP_PREFIX_LENGTH is the octet length of the keystream prefix as
defined above, a parameter of AUTH_ALG
The distinct session authentication keys for SRTP/SRTCP are by
default derived as specified in Section 4.3.
The values of n_a, n_tag, and SRTP_PREFIX_LENGTH MUST be fixed for
any particular fixed value of the key.
We describe the process of computing authentication tags as follows.
The sender computes the tag of M and appends it to the packet. The
SRTP receiver verifies a message/authentication tag pair by computing
a new authentication tag over M using the selected algorithm and key,
and then compares it to the tag associated with the received message.
If the two tags are equal, then the message/tag pair is valid;
otherwise, it is invalid and the error audit message "AUTHENTICATION
FAILURE" MUST be returned.
The pre-defined authentication transform for SRTP is HMAC-SHA1
[RFC2104]. With HMAC-SHA1, the SRTP_PREFIX_LENGTH (Figure 3) SHALL
be 0. For SRTP (respectively SRTCP), the HMAC SHALL be applied to
the session authentication key and M as specified above, i.e.,
HMAC(k_a, M). The HMAC output SHALL then be truncated to the n_tag
left-most bits.
Baugher, et al. Standards Track [Page 25]
RFC 3711 SRTP March 2004
Regardless of the encryption or message authentication transform that
is employed (it may be an SRTP pre-defined transform or newly
introduced according to Section 6), interoperable SRTP
implementations MUST use the SRTP key derivation to generate session
keys. Once the key derivation rate is properly signaled at the start
of the session, there is no need for extra communication between the
parties that use SRTP key derivation.
packet index ---+
|
v
+-----------+ master +--------+ session encr_key
| ext | key | |---------->
| key mgmt |-------->| key | session auth_key
| (optional | | deriv |---------->
| rekey) |-------->| | session salt_key
| | master | |---------->
+-----------+ salt +--------+
Figure 5: SRTP key derivation.
At least one initial key derivation SHALL be performed by SRTP, i.e.,
the first key derivation is REQUIRED. Further applications of the
key derivation MAY be performed, according to the
"key_derivation_rate" value in the cryptographic context. The key
derivation function SHALL initially be invoked before the first
packet and then, when r > 0, a key derivation is performed whenever
index mod r equals zero. This can be thought of as "refreshing" the
session keys. The value of "key_derivation_rate" MUST be kept fixed
for the lifetime of the associated master key.
Interoperable SRTP implementations MAY also derive session salting
keys for encryption transforms, as is done in both of the pre-
defined transforms.
Let m and n be positive integers. A pseudo-random function family is
a set of keyed functions {PRF_n(k,x)} such that for the (secret)
random key k, given m-bit x, PRF_n(k,x) is an n-bit string,
computationally indistinguishable from random n-bit strings, see
[HAC]. For the purpose of key derivation in SRTP, a secure PRF with
m = 128 (or more) MUST be used, and a default PRF transform is
defined in Section 4.3.3.
Baugher, et al. Standards Track [Page 26]
RFC 3711 SRTP March 2004
Let "a DIV t" denote integer division of a by t, rounded down, and
with the convention that "a DIV 0 = 0" for all a. We also make the
convention of treating "a DIV t" as a bit string of the same length
as a, and thus "a DIV t" will in general have leading zeros.
Key derivation SHALL be defined as follows in terms of <label>, an
8-bit constant (see below), master_salt and key_derivation_rate, as
determined in the cryptographic context, and index, the packet index
(i.e., the 48-bit ROC || SEQ for SRTP):
* Let r = index DIV key_derivation_rate (with DIV as defined above).
* Let key_id = <label> || r.
* Let x = key_id XOR master_salt, where key_id and master_salt are
aligned so that their least significant bits agree (right-
alignment).
<label> MUST be unique for each type of key to be derived. We
currently define <label> 0x00 to 0x05 (see below), and future
extensions MAY specify new values in the range 0x06 to 0xff for other
purposes. The n-bit SRTP key (or salt) for this packet SHALL then be
derived from the master key, k_master as follows:
PRF_n(k_master, x).
(The PRF may internally specify additional formatting and padding of
x, see e.g., Section 4.3.3 for the default PRF.)
The session keys and salt SHALL now be derived using:
- k_e (SRTP encryption): <label> = 0x00, n = n_e.
- k_a (SRTP message authentication): <label> = 0x01, n = n_a.
- k_s (SRTP salting key): <label> = 0x02, n = n_s.
where n_e, n_s, and n_a are from the cryptographic context.
The master key and master salt MUST be random, but the master salt
MAY be public.
Note that for a key_derivation_rate of 0, the application of the key
derivation SHALL take place exactly once.
The definition of DIV above is purely for notational convenience.
For a non-zero t among the set of allowed key derivation rates, "a
DIV t" can be implemented as a right-shift by the base-2 logarithm of
Baugher, et al. Standards Track [Page 27]
RFC 3711 SRTP March 2004
t. The derivation operation is further facilitated if the rates are
chosen to be powers of 256, but that granularity was considered too
coarse to be a requirement of this specification.
The upper limit on the number of packets that can be secured using
the same master key (see Section 9.2) is independent of the key
derivation.
SRTCP SHALL by default use the same master key (and master salt) as
SRTP. To do this securely, the following changes SHALL be done to
the definitions in Section 4.3.1 when applying session key derivation
for SRTCP.
Replace the SRTP index by the 32-bit quantity: 0 || SRTCP index
(i.e., excluding the E-bit, replacing it with a fixed 0-bit), and use
<label> = 0x03 for the SRTCP encryption key, <label> = 0x04 for the
SRTCP authentication key, and, <label> = 0x05 for the SRTCP salting
key.
The currently defined PRF, keyed by 128, 192, or 256 bit master key,
has input block size m = 128 and can produce n-bit outputs for n up
to 2^23. PRF_n(k_master,x) SHALL be AES in Counter Mode as described
in Section 4.1.1, applied to key k_master, and IV equal to (x*2^16),
and with the output keystream truncated to the n first (left-most)
bits. (Requiring n/128, rounded up, applications of AES.)
The default transforms also are mandatory-to-implement transforms in
SRTP. Of course, "mandatory-to-implement" does not imply
"mandatory-to-use". Table 1 summarizes the pre-defined transforms.
The default values below are valid for the pre-defined transforms.
mandatory-to-impl. optional default
encryption AES-CM, NULL AES-f8 AES-CM
message integrity HMAC-SHA1 - HMAC-SHA1
key derivation (PRF) AES-CM - AES-CM
Table 1: Mandatory-to-implement, optional and default transforms in
SRTP and SRTCP.
Baugher, et al. Standards Track [Page 28]
RFC 3711 SRTP March 2004
AES running in Segmented Integer Counter Mode, as defined in Section
4.1.1, SHALL be the default encryption algorithm. The default key
lengths SHALL be 128-bit for the session encryption key (n_e). The
default session salt key-length (n_s) SHALL be 112 bits.
The NULL cipher SHALL also be mandatory-to-implement.
HMAC-SHA1, as defined in Section 4.2.1, SHALL be the default message
authentication code. The default session authentication key-length
(n_a) SHALL be 160 bits, the default authentication tag length
(n_tag) SHALL be 80 bits, and the SRTP_PREFIX_LENGTH SHALL be zero
for HMAC-SHA1. In addition, for SRTCP, the pre-defined HMAC-SHA1
MUST NOT be applied with a value of n_tag, nor n_a, that are smaller
than these defaults. For SRTP, smaller values are NOT RECOMMENDED,
but MAY be used after careful consideration of the issues in Section
7.5 and 9.5.
The AES Counter Mode based key derivation and PRF defined in Sections
4.3.1 to 4.3.3, using a 128-bit master key, SHALL be the default
method for generating session keys. The default master salt length
SHALL be 112 bits and the default key-derivation rate SHALL be zero.
Section 4 provides examples of the level of detail needed for
defining transforms. Whenever a new transform is to be added to
SRTP, a companion standard track RFC MUST be written to exactly
define how the new transform can be used with SRTP (and SRTCP). Such
a companion RFC SHOULD avoid overlap with the SRTP protocol document.
Note however, that it MAY be necessary to extend the SRTP or SRTCP
cryptographic context definition with new parameters (including fixed
or default values), add steps to the packet processing, or even add
fields to the SRTP/SRTCP packets. The companion RFC SHALL explain
any known issues regarding interactions between the transform and
other aspects of SRTP.
Each new transform document SHOULD specify its key attributes, e.g.,
size of keys (minimum, maximum, recommended), format of keys,
recommended/required processing of input keying material,
requirements/recommendations on key lifetime, re-keying and key
derivation, whether sharing of keys between SRTP and SRTCP is allowed
or not, etc.
Baugher, et al. Standards Track [Page 29]
RFC 3711 SRTP March 2004
An added message integrity transform SHOULD define a minimum
acceptable key/tag size for SRTCP, equivalent in strength to the
minimum values as defined in Section 5.2.
Key derivation reduces the burden on the key establishment. As many
as six different keys are needed per crypto context (SRTP and SRTCP
encryption keys and salts, SRTP and SRTCP authentication keys), but
these are derived from a single master key in a cryptographically
secure way. Thus, the key management protocol needs to exchange only
one master key (plus master salt when required), and then SRTP itself
derives all the necessary session keys (via the first, mandatory
application of the key derivation function).
Multiple applications of the key derivation function are optional,
but will give security benefits when enabled. They prevent an
attacker from obtaining large amounts of ciphertext produced by a
single fixed session key. If the attacker was able to collect a
large amount of ciphertext for a certain session key, he might be
helped in mounting certain attacks.
Multiple applications of the key derivation function provide
backwards and forward security in the sense that a compromised
session key does not compromise other session keys derived from the
same master key. This means that the attacker who is able to recover
a certain session key, is anyway not able to have access to messages
secured under previous and later session keys (derived from the same
master key). (Note that, of course, a leaked master key reveals all
the session keys derived from it.)
Considerations arise with high-rate key refresh, especially in large
multicast settings, see Section 11.
The master salt guarantees security against off-line key-collision
attacks on the key derivation that might otherwise reduce the
effective key size [MF00].
Baugher, et al. Standards Track [Page 30]
RFC 3711 SRTP March 2004
The derived session salting key used in the encryption, has been
introduced to protect against some attacks on additive stream
ciphers, see Section 9.2. The explicit inclusion method of the salt
in the IV has been selected for ease of hardware implementation.
The particular definition of the keystream given in Section 4.1 (the
keystream prefix) is to give provision for particular universal hash
functions, suitable for message authentication in the Wegman-Carter
paradigm [WC81]. Such functions are provably secure, simple, quick,
and especially appropriate for Digital Signal Processors and other
processors with a fast multiply operation.
No authentication transforms are currently provided in SRTP other
than HMAC-SHA1. Future transforms, like the above mentioned
universal hash functions, MAY be added following the guidelines in
Section 6.
Note that in pair-wise communications, integrity and data origin
authentication are provided together. However, in group scenarios
where the keys are shared between members, the MAC tag only proves
that a member of the group sent the packet, but does not prevent
against a member impersonating another. Data origin authentication
(DOA) for multicast and group RTP sessions is a hard problem that
needs a solution; while some promising proposals are being
investigated [PCST1] [PCST2], more work is needed to rigorously
specify these technologies. Thus SRTP data origin authentication in
groups is for further study.
DOA can be done otherwise using signatures. However, this has high
impact in terms of bandwidth and processing time, therefore we do not
offer this form of authentication in the pre-defined packet-integrity
transform.
The presence of mixers and translators does not allow data origin
authentication in case the RTP payload and/or the RTP header are
manipulated. Note that these types of middle entities also disrupt
end-to-end confidentiality (as the IV formation depends e.g., on the
RTP header preservation). A certain trust model may choose to trust
the mixers/translators to decrypt/re-encrypt the media (this would
imply breaking the end-to-end security, with related security
implications).
Baugher, et al. Standards Track [Page 31]
RFC 3711 SRTP March 2004
As shown in Figure 1, the authentication tag is RECOMMENDED in SRTP.
A full 80-bit authentication-tag SHOULD be used, but a shorter tag or
even a zero-length tag (i.e., no message authentication) MAY be used
under certain conditions to support either of the following two
application environments.
1. Strong authentication can be impractical in environments where
bandwidth preservation is imperative. An important special
case is wireless communication systems, in which bandwidth is a
scarce and expensive resource. Studies have shown that for
certain applications and link technologies, additional bytes
may result in a significant decrease in spectrum efficiency
[SWO]. Considerable effort has been made to design IP header
compression techniques to improve spectrum efficiency
[RFC3095]. A typical voice application produces 20 byte
samples, and the RTP, UDP and IP headers need to be jointly
compressed to one or two bytes on average in order to obtain
acceptable wireless bandwidth economy [RFC3095]. In this case,
strong authentication would impose nearly fifty percent
overhead.
2. Authentication is impractical for applications that use data
links with fixed-width fields that cannot accommodate the
expansion due to the authentication tag. This is the case for
some important existing wireless channels. For example, zero-
byte header compression is used to adapt EVRC/SMV voice with
the legacy IS-95 bearer channel in CDMA2000 VoIP services. It
was found that not a single additional octet could be added to
the data, which motivated the creation of a zero-byte profile
for ROHC [RFC3242].
A short tag is secure for a restricted set of applications. Consider
a voice telephony application, for example, such as a G.729 audio
codec with a 20-millisecond packetization interval, protected by a
32-bit message authentication tag. The likelihood of any given
packet being successfully forged is only one in 2^32. Thus an
adversary can control no more than 20 milliseconds of audio output
during a 994-day period, on average. In contrast, the effect of a
single forged packet can be much larger if the application is
stateful. A codec that uses relative or predictive compression
across packets will propagate the maliciously generated state,
affecting a longer duration of output.
Baugher, et al. Standards Track [Page 32]
RFC 3711 SRTP March 2004
Certainly not all SRTP or telephony applications meet the criteria
for short or zero-length authentication tags. Section 9.5.1
discusses the risks of weak or no message authentication, and section
9.5 describes the circumstances when it is acceptable and when it is
unacceptable.
There are emerging key management standards [MIKEY] [KEYMGT] [SDMS]
for establishing an SRTP cryptographic context (e.g., an SRTP master
key). Both proprietary and open-standard key management methods are
likely to be used for telephony applications [MIKEY] [KINK] and
multicast applications [GDOI]. This section provides guidance for
key management systems that service SRTP session.
For initialization, an interoperable SRTP implementation SHOULD be
given the SSRC and MAY be given the initial RTP sequence number for
the RTP stream by key management (thus, key management has a
dependency on RTP operational parameters). Sending the RTP sequence
number in the key management may be useful e.g., when the initial
sequence number is close to wrapping (to avoid synchronization
problems), and to communicate the current sequence number to a
joining endpoint (to properly initialize its replay list).
If the pre-defined transforms are used, SRTP allows sharing of the
same master key between SRTP/SRTCP streams belonging to the same RTP
session.
First, sharing between SRTP streams belonging to the same RTP session
is secure if the design of the synchronization mechanism, i.e., the
IV, avoids keystream re-use (the two-time pad, Section 9.1). This is
taken care of by the fact that RTP provides for unique SSRCs for
streams belonging to the same RTP session. See Section 9.1 for
further discussion.
Second, sharing between SRTP and the corresponding SRTCP is secure.
The fact that an SRTP stream and its associated SRTCP stream both
carry the same SSRC does not constitute a problem for the two-time
pad due to the key derivation. Thus, SRTP and SRTCP corresponding to
one RTP session MAY share master keys (as they do by default).
Note that message authentication also has a dependency on SSRC
uniqueness that is unrelated to the problem of keystream reuse: SRTP
streams authenticated under the same key MUST have a distinct SSRC in
order to identify the sender of the message. This requirement is
needed because the SSRC is the cryptographically authenticated field
Baugher, et al. Standards Track [Page 33]
RFC 3711 SRTP March 2004
used to distinguish between different SRTP streams. Were two streams
to use identical SSRC values, then an adversary could substitute
messages from one stream into the other without detection.
SRTP/SRTCP MUST NOT share master keys under any other circumstances
than the ones given above, i.e., between SRTP and its corresponding
SRTCP, and, between streams belonging to the same RTP session.
The recommended way for a particular key management system to provide
re-key within SRTP is by associating a master key in a crypto context
with an MKI.
This provides for easy master key retrieval (see Scenarios in Section
11), but has the disadvantage of adding extra bits to each packet.
As noted in Section 7.5, some wireless links do not cater for added
bits, therefore SRTP also defines a more economic way of triggering
re-keying, via use of <From, To>, which works in some specific,
simple scenarios (see Section 8.1.1).
SRTP senders SHALL count the amount of SRTP and SRTCP traffic being
used for a master key and invoke key management to re-key if needed
(Section 9.2). These interactions are defined by the key management
interface to SRTP and are not defined by this protocol specification.
In addition to the use of the MKI, SRTP defines another optional
mechanism for master key retrieval, the <From, To>. The <From, To>
specifies the range of SRTP indices (a pair of sequence number and
ROC) within which a certain master key is valid, and is (when used)
part of the crypto context. By looking at the 48-bit SRTP index of
the current SRTP packet, the corresponding master key can be found by
determining which From-To interval it belongs to. For SRTCP, the
most recently observed/used SRTP index (which can be obtained from
the cryptographic context) is used for this purpose, even though
SRTCP has its own (31-bit) index (see caveat below).
This method, compared to the MKI, has the advantage of identifying
the master key and defining its lifetime without adding extra bits to
each packet. This could be useful, as already noted, for some
wireless links that do not cater for added bits. However, its use
SHOULD be limited to specific, very simple scenarios. We recommend
to limit its use when the RTP session is a simple unidirectional or
bi-directional stream. This is because in case of multiple streams,
it is difficult to trigger the re-key based on the <From, To> of a
single RTP stream. For example, if several streams share a master
Baugher, et al. Standards Track [Page 34]
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key, there is no simple one-to-one correspondence between the index
sequence space of a certain stream, and the index sequence space on
which the <From, To> values are based. Consequently, when a master
key is shared between streams, one of these streams MUST be
designated by key management as the one whose index space defines the
re-keying points. Also, the re-key triggering on SRTCP is based on
the correspondent SRTP stream, i.e., when the SRTP stream changes the
master key, so does the correspondent SRTCP. This becomes obviously
more and more complex with multiple streams.
The default values for the <From, To> are "from the first observed
packet" and "until further notice". However, the maximum limit of
SRTP/SRTCP packets that are sent under each given master/session key
(Section 9.2) MUST NOT be exceeded.
In case the <From, To> is used as key retrieval, then the MKI is not
inserted in the packet (and its indicator in the crypto context is
zero). However, using the MKI does not exclude using <From, To> key
lifetime simultaneously. This can for instance be useful to signal
at the sender side at which point in time an MKI is to be made
active.
The table below lists all SRTP parameters that key management can
supply. For reference, it also provides a summary of the default and
mandatory-to-support values for an SRTP implementation as described
in Section 5.
Baugher, et al. Standards Track [Page 35]
RFC 3711 SRTP March 2004
Parameter Mandatory-to-support Default
--------- -------------------- -------
SRTP and SRTCP encr transf. AES_CM, NULL AES_CM
(Other possible values: AES_f8)
SRTP and SRTCP auth transf. HMAC-SHA1 HMAC-SHA1
SRTP and SRTCP auth params:
n_tag (tag length) 80 80
SRTP prefix_length 0 0
Key derivation PRF AES_CM AES_CM
Key material params
(for each master key):
master key length 128 128
n_e (encr session key length) 128 128
n_a (auth session key length) 160 160
master salt key
length of the master salt 112 112
n_s (session salt key length) 112 112
key derivation rate 0 0
key lifetime
SRTP-packets-max-lifetime 2^48 2^48
SRTCP-packets-max-lifetime 2^31 2^31
from-to-lifetime <From, To>
MKI indicator 0 0
length of the MKI 0 0
value of the MKI
Crypto context index params:
SSRC value
ROC
SEQ
SRTCP Index
Transport address
Port number
Relation to other RTP profiles:
sender's order between FEC and SRTP FEC-SRTP FEC-SRTP
(see Section 10)
Baugher, et al. Standards Track [Page 36]
RFC 3711 SRTP March 2004
Any fixed keystream output, generated from the same key and index
MUST only be used to encrypt once. Re-using such keystream (jokingly
called a "two-time pad" system by cryptographers), can seriously
compromise security. The NSA's VENONA project [C99] provides a
historical example of such a compromise. It is REQUIRED that
automatic key management be used for establishing and maintaining
SRTP and SRTCP keying material; this requirement is to avoid
keystream reuse, which is more likely to occur with manual key
management. Furthermore, in SRTP, a "two-time pad" is avoided by
requiring the key, or some other parameter of cryptographic
significance, to be unique per RTP/RTCP stream and packet. The pre-
defined SRTP transforms accomplish packet-uniqueness by including the
packet index and stream-uniqueness by inclusion of the SSRC.
The pre-defined transforms (AES-CM and AES-f8) allow master keys to
be shared across streams belonging to the same RTP session by the
inclusion of the SSRC in the IV. A master key MUST NOT be shared
among different RTP sessions.
Thus, the SSRC MUST be unique between all the RTP streams within the
same RTP session that share the same master key. RTP itself provides
an algorithm for detecting SSRC collisions within the same RTP
session. Thus, temporary collisions could lead to temporary two-time
pad, in the unfortunate event that SSRCs collide at a point in time
when the streams also have identical sequence numbers (occurring with
probability roughly 2^(-48)). Therefore, the key management SHOULD
take care of avoiding such SSRC collisions by including the SSRCs to
be used in the session as negotiation parameters, proactively
assuring their uniqueness. This is a strong requirements in
scenarios where for example, there are multiple senders that can
start to transmit simultaneously, before SSRC collision are detected
at the RTP level.
Note also that even with distinct SSRCs, extensive use of the same
key might improve chances of probabilistic collision and time-
memory-tradeoff attacks succeeding.
As described, master keys MAY be shared between streams belonging to
the same RTP session, but it is RECOMMENDED that each SSRC have its
own master key. When master keys are shared among SSRC participants
and SSRCs are managed by a key management module as recommended
above, the RECOMMENDED policy for an SSRC collision error is for the
participant to leave the SRTP session as it is a sign of malfunction.
Baugher, et al. Standards Track [Page 37]
RFC 3711 SRTP March 2004
The effective key size is determined (upper bounded) by the size of
the master key and, for encryption, the size of the salting key. Any
additive stream cipher is vulnerable to attacks that use statistical
knowledge about the plaintext source to enable key collision and
time-memory tradeoff attacks [MF00] [H80] [BS00]. These attacks take
advantage of commonalities among plaintexts, and provide a way for a
cryptanalyst to amortize the computational effort of decryption over
many keys, or over many bytes of output, thus reducing the effective
key size of the cipher. A detailed analysis of these attacks and
their applicability to the encryption of Internet traffic is provided
in [MF00]. In summary, the effective key size of SRTP when used in a
security system in which m distinct keys are used, is equal to the
key size of the cipher less the logarithm (base two) of m.
Protection against such attacks can be provided simply by increasing
the size of the keys used, which here can be accomplished by the use
of the salting key. Note that the salting key MUST be random but MAY
be public. A salt size of (the suggested) size 112 bits protects
against attacks in scenarios where at most 2^112 keys are in use.
This is sufficient for all practical purposes.
Implementations SHOULD use keys that are as large as possible.
Please note that in many cases increasing the key size of a cipher
does not affect the throughput of that cipher.
The use of the SRTP and SRTCP indices in the pre-defined transforms
fixes the maximum number of packets that can be secured with the same
key. This limit is fixed to 2^48 SRTP packets for an SRTP stream,
and 2^31 SRTCP packets, when SRTP and SRTCP are considered
independently. Due to for example re-keying, reaching this limit may
or may not coincide with wrapping of the indices, and thus the sender
MUST keep packet counts. However, when the session keys for related
SRTP and SRTCP streams are derived from the same master key (the
default behavior, Section 4.3), the upper bound that has to be
considered is in practice the minimum of the two quantities. That
is, when 2^48 SRTP packets or 2^31 SRTCP packets have been secured
with the same key (whichever occurs before), the key management MUST
be called to provide new master key(s) (previously stored and used
keys MUST NOT be used again), or the session MUST be terminated. If
a sender of RTCP discovers that the sender of SRTP (or SRTCP) has not
updated the master or session key prior to sending 2^48 SRTP (or 2^31
SRTCP) packets belonging to the same SRTP (SRTCP) stream, it is up to
the security policy of the RTCP sender how to behave, e.g., whether
an RTCP BYE-packet should be sent and/or if the event should be
logged.
Baugher, et al. Standards Track [Page 38]
RFC 3711 SRTP March 2004
Note: in most typical applications (assuming at least one RTCP packet
for every 128,000 RTP packets), it will be the SRTCP index that first
reaches the upper limit, although the time until this occurs is very
long: even at 200 SRTCP packets/sec, the 2^31 index space of SRTCP is
enough to secure approximately 4 months of communication.
Note that if the master key is to be shared between SRTP streams
within the same RTP session (Section 9.1), although the above bounds
are on a per stream (i.e., per SSRC) basis, the sender MUST base re-
key decision on the stream whose sequence number space is the first
to be exhausted.
Key derivation limits the amount of plaintext that is encrypted with
a fixed session key, and made available to an attacker for analysis,
but key derivation does not extend the master key's lifetime. To see
this, simply consider our requirements to avoid two-time pad: two
distinct packets MUST either be processed with distinct IVs, or with
distinct session keys, and both the distinctness of IV and of the
session keys are (for the pre-defined transforms) dependent on the
distinctness of the packet indices.
Note that with the key derivation, the effective key size is at most
that of the master key, even if the derived session key is
considerably longer. With the pre-defined authentication transform,
the session authentication key is 160 bits, but the master key by
default is only 128 bits. This design choice was made to comply with
certain recommendations in [RFC2104] so that an existing HMAC
implementation can be plugged into SRTP without problems. Since the
default tag size is 80 bits, it is, for the applications in mind,
also considered acceptable from security point of view. Users having
concerns about this are RECOMMENDED to instead use a 192 bit master
key in the key derivation. It was, however, chosen not to mandate
192-bit keys since existing AES implementations to be used in the
key-derivation may not always support key-lengths other than 128
bits. Since AES is not defined (or properly analyzed) for use with
160 bit keys it is NOT RECOMMENDED that ad-hoc key-padding schemes
are used to pad shorter keys to 192 or 256 bits.
SRTP's pre-defined ciphers are "seekable" stream ciphers, i.e.,
ciphers able to efficiently seek to arbitrary locations in their
keystream (so that the encryption or decryption of one packet does
not depend on preceding packets). By using seekable stream ciphers,
SRTP avoids the denial of service attacks that are possible on stream
ciphers that lack this property. It is important to be aware that,
as with any stream cipher, the exact length of the payload is
revealed by the encryption. This means that it may be possible to
Baugher, et al. Standards Track [Page 39]
RFC 3711 SRTP March 2004
deduce certain "formatting bits" of the payload, as the length of the
codec output might vary due to certain parameter settings etc. This,
in turn, implies that the corresponding bit of the keystream can be
deduced. However, if the stream cipher is secure (counter mode and
f8 are provably secure under certain assumptions [BDJR] [KSYH] [IK]),
knowledge of a few bits of the keystream will not aid an attacker in
predicting subsequent keystream bits. Thus, the payload length (and
information deducible from this) will leak, but nothing else.
As some RTP packet could contain highly predictable data, e.g., SID,
it is important to use a cipher designed to resist known plaintext
attacks (which is the current practice).
In SRTP, RTP headers are sent in the clear to allow for header
compression. This means that data such as payload type,
synchronization source identifier, and timestamp are available to an
eavesdropper. Moreover, since RTP allows for future extensions of
headers, we cannot foresee what kind of possibly sensitive
information might also be "leaked".
SRTP is a low-cost method, which allows header compression to reduce
bandwidth. It is up to the endpoints' policies to decide about the
security protocol to employ. If one really needs to protect headers,
and is allowed to do so by the surrounding environment, then one
should also look at alternatives, e.g., IPsec [RFC2401].
SRTP messages are subject to attacks on their integrity and source
identification, and these risks are discussed in Section 9.5.1. To
protect against these attacks, each SRTP stream SHOULD be protected
by HMAC-SHA1 [RFC2104] with an 80-bit output tag and a 160-bit key,
or a message authentication code with equivalent strength. Secure
RTP SHOULD NOT be used without message authentication, except under
the circumstances described in this section. It is important to note
that encryption algorithms, including AES Counter Mode and f8, do not
provide message authentication. SRTCP MUST NOT be used with weak (or
NULL) authentication.
SRTP MAY be used with weak authentication (e.g., a 32-bit
authentication tag), or with no authentication (the NULL
authentication algorithm). These options allow SRTP to be used to
provide confidentiality in situations where
* weak or null authentication is an acceptable security risk, and
* it is impractical to provide strong message authentication.
Baugher, et al. Standards Track [Page 40]
RFC 3711 SRTP March 2004
These conditions are described below and in Section 7.5. Note that
both conditions MUST hold in order for weak or null authentication to
be used. The risks associated with exercising the weak or null
authentication options need to be considered by a security audit
prior to their use for a particular application or environment given
the risks, which are discussed in Section 9.5.1.
Weak authentication is acceptable when the RTP application is such
that the effect of a small fraction of successful forgeries is
negligible. If the application is stateless, then the effect of a
single forged RTP packet is limited to the decoding of that
particular packet. Under this condition, the size of the
authentication tag MUST ensure that only a negligible fraction of the
packets passed to the RTP application by the SRTP receiver can be
forgeries. This fraction is negligible when an adversary, if given
control of the forged packets, is not able to make a significant
impact on the output of the RTP application (see the example of
Section 7.5).
Weak or null authentication MAY be acceptable when it is unlikely
that an adversary can modify ciphertext so that it decrypts to an
intelligible value. One important case is when it is difficult for
an adversary to acquire the RTP plaintext data, since for many
codecs, an adversary that does not know the input signal cannot
manipulate the output signal in a controlled way. In many cases it
may be difficult for the adversary to determine the actual value of
the plaintext. For example, a hidden snooping device might be
required in order to know a live audio or video signal. The
adversary's signal must have a quality equivalent to or greater than
that of the signal under attack, since otherwise the adversary would
not have enough information to encode that signal with the codec used
by the victim. Plaintext prediction may also be especially difficult
for an interactive application such as a telephone call.
Weak or null authentication MUST NOT be used when the RTP application
makes data forwarding or access control decisions based on the RTP
data. In such a case, an attacker may be able to subvert
confidentiality by causing the receiver to forward data to an
attacker. See Section 3 of [B96] for a real-life example of such
attacks.
Null authentication MUST NOT be used when a replay attack, in which
an adversary stores packets then replays them later in the session,
could have a non-negligible impact on the receiver. An example of a
successful replay attack is the storing of the output of a
surveillance camera for a period of time, later followed by the
Baugher, et al. Standards Track [Page 41]
RFC 3711 SRTP March 2004
injection of that output to the monitoring station to avoid
surveillance. Encryption does not protect against this attack, and
non-null authentication is REQUIRED in order to defeat it.
If existential message forgery is an issue, i.e., when the accuracy
of the received data is of non-negligible importance, null
authentication MUST NOT be used.
During a security audit considering the use of weak or null
authentication, it is important to keep in mind the following attacks
which are possible when no message authentication algorithm is used.
An attacker who cannot predict the plaintext is still always able to
modify the message sent between the sender and the receiver so that
it decrypts to a random plaintext value, or to send a stream of bogus
packets to the receiver that will decrypt to random plaintext values.
This attack is essentially a denial of service attack, though in the
absence of message authentication, the RTP application will have
inputs that are bit-wise correlated with the true value. Some
multimedia codecs and common operating systems will crash when such
data are accepted as valid video data. This denial of service attack
may be a much larger threat than that due to an attacker dropping,
delaying, or re-ordering packets.
An attacker who cannot predict the plaintext can still replay a
previous message with certainty that the receiver will accept it.
Applications with stateless codecs might be robust against this type
of attack, but for other, more complex applications these attacks may
be far more grave.
An attacker who can predict the plaintext can modify the ciphertext
so that it will decrypt to any value of her choosing. With an
additive stream cipher, an attacker will always be able to change
individual bits.
An attacker may be able to subvert confidentiality due to the lack of
authentication when a data forwarding or access control decision is
made on decrypted but unauthenticated plaintext. This is because the
receiver may be fooled into forwarding data to an attacker, leading
to an indirect breach of confidentiality (see Section 3 of [B96]).
This is because data-forwarding decisions are made on the decrypted
plaintext; information in the plaintext will determine to what subnet
(or process) the plaintext is forwarded in ESP [RFC2401] tunnel mode
(respectively, transport mode). When Secure RTP is used without
Baugher, et al. Standards Track [Page 42]
RFC 3711 SRTP March 2004
message authentication, it should be verified that the application
does not make data forwarding or access control decisions based on
the decrypted plaintext.
Some cipher modes of operation that require padding, e.g., standard
cipher block chaining (CBC) are very sensitive to attacks on
confidentiality if certain padding types are used in the absence of
integrity. The attack [V02] shows that this is indeed the case for
the standard RTP padding as discussed in reference to Figure 1, when
used together with CBC mode. Later transform additions to SRTP MUST
therefore carefully consider the risk of using this padding without
proper integrity protection.
The IV formation of the f8-mode gives implicit authentication (IHA)
of the RTP header, even when message authentication is not used.
When IHA is used, an attacker that modifies the value of the RTP
header will cause the decryption process at the receiver to produce
random plaintext values. While this protection is not equivalent to
message authentication, it may be useful for some applications.
The default processing when using Forward Error Correction (e.g., RFC
2733) processing with SRTP SHALL be to perform FEC processing prior
to SRTP processing on the sender side and to perform SRTP processing
prior to FEC processing on the receiver side. Any change to this
ordering (reversing it, or, placing FEC between SRTP encryption and
SRTP authentication) SHALL be signaled out of band.
SRTP can be used as security protocol for the RTP/RTCP traffic in
many different scenarios. SRTP has a number of configuration
options, in particular regarding key usage, and can have impact on
the total performance of the application according to the way it is
used. Hence, the use of SRTP is dependent on the kind of scenario
and application it is used with. In the following, we briefly
illustrate some use cases for SRTP, and give some guidelines for
recommended setting of its options.
A typical example would be a voice call or video-on-demand
application.
Baugher, et al. Standards Track [Page 43]
RFC 3711 SRTP March 2004
Consider one bi-directional RTP stream, as one RTP session. It is
possible for the two parties to share the same master key in the two
directions according to the principles of Section 9.1. The first
round of the key derivation splits the master key into any or all of
the following session keys (according to the provided security
functions):
SRTP_encr_key, SRTP_auth_key, SRTCP_encr_key, and SRTCP_auth key.
(For simplicity, we omit discussion of the salts, which are also
derived.) In this scenario, it will in most cases suffice to have a
single master key with the default lifetime. This guarantees
sufficiently long lifetime of the keys and a minimum set of keys in
place for most practical purposes. Also, in this case RTCP
protection can be applied smoothly. Under these assumptions, use of
the MKI can be omitted. As the key-derivation in combination with
large difference in the packet rate in the respective directions may
require simultaneous storage of several session keys, if storage is
an issue, we recommended to use low-rate key derivation.
The same considerations can be extended to the unicast scenario with
multiple RTP sessions, where each session would have a distinct
master key.
Just as with (unprotected) RTP, a scalability issue arises in big
groups due to the possibly very large amount of SRTCP Receiver
Reports that the sender might need to process. In SRTP, the sender
may have to keep state (the cryptographic context) for each receiver,
or more precisely, for the SRTCP used to protect Receiver Reports.
The overhead increases proportionally to the size of the group. In
particular, re-keying requires special concern, see below.
Consider first a small group of receivers. There are a few possible
setups with the distribution of master keys among the receivers.
Given a single RTP session, one possibility is that the receivers
share the same master key as per Section 9.1 to secure all their
respective RTCP traffic. This shared master key could then be the
same one used by the sender to protect its outbound SRTP traffic.
Alternatively, it could be a master key shared only among the
receivers and used solely for their SRTCP traffic. Both alternatives
require the receivers to trust each other.
Considering SRTCP and key storage, it is recommended to use low-rate
(or zero) key_derivation (except the mandatory initial one), so that
the sender does not need to store too many session keys (each SRTCP
stream might otherwise have a different session key at a given point
Baugher, et al. Standards Track [Page 44]
RFC 3711 SRTP March 2004
in time, as the SRTCP sources send at different times). Thus, in
case key derivation is wanted for SRTP, the cryptographic context for
SRTP can be kept separate from the SRTCP crypto context, so that it
is possible to have a key_derivation_rate of 0 for SRTCP and a non-
zero value for SRTP.
Use of the MKI for re-keying is RECOMMENDED for most applications
(see Section 8.1).
If there are more than one SRTP/SRTCP stream (within the same RTP
session) that share the master key, the upper limit of 2^48 SRTP
packets / 2^31 SRTCP packets means that, before one of the streams
reaches its maximum number of packets, re-keying MUST be triggered on
ALL streams sharing the master key. (From strict security point of
view, only the stream reaching the maximum would need to be re-keyed,
but then the streams would no longer be sharing master key, which is
the intention.) A local policy at the sender side should force
rekeying in a way that the maximum packet limit is not reached on any
of the streams. Use of the MKI for re-keying is RECOMMENDED.
In large multicast with one sender, the same considerations as for
the small group multicast hold. The biggest issue in this scenario
is the additional load placed at the sender side, due to the state
(cryptographic contexts) that has to be maintained for each receiver,
sending back RTCP Receiver Reports. At minimum, a replay window
might need to be maintained for each RTCP source.
Re-keying may occur due to access control (e.g., when a member is
removed during a multicast RTP session), or for pure cryptographic
reasons (e.g., the key is at the end of its lifetime). When using
SRTP default transforms, the master key MUST be replaced before any
of the index spaces are exhausted for any of the streams protected by
one and the same master key.
How key management re-keys SRTP implementations is out of scope, but
it is clear that there are straightforward ways to manage keys for a
multicast group. In one-sender multicast, for example, it is
typically the responsibility of the sender to determine when a new
key is needed. The sender is the one entity that can keep track of
when the maximum number of packets has been sent, as receivers may
join and leave the session at any time, there may be packet loss and
delay etc. In scenarios other than one-sender multicast, other
methods can be used. Here, one must take into consideration that key
exchange can be a costly operation, taking several seconds for a
single exchange. Hence, some time before the master key is
exhausted/expires, out-of-band key management is initiated, resulting
Baugher, et al. Standards Track [Page 45]
RFC 3711 SRTP March 2004
in a new master key that is shared with the receiver(s). In any
event, to maintain synchronization when switching to the new key,
group policy might choose between using the MKI and the <From, To>,
as described in Section 8.1.
For access control purposes, the <From, To> periods are set at the
desired granularity, dependent on the packet rate. High rate re-
keying can be problematic for SRTCP in some large-group scenarios.
As mentioned, there are potential problems in using the SRTP index,
rather than the SRTCP index, for determining the master key. In
particular, for short periods during switching of master keys, it may
be the case that SRTCP packets are not under the current master key
of the correspondent SRTP. Therefore, using the MKI for re-keying in
such scenarios will produce better results.
The description of these scenarios highlights some recommendations on
the use of SRTP, mainly related to re-keying and large scale
multicast:
- Do not use fast re-keying with the <From, To> feature. It may, in
particular, give problems in retrieving the correct SRTCP key, if
an SRTCP packet arrives close to the re-keying time. The MKI
SHOULD be used in this case.
- If multiple SRTP streams in the same RTP session share the same
master key, also moderate rate re-keying MAY have the same
problems, and the MKI SHOULD be used.
- Though offering increased security, a non-zero key_derivation_rate
is NOT RECOMMENDED when trying to minimize the number of keys in
use with multiple streams.
The RTP specification establishes a registry of profile names for use
by higher-level control protocols, such as the Session Description
Protocol (SDP), to refer to transport methods. This profile
registers the name "RTP/SAVP".
SRTP uses cryptographic transforms which a key management protocol
signals. It is the task of each particular key management protocol
to register the cryptographic transforms or suites of transforms with
IANA. The key management protocol conveys these protocol numbers,
not SRTP, and each key management protocol chooses the numbering
scheme and syntax that it requires.
Baugher, et al. Standards Track [Page 46]
RFC 3711 SRTP March 2004
Specification of a key management protocol for SRTP is out of scope
here. Section 8.2, however, provides guidance on the parameters that
need to be defined for the default and mandatory transforms.
David Oran (Cisco) and Rolf Blom (Ericsson) are co-authors of this
document but their valuable contributions are acknowledged here to
keep the length of the author list down.
The authors would in addition like to thank Magnus Westerlund, Brian
Weis, Ghyslain Pelletier, Morgan Lindqvist, Robert Fairlie-
Cuninghame, Adrian Perrig, the AVT WG and in particular the chairmen
Colin Perkins and Stephen Casner, the Transport and Security Area
Directors, and Eric Rescorla for their reviews and support.
[AES] NIST, "Advanced Encryption Standard (AES)", FIPS PUB 197,
http://www.nist.gov/aes/
[RFC2104] Krawczyk, H., Bellare, M. and R. Canetti, "HMAC: Keyed-
Hashing for Message Authentication", RFC 2104, February
1997.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2401] Kent, S. and R. Atkinson, "Security Architecture for
Internet Protocol", RFC 2401, November 1998.
[RFC2828] Shirey, R., "Internet Security Glossary", FYI 36, RFC 2828,
May 2000.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
"RTP: A Transport Protocol for Real-time Applications", RFC
3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", RFC 3551, July
2003.
Baugher, et al. Standards Track [Page 47]
RFC 3711 SRTP March 2004
[AES-CTR] Lipmaa, H., Rogaway, P. and D. Wagner, "CTR-Mode
Encryption", NIST, http://csrc.nist.gov/encryption/modes/
workshop1/papers/lipmaa-ctr.pdf
[B96] Bellovin, S., "Problem Areas for the IP Security
Protocols," in Proceedings of the Sixth Usenix Unix
Security Symposium, pp. 1-16, San Jose, CA, July 1996
(http://www.research.att.com/~smb/papers/index.html).
[BDJR] Bellare, M., Desai, A., Jokipii, E. and P. Rogaway, "A
Concrete Treatment of Symmetric Encryption: Analysis of DES
Modes of Operation", Proceedings 38th IEEE FOCS, pp. 394-
403, 1997.
[BS00] Biryukov, A. and A. Shamir, "Cryptanalytic Time/Memory/Data
Tradeoffs for Stream Ciphers", Proceedings, ASIACRYPT 2000,
LNCS 1976, pp. 1-13, Springer Verlag.
[C99] Crowell, W. P., "Introduction to the VENONA Project",
http://www.nsa.gov:8080/docs/venona/index.html.
[CTR] Dworkin, M., NIST Special Publication 800-38A,
"Recommendation for Block Cipher Modes of Operation:
Methods and Techniques", 2001.
http://csrc.nist.gov/publications/nistpubs/800-38a/sp800-
38a.pdf.
[f8-a] 3GPP TS 35.201 V4.1.0 (2001-12) Technical Specification 3rd
Generation Partnership Project; Technical Specification
Group Services and System Aspects; 3G Security;
Specification of the 3GPP Confidentiality and Integrity
Algorithms; Document 1: f8 and f9 Specification (Release
4).
[f8-b] 3GPP TR 33.908 V4.0.0 (2001-09) Technical Report 3rd
Generation Partnership Project; Technical Specification
Group Services and System Aspects; 3G Security; General
Report on the Design, Specification and Evaluation of 3GPP
Standard Confidentiality and Integrity Algorithms (Release
4).
[GDOI] Baugher, M., Weis, B., Hardjono, T. and H. Harney, "The
Group Domain of Interpretation, RFC 3547, July 2003.
Baugher, et al. Standards Track [Page 48]
RFC 3711 SRTP March 2004
[HAC] Menezes, A., Van Oorschot, P. and S. Vanstone, "Handbook
of Applied Cryptography", CRC Press, 1997, ISBN 0-8493-
8523-7.
[H80] Hellman, M. E., "A cryptanalytic time-memory trade-off",
IEEE Transactions on Information Theory, July 1980, pp.
401-406.
[IK] T. Iwata and T. Kohno: "New Security Proofs for the 3GPP
Confidentiality and Integrity Algorithms", Proceedings of
FSE 2004.
[KINK] Thomas, M. and J. Vilhuber, "Kerberized Internet
Negotiation of Keys (KINK)", Work in Progress.
[KEYMGT] Arrko, J., et al., "Key Management Extensions for Session
Description Protocol (SDP) and Real Time Streaming Protocol
(RTSP)", Work in Progress.
[KSYH] Kang, J-S., Shin, S-U., Hong, D. and O. Yi, "Provable
Security of KASUMI and 3GPP Encryption Mode f8",
Proceedings Asiacrypt 2001, Springer Verlag LNCS 2248, pp.
255-271, 2001.
[MIKEY] Arrko, J., et. al., "MIKEY: Multimedia Internet KEYing",
Work in Progress.
[MF00] McGrew, D. and S. Fluhrer, "Attacks on Encryption of
Redundant Plaintext and Implications on Internet Security",
the Proceedings of the Seventh Annual Workshop on Selected
Areas in Cryptography (SAC 2000), Springer-Verlag.
[PCST1] Perrig, A., Canetti, R., Tygar, D. and D. Song, "Efficient
and Secure Source Authentication for Multicast", in Proc.
of Network and Distributed System Security Symposium NDSS
2001, pp. 35-46, 2001.
[PCST2] Perrig, A., Canetti, R., Tygar, D. and D. Song, "Efficient
Authentication and Signing of Multicast Streams over Lossy
Channels", in Proc. of IEEE Security and Privacy Symposium
S&P2000, pp. 56-73, 2000.
[RFC1750] Eastlake, D., Crocker, S. and J. Schiller, "Randomness
Recommendations for Security", RFC 1750, December 1994.
[RFC2675] Borman, D., Deering, S. and R. Hinden, "IPv6 Jumbograms",
RFC 2675, August 1999.
Baugher, et al. Standards Track [Page 49]
RFC 3711 SRTP March 2004
[RFC3095] Bormann, C., Burmeister, C., Degermark, M., Fukuhsima, H.,
Hannu, H., Jonsson, L-E., Hakenberg, R., Koren, T., Le, K.,
Liu, Z., Martensson, A., Miyazaki, A., Svanbro, K., Wiebke,
T., Yoshimura, T. and H. Zheng, "RObust Header Compression:
Framework and Four Profiles: RTP, UDP, ESP, and
uncompressed (ROHC)", RFC 3095, July 2001.
[RFC3242] Jonsson, L-E. and G. Pelletier, "RObust Header Compression
(ROHC): A Link-Layer Assisted Profile for IP/UDP/RTP ", RFC
3242, April 2002.
[SDMS] Andreasen, F., Baugher, M. and D. Wing, "Session
Description Protocol Security Descriptions for Media
Streams", Work in Progress.
[SWO] Svanbro, K., Wiorek, J. and B. Olin, "Voice-over-IP-over-
wireless", Proc. PIMRC 2000, London, Sept. 2000.
[V02] Vaudenay, S., "Security Flaws Induced by CBC Padding -
Application to SSL, IPsec, WTLS...", Advances in
Cryptology, EUROCRYPT'02, LNCS 2332, pp. 534-545.
[WC81] Wegman, M. N., and J.L. Carter, "New Hash Functions and
Their Use in Authentication and Set Equality", JCSS 22,
265-279, 1981.
Baugher, et al. Standards Track [Page 50]
RFC 3711 SRTP March 2004
Appendix A: Pseudocode for Index Determination
The following is an example of pseudo-code for the algorithm to
determine the index i of an SRTP packet with sequence number SEQ. In
the following, signed arithmetic is assumed.
if (s_l < 32,768)
if (SEQ - s_l > 32,768)
set v to (ROC-1) mod 2^32
else
set v to ROC
endif
else
if (s_l - 32,768 > SEQ)
set v to (ROC+1) mod 2^32
else
set v to ROC
endif
endif
return SEQ + v*65,536
Appendix B: Test Vectors
All values are in hexadecimal.
Keystream segment length: 1044512 octets (65282 AES blocks)
Session Key: 2B7E151628AED2A6ABF7158809CF4F3C
Rollover Counter: 00000000
Sequence Number: 0000
SSRC: 00000000
Session Salt: F0F1F2F3F4F5F6F7F8F9FAFBFCFD0000 (already shifted)
Offset: F0F1F2F3F4F5F6F7F8F9FAFBFCFD0000
Counter Keystream
F0F1F2F3F4F5F6F7F8F9FAFBFCFD0000 E03EAD0935C95E80E166B16DD92B4EB4
F0F1F2F3F4F5F6F7F8F9FAFBFCFD0001 D23513162B02D0F72A43A2FE4A5F97AB
F0F1F2F3F4F5F6F7F8F9FAFBFCFD0002 41E95B3BB0A2E8DD477901E4FCA894C0
... ...
F0F1F2F3F4F5F6F7F8F9FAFBFCFDFEFF EC8CDF7398607CB0F2D21675EA9EA1E4
F0F1F2F3F4F5F6F7F8F9FAFBFCFDFF00 362B7C3C6773516318A077D7FC5073AE
F0F1F2F3F4F5F6F7F8F9FAFBFCFDFF01 6A2CC3787889374FBEB4C81B17BA6C44
Nota Bene: this test case is contrived so that the latter part of the
keystream segment coincides with the test case in Section F.5.1 of
[CTR].
Baugher, et al. Standards Track [Page 52]
RFC 3711 SRTP March 2004
This section provides test data for the default key derivation
function, which uses AES-128 in Counter Mode. In the following, we
walk through the initial key derivation for the AES-128 Counter Mode
cipher, which requires a 16 octet session encryption key and a 14
octet session salt, and an authentication function which requires a
94-octet session authentication key. These values are called the
cipher key, the cipher salt, and the auth key in the following.
Since this is the initial key derivation and the key derivation rate
is equal to zero, the value of (index DIV key_derivation_rate) is
zero (actually, a six-octet string of zeros). In the following, we
shorten key_derivation_rate to kdr.
The inputs to the key derivation function are the 16 octet master key
and the 14 octet master salt:
master key: E1F97A0D3E018BE0D64FA32C06DE4139
master salt: 0EC675AD498AFEEBB6960B3AABE6
We first show how the cipher key is generated. The input block for
AES-CM is generated by exclusive-oring the master salt with the
concatenation of the encryption key label 0x00 with (index DIV kdr),
then padding on the right with two null octets (which implements the
multiply-by-2^16 operation, see Section 4.3.3). The resulting value
is then AES-CM- encrypted using the master key to get the cipher key.
index DIV kdr: 000000000000
label: 00
master salt: 0EC675AD498AFEEBB6960B3AABE6
-----------------------------------------------
xor: 0EC675AD498AFEEBB6960B3AABE6 (x, PRF input)
x*2^16: 0EC675AD498AFEEBB6960B3AABE60000 (AES-CM input)
cipher key: C61E7A93744F39EE10734AFE3FF7A087 (AES-CM output)
Baugher, et al. Standards Track [Page 53]
RFC 3711 SRTP March 2004
Next, we show how the cipher salt is generated. The input block for
AES-CM is generated by exclusive-oring the master salt with the
concatenation of the encryption salt label. That value is padded and
encrypted as above.
index DIV kdr: 000000000000
label: 02
master salt: 0EC675AD498AFEEBB6960B3AABE6
----------------------------------------------
xor: 0EC675AD498AFEE9B6960B3AABE6 (x, PRF input)
x*2^16: 0EC675AD498AFEE9B6960B3AABE60000 (AES-CM input)
30CBBC08863D8C85D49DB34A9AE17AC6 (AES-CM ouptut)
cipher salt: 30CBBC08863D8C85D49DB34A9AE1
We now show how the auth key is generated. The input block for AES-
CM is generated as above, but using the authentication key label.
index DIV kdr: 000000000000
label: 01
master salt: 0EC675AD498AFEEBB6960B3AABE6
-----------------------------------------------
xor: 0EC675AD498AFEEAB6960B3AABE6 (x, PRF input)
x*2^16: 0EC675AD498AFEEAB6960B3AABE60000 (AES-CM input)
Below, the auth key is shown on the left, while the corresponding AES
input blocks are shown on the right.
auth key AES input blocks
CEBE321F6FF7716B6FD4AB49AF256A15 0EC675AD498AFEEAB6960B3AABE60000
6D38BAA48F0A0ACF3C34E2359E6CDBCE 0EC675AD498AFEEAB6960B3AABE60001
E049646C43D9327AD175578EF7227098 0EC675AD498AFEEAB6960B3AABE60002
6371C10C9A369AC2F94A8C5FBCDDDC25 0EC675AD498AFEEAB6960B3AABE60003
6D6E919A48B610EF17C2041E47403576 0EC675AD498AFEEAB6960B3AABE60004
6B68642C59BBFC2F34DB60DBDFB2 0EC675AD498AFEEAB6960B3AABE60005
Baugher, et al. Standards Track [Page 54]
RFC 3711 SRTP March 2004
Authors' Addresses
Questions and comments should be directed to the authors and
avt@ietf.org:
Mark Baugher
Cisco Systems, Inc.
5510 SW Orchid Street
Portland, OR 97219 USA
Phone: +1 408-853-4418
EMail: mbaugher@cisco.com
Elisabetta Carrara
Ericsson Research
SE-16480 Stockholm
Sweden
Phone: +46 8 50877040
EMail: elisabetta.carrara@ericsson.com
David A. McGrew
Cisco Systems, Inc.
San Jose, CA 95134-1706
USA
Phone: +1 301-349-5815
EMail: mcgrew@cisco.com
Mats Naslund
Ericsson Research
SE-16480 Stockholm
Sweden
Phone: +46 8 58533739
EMail: mats.naslund@ericsson.com
Karl Norrman
Ericsson Research
SE-16480 Stockholm
Sweden
Phone: +46 8 4044502
EMail: karl.norrman@ericsson.com
Baugher, et al. Standards Track [Page 55]
RFC 3711 SRTP March 2004
Full Copyright Statement
Copyright (C) The Internet Society (2004). This document is subject
to the rights, licenses and restrictions contained in BCP 78 and
except as set forth therein, the authors retain all their rights.
This document and the information contained herein are provided on an
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Baugher, et al. Standards Track [Page 56]