Internet Engineering Task Force Rolf Blom, Ericsson AVT Working Group Elisabetta Carrara, Ericsson INTERNET-DRAFT David A. McGrew, Cisco Expires: July 2001 Mats Naslund, Ericsson Karl Norrman, Ericsson David Oran, Cisco February 2001 The Secure Real Time Transport Protocol Status of this memo This document is an Internet-Draft and is in full conformance with all provisions of Section 10 of RFC2026. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet-Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or cite them other than as "work in progress". The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/lid-abstracts.txt The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html Abstract This document describes the Secure Real Time Transport Protocol (SRTP), a profile of the Real Time Transport Protocol (RTP) which can provide privacy, message authentication, replay protection, and implicit header authentication. SRTP can achieve high throughput and low packet expansion by using an additive stream cipher for encryption, a universal hashing based Blom et al. [Page 1] INTERNET-DRAFT SRTP February 2001 function for message authentication, and an 'implicit' index for sequencing based on the RTP sequence number. In addition, SRTP proves to be a suitable protection for heterogenous environments, i.e. environments including both wired and wireless links. TABLE OF CONTENTS 1. Notational Conventions.........................................2 2. Goals..........................................................3 3. SRTP Overview..................................................4 3.1 SRTP Cryptographic Contexts...................................5 3.2 Mapping SRTP Packets to Cryptographic Contexts................5 3.3 SRTP Packet Processing........................................6 3.4 Cryptographic Algorithms......................................7 4. Synchronization................................................8 4.1. IV Formation for Implicit Header Authentication .............9 5. Replay Protection.............................................10 6. Encryption....................................................10 6.1 Defined Ciphers..............................................11 6.1.1. Counter Mode AES..........................................11 6.1.2. AES in f8-Mode............................................12 6.1.3. NULL Cipher...............................................13 7. Message Authentication........................................13 7.1 Default MAC: UMAC............................................14 8. SRTP Parameters...............................................14 9. Secure RTCP...................................................15 10. Rationale....................................................17 10.1 Synchronization.............................................18 10.2 Replay Protection...........................................18 10.3 Source Origin Authentication................................18 10.4. Choice of Encryption Transform.............................19 11. Security Considerations......................................20 11.1. SSRC collision.............................................21 11.2. Confidentiality of the RTP Payload.........................21 11.3. Confidentiality of the RTP Header..........................22 11.4. Integrity of RTP headers...................................22 12. Multicast and Multi-unicast..................................22 13. Acknowledgements.............................................23 14. Author's Addresses...........................................23 15. References...................................................23 APPENDIX A: Test Vectors.........................................25 1. Notational Conventions The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC-2119 [B97]. Blom et al. [Page 2] INTERNET-DRAFT SRTP February 2001 By convention, the most left bit (byte) is the most significant one. By XOR we mean bitwise addition modulo 2 of binary strings, and || denotes concatenation. E.g. if C = A || B, then the most significant bits of C are the same as those of A, and the least significant bits of C equals those of B. 2. Goals The security goals for SRTP are to ensure: * the privacy of the RTP payload, * the authentication of the entire RTP packet, including protection against replayed RTP packets, and * implicit authentication of the header. Each of the security services described above is optional. Any combination of options can be provided, except the single option of implicit header authentication. Source origin authentication (e.g., digitally signed packets) may be desirable in some situations, but this goal is deferred from consideration in this document. See Section 10.3 for a discussion on this point. Other goals for the protocol are: * a low computational cost, * a low footprint (i.e., small code size and data memory for key schedules and replay lists), * limited packet expansion, * no error propagation (e.g., changing a single bit of an SRTP packet should change no more than one bit of the corresponding RTP packet), * the preservation of RTP header compression efficiency, * to allow cryptographic keys to be used by multiple RTP sessions simultaneously, * independence from the underlying transport used by RTP. These properties ensures that SRTP is a suitable protection scheme for both wired and wireless scenarios. Blom et al. [Page 3] INTERNET-DRAFT SRTP February 2001 3. SRTP Overview RTP is the Real Time Protocol [SCFJ96]. We define SRTP as a profile of RTP, in an analogous way to RFC1890 which defines the audio/video profile for RTP. Conceptually, we consider a 'bump in the stack' implementation which resides between the RTP application and the transport layer, which intercepts RTP packets and then forwards an equivalent SRTP packet on the sending side, and which intercepts SRTP packets and passes an equivalent RTP packet up the stack on the receiving side. 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 | | | | .... | +-+>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | | | authentication tag (optional) | | | | | | | .... | | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | | | +- Encrypted Portion +---- Authenticated Portion Figure 1. The format of an SRTP packet. The format of an SRTP packet is illustrated in Figure 1. The optional authentication tag is the only field defined by SRTP that is not in RTP. It provides data origin authentication of the header and payload, and it indirectly provides replay protection by Blom et al. [Page 4] INTERNET-DRAFT SRTP February 2001 authenticating the sequence number. The Encrypted Portion of an SRTP packet consists of the RTP payload of the equivalent RTP packet. The Authenticated Portion of an SRTP packet consists of the entire equivalent RTP packet. 3.1 SRTP Cryptographic Contexts Each SRTP session requires the sender and receiver to maintain cryptographic state information. This information is called the cryptographic context, and it consists of: * an encryption key k_e, and a optionally "salting key" k_s. These keys must be randomly and independently chosen. * a 32-bit rollover counter r (which records how many times the 16-bit RTP sequence number has been reset to zero after passing through 65,535), * an 8-bit FLAG used to signal additional information, * the mode of operation for the encryption scheme, and * the cipher. In addition, when authentication and replay protection are provided: * a message authentication key k_a, * a sequence number s_l (which is the last received and authenticated sequence number for the receiver, and is the last sequence number sent for the sender), and * a replay list L (maintained by the receiver only). 3.2 Mapping SRTP Packets to Cryptographic Contexts In this section we define the mapping of RTP and SRTP packets to the cryptographic contexts used to protect them. The RTP synchronization source (SSRC) identifier is used, along with the RTP transport address (e.g., the Destination IP Address and Port Number) by a receiver to identify the proper cryptographic context for each packet. Recall that an RTP session is defined [SCFJ96] 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 Blom et al. [Page 5] INTERNET-DRAFT SRTP February 2001 session could include an audio RTP session, a video RTP session, and a text RTP session. An SSRC identifier is unique inside an RTP session, and all packets with the same SSRC form part of the same timing and sequence number space. Thus, the SSRC field and transport address information can be used by an SRTP receiver (or by a bump in the stack implementation on the sender's side) to identify the proper cryptographic context within that session. Note though that, for instance in a multicast scenario, the RTP anti-collision mechanism for SSRCs may force these identifiers to change over time, see discussion in Section 12. SRTP may allow the different RTP sessions to use identical cryptographic keys. This is possible if the design of the synchronization mechanism (i.e., the IV in the case of the F8 and Counter Modes) avoids keystream re-use (the two-time pad, Section 11) and with uniqueness requirements on SSRC beyond that dictated by the RTP standard, see Section 12. However, different multimedia sessions SHOULD use different keys. The authentication and encryption keys of each context MUST remain fixed for the duration of that context. This ensures that incorrect keys will not be used by the receiver due to a synchronization error. 3.3 SRTP Packet Processing When Generic Forward Error Correction is performed as specified in RFC 2733, then the security processing takes place before FEC on the sender's side, and after FEC on the receiver's side. To construct a proper SRTP packet, given an RTP packet, the sender does the following: 1. Determine which cryptographic context to use by checking the SSRC field of the RTP packet, and the Transport Address information of that packet (e.g., the Destination IP Address and Port Number). 2. Determine the index of the SRTP packet as described in Section 4, using the rollover counter in the cryptographic context and the sequence number in the RTP packet. Form the current initialization vector (IV). If Implicit Header Authentication is provided, this can be done as described in Section 4.1. 3. Encrypt the Encrypted Portion of the packet, as described in Section 6, using the IV determined in Step 2 and the encryption key and salting key in the context found in Step 1. 4. If authentication is provided, compute the authentication tag for the Authenticated Portion of the packet, as described in Section 7, using the index determined in Step 2 and the authentication key in Blom et al. [Page 6] INTERNET-DRAFT SRTP February 2001 the context found in Step 1. Note that the Encrypted Portion is encrypted before the authentication tag is computed. To authenticate and decrypt a SRTP packet, the receiver does the following: 1. Determine which cryptographic context to use by checking the SSRC field of the RTP packet and the transport address information of the underlying transport header (e.g., the Destination IP Address and Port Number). 2. Determine the index of the SRTP packet from the rollover counter in the cryptographic context and the sequence number in the RTP packet, as described in Section 4. Form the current IV in the same way as done in Step 2 in the encryption process. 3. If authentication is provided, check the Replay List to ensure that no packet with that index has been received and authenticated before, as described in Section 5. If that index is in the list, then the packet has been replayed and is invalid. It MUST be discarded, and the event SHOULD be logged. Compute the authentication tag for the Authenticated Portion of the packet, as described in Section 7, using the index determined in Step 2 and the authentication key in the context found in Step 1. Note that the Encrypted Portion is not decrypted before the authentication tag is computed. If the authentication tag that is computed matches that in the SRTP packet, then the packet is accepted and the index is added to the Replay List. Otherwise, the packet is invalid: it MUST be discarded, and the event SHOULD be logged. 4. Decrypt the Encrypted Portion of the packet, as described in Section 6, using the IV determined in Step 2 and the encryption key and salting key in the context found in Step 1. The processing occurring when replay protection is activated has been chosen to maximize resistance to denial of service attacks (i.e., to minimize the receiver's effort in processing spurious packets). 3.4 Cryptographic Algorithms Default encryption and authentication algorithms are specified in Sections 6.1 and 7.1. While there are numerous encryption and message authentication algorithms that can be used in SRTP, we define default algorithms in order to avoid the complexity of specifying the encodings for the signaling of algorithm and parameter identifiers. Blom et al. [Page 7] INTERNET-DRAFT SRTP February 2001 4. Synchronization SRTP implementations use an 'implicit' packet index for sequencing. Receiver-side implementations use the RTP sequence number to reconstruct the correct index (that is, location in the sequence of all RTP packets). The index is defined as s + r * 65,536, where the sequence number is s and the rollover counter is r. A robust approach for the proper use of a rollover counter requires that its handling and use be well defined. In particular, out-of- order RTP packets with sequence numbers close to 65,536 or zero must be properly dealt with. A receiver reconstructs the index i of a packet with sequence number s using the estimate i = 65,536 * t + s, where t is chosen from the set { r-1, r, r+1 } such that i is closest to the value 65,536 * r + s_l. If the value r+1 is used, then the rollover counter r in the cryptographic context is incremented by one. The pseudocode for the algorithm to process a packet with sequence number s follows: if (s_l < 32,768) if (s - s_l > 32,768) set i to s + 65,536 * (r-1) else set i to s + 65,536 * r endif else if (s_l - 32,768 > s) set r to r + 1 endif set i to s + r * 65,536 endif set s_l to s The index i is used in replay protection (Section 5) when authentication is provided, in encryption (Section 6), and in message authentication (Section 7). This algorithm should be extended by using the information in the authenticated RTCP reports. When RTP authentication is not present, robust synchronization is not possible. In this case, transmission errors or an active attacker may force the receiver to erroneously update his rollover counter and thus to become completely out of synch. It is not possible to protect Blom et al. [Page 8] INTERNET-DRAFT SRTP February 2001 against active attackers in such case, but it is possible to have an update policy for the rollover counter which, except in rare cases, is robust with respect to random bit errors. As the rollover counter is 32 bits long, the maximum number of packets in any given SRTP session is 2^48 = 281,474,976,710,656. After that number of SRTP packets have been sent, the sender MUST not send any more packets with that cryptographic context. This limitation enforces a security benefit by providing an upper bound on the amount of traffic that can pass before cryptographic keys are changed. Other approaches to sequencing were considered and rejected; please see Section 10.1 for our rationale. 4.1. IV Formation for Implicit Header Authentication There may be several alternatives for the Initialization Vector (IV) formation. To guarantee synchronization and avoid keystream re-use, we only require the SSRC, rollover counter and sequence number, or some function thereof (possibly combined with re-keying mechanisms), to be part of the IV. Below, we give a concrete proposal which also provides 'implicit' header authentication, and works with every cipher having at least 128-bit block size. This particular solution also gives a high degree of agreement between bit ordering in the RTP packet header and the IV, simplifying data copying. When implicit header authentication is provided, data from each RTP packet to be encrypted and transmitted, must be included in the(IV). This IV shall be computed and supplied as input to the ciphering algorithm. This shall be done by taking information of said RTP packet, the FLAG, and the rollover counter value, and computing the 128-bit IV: IV = ROC || FLAG || M || PT || SEQ || TS || SSRC where TS (Timestamp, 32 bits), SEQ (Sequence Number, 16 bits), M (Marker Bit, 1 bit), PT (Payload Type, 7 bits), and SSRC (Synchronization Source, 32 bits) are taken from the current RTP header. ROC is the 32-bit rollover counter from the identified context. FLAG is a 8-bit value which is used to signal additional information. Currently, the only value defined (for RTP) is FLAG = 00..0. The value 00..01 is reserved for RTCP and MUST not be used with RTP. With this IV formation, the number of SRTP packets encrypted with any fixed encryption key MUST therefore be no more than 2^48. Otherwise, the size of the ROC ..||..SEQ .. field will not be large enough to avoid keystream reuse. Blom et al. [Page 9] INTERNET-DRAFT SRTP February 2001 5. Replay Protection A packet is 'replayed' when it is stored by an adversary, and then re-injected onto the network. SRTP provides protection against such attacks whenever authentication is provided, through the storage of the indices of the most recently received and authenticated packets. 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. SRTP packet indices which are less than s_l * 65,536 - SRTP-WINDOW-SIZE MAY be assumed to have been received, where SRTP- WINDOW_SIZE is a parameter that MUST be at least 64, and which MAY be set to a higher value. 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 [KA98a]. 6. Encryption Encryption uses a 'seekable' additive stream cipher, following the Stream Cipher ESP [sc-esp]. The stream ciphers that can be used must be able to efficiently seek to arbitrary locations in their keystream. Ciphers that can do this include SEAL [RC94, RC98], LEVIATHAN [MF00b], and any block cipher run in suitable mode. In particular, AES in counter mode will provide good security, reasonable performance, and conform to emerging U.S. Federal standards. Another mode which fulfils the requirements is f8 mode [ES3D], used together with AES. SRTP encryption consists of generating a keystream segment corresponding to the index of the packet, and then bitwise exclusive- oring that keystream segment into the RTP packet, starting at the first bit of the RTP payload. Decryption is then done the same way, but swapping the roles of the plaintext and ciphertext. The definition of how the keystream is generated, given the index, depends on the cipher and its mode of operation. Such a cipher shows features which are desired in a general scenario, e.g. low computational cost, and speed. It also shows properties which fulfil additional requirements posed by the cellular environment [BCNN00], i.e. preservation of RTP header compression efficiency, and absence of error propagation and message expansion. Hence, we conclude that the proposed profile can be applied to the most general heterogenous environment. Blom et al. [Page 10] INTERNET-DRAFT SRTP February 2001 6.1 Defined Ciphers The default cipher is the Advanced Encryption Standard (AES), and we define two modes of running AES, Counter Mode AES and AES in f8-Mode. Both of these modes provide implicit header authentication through the use of the IV formation described in Section 4.1. The NULL cipher is also defined, to be used when encryption is not required. 6.1.1. Counter Mode AES The default cipher SHALL be AES used in the Segmented Integer Counter Mode (SICM) [M00], with a 128-bit key size and a 128-bit block size. Conceptually, counter mode consists of encrypting successive integers. The actual definition is somewhat more complicated, in order to avoid 128 bit integer arithmetic and to randomize the starting point of the integer sequence. Each packet is encrypted with a distinct keystream segment, which is computed as follows. The 128-bit block is divided into three parts: a 64-bit segment prefix, a 32-bit block index, which is incremented to generate a keystream segment, and a 32-bit segment suffix. The segment prefix/suffix pair is unique for each keystream segment. A keystream segment is the concatenation of the output blocks of the cipher in encrypt mode, in which the block indicies are in increasing order. Symbolically, each keystream segment looks like E(A || B || C) || E(A || B + 1 mod 2^32 || C) || E(A || B + 2 mod 2^32 || C) .. where A, B, and C are segment prefix, block index, and segment suffix, respectively, determined as given below. The offsets are computed from the salting key k_s and the IV (from Section 4.1) by exclusive-oring k_s and the IV, and setting A to the first 64 bits of the result, B as the following 32 and C to the remaining 32 bits of the result. Symbolically, A || B || C = IV XOR k_s. If k_s is less than 128 bits long, then k_s is concatenated with itself as many times as needed in order to form the salt which is added to the IV. If no salting key is used, this is interpreted as k_s = 0. Note that the segment prefix/suffix pair is distinct for each packet which is encrypted, thus ensuring that keystream segments are distinct and non-overlapping. Blom et al. [Page 11] INTERNET-DRAFT SRTP February 2001 The restriction on the maximunm number of RTP packets above ensures the security of the encryption method by limiting the effectiveness of probabilistic attacks [BR98]. The AES has a block size of 128 bits, so 2^32 output blocks are sufficient to generate the 2^7 * 2^32 = 549755813888 bits of keystream needed to encrypt the largest possible RTP packet. 6.1.2. AES in f8-Mode To encrypt UMTS (Universal Mobile Telecommunications System, as 3G networks) data, a solution (see [ES3D]) 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 define the use of AES as default block cipher to be used in f8-Mode for RTP encryption, with 128-bit key and block size. Figure 2 shows the structure of an arbitrary b-bit block size cipher, E, running in what we shall call "f8-mode of operation". | | \|/ +------+ | | --->| E | | | | | +------+ | | m --> * |--------------------------- ... -------| _____ | IV' | | | | | | ct=1 --> * ct=2 --> * ... ct=L-1 --> * | | | | | | | --> * --> * ... --> * | \|/ | \|/ | \|/ | \|/ | +------+ | +------+ | +------+ | +------+ | | | | | | | | | | | | k -------->| E | | | E | | | E | | | E | | | | | | | | | | | | +------+ | +------+ | +------+ | +------+ | | | | | | | |------ |-------- | ... ---- | | | | | \|/ \|/ \|/ \|/ S(0) S(1) S(2) . . . S(L-1) Blom et al. [Page 12] INTERNET-DRAFT SRTP February 2001 Figure 2. f8-mode of operation (asterisk, *, denotes bitwise XOR). Let E(k,B) be the 128-bit output of E in encrypt mode when applied to the 128-bit key k and 128-bit plaintext block B. Let ct, IV, IV', S(j), and m denote 128-bit blocks, determined below. The S() keystream for an n-bit message is defined by setting IV' = E(k XOR m, IV), and ct = S(-1) = 00..0. For j = 0,1,.., L-1 where L = n/128 (rounded up to nearest integer) compute S(j) = E(k,IV' XOR ct XOR S(j-1)), (Eq. 1) ct = ct + 1 mod 2^128 (Eq. 2) Notice that the IV (as defined in Section 4.1) is not used directly. Instead it is fed through E under another key to produce an internal, "salted" value (denoted IV') to prevent an attacker from gaining known input/ouput pairs, and the roll of the internal counter is to prevent short keystream cycles. The value of the key mask m is defined to be m = k_s || 0x555..5, i.e. the salting key, padded with the the binary pattern 0101.. to fill the 128-bit key size. (If no salting key is used, m = 0x55..5.) The maxium allowable packet size can be determined as follows. The AES has a block size of 128 bits. Assuming that AES behaves like a random function, it is (heuristically) secure to generate about 2^64 output blocks, which is sufficient to generate the 2^71 bits of keystream. In practise though, the counter ct above will often be sufficient if implemented as a 16- or 32-bit counter. In fact, for some security margin, other methods SHOULD be used if packets of size exceeding 2^32 * 128 = 549755813888 bits are to be encrypted. 6.1.3. NULL Cipher The NULL cipher is used when no confidentiality is requested. It simply copies the plaintext input into the ciphertext output. 7. Message Authentication Message integrity and authentication (hereafter referred to as just "authentication") are optional functions provided by SRTP. Authentication can be provided by any message authentication code, though the default value is UMAC [KBHHKR00]. Blom et al. [Page 13] INTERNET-DRAFT SRTP February 2001 The authentication tag is computed by applying the UMAC function to the Authenticated Portion of the SRTP packet. The authentication tag is appended to the RTP packet. This expansion of the RTP packet may cause the packet size to exceed the Maximum transmission Unit (MTU) of a network interface on its path, especially in circumstances when the application is attempting to 'optimize' the size of packets. MTU path discovery SHOULD be used to avoid this problem. Authentication SHOULD be provided by SRTP. The fact that authentication is optional is motivated by the fact that, while the function is typically highly desired, there are certain cases (notably in the cellular environment) where it has an impact in terms of cost, as motivated in [BCNN00]. In those cases, it is up to the user security profile to request authentication. 7.1 Default MAC: UMAC The default message authentication code is UMAC [KBHHKR00], which has proven security properties and is quite fast. Furthermore, it can be used with short (e.g., two or four byte) authentication tags, as well as larger tags. UMAC is a parameterized algorithm (see Section 2.1 of [KBHHKR00]). The default selection of UMAC parameters for SRTP are: WORD-LEN 2 UMAC-OUTPUT-LEN 4 L1-KEY-LEN 128 UMAC-KEY-LEN 16 ENDIAN-FAVORITE BIG L1-OPERATIONS-SIGN SIGNED This choice of parameters is intended to work well on low-power processors, to minimize packet expansion, and to minimize the size of the cryptographic context. The WORD-LEN of two will work well on 16 bit and higher processors. The packet expansion is determined by the UMAC-OUTPUT-LEN to be only four bytes. The storage requirement, per cryptographic context, is 144 bytes. These parameters ensure a forgery probability of no greater than 1/2^30 for each individual packet. Please see the security considerations section in [KBHHKR00] and the references therein for a more detailed discussion. 8. SRTP Parameters The SRTP-WINDOW-SIZE is defined to be at least 64 (Section 5). Blom et al. [Page 14] INTERNET-DRAFT SRTP February 2001 The current defined modes are Counter Mode (default), f8 Mode (Section 6), and the NULL Cipher. The default cipher is AES (Section 6), used with a block- and encryption key size of 128 bits. 9. Secure RTCP Secure RTCP follows the definition of Secure RTP, but defines the index and IV differently. In order to differentiate these quantities, we refer to it as the SRTCP index and IV. SRTCP is defined as a profile of RTCP, and it adds two new fields to the RTCP packet definition, the SRTCP index and the authentication tag. Those fields are appended to an RTCP packet in order to form an equivalent SRTCP packet, so that they follow any other profile- specific extensions. An SRTCP packet is illustrated in Figure 3. 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=200 | length | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | | SSRC of sender | | +>+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | | | ... | | | | sender info | | | | ... | | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | | | ... | | | | report block 1 | | | | ... | | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | | | ... | | | | report block 2 | | | | ... | | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | | | | | | | ... | | | | | | | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | | | ... | | | | profile-specific extensions | | | | ... | | +>+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | | | SRTCP index | +-|>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | | | ... | | | | authentication tag | | | | ... | | | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | | Blom et al. [Page 15] INTERNET-DRAFT SRTP February 2001 | +-- Encrypted Portion +---- Authenticated Portion Figure 3. The format of a Secure RTCP packet, after Section 6.3.1 of [SCFJ96]. In this case, the underlying RTCP packet is a sender report packet; the SRTP format is identical for other RTCP packet types. The SRTCP index is a 32-bit value. As we allow both encrypted and non-encrypted packets belonging to the same flow (see discussion below), indices with their most significant bit set to '1' are reserved for encrypted packets, and indices with most significant bit set to '0' are used for non-encrypted packets. With this restriction, the rest of the bits are set to zero before the first SRTCP packet is sent, and is incremented by one after each SRTCP is sent. Except for differences in the most significant bit, SRTCP indices form a strictly increasing sequence. The index is explicitly included in each packet, in contrast to the 'implicit' index approach used for SRTP. SRTCP packet processing is identical to that of SRTP packet processing, with the following changes: * SRTCP replay protection is as defined in Section 5, but using the the SRTCP index as the index i. * SRTCP encryption is as defined in Section 6, but using the definition of the SRTCP Encrypted Portion as defined in this section, using the SRTCP index as the index i, and the IV as defined in this section. * The SRTCP authentication tag is defined as in Section 7, but applying the UMAC function to the Authenticated Portion of the SRTCP packet as defined in this section, and using the SRTCP index as the index i. * SRCTP decryption is performed as in Section 6, but only if the SRTCP index has its most significant bit equal to 1. If so, the encrypted portion is decrypted, using the SRTCP index as the index i, and the IV as defined in this section. In case the most significant bit of the index is 0, the payload is simply copied. The IV for ciphers using 128-bit block size is formed in the following way: IV = SRTCP index || FLAG || PT || 0..0 || SSRC where PT (Payload Type, 8 bit), and SSRC (Synchronization Source, 32 bits) are taken from the first header in the RTCP compound packet. SRTCP index is the added 32-bit index to the packet. A pad of 48 zeros is inserted between the PT and the SSRC. Blom et al. [Page 16] INTERNET-DRAFT SRTP February 2001 FLAG is a 8-bit value which is used to signal additional information. Currently, the only value defined (for RTCP) is FLAG = 00..01. The value 0..0 is reserved for RTP and MUST not be used for RTCP. This allows to use the same key for related RTP and RTCP flows (being the IV unique). Then this IV is treated in the same way as defined in Section 6, according to the chosen encryption mode. The encryption prefix (Section 6.1 of [SCFJ96]), which is a random 32-bit quantity intended to improve privacy, SHOULD NOT be used. This is because SRTP encryption uses an additive stream cipher, and thus the prefix offers no benefit. The maximum number of SRTCP packets is limited to 2^31 = 2,147,483,648. The last RTCP packet MUST contain an RTCP BYE. SRTCP senders MUST send an RTCP BYE in the final packet, if the maximum number of SRTCP packets is reached. Similarly, SRTCP receivers MUST act as though the last RTCP packet included a BYE, even if no BYE was included in the packet, if the maximum number of SRTCP packets is reached. Authentication MUST be required for RTCP, being it the control protocol (e.g., it has a BYE packet). Moreover, the cost for RTCP authentication is not of the same order of RTP authentication, being the session bandwidth allocated to RTCP recommended at 5%. However, when adding authentication to RTCP, the overhead in bandwidth SHOULD be considered (it will be more than 5%). It is allowed to split a compound RTCP packet into two lower-layer packets, one to be encrypted and one to be sent in the clear, as described in Section 9.1 of [SCFJ96]. Encryption/non-encryption is signaled by the most significant bit of the SRTCP index as described above. 10. Rationale SRTP achieves high throughput and low packet expansion by using fast stream ciphers for encryption, an implicit index for synchronization, and universal hash functions for message authentication. SRTP shows to be a suitable choice for the most general scenario, and to fit also the most demanding one, conversation multimedia over wireless, having it the necessary robustness properties. Only a single header extension may be appended to the RTP data header, so the use of a header extension for SRTP was avoided. SRTP and SRTCP are defined as profiles of RTP and RTCP, respectively. Blom et al. [Page 17] INTERNET-DRAFT SRTP February 2001 10.1 Synchronization RTP runs over unreliable transport. Thus, maintaining synchronization of the cryptographic context between the sender and receiver is a conspicuous challenge. Because of the requirement to minimize packet expansion, no explicit sequencing information should be added. RTP packets contain two fields for synchronization purposes, the timestamp and the sequence number. The timestamp field could be used for cryptographic synchronization in some circumstances. However, this field is not appropriate for such use. From [SCFJ96]: Several consecutive RTP packets may have equal timestamps if they are (logically) generated at once, e.g., belong to the same video frame. Consecutive RTP packets may contain timestamps that are not monotonic if the data is not transmitted in the order it was sampled, as in the case of MPEG interpolated video frames. The RTP sequence number might be directly used as a unique identifier for SRTP packets. However, it has only sixteen bits, which would limit the duration of an SRTP security association to only 64,536 packets, asking therefore for periodically rekeying. The 'implicit index' approach works as long as the reorder and loss of the packets is not too great. In particular, 32,768 packets would need to be lost, or a packet would need to be 32,768 packets out of sequence in order for synchronization to be lost. Such drastic loss or reorder is likely to disrupt the RTP application itself. When a participant joins an SRTP session while that session is in progress, the entire cryptographic context except for the replay list is sent to that participant. This step is essential for security. See also Section 12. 10.2 Replay Protection Replay protection is undoubtedly important for multimedia data, and SHOULD be provided. Otherwise, it would be possible for an adversary to perform simple manipulations on data that subverted security. For example, in a voice application, the phrase "yes" could be substituted for "no" if replay protection were not present. However, there are certain scenarios, e.g. conversation multimedia, where it may be difficult to perform such a kind of attacks. Moreover, to be useful, replay protection needs to be based on an authentication mechanism (i.e., authentication of the sequence number of the RTP header), and this has a cost when cellular links are involved on the path. 10.3 Source Origin Authentication Blom et al. [Page 18] INTERNET-DRAFT SRTP February 2001 'Source origin authentication' was listed as an option in the security goals, not because it is not an appropriate goal, but because it may not be achievable. This goal may be desirable in some circumstances, such as multicast environments in which the sender is more trusted than the receivers, or when translators or mixers (Section 2.3 of [SCFJ96]) are used. However, it is not clear that this capability can always be provided, as mixers and translators can change the payload. Furthermore, this security service essentially requires digital signatures (at least if collusion resistance is required [BF00]). Two examples of the multicast scenario mentioned above are a military commander addressing his troops over RTP, and financial market data sent over RTP. In these situations, a 'stream signing' method can provide digital signatures on the entire RTP packets. An extensive literature on such methods is developing, and it is reasonable to expect that one of these methods can be reduced to practice and specified for RTP. This suggests that it should be left as an option in the current specification. A future effort can define a stream signing method as an authentication type for RTP, which could be used as a replacement for a message integrity transform. Examples of the mixer and translator scenarios include a translator re-encoding data at a lower rate or in a different encoding, and a mixer combining the audio streams of multiple speakers in a teleconference. In these cases, it is not clear that meaningful source origin authentication is possible, as the data that is received is not the same as the data that is signed. If the translator is trusted by the receivers, then it could sign or re-sign the data streams, but this scenario may not be prevalent. It may be possible to devise a signing scheme that authenticates the source but not the content (enabling the receivers to know that "John is one of the people talking", but not providing authentication on who said what) by signing the concatenation of the Contributing source (CSRC) field and some sequencing information (e.g., a timestamp or sequence number), but such schemes require synchronization between the senders. This synchronization is not required by the RTP protocol itself, and may be difficult or impossible to arrange. 10.4 Choice of Encryption Transform When adopting a block cipher mode to produce keystreams, the central ingredient is the block cipher which is its core. As far as modern cryptology knows, the security basically stands (and falls) with the security of the block cipher. This means that if a weakness is found, replacing the block cipher with a new one will most likely remedy the security problems. We define AES (Rijndael) [AES] as default block cipher, as it is widely believed to be secure. Blom et al. [Page 19] INTERNET-DRAFT SRTP February 2001 11. Security Considerations The security of UMAC is well understood, and is described in [KBHHKR00]. Additive ciphers do not provide any security service other than privacy. In particular, they do not provide message authentication (see [RK99] or [S96] for a discussion of this security service). However, SRTP uses a message authentication code to provide that security service. By using 'seekable' stream ciphers, SRTP avoids the denial of service attacks that are possible on stream ciphers that lack this property (these attacks are described in Section 3.4 of [B96]). No bit of keystream in an additive stream cipher should ever be used to encrypt multiple distinct plaintext bits. Such keystream reuse (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. In SRTP, a 'two- time pad' is avoided by requiring the key or the IV to be unique. An SSRC is mapped to a unique crypto context. Multiple crypto contexts may contain identical keys; in this case, each context together with data from the RTP header MUST produce a unique IV (which is typically assured by plugging the unique SSRC in the IV). If manual keying is used, two different cryptographic contexts might accidentally use the same encryption key with non-negligible probability, through manual error or procedural inadequacies. Thus, manual keying SHOULD NOT be used for SRTP (or SRTCP). An 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,Bi96]. 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, 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". In order to provide an effective key size of n bits in a deployment in which 2^m SRTP/SRTCP cryptographic contexts will be created, the true key size will need to be n+m bits. The value of m SHOULD be 32 bits for networks with 50,000 connections (fully meshed networks Blom et al. [Page 20] INTERNET-DRAFT SRTP February 2001 with up to 200 devices), and SHOULD be 64 bits for networks with 49e+12 connections (fully meshed networks with up to 7,000,000 devices). These choices of m ensures that key collision attacks amortized over a ten year period offer no advantage over exhaustive search, when new SRTP keys are established for every connection every hour (note that such an attack requires the storage of all network traffic over the ten year period). These choices will suffice for many networks, though SRTP deployments with more stringent security requirements will need to make a detailed assessment of those requirements with respect to the attacks described in [MF00]. 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. It is an important point that the m bits of 'extra' key provided to thwart these attacks need not be private. In jurisdictions with mandated limits on the length of a secret key, the additional key bits could be made public. This is because those bits are functionally equivalent to the 'salt' that is used to protect passwords from dictionary attacks. The fact that the 'extra' key bits are distinct for many different keys defeats the key collision and time-memory tradeoff attacks by reducing the number of keys over which cryptanalytic computation can be amortized. Note that other security protocols which use additive ciphers for the encryption of Internet traffic (e.g., SSL, TLS, SSH, IPSEC) are also vulnerable to the attacks described in [MF00]. Those attacks are generic to additive encryption of redundant plaintext, and are not particular to SRTP. 11.1 SSRC collision Assume that two or more communication parties use the same key. Though RTP implements an SSRC collision detection mechanism, it is impossible to guarantee that two parties do not accidently choose the same SSRC and send a few packets before the collision is detected. In a very unfortunate case, the IV formation in Section 4.1 could in fact make the keystreams collide and we have a 'two-time pad'. This is probably a bigger problem in the case of group communication when a single group key is desired. See also some administrative issues with SSRC collisions in Section 12. 11.2. Confidentiality of the RTP Payload 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 deduce certain "formatting bits" of the payload, as the length of the CODEC output might vary due to certain Blom et al. [Page 21] INTERNET-DRAFT SRTP February 2001 parameter settings etc. This, in turn, implies that the corresponding bit of the keystream can be deduced. However, if the stream cipher is secure, knowledge of a few bits of the keystream will not aid an attacker in predicting the following keystream bits. Thus, the payload length (and information deducible from this) will leak, but nothing else. 11.3. Confidentiality of the RTP Header With our proposal, 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". Our proposal is a low-cost method, which allows header compression to reduce bandwidth. It is up to the endpoints policies to decide about the security scheme to employ. If the header compression is omitted, other solutions might be applicable, e.g. [sc-esp]. In other words, we provide a solution that works in the most general scenario, even in the most demanding one (like conversational multimedia over low- bandwidth, unreliable media. Of course the solution will then also work in less restricted environments, but we suggest that if one really needs to protect headers, and is allowed to do so by the surrounding environment, then he should also look at alternatives. In addition, we strongly recommend the use of profiles to select the right trade-off for the required level of security. 11.4 Integrity of RTP headers The IV formation in Section 4.1, which depends on the RTP header, provides an 'implicit' authentication of that header, which is useful when the authentication option is not present. This is because any attacks which modify the header of such a packet will cause the SRTP receiver to use an incorrect IV in the decryption step, with the result that the decrypted RTP payload will be essentially random. 12. Multicast and Multi-unicast The scheme described here can be used in case a single, unique key (a single pair, encryption group key and authentication group key) is to be used inside a multimedia session, for a low complexity key management. However, it then becomes necessary to have a way to assure that each SSRC is unique inside that multimedia session. This is a light and feasible solution in several scenarios, e.g. one sender only, streaming, and unicast. Blom et al. [Page 22] INTERNET-DRAFT SRTP February 2001 In multicast and multi-unicast, to use the same group key for the multimedia session, there should be a way to guarantee uniqueness of the SSRC before starting sending. Otherwise, the triggering of the anti-collision mechanism will ask for a change in the SSRCs of the parties that happened to have the same SSRC, hence giving trouble in pointing to the right context. The problem remains how to address the context database after the anti-collision algorithm has changed the SSRCs. Section 3.3 defines the use of SSRC and Transport Address of that packet as selectors to the database. In case of UDP, the unchanged transport address can be a good indicator that a collision, followed by anti-collision triggering, has happened. So, simply try decryptions until a RTCP message confirms the change in the SSRC on that transport address and then update the database selector triplet. If the requirement of unique SSRC inside that multimedia session cannot be guaranteed (e.g., for large groups), then a unique key per sender should be used. The additional requirement is to have SSRC unique per sender, which appears to be feasible enough. However, the same consideration on the anti-collision algorithm triggerring applies. 13. Acknowledgements The authors would like to thank Brian Weis and Magnus Westerlund for their reviews and comments. 14. Author's Addresses Questions and comments about this memo can be directed to: David A. McGrew David Oran Cisco Systems, Inc. San Jose, CA 95134-1706 USA mcgrew@cisco.com, oran@cisco.com Rolf Blom Elisabetta Carrara Mats Naslund Karl Norrman Ericsson Research {rolf.blom, elisabetta.carrara, mats.naslund, karl.norrman}@era.ericsson.se 15. References Blom et al. [Page 23] INTERNET-DRAFT SRTP February 2001 [AES] NIST, "Advanced Encryption Standard (AES)", http://csrc.nist.gov/encryption/aes/ [B97] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", RFC 2119, March 1997. [BCNN00] Blom, R., Carrara, E., Naslund, M., and Norrman, K., "Conversational Multimedia Security in 3G Networks", Internet Draft, November 2000, . [BF00] Boneh, D., and Franklin, M., "Message Authentication in a Multicast Environment", the Proceedings of the Seventh Annual Workshop on Selected Areas in Cryptography (SAC 2000), Springer- Verlag. [C99] Crowell, W. P., "Introduction to the VENONA Project", http://www.nsa.gov:8080/docs/venona/index.html. [ES3D] ETSI SAGE 3GPP Standard Algorithms Task Force, "Security Algorithms Group of Experts (SAGE); General Report on the Design, Specification and Evaluation of 3GPP Standard Confidentiality and Integrity Algorithms", Public report, Draft Version 1.0, Dec 1999. [ES3E] ETSI SAGE 3GPP Standard Algorithms Task Force, "Security Algorithms Group of Experts (SAGE) Report on the Evaluation of 3GPP Standard Confidentiality and Integrity Algorithms", Public report, Draft Version 1.0, Dec 1999. [HAC] Menezes, A., Van Oorschot, P., and Vanstone, S., "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. [KA98a] Kent, S., and R. Atkinson, "Security Architecture for IP", RFC 2401, November 1998. [KBHHKR00] Krovetz, T., Black, J., Halevi, S., Hevia, A., Krawczyk, H., Rogaway, P., "UMAC: Message Authentication Code using Universal Hashing", Internet Draft, October 2000, . [LRW00] Lipmaa, H., Rogaway, P., and Wagner, D., "Comments to NIST Concerning AES Modes of Operation: CTR-Mode Encryption", NIST Workshop on AES Modes of Operation, http://csrc.nist.gov/encryption/aes/modes/lipmaa-ctr.pdf [M00] McGrew, D., "Segmented Integer Counter Mode: Specification and Rationale", NIST Workshop on AES Modes of Operation, http://www.mindspring.com/~dmcgrew/sic-mode.pdf Blom et al. [Page 24] INTERNET-DRAFT SRTP February 2001 [MF00] McGrew, D., and Fluhrer, S., "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. [MF00b] McGrew, D., and Fluhrer, S., "The Stream Cipher LEVIATHAN: Specification and Supporting Documentation", Submission to the New European Schemes for Signatures, Integrity, and Encryption (NESSIE) Process, October, 2000http://www.cryptonessie.org/. [R92] Rueppel, R., "Stream Ciphers", Chapter 2 of Simmons, G., "Contemporary Cryptology: the Science of Information Integrity," 1992, IEEE Press. [RC94] Rogaway, P. and Coppersmith, D., "A Software-Optimized Encryption Algorithm", Proceedings of the 1994 Fast Software Encryption Workshop, Lecture Notes In Computer Science, Volume 809, Springer-Verlag, 1994, pp. 56-63. [RC98] Rogaway, P. and Coppersmith, D., "A Software-Optimized Encryption Algorithm", Journal of Cryptology, Volume 11, Number 4, Springer-Verlag, 1998, Pages 273-287. Also available on the Internet at http://www.cs.ucdavis.edu/~rogaway/papers/seal-abstract.html. [RK99] Rescorla, E., and Korver, B., "Guidelines for Writing RFC Text on Security Considerations," draft-rescorla-sec-cons-00.txt [S96] Schneier, B. "Applied Cryptography: Protocols, Algorithms, and Source Code in C", Wiley, 1996. [sc-esp] McGrew, D., Fluhrer, S., Peyravian, M., "The Stream Cipher Encapsulating Security Payload", Internet Draft, July 2000 [SCFJ96] Schulzrinne, H., Casner, S., Frederick, R., Jacobson, V., "RTP: A Transport Protocol for Real-Time Applications", IETF Request For Comments RFC 1889. Appendix A. Test vectors We include in the following some test vectors for f8-AES. key: 234829008467be186c3de14aae72d62c salting key || 0x555... : 32f2870d555555555555555555555555 Blom et al. [Page 25] INTERNET-DRAFT SRTP February 2001 AES-internal expanded key: 23482900 8467be18 6c3de14a ae72d62c 62be58e4 e6d9e6fc 8ae407b6 2496d19a f080e0d2 1659062e 9cbd0198 b82bd002 05f097be 13a99190 8f149008 373f400a 78f9f024 6b5061b4 e444f1bc d37bb1b6 4931be42 2261dff6 c6252e4a 155e9ffc 31ea0e1b 138bd1ed d5aeffa7 c0f0605b fd3a37a1 eeb1e64c 3b1f19eb fbef79b0 a28cd0ae 4c3d36e2 77222f09 8ccd56b9 043d86ca 4800b028 3f229f21 b3efc998 ede0c0a7 a5e0708f 9ac2efae 292d2636 AES-internal expanded salting key || 555...: 32f2870d 55555555 55555555 55555555 cf0e7bf1 9a5b2ea4 cf0e7bf1 9a5b2ea4 f43f3249 6e641ced a16a671c 3b3149b8 37045eab 59604246 f80a255a c33b6ce2 dd54c685 843484c3 7c3ea199 bf05cd7b a6e9e78d 22dd634e 5ee3c2d7 e1e60fac 089f7675 2a42153b 74a1d7ec 9547d840 e8fe7f5f c2bc6a64 b61dbd88 235a65c8 d6b39779 140ffd1d a2124095 8148255d 9f8cdb75 8b832668 299166fd a8d943a0 9c963bb7 17151ddf 3e847b22 965d3882 RTP-packet header fields: version = 2 padding = 0 extension = 0 CSRC count = 0 marker bit = 0 payload type = 6e sequence no. = 5cba timestamp = 50681de5 SSRC = 5c621599 Data from Cryptographic context: FLAG = 0 Rollover counter = d462564a IV: d462564a006e5cba50681de55c621599 IV': 4fee844eedb458a3e2b0c7ed43888cc1 Encryption of bits 0 to 127: Blom et al. [Page 26] INTERNET-DRAFT SRTP February 2001 ct: 0 S(-1) : 00000000000000000000000000000000 S(-1) XOR IV' : 4fee844eedb458a3e2b0c7ed43888cc1 S(-1) XOR IV' XOR ct : 4fee844eedb458a3e2b0c7ed43888cc1 plain text P[0..127] : 6e915f07cd6f1c0d44afaab4961c7d31 final keystream S(0) : b2d3b3d7e16092de379e33b350582e63 cipher text C[0..127] : dc42ecd02c0f8ed373319907c6445352 Encryption of bits 128 to 255: ct: 1 S(0) : b2d3b3d7e16092de379e33b350582e63 S(0) XOR IV' : fd3d37990cd4ca7dd52ef45e13d0a2a2 S(0) XOR IV' XOR ct : fd3d37990cd4ca7dd52ef45e13d0a2a3 plain text P[128..255] : 7b9daad84352a6d4bcdf501a560832a0 final keystream S(1) : b1ce287dc53c1975de3d7d0500f780ba cipher text C[128..255] : ca5382a5866ebfa162e22d1f56ffb21a ------------------------------------------------------------ This Internet-Draft expires in July 2001. Blom et al. [Page 27]