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Run idnits with the --verbose option for more detailed information about the items above. -------------------------------------------------------------------------------- 2 Network Working Group M. Westerlund 3 Internet-Draft Ericsson 4 Intended status: Informational C. Perkins 5 Expires: May 16, 2014 University of Glasgow 6 November 12, 2013 8 Options for Securing RTP Sessions 9 draft-ietf-avtcore-rtp-security-options-09 11 Abstract 13 The Real-time Transport Protocol (RTP) is used in a large number of 14 different application domains and environments. This heterogeneity 15 implies that different security mechanisms are needed to provide 16 services such as confidentiality, integrity and source authentication 17 of RTP/RTCP packets suitable for the various environments. The range 18 of solutions makes it difficult for RTP-based application developers 19 to pick the most suitable mechanism. This document provides an 20 overview of a number of security solutions for RTP, and gives 21 guidance for developers on how to choose the appropriate security 22 mechanism. 24 Status of This Memo 26 This Internet-Draft is submitted in full conformance with the 27 provisions of BCP 78 and BCP 79. 29 Internet-Drafts are working documents of the Internet Engineering 30 Task Force (IETF). Note that other groups may also distribute 31 working documents as Internet-Drafts. The list of current Internet- 32 Drafts is at http://datatracker.ietf.org/drafts/current/. 34 Internet-Drafts are draft documents valid for a maximum of six months 35 and may be updated, replaced, or obsoleted by other documents at any 36 time. It is inappropriate to use Internet-Drafts as reference 37 material or to cite them other than as "work in progress." 39 This Internet-Draft will expire on May 16, 2014. 41 Copyright Notice 43 Copyright (c) 2013 IETF Trust and the persons identified as the 44 document authors. All rights reserved. 46 This document is subject to BCP 78 and the IETF Trust's Legal 47 Provisions Relating to IETF Documents 48 (http://trustee.ietf.org/license-info) in effect on the date of 49 publication of this document. Please review these documents 50 carefully, as they describe your rights and restrictions with respect 51 to this document. Code Components extracted from this document must 52 include Simplified BSD License text as described in Section 4.e of 53 the Trust Legal Provisions and are provided without warranty as 54 described in the Simplified BSD License. 56 Table of Contents 58 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 59 2. Background . . . . . . . . . . . . . . . . . . . . . . . . . 4 60 2.1. Point-to-Point Sessions . . . . . . . . . . . . . . . . . 4 61 2.2. Sessions Using an RTP Mixer . . . . . . . . . . . . . . . 4 62 2.3. Sessions Using an RTP Translator . . . . . . . . . . . . 5 63 2.3.1. Transport Translator (Relay) . . . . . . . . . . . . 5 64 2.3.2. Gateway . . . . . . . . . . . . . . . . . . . . . . . 6 65 2.3.3. Media Transcoder . . . . . . . . . . . . . . . . . . 7 66 2.4. Any Source Multicast . . . . . . . . . . . . . . . . . . 7 67 2.5. Source-Specific Multicast . . . . . . . . . . . . . . . . 7 68 3. Security Options . . . . . . . . . . . . . . . . . . . . . . 9 69 3.1. Secure RTP . . . . . . . . . . . . . . . . . . . . . . . 9 70 3.1.1. Key Management for SRTP: DTLS-SRTP . . . . . . . . . 11 71 3.1.2. Key Management for SRTP: MIKEY . . . . . . . . . . . 12 72 3.1.3. Key Management for SRTP: Security Descriptions . . . 14 73 3.1.4. Key Management for SRTP: Encrypted Key Transport . . 15 74 3.1.5. Key Management for SRTP: Other systems . . . . . . . 15 75 3.2. RTP Legacy Confidentiality . . . . . . . . . . . . . . . 16 76 3.3. IPsec . . . . . . . . . . . . . . . . . . . . . . . . . . 16 77 3.4. RTP over TLS over TCP . . . . . . . . . . . . . . . . . . 16 78 3.5. RTP over Datagram TLS (DTLS) . . . . . . . . . . . . . . 17 79 3.6. Media Content Security/Digital Rights Management . . . . 17 80 3.6.1. ISMA Encryption and Authentication . . . . . . . . . 18 81 4. Securing RTP Applications . . . . . . . . . . . . . . . . . . 18 82 4.1. Application Requirements . . . . . . . . . . . . . . . . 18 83 4.1.1. Confidentiality . . . . . . . . . . . . . . . . . . . 19 84 4.1.2. Integrity . . . . . . . . . . . . . . . . . . . . . . 20 85 4.1.3. Source Authentication . . . . . . . . . . . . . . . . 20 86 4.1.4. Identity . . . . . . . . . . . . . . . . . . . . . . 22 87 4.1.5. Privacy . . . . . . . . . . . . . . . . . . . . . . . 22 88 4.2. Application Structure . . . . . . . . . . . . . . . . . . 23 89 4.3. Automatic Key Management . . . . . . . . . . . . . . . . 23 90 4.4. End-to-End Security vs Tunnels . . . . . . . . . . . . . 24 91 4.5. Plain Text Keys . . . . . . . . . . . . . . . . . . . . . 24 92 4.6. Interoperability . . . . . . . . . . . . . . . . . . . . 25 93 5. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 25 94 5.1. Media Security for SIP-established Sessions using DTLS- 95 SRTP . . . . . . . . . . . . . . . . . . . . . . . . . . 25 96 5.2. Media Security for WebRTC Sessions . . . . . . . . . . . 26 97 5.3. IP Multimedia Subsystem (IMS) Media Security . . . . . . 27 98 5.4. 3GPP Packet Based Streaming Service (PSS) . . . . . . . . 28 99 5.5. RTSP 2.0 . . . . . . . . . . . . . . . . . . . . . . . . 29 100 6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 29 101 7. Security Considerations . . . . . . . . . . . . . . . . . . . 30 102 8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 30 103 9. Informative References . . . . . . . . . . . . . . . . . . . 30 104 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 35 106 1. Introduction 108 Real-time Transport Protocol (RTP) [RFC3550] is widely used in a 109 large variety of multimedia applications, including Voice over IP 110 (VoIP), centralized multimedia conferencing, sensor data transport, 111 and Internet television (IPTV) services. These applications can 112 range from point-to-point phone calls, through centralised group 113 teleconferences, to large-scale television distribution services. 114 The types of media can vary significantly, as can the signalling 115 methods used to establish the RTP sessions. 117 This multi-dimensional heterogeneity has so far prevented development 118 of a single security solution that meets the needs of the different 119 applications. Instead significant number of different solutions have 120 been developed to meet different sets of security goals. This makes 121 it difficult for application developers to know what solutions exist, 122 and whether their properties are appropriate. This memo gives an 123 overview of the available RTP solutions, and provides guidance on 124 their applicability for different application domains. It also 125 attempts to provide indication of actual and intended usage at time 126 of writing as additional input to help with considerations such as 127 interoperability, availability of implementations etc. The guidance 128 provided is not exhaustive, and this memo does not provide normative 129 recommendations. 131 It is important that application developers consider the security 132 goals and requirements for their application. The IETF considers it 133 important that protocols implement, and makes available to the user, 134 secure modes of operation [RFC3365]. Because of the heterogeneity of 135 RTP applications and use cases, however, a single security solution 136 cannot be mandated [I-D.ietf-avt-srtp-not-mandatory]. Instead, 137 application developers need to select mechanisms that provide 138 appropriate security for their environment. It is strongly 139 encouraged that common mechanisms are used by related applications in 140 common environments. The IETF publishes guidelines for specific 141 classes of applications, so it is worth searching for such 142 guidelines. 144 The remainder of this document is structured as follows. Section 2 145 provides additional background. Section 3 outlines the available 146 security mechanisms at the time of this writing, and lists their key 147 security properties and constraints. That is followed by guidelines 148 and important aspects to consider when securing an RTP application in 149 Section 4. Finally, we give some examples of application domains 150 where guidelines for security exist in Section 5. 152 2. Background 154 RTP can be used in a wide variety of topologies due to its support 155 for point-to-point sessions, multicast groups, and other topologies 156 built around different types of RTP middleboxes. In the following we 157 review the different topologies supported by RTP to understand their 158 implications for the security properties and trust relations that can 159 exist in RTP sessions. 161 2.1. Point-to-Point Sessions 163 The most basic use case is two directly connected end-points, shown 164 in Figure 1, where A has established an RTP session with B. In this 165 case the RTP security is primarily about ensuring that any third 166 party can't compromise the confidentiality and integrity of the media 167 communication. This requires confidentiality protection of the RTP 168 session, integrity protection of the RTP/RTCP packets, and source 169 authentication of all the packets to ensure no man-in-the-middle 170 attack is taking place. 172 The source authentication can also be tied to a user or an end- 173 point's verifiable identity to ensure that the peer knows who they 174 are communicating with. Here the combination of the security 175 protocol protecting the RTP session and its RTP and RTCP traffic and 176 the key-management protocol becomes important in which security 177 statements one can do. 179 +---+ +---+ 180 | A |<------->| B | 181 +---+ +---+ 183 Figure 1: Point-to-point topology 185 2.2. Sessions Using an RTP Mixer 187 An RTP mixer is an RTP session-level middlebox that one can build a 188 multi-party RTP based conference around. The RTP mixer might 189 actually perform media mixing, like mixing audio or compositing video 190 images into a new media stream being sent from the mixer to a given 191 participant; or it might provide a conceptual stream, for example the 192 video of the current active speaker. From a security point of view, 193 the important features of an RTP mixer is that it generates a new 194 media stream, and has its own source identifier, and does not simply 195 forward the original media. 197 An RTP session using a mixer might have a topology like that in 198 Figure 2. In this example, participants A through D each send 199 unicast RTP traffic to the RTP mixer, and receive an RTP stream from 200 the mixer, comprising a mixture of the streams from the other 201 participants. 203 +---+ +------------+ +---+ 204 | A |<---->| |<---->| B | 205 +---+ | | +---+ 206 | Mixer | 207 +---+ | | +---+ 208 | C |<---->| |<---->| D | 209 +---+ +------------+ +---+ 211 Figure 2: Example RTP mixer Topology 213 A consequence of an RTP mixer having its own source identifier, and 214 acting as an active participant towards the other end-points is that 215 the RTP mixer needs to be a trusted device that has access to the 216 security context(s) established. The RTP mixer can also become a 217 security enforcing entity. For example, a common approach to secure 218 the topology in Figure 2 is to establish a security context between 219 the mixer and each participant independently, and have the mixer 220 source authenticate each peer. The mixer then ensures that one 221 participant cannot impersonate another. 223 2.3. Sessions Using an RTP Translator 225 RTP translators are middleboxes that provide various levels of in- 226 network media translation and transcoding. Their security properties 227 vary widely, depending on which type of operations they attempt to 228 perform. We identify three different categories of RTP translator: 229 transport translators, gateways, and media transcoders. We discuss 230 each in turn. 232 2.3.1. Transport Translator (Relay) 234 A transport translator [RFC5117] operates on a level below RTP and 235 RTCP. It relays the RTP/RTCP traffic from one end-point to one or 236 more other addresses. This can be done based only on IP addresses 237 and transport protocol ports, with each receive port on the 238 translator can have a very basic list of where to forward traffic. 239 Transport translators also need to implement ingress filtering to 240 prevent random traffic from being forwarded that isn't coming from a 241 participant in the conference. 243 Figure 3 shows an example transport translator, where traffic from 244 any one of the four participants will be forwarded to the other three 245 participants unchanged. The resulting topology is very similar to 246 Any Source Multicast (ASM) session (as discussed in Section 2.4), but 247 implemented at the application layer. 249 +---+ +------------+ +---+ 250 | A |<---->| |<---->| B | 251 +---+ | Relay | +---+ 252 | Translator | 253 +---+ | | +---+ 254 | C |<---->| |<---->| D | 255 +---+ +------------+ +---+ 257 Figure 3: RTP relay translator topology 259 A transport translator can often operate without needing access to 260 the security context, as long as the security mechanism does not 261 provide protection over the transport-layer information. A transport 262 translator does, however, make the group communication visible, and 263 so can complicate keying and source authentication mechanisms. This 264 is further discussed in Section 2.4. 266 2.3.2. Gateway 268 Gateways are deployed when the endpoints are not fully compatible. 269 Figure 4 shows an example topology. The functions a gateway provides 270 can be diverse, and range from transport layer relaying between two 271 domains not allowing direct communication, via transport or media 272 protocol function initiation or termination, to protocol or media 273 encoding translation. The supported security protocol might even be 274 one of the reasons a gateway is needed. 276 +---+ +-----------+ +---+ 277 | A |<---->| Gateway |<---->| B | 278 +---+ +-----------+ +---+ 280 Figure 4: RTP gateway topology 282 The choice of security protocol, and the details of the gateway 283 function, will determine if the gateway needs to be trusted with 284 access to the application security context. Many gateways need to be 285 trusted by all peers to perform the translation; in other cases some 286 or all peers might not be aware of the presence of the gateway. The 287 security protocols have different properties depending on the degree 288 of trust and visibility needed. Ensuring communication is possible 289 without trusting the gateway can be strong incentive for accepting 290 different security properties. Some security solutions will be able 291 to detect the gateways as manipulating the media stream, unless the 292 gateway is a trusted device. 294 2.3.3. Media Transcoder 296 A Media transcoder is a special type of gateway device that changes 297 the encoding of the media being transported by RTP. The discussion 298 in Section 2.3.2 applies. A media transcoder alters the media data, 299 and thus needs to be trusted with access to the security context. 301 2.4. Any Source Multicast 303 Any Source Multicast [RFC1112] is the original multicast model where 304 any multicast group participant can send to the multicast group, and 305 get their packets delivered to all group members (see Figure 5). 306 This form of communication has interesting security properties, due 307 to the many-to-many nature of the group. Source authentication is 308 important, but all participants with access to group security context 309 will have the necessary secrets to decrypt and verify integrity of 310 the traffic. Thus use of any group security context fails if the 311 goal is to separate individual sources; alternate solutions are 312 needed. 314 +-----+ 315 +---+ / \ +---+ 316 | A |----/ \---| B | 317 +---+ / Multi- \ +---+ 318 + Cast + 319 +---+ \ Network / +---+ 320 | C |----\ /---| D | 321 +---+ \ / +---+ 322 +-----+ 324 Figure 5: Any source multicast (ASM) group 326 In addition the potential large size of multicast groups creates some 327 considerations for the scalability of the solution and how the key- 328 management is handled. 330 2.5. Source-Specific Multicast 332 Source-Specific Multicast [RFC4607] allows only a specific end-point 333 to send traffic to the multicast group, irrespective of the number of 334 RTP media sources. The end-point is known as the media Distribution 335 Source. For RTP session to function correctly with RTCP over an SSM 336 session extensions have been defined in [RFC5760]. Figure 6 shows a 337 sample SSM-based RTP session where several media sources, MS1...MSm, 338 all send media to a Distribution Source, which then forwards the 339 media data to the SSM group for delivery to the receivers, R1...Rn, 340 and the Feedback Targets, FT1...FTn. RTCP reception quality feedback 341 is sent unicast from each receiver to one of the Feedback Targets. 342 The feedback targets aggregate reception quality feedback and forward 343 it upstream towards the distribution source. The distribution source 344 forwards (possibly aggregated and summarised) reception feedback to 345 the SSM group, and back to the original media sources. The feedback 346 targets are also members of the SSM group and receive the media data, 347 so they can send unicast repair data to the receivers in response to 348 feedback if appropriate. 350 +-----+ +-----+ +-----+ 351 | MS1 | | MS2 | .... | MSm | 352 +-----+ +-----+ +-----+ 353 ^ ^ ^ 354 | | | 355 V V V 356 +---------------------------------+ 357 | Distribution Source | 358 +--------+ | 359 | FT Agg | | 360 +--------+------------------------+ 361 ^ ^ | 362 : . | 363 : +...................+ 364 : | . 365 : / \ . 366 +------+ / \ +-----+ 367 | FT1 |<----+ +----->| FT2 | 368 +------+ / \ +-----+ 369 ^ ^ / \ ^ ^ 370 : : / \ : : 371 : : / \ : : 372 : : / \ : : 373 : ./\ /\. : 374 : /. \ / .\ : 375 : V . V V . V : 376 +----+ +----+ +----+ +----+ 377 | R1 | | R2 | ... |Rn-1| | Rn | 378 +----+ +----+ +----+ +----+ 380 Figure 6: Example SSM-based RTP session with two feedback targets 382 The use of SSM makes it more difficult to inject traffic into the 383 multicast group, but not impossible. Source authentication 384 requirements apply for SSM sessions too, and an individual 385 verification of who sent the RTP and RTCP packets is needed. An RTP 386 session using SSM will have a group security context that includes 387 the media sources, distribution source, feedback targets, and the 388 receivers. Each has a different role and will be trusted to perform 389 different actions. For example, the distribution source will need to 390 authenticate the media sources to prevent unwanted traffic being 391 distributed via the SSM group. Similarly, the receivers need to 392 authenticate both the distribution source and their feedback target, 393 to prevent injection attacks from malicious devices claiming to be 394 feedback targets. An understanding of the trust relationships and 395 group security context is needed between all components of the 396 system. 398 3. Security Options 400 This section provides an overview of security requirements, and the 401 current RTP security mechanisms that implement those requirements. 402 This cannot be a complete survey, since new security mechanisms are 403 defined regularly. The goal is to help applications designer by 404 reviewing the types of solution that are available. This section 405 will use a number of different security related terms, described in 406 the Internet Security Glossary, Version 2 [RFC4949]. 408 3.1. Secure RTP 410 The Secure RTP (SRTP) protocol [RFC3711] is one of the most commonly 411 used mechanisms to provide confidentiality, integrity protection, 412 source authentication and replay protection for RTP. SRTP was 413 developed with RTP header compression and third party monitors in 414 mind. Thus the RTP header is not encrypted in RTP data packets, and 415 the first 8 bytes of the first RTCP packet header in each compound 416 RTCP packet are not encrypted. The entirety of RTP packets and 417 compound RTCP packets are integrity protected. This allows RTP 418 header compression to work, and lets third party monitors determine 419 what RTP traffic flows exist based on the SSRC fields, but protects 420 the sensitive content. 422 SRTP works with transforms where different combinations of encryption 423 algorithm, authentication algorithm, and pseudo-random function can 424 be used, and the authentication tag length can be set to any value. 425 SRTP can also be easily extended with additional cryptographic 426 transforms. This gives flexibility, but requires more security 427 knowledge by the application developer. To simplify things, SDP 428 Security Descriptions (see Section 3.1.3) and DTLS-SRTP (see 429 Section 3.1.1) use pre-defined combinations of transforms, known as 430 SRTP crypto suites and SRTP protection profiles, that bundle together 431 transforms and other parameters, making them easier to use but 432 reducing flexibility. The MIKEY protocol (see Section 3.1.2) 433 provides flexibility to negotiate the full selection of transforms. 434 At the time of this writing, the following transforms, SRTP crypto 435 suites, and SRTP protection profiles are defined or under definition: 437 AES-CM and HMAC-SHA-1: AES Counter Mode encryption with 128-bit keys 438 combined with 160-bit keyed HMAC-SHA-1 with 80-bit authentication 439 tag. This is the default cryptographic transform that needs to be 440 supported. The transforms are defined in SRTP [RFC3711], with the 441 corresponding SRTP crypto suite in [RFC4568] and SRTP protection 442 profile in [RFC5764]. 444 AES-f8 and HMAC-SHA-1: AES f8 mode encryption using 128-bit keys 445 combined with keyed HMAC-SHA-1 using 80-bit authentication. The 446 transforms are defined in [RFC3711], with the corresponding SRTP 447 crypto suite in [RFC4568]. The corresponding SRTP protection 448 profile is not defined. 450 SEED: A Korean national standard cryptographic transform that is 451 defined to be used with SRTP in [RFC5669]. Three options are 452 defined, one using SHA-1 authentication, one using Counter mode 453 with CBC-MAC, and finally one using Galois Counter mode. 455 ARIA: A Korean block cipher [I-D.ietf-avtcore-aria-srtp], that 456 supports 128-, 192- and 256- bit keys. It also defines three 457 options, Counter mode where combined with HMAC-SHA-1 with 80 or 32 458 bits authentication tags, Counter mode with CBC-MAC and Galois 459 Counter mode. It also defines a different key derivation function 460 than the AES based systems. 462 AES-192-CM and AES-256-CM: Cryptographic transforms for SRTP based 463 on AES-192 and AES-256 counter mode encryption and 160-bit keyed 464 HMAC-SHA-1 with 80- and 32-bit authentication tags. These provide 465 192- and 256-bit encryption keys, but otherwise match the default 466 128-bit AES-CM transform. The transforms are defined in [RFC3711] 467 and [RFC6188], with the SRTP crypto suites in [RFC6188]. 469 AES-GCM and AES-CCM: AES Galois Counter Mode and AES Counter with 470 CBC MAC for AES-128 and AES-256. This authentication is included 471 in the cipher text which becomes expanded with the length of the 472 authentication tag instead of using the SRTP authentication tag. 473 This is defined in [I-D.ietf-avtcore-srtp-aes-gcm]. 475 NULL: SRTP [RFC3711] also provides a NULL cipher that can be used 476 when no confidentiality for RTP/RTCP is requested. The 477 corresponding SRTP protection profile is defined in [RFC5764]. 479 The source authentication guarantees provided by SRTP depend on the 480 cryptographic transform and key-management used. Some transforms 481 give strong source authentication even in multiparty sessions; others 482 give weaker guarantees and can authenticate group membership but not 483 sources. TESLA [RFC4383] offers a complement to the regular 484 symmetric keyed authentication transforms, like HMAC-SHA-1, and can 485 provide per-source authentication in some group communication 486 scenarios. The downside is need for buffering the packets for a 487 while before authenticity can be verified. 489 [RFC4771] defines a variant of the authentication tag that enables a 490 receiver to obtain the Roll over Counter for the RTP sequence number 491 that is part of the Initialization vector (IV) for many cryptographic 492 transforms. This enables quicker and easier options for joining a 493 long lived secure RTP group, for example a broadcast session. 495 RTP header extensions are normally carried in the clear and only 496 integrity protected in SRTP. This can be problematic in some cases, 497 so [RFC6904] defines an extension to also encrypt selected header 498 extensions. 500 SRTP is specified and deployed in a number of RTP usage contexts; 501 Significant support in SIP-established VoIP clients including IMS; 502 RTSP [I-D.ietf-mmusic-rfc2326bis] and RTP based media streaming. 503 Thus SRTP in general is widely deployed. When it comes to 504 cryptographic transforms the default (AES-CM and HMAC-SHA-1) is the 505 most commonly used, but it might be expected that AES-GCM, 506 AES-192-CM, and AES-256-CM will gain usage in future, especially due 507 to the AES- and GCM-specific instructions in new CPUs. 509 SRTP does not contain an integrated key-management solution, and 510 instead relies on an external key management protocol. There are 511 several protocols that can be used. The following sections outline 512 some popular schemes. 514 3.1.1. Key Management for SRTP: DTLS-SRTP 516 A Datagram Transport Layer Security extension exists for establishing 517 SRTP keys [RFC5763][RFC5764]. This extension provides secure key- 518 exchange between two peers, enabling perfect forward secrecy and 519 binding strong identity verification to an end-point. The default 520 key generation will generate a key that contains material contributed 521 by both peers. The key-exchange happens in the media plane directly 522 between the peers. The common key-exchange procedures will take two 523 round trips assuming no losses. TLS resumption can be used when 524 establishing additional media streams with the same peer, and reduces 525 the set-up time to one RTT for these streams (see [RFC5764] for a 526 discussion of TLS resumption in this context). 528 The actual security properties of an established SRTP session using 529 DTLS will depend on the cipher suites offered and used, as well as 530 the mechanism for identifying the end-points of the hand-shake. For 531 example some cipher suits provide perfect forward secrecy (PFS), 532 while other do not. When using DTLS, the application designer needs 533 to select which cipher suites DTLS-SRTP can offer and accept so that 534 the desired security properties are achieved. The next choice is how 535 to verify the identity of the peer end-point. One choice can be to 536 rely on the certificates and use a PKI to verify them to make an 537 identity assertion. However, this is not the most common way, 538 instead self-signed certificate are common to use, and instead 539 establish trust through signalling or other third party solutions. 541 DTLS-SRTP key management can use the signalling protocol in four 542 ways. First, to agree on using DTLS-SRTP for media security. 543 Secondly, to determine the network location (address and port) where 544 each side is running a DTLS listener to let the parts perform the 545 key-management handshakes that generate the keys used by SRTP. 546 Thirdly, to exchange hashes of each side's certificates to bind these 547 to the signalling, and ensure there is no man-in-the-middle attack. 548 This assumes that one can trust the signalling solution to be 549 resistant to modification, and not be in collaboration with an 550 attacker. Finally to provide an assertable identity, e.g. [RFC4474] 551 that can be used to prevent modification of the signalling and the 552 exchange of certificate hashes. That way enabling binding between 553 the key-exchange and the signalling. 555 This usage is well defined for SIP/SDP in [RFC5763], and in most 556 cases can be adopted for use with other bi-directional signalling 557 solutions. It is to be noted that there is work underway to revisit 558 the SIP Identity mechanism [RFC4474] in the IETF STIR working group. 560 The main question regarding DTLS-SRTP's security properties is how 561 one verifies any peer identity or at least prevents man-in-the-middle 562 attacks. This do requires trust in some DTLS-SRTP external party, 563 either a PKI, a signalling system or some identity provider. 565 DTLS-SRTP usage is clearly on the rise. It is mandatory to support 566 in WebRTC. It has growing support among SIP end-points. DTLS-SRTP 567 was developed in IETF primarily to meet security requirements for 568 SIP. 570 3.1.2. Key Management for SRTP: MIKEY 571 Multimedia Internet Keying (MIKEY) [RFC3830] is a keying protocol 572 that has several modes with different properties. MIKEY can be used 573 in point-to-point applications using SIP and RTSP (e.g., VoIP calls), 574 but is also suitable for use in broadcast and multicast applications, 575 and centralized group communications. 577 MIKEY can establish multiple security contexts or cryptographic 578 sessions with a single message. It is useable in scenarios where one 579 entity generates the key and needs to distribute the key to a number 580 of participants. The different modes and the resulting properties 581 are highly dependent on the cryptographic method used to establish 582 the session keys actually used by the security protocol, like SRTP. 584 MIKEY has the following modes of operation: 586 Pre-Shared Key: Uses a pre-shared secret for symmetric key crypto 587 used to secure a keying message carrying the already generated 588 session key. This system is the most efficient from the 589 perspective of having small messages and processing demands. The 590 downside is scalability, where usually the effort for the 591 provisioning of pre-shared keys is only manageable if the number 592 of endpoints is small. 594 Public Key encryption: Uses a public key crypto to secure a keying 595 message carrying the already-generated session key. This is more 596 resource intensive but enables scalable systems. It does require 597 a public key infrastructure to enable verification. 599 Diffie-Hellman: Uses Diffie-Hellman key-agreement to generate the 600 session key, thus providing perfect forward secrecy. The downside 601 is high resource consumption in bandwidth and processing during 602 the MIKEY exchange. This method can't be used to establish group 603 keys as each pair of peers performing the MIKEY exchange will 604 establish different keys. 606 HMAC-Authenticated Diffie-Hellman: [RFC4650] defines a variant of 607 the Diffie-Hellman exchange that uses a pre-shared key in a keyed 608 HMAC to verify authenticity of the keying material instead of a 609 digital signature as in the previous method. This method is still 610 restricted to point-to-point usage. 612 RSA-R: MIKEY-RSA in Reverse mode [RFC4738] is a variant of the 613 public key method which doesn't rely on the initiator of the key- 614 exchange knowing the responder's certificate. This method lets 615 both the initiator and the responder to specify the session keying 616 material depending on use case. Usage of this mode requires one 617 round-trip time. 619 TICKET: [RFC6043] is a MIKEY extension using a trusted centralized 620 key management service (KMS). The Initiator and Responder do not 621 share any credentials; instead, they trust a third party, the KMS, 622 with which they both have or can establish shared credentials. 624 IBAKE: [RFC6267] uses a key management services (KMS) infrastructure 625 but with lower demand on the KMS. Claims to provides both perfect 626 forward and backwards secrecy, the exact meaning is unclear (See 627 Perfect Forward Secrecy in [RFC4949]). 629 SAKKE: [RFC6509] provides Sakai-Kasahara Key Encryption in MIKEY. 630 Based on Identity based Public Key Cryptography and a KMS 631 infrastructure to establish a shared secret value and certificate 632 less signatures to provide source authentication. Its features 633 include simplex transmission, scalability, low-latency call set- 634 up, and support for secure deferred delivery. 636 MIKEY messages have several different transports. [RFC4567] defines 637 how MIKEY messages can be embedded in general SDP for usage with the 638 signalling protocols SIP, SAP and RTSP. There also exist a 3GPP 639 defined usage of MIKEY that sends MIKEY messages directly over UDP 640 [T3GPP.33.246] to key the receivers of Multimedia Broadcast and 641 Multicast Service (MBMS) [T3GPP.26.346]. [RFC3830] defines the 642 application/mikey media type allowing MIKEY to be used in, e.g., 643 email and HTTP. 645 Based on the many choices it is important to consider the properties 646 needed in ones solution and based on that evaluate which modes that 647 are candidates for ones usage. More information on the applicability 648 of the different MIKEY modes can be found in [RFC5197]. 650 MIKEY with pre-shared keys are used by 3GPP MBMS [T3GPP.33.246] and 651 IMS media security [T3GPP.33.328] specifies the use of the TICKET 652 mode transported over SIP and HTTP. RTSP 2.0 653 [I-D.ietf-mmusic-rfc2326bis] specifies use of the RSA-R mode. There 654 are some SIP end-points that support MIKEY. The modes they use are 655 unknown to the authors. 657 3.1.3. Key Management for SRTP: Security Descriptions 659 [RFC4568] provides a keying solution based on sending plain text keys 660 in SDP [RFC4566]. It is primarily used with SIP and the SDP Offer/ 661 Answer model, and is well-defined in point-to-point sessions where 662 each side declares its own unique key. Using Security Descriptions 663 to establish group keys is less well defined, and can have security 664 issues since it's difficult to guarantee unique SSRCs (as needed to 665 avoid a "two-time pad" attack - see Section 9 of [RFC3711]). 667 Since keys are transported in plain text in SDP, they can easily be 668 intercepted unless the SDP carrying protocol provides strong end-to- 669 end confidentiality and authentication guarantees. This is not 670 normally the case, where instead hop-by-hop security is provided 671 between signalling nodes using TLS. This leaves the keying material 672 sensitive to capture by the traversed signalling nodes. Thus, in 673 most cases, the security properties of security descriptions are 674 weak. The usage of security descriptions usually requires additional 675 security measures, e.g. the signalling nodes be trusted and protected 676 by strict access control. Usage of security descriptions requires 677 careful design in order to ensure that the security goals can be met. 679 Security Descriptions is the most commonly deployed keying solution 680 for SIP-based end-points, where almost all end-points that support 681 SRTP also support Security Descriptions. It is also used for access 682 protection in IMS Media Security [T3GPP.33.328]. 684 3.1.4. Key Management for SRTP: Encrypted Key Transport 686 Encrypted Key Transport (EKT) [I-D.ietf-avtcore-srtp-ekt] is an SRTP 687 extension that enables group keying despite using a keying mechanism 688 like DTLS-SRTP that doesn't support group keys. It is designed for 689 centralized conferencing, but can also be used in sessions where end- 690 points connect to a conference bridge or a gateway, and need to be 691 provisioned with the keys each participant on the bridge or gateway 692 uses to avoid decryption and encryption cycles on the bridge or 693 gateway. This can enable interworking between DTLS-SRTP and other 694 keying systems where either party can set the key (e.g., interworking 695 with security descriptions). 697 The mechanism is based on establishing an additional EKT key which 698 everyone uses to protect their actual session key. The actual 699 session key is sent in a expanded authentication tag to the other 700 session participants. This key is only sent occasionally or 701 periodically depending on use cases and depending on what 702 requirements exist for timely delivery or notification. 704 The only known deployment of EKT so far are in some Cisco video 705 conferencing products. 707 3.1.5. Key Management for SRTP: Other systems 709 The ZRTP [RFC6189] key-management system for SRTP was proposed as an 710 alternative to DTLS-SRTP. ZRTP provides best effort encryption 711 independent of the signalling protocol and utilizes key continuity, 712 Short Authentication Strings, or a PKI for authentication. ZRTP 713 wasn't adopted as an IETF standards track protocol, but was instead 714 published as an informational RFC. Commercial implementations exist. 716 Additional proprietary solutions are also known to exist. 718 3.2. RTP Legacy Confidentiality 720 Section 9 of the RTP standard [RFC3550] defines a DES or 3DES based 721 encryption of RTP and RTCP packets. This mechanism is keyed using 722 plain text keys in SDP [RFC4566] using the "k=" SDP field. This 723 method can provide confidentiality but, as discussed in Section 9 of 724 [RFC3550], it has extremely weak security properties and is not to be 725 used. 727 3.3. IPsec 729 IPsec [RFC4301] can be used in either tunnel or transport mode to 730 protect RTP and RTCP packets in transit from one network interface to 731 another. This can be sufficient when the network interfaces have a 732 direct relation, or in a secured environment where it can be 733 controlled who can read the packets from those interfaces. 735 The main concern with using IPsec to protect RTP traffic is that in 736 most cases using a VPN approach that terminates the security 737 association at some node prior to the RTP end-point leaves the 738 traffic vulnerable to attack between the VPN termination node and the 739 end-point. Thus usage of IPsec requires careful thought and design 740 of its usage so that it meets the security goals. A important 741 question is how one ensures the IPsec terminating peer and the 742 ultimate destination are the same. Applications can have issues 743 using existing APIs with determining if IPsec is being used or not, 744 and when used who the authenticated peer entity is. 746 IPsec with RTP is more commonly used as a security solution between 747 infrastructure nodes that exchange many RTP sessions and media 748 streams. The establishment of a secure tunnel between such nodes 749 minimizes the key-management overhead. 751 3.4. RTP over TLS over TCP 753 Just as RTP can be sent over TCP [RFC4571], it can also be sent over 754 TLS over TCP [RFC4572], using TLS to provide point-to-point security 755 services. The security properties TLS provides are confidentiality, 756 integrity protection and possible source authentication if the client 757 or server certificates are verified and provide a usable identity. 758 When used in multi-party scenarios using a central node for media 759 distribution, the security provide is only between the central node 760 and the peers, so the security properties for the whole session are 761 dependent on what trust one can place in the central node. 763 RTSP 1.0 [RFC2326] and 2.0 [I-D.ietf-mmusic-rfc2326bis] specifies the 764 usage of RTP over the same TLS/TCP connection that the RTSP messages 765 are sent over. It appears that RTP over TLS/TCP is also used in some 766 proprietary solutions that uses TLS to bypass firewalls. 768 3.5. RTP over Datagram TLS (DTLS) 770 Datagram Transport Layer Security (DTLS) [RFC6347] is a based on TLS 771 [RFC5246], but designed to work over a unreliable datagram oriented 772 transport rather than requiring reliable byte stream semantics from 773 the transport protocol. Accordingly, DTLS can provide point-to-point 774 security for RTP flows analogous to that provided by TLS, but over an 775 datagram transport such as UDP. The two peers establish an DTLS 776 association between each other, including the possibility to do 777 certificate-based source authentication when establishing the 778 association. All RTP and RTCP packets flowing will be protected by 779 this DTLS association. 781 Note that using DTLS for RTP flows is different to using DTLS-SRTP 782 key management. DTLS-SRTP uses the same key-management steps as 783 DTLS, but uses SRTP for the per packet security operations. Using 784 DTLS for RTP flows uses the normal datagram TLS data protection, 785 wrapping complete RTP packets. When using DTLS for RTP flows, the 786 RTP and RTCP packets are completely encrypted with no headers in the 787 clear; when using DTLS-SRTP, the RTP headers are in the clear and 788 only the payload data is encrypted. 790 DTLS can use similar techniques to those available for DTLS-SRTP to 791 bind a signalling-side agreement to communicate to the certificates 792 used by the end-point when doing the DTLS handshake. This enables 793 use without having a certificate-based trust chain to a trusted 794 certificate root. 796 There does not appear to be significant usage of DTLS for RTP. 798 3.6. Media Content Security/Digital Rights Management 800 Mechanisms have been defined that encrypt only the media content, 801 operating within the RTP payload data and leaving the RTP headers and 802 RTCP unaffected. There are several reasons why this might be 803 appropriate, but a common rationale is to ensure that the content 804 stored by RTSP streaming servers has the media content in a protected 805 format that cannot be read by the streaming server (this is mostly 806 done in the context of Digital Rights Management). These approaches 807 then use a key-management solution between the rights provider and 808 the consuming client to deliver the key used to protect the content 809 and do not give the media server access to the security context. 810 Such methods have several security weaknesses such as the fact that 811 the same key is handed out to a potentially large group of receiving 812 clients, increasing the risk of a leak. 814 Use of this type of solution can be of interest in environments that 815 allow middleboxes to rewrite the RTP headers and select which streams 816 are delivered to an end-point (e.g., some types of centralised video 817 conference systems). The advantage of encrypting and possibly 818 integrity protecting the payload but not the headers is that the 819 middlebox can't eavesdrop on the media content, but can still provide 820 stream switching functionality. The downside of such a system is 821 that it likely needs two levels of security: the payload level 822 solution to provide confidentiality and source authentication, and a 823 second layer with additional transport security ensuring source 824 authentication and integrity of the RTP headers associated with the 825 encrypted payloads. This can also results in the need to have two 826 different key-management systems as the entity protecting the packets 827 and payloads are different with different set of keys. 829 The aspect of two tiers of security are present in ISMACryp (see 830 Section 3.6.1) and the deprecated 3GPP Packet Based Streaming Service 831 Annex.K [T3GPP.26.234R8] solution. 833 3.6.1. ISMA Encryption and Authentication 835 The Internet Streaming Media Alliance (ISMA) has defined ISMA 836 Encryption and Authentication 2.0 [ISMACryp2]. This specification 837 defines how one encrypts and packetizes the encrypted application 838 data units (ADUs) in an RTP payload using the MPEG-4 Generic payload 839 format [RFC3640]. The ADU types that are allowed are those that can 840 be stored as elementary streams in an ISO Media File format based 841 file. ISMACryp uses SRTP for packet level integrity and source 842 authentication from a streaming server to the receiver. 844 Key-management for a ISMACryp based system can be achieved through 845 Open Mobile Alliance (OMA) Digital Rights Management 2.0 [OMADRMv2], 846 for example. 848 4. Securing RTP Applications 850 In the following we provide guidelines for how to choose appropriate 851 security mechanisms for RTP applications. 853 4.1. Application Requirements 855 This section discusses a number of application requirements that need 856 be considered. An application designer choosing security solutions 857 requires a good understanding of what level of security is needed and 858 what behaviour they strive to achieve. 860 4.1.1. Confidentiality 862 When it comes to confidentiality of an RTP session there are several 863 aspects to consider: 865 Probability of compromise: When using encryption to provide media 866 confidentiality, it is necessary to have some rough understanding 867 of the security goal and how long one expect the protected content 868 to remain confidential. National or other regulations might 869 provide additional requirements on a particular usage of an RTP. 870 From that, one can determine which encryption algorithms are to be 871 used from the set of available transforms. 873 Potential for other leakage: RTP based security in most of its forms 874 simply wraps RTP and RTCP packets into cryptographic containers. 875 This commonly means that the size of the original RTP payload is 876 visible to observers of the protected packet flow. This can 877 provide information to those observers. A well-documented case is 878 the risk with variable bit-rate speech codecs that produce 879 different sized packets based on the speech input [RFC6562]. 880 Potential threats such as these need to be considered and, if they 881 are significant, then restrictions will be needed on mode choices 882 in the codec, or additional padding will need to be added to make 883 all packets equal size and remove the informational leakage. 885 Another case is RTP header extensions. If SRTP is used, header 886 extensions are normally not protected by the security mechanism 887 protecting the RTP payload. If the header extension carries 888 information that is considered sensitive, then the application 889 needs to be modified to ensure that mechanisms used to protect 890 against such information leakage are employed. 892 Who has access: When considering the confidentiality properties of a 893 system, it is important to consider where the media handled in the 894 clear. For example, if the system is based on an RTP mixer that 895 needs the keys to decrypt the media, process, and repacketize it, 896 then is the mixer providing the security guarantees expected by 897 the other parts of the system? Furthermore, it is important to 898 consider who has access to the keys. The policies for the 899 handling of the keys, and who can access the keys, need to be 900 considered along with the confidentiality goals. 902 As can be seen the actual confidentiality level has likely more to do 903 with the application's usage of centralized nodes, and the details of 904 the key-management solution chosen, than with the actual choice of 905 encryption algorithm (although, of course, the encryption algorithm 906 needs to be chosen appropriately for the desired security level). 908 4.1.2. Integrity 910 Protection against modification of content by a third party, or due 911 to errors in the network, is another factor to consider. The first 912 aspect that one considers is what resilience one has against 913 modifications to the content. Some media types are extremely 914 sensitive to network bit errors, whereas others might be able to 915 tolerate some degree of data corruption. Equally important is to 916 consider the sensitivity of the content, who is providing the 917 integrity assertion, what is the source of the integrity tag, and 918 what are the risks of modifications happening prior to that point 919 where protection is applied? These issues affect what cryptographic 920 algorithm is used, and the length of the integrity tags, and whether 921 the entire payload is protected. 923 RTP applications that rely on central nodes need to consider if hop- 924 by-hop integrity is acceptable, or if true end-to-end integrity 925 protection is needed? Is it important to be able to tell if a 926 middlebox has modified the data? There are some uses of RTP that 927 require trusted middleboxes that can modify the data in a way that 928 doesn't break integrity protection as seen by the receiver, for 929 example local advertisement insertion in IPTV systems; there are also 930 uses where it is essential that such in-network modification be 931 detectable. RTP can support both, with appropriate choices of 932 security mechanisms. 934 Integrity of the data is commonly closely tied to the question of 935 source authentication. That is, it becomes important to know who 936 makes an integrity assertion for the data. 938 4.1.3. Source Authentication 940 Source authentication is about determining who sent a particular RTP 941 or RTCP packet. It is normally closely tied with integrity, since a 942 receiver generally also wants to ensure that the data received is 943 what the source really sent, so source authentication without 944 integrity is not particularly useful. Similarly, integrity 945 protection without source authentication is also not particularly 946 useful; a claim that a packet is unchanged that cannot itself be 947 validated as from the source (or some from other known and trusted 948 party) is meaningless. 950 Source authentication can be asserted in several different ways: 952 Base level: Using cryptographic mechanisms that give authentication 953 with some type of key-management provide an implicit method for 954 source authentication. Assuming that the mechanism has sufficient 955 strength to not be circumvented in the time frame when you would 956 accept the packet as valid, it is possible to assert a source- 957 authenticated statement; this message is likely from a source that 958 has the cryptographic key(s) to this communication. 960 What that assertion actually means is highly dependent on the 961 application and how it handles the keys. If only the two peers 962 have access to the keys, this can form a basis for a strong trust 963 relationship that traffic is authenticated coming from one of the 964 peers. However, in a multi-party scenario where security contexts 965 are shared among participants, most base-level authentication 966 solutions can't even assert that this packet is from the same 967 source as the previous packet. 969 Binding the source and the signalling: A step up in the assertion 970 that can be done in base-level systems is to tie the signalling to 971 the key-exchange. Here, the goal is to at least be able to assert 972 that the source of the packets is the same entity that the 973 receiver established the session with. How feasible this is 974 depends on the properties of the key-management system, the 975 ability to tie the signalling to a particular source, and the 976 degree of trust the receiver places on the different nodes 977 involved. 979 For example, systems where the key-exchange is done using the 980 signalling systems, such as Security Descriptions [RFC4568], 981 enable a direct binding between signalling and key-exchange. In 982 such systems, the actual security depends on the trust one can 983 place in the signalling system to correctly associate the peer's 984 identity with the key-exchange. 986 Using Identities: If the applications have access to a system that 987 can provide verifiable identities, then the source authentication 988 can be bound to that identity. For example, in a point-to-point 989 communication even symmetric key crypto, where the key-management 990 can assert that the key has only been exchanged with a particular 991 identity, can provide a strong assertion about the source of the 992 traffic. SIP identity [RFC4474] provides one example of how this 993 can be done, and could be used to bind DTLS-SRTP certificates to 994 the identity provider's public key to authenticate the source of a 995 DTLS-SRTP flow. 997 Note that all levels of the system need to have matching 998 capability to assert identity. If the signalling can assert that 999 only a given entity in a multiparty session has a key, then the 1000 media layer might be able to provide guarantees about the identity 1001 of the media sender. However, using an signalling authentication 1002 mechanism built on a group key can limit the media layer to 1003 asserting only group membership. 1005 4.1.4. Identity 1007 There exist many different types of identity systems with different 1008 properties (e.g., SIP identity [RFC4474]). In the context of RTP 1009 applications, the most important property is the possibility to 1010 perform source authentication and verify such assertions in relation 1011 to any claimed identities. What an identity really is can also vary 1012 but, in the context of communication, one of the most obvious is the 1013 identity of the human user one communicates with. However, the human 1014 user can also have additional identities in a particular role. For 1015 example, the human Alice, can also be a police officer and in some 1016 cases her identity as police officer will be more relevant then that 1017 she is Alice. This is common in contact with organizations, where it 1018 is important to prove the persons right to represent the 1019 organization. Some examples of identity mechanisms that can be used: 1021 Certificate based: A certificate is used to prove the identity, by 1022 having access to the private part of the certificate one can 1023 perform signing to assert ones identity. Any entity interested in 1024 verifying the assertion then needs the public part of the 1025 certificate. By having the certificate, one can verify the 1026 signature against the certificate. The next step is to determine 1027 if one trusts the certificate's trust chain. Commonly by 1028 provisioning the verifier with the public part of a root 1029 certificate, this enables the verifier to verify a trust chain 1030 from the root certificate down to the identity certificate. 1031 However, the trust is based on all steps in the certificate chain 1032 being verifiable and trusted. Thus provisioning of root 1033 certificates and the ability to revoke compromised certificates 1034 are aspects that will require infrastructure. 1036 Online Identity Providers: An online identity provider (IdP) can 1037 authenticate a user's right to use an identity, then perform 1038 assertions on their behalf or provision the requester with short- 1039 term credentials to assert their identity. The verifier can then 1040 contact the IdP to request verification of a particular identity. 1041 Here the trust is highly dependent on how much one trusts the IdP. 1042 The system also becomes dependent on having access to the relevant 1043 IdP. 1045 In all of the above examples, an important part of the security 1046 properties are related to the method for authenticating the access to 1047 the identity. 1049 4.1.5. Privacy 1051 RTP applications need to consider what privacy goals they have. As 1052 RTP applications communicate directly between peers in many cases, 1053 the IP addresses of any communication peer will be available. The 1054 main privacy concern with IP addresses is related to geographical 1055 location and the possibility to track a user of an end-point. The 1056 main way of avoid such concerns is the introduction of relay (e.g., a 1057 TURN server [RFC5766]) or centralized media mixers or forwarders that 1058 hides the address of a peer from any other peer. The security and 1059 trust placed in these relays obviously needs to be carefully 1060 considered. 1062 RTP itself can contribute to enabling a particular user to be tracked 1063 between communication sessions if the CNAME is generated according to 1064 the RTP specification in the form of user@host. Such RTCP CNAMEs are 1065 likely long term stable over multiple sessions, allowing tracking of 1066 users. This can be desirable for long-term fault tracking and 1067 diagnosis, but clearly has privacy implications. Instead 1068 cryptographically random ones could be used as defined by Guidelines 1069 for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs) 1070 [RFC7022]. 1072 If there exist privacy goals, these need to be considered, and the 1073 system designed with them in mind. In addition certain RTP features 1074 might have to be configured to safeguard privacy, or have 1075 requirements on how the implementation is done. 1077 4.2. Application Structure 1079 When it comes to RTP security, the most appropriate solution is often 1080 highly dependent on the topology of the communication session. The 1081 signalling also impacts what information can be provided, and if this 1082 can be instance specific, or common for a group. In the end the key- 1083 management system will highly affect the security properties achieved 1084 by the application. At the same time, the communication structure of 1085 the application limits what key management methods are applicable. 1086 As different key-management have different requirements on underlying 1087 infrastructure it is important to take that aspect into consideration 1088 early in the design. 1090 4.3. Automatic Key Management 1092 The Guidelines for Cryptographic Key Management [RFC4107] provide an 1093 overview of why automatic key management is important. They also 1094 provide a strong recommendation on using automatic key management. 1095 Most of the security solutions reviewed in this document provide or 1096 support automatic key management, at least to establish session keys. 1097 In some more long term use cases, credentials might in certain cases 1098 need to be be manually deployed. 1100 For SRTP an important aspect of automatic key management is to ensure 1101 that two time pads do not occur, in particular by preventing multiple 1102 end points using the same session key and SSRC. In these cases 1103 automatic key management methods can have strong dependencies on 1104 signalling features to function correctly. If those dependencies 1105 can't be fulfilled, additional constrains on usage, e.g., per-end 1106 point session keys, might be needed to avoid the issue. 1108 When selecting security mechanisms for an RTP application it is 1109 important to consider the properties of the key management. Using 1110 key management that is both automatic and integrated will provide 1111 minimal interruption for the user, and is important to ensure that 1112 security can, and will remain, to be on by default. 1114 4.4. End-to-End Security vs Tunnels 1116 If the security mechanism only provides a secured tunnel, for example 1117 like some common uses of IPSec Section 3.3, it is important to 1118 consider the full end-to-end properties of the system. How does one 1119 ensure that the path from the endpoint to the local tunnel ingress/ 1120 egress is secure and can be trusted (and similarly for the other end 1121 of the tunnel)? How does one handle the source authentication of the 1122 peer, as the security protocol identifies the other end of the 1123 tunnel. These are some of the issues that arise when one considers a 1124 tunnel based security protocol rather than an end-to-end. Even with 1125 clear requirements and knowledge that one still can achieve the 1126 security properties using a tunnel based solution, one ought to 1127 prefer to use end-to-end mechanisms, as they are much less likely to 1128 violate any assumptions made about deployment. These assumptions can 1129 also be difficult to automatically verify. 1131 4.5. Plain Text Keys 1133 Key management solutions that use plain text keys, like SDP Security 1134 Descriptions (Section 3.1.3), require care to ensure a secure 1135 transport of the signalling messages that contain the plain text 1136 keys. For plain text keys the security properties of the system 1137 depend on how securely the plain text keys are protected end-to-end 1138 between the sender and receiver(s). Not only does one need to 1139 consider what transport protection is provided for the signalling 1140 message including the keys, but also the degree to which any 1141 intermediaries in the signalling are trusted. Untrusted 1142 intermediaries can perform man in the middle attacks on the 1143 communication, or can log the keys with the result in encryption 1144 being compromised significantly after the actual communication 1145 occurred. 1147 4.6. Interoperability 1149 Few RTP applications exist as independent applications that never 1150 interoperate with anything else. Rather, they enable communication 1151 with a potentially large number of other systems. To minimize the 1152 number of security mechanisms that need to be implemented, it is 1153 important to consider if one can use the same security mechanisms as 1154 other applications. This can also reduce problems of determining 1155 what security level is actually negotiated in a particular session. 1157 The desire to be interoperable can, in some cases, be in conflict 1158 with the security requirements of an application. To meet the 1159 security goals, it might be necessary to sacrifice interoperability. 1160 Alternatively, one can implement multiple security mechanisms, this 1161 however introduces the complication of ensuring that the user 1162 understands what it means to use a particular security system. In 1163 addition, the application can then become vulnerable to bid-down 1164 attack. 1166 5. Examples 1168 In the following we describe a number of example security solutions 1169 for applications using RTP services or frameworks. These examples 1170 are provided to illustrate the choices available. They are not 1171 normative recommendations for security. 1173 5.1. Media Security for SIP-established Sessions using DTLS-SRTP 1175 The IETF evaluated media security for RTP sessions established using 1176 point-to-point SIP sessions in 2009. A number of requirements were 1177 determined, and based on those, the existing solutions for media 1178 security and especially the keying methods were analysed. The 1179 resulting requirements and analysis were published in [RFC5479]. 1180 Based on this analysis and working group discussion, DTLS-SRTP was 1181 determined to be the best solution. 1183 The security solution for SIP using DTLS-SRTP is defined in the 1184 Framework for Establishing a Secure Real-time Transport Protocol 1185 (SRTP) Security Context Using Datagram Transport Layer Security 1186 (DTLS) [RFC5763]. On a high level the framework uses SIP with SDP 1187 offer/answer procedures to exchange the network addresses where the 1188 server end-point will have a DTLS-SRTP enable server running. The 1189 SIP signalling is also used to exchange the fingerprints of the 1190 certificate each end-point will use in the DTLS establishment 1191 process. When the signalling is sufficiently completed, the DTLS- 1192 SRTP client performs DTLS handshakes and establishes SRTP session 1193 keys. The clients also verify the fingerprints of the certificates 1194 to verify that no man in the middle has inserted themselves into the 1195 exchange. 1197 DTLS has a number of good security properties. For example, to 1198 enable a man in the middle someone in the signalling path needs to 1199 perform an active action and modify both the signalling message and 1200 the DTLS handshake. There also exists solutions that enables the 1201 fingerprints to be bound to identities. SIP Identity provides an 1202 identity established by the first proxy for each user [RFC4474]. 1203 This reduces the number of nodes the connecting user User Agent has 1204 to trust to include just the first hop proxy, rather than the full 1205 signalling path. The biggest security weakness of this system is its 1206 dependency on the signalling. SIP signalling passes multiple nodes 1207 and there is usually no message security deployed, only hop-by-hop 1208 transport security, if any, between the nodes. 1210 5.2. Media Security for WebRTC Sessions 1212 Web Real-Time Communication (WebRTC) [I-D.ietf-rtcweb-overview] is a 1213 solution providing JavaScript web applications with real-time media 1214 directly between browsers. Media is transported using RTP protected 1215 using a mandatory application of SRTP [RFC3711], with keying done 1216 using DTLS-SRTP [RFC5764]. The security configuration is further 1217 defined in the WebRTC Security Architecture 1218 [I-D.ietf-rtcweb-security-arch]. 1220 A hash of the peer's certificate is provided to the JavaScript web 1221 application, allowing that web application to verify identity of the 1222 peer. There are several ways in which the certificate hashes can be 1223 verified. An approach identified in the WebRTC security architecture 1224 [I-D.ietf-rtcweb-security-arch] is to use an identity provider. In 1225 this solution the Identity Provider, which is a third party to the 1226 web application, signs the DTLS-SRTP hash combined with a statement 1227 on the validity of the user identity that has been used to sign the 1228 hash. The receiver of such an identity assertion can then 1229 independently verify the user identity to ensure that it is the 1230 identity that the receiver intended to communicate with, and that the 1231 cryptographic assertion holds; this way a user can be certain that 1232 the application also can't perform a MITM and acquire the keys to the 1233 media communication. Other ways of verifying the certificate hashes 1234 exist, for example they could be verified against a hash carried in 1235 some out of band channel (e.g., compare with a hash printed on a 1236 business card), or using a verbal short authentication string (e.g., 1237 as in ZRTP [RFC6189]), or using hash continuity. 1239 In the development of WebRTC there has also been attention given to 1240 privacy considerations. The main RTP-related concerns that have been 1241 raised are: 1243 Location Disclosure: As ICE negotiation [RFC5245] provides IP 1244 addresses and ports for the browser, this leaks location 1245 information in the signalling to the peer. To prevent this one 1246 can block the usage of any ICE candidate that isn't a relay 1247 candidate, i.e. where the IP and port provided belong to the 1248 service providers media traffic relay. 1250 Prevent tracking between sessions: static RTP CNAMEs and DTLS-SRTP 1251 certificates provide information that is re-used between session 1252 instances. Thus to prevent tracking, such information is ought 1253 not be re-used between sessions, or the information ought not sent 1254 in the clear. Note, that generating new certificates each time 1255 prevents continuity in authentication, however, as WebRTC users 1256 are expected to use multiple devices to access the same 1257 communication service, such continuity can't be expected anyway, 1258 instead the above described identity mechanism has to be relied 1259 on. 1261 Note: The above cases are focused on providing privacy from other 1262 parties, not on providing privacy from the web server that provides 1263 the WebRTC Javascript application. 1265 5.3. IP Multimedia Subsystem (IMS) Media Security 1267 In IMS, the core network is controlled by a single operator, or by 1268 several operators with high trust in each other. Except for some 1269 types of accesses, the operator is in full control, and no packages 1270 are routed over the Internet. Nodes in the core network offer 1271 services such as voice mail, interworking with legacy systems (PSTN, 1272 GSM, and 3G), and transcoding. End-points are authenticated during 1273 the SIP registration using either IMS-AKA (using SIM credentials) or 1274 SIP Digest (using password). 1276 In IMS media security [T3GPP.33.328], end-to-end encryption is 1277 therefore not seen as needed or desired as it would hinder for 1278 example interworking and transcoding, making calls between 1279 incompatible terminals impossible. Because of this IMS media 1280 security mostly uses end-to-access-edge security where SRTP is 1281 terminated in the first node in the core network. As the SIP 1282 signaling is trusted and encrypted (with TLS or IPsec), security 1283 descriptions [RFC4568] is considered to give good protection against 1284 eavesdropping over the accesses that are not already encrypted (GSM, 1285 3G, LTE). Media source authentication is based on knowledge of the 1286 SRTP session key and trust in that the IMS network will only forward 1287 media from the correct end-point. 1289 For enterprises and government agencies, which might have weaker 1290 trust in the IMS core network and can be assumed to have compatible 1291 terminals, end-to-end security can be achieved by deploying their own 1292 key management server. 1294 Work on Interworking with WebRTC is currently ongoing; the security 1295 will still be end-to-access-edge, but using DTLS-SRTP [RFC5763] 1296 instead of security descriptions. 1298 5.4. 3GPP Packet Based Streaming Service (PSS) 1300 The 3GPP Release 11 PSS specification of the Packet Based Streaming 1301 Service (PSS) [T3GPP.26.234R11] defines, in Annex R, a set of 1302 security mechanisms. These security mechanisms are concerned with 1303 protecting the content from being copied, i.e. Digital Rights 1304 Management. To meet these goals with the specified solution, the 1305 client implementation and the application platform are trusted to 1306 protect against access and modification by an attacker. 1308 PSS is RTSP 1.0 [RFC2326] controlled media streaming over RTP. Thus 1309 an RTSP client whose user wants to access a protected content will 1310 request a session description (SDP [RFC4566]) for the protected 1311 content. This SDP will indicate that the media is ISMACryp 2.0 1312 [ISMACryp2] protected media encoding application units (AUs). The 1313 key(s) used to protect the media are provided in either of two ways. 1314 If a single key is used then the client uses some DRM system to 1315 retrieve the key as indicated in the SDP. Commonly OMA DRM v2 1316 [OMADRMv2] will be used to retrieve the key. If multiple keys are to 1317 be used, then an additional RTSP stream for key-updates in parallel 1318 with the media streams is established, where key updates are sent to 1319 the client using Short Term Key Messages defined in the "Service and 1320 Content Protection for Mobile Broadcast Services" section of the OMA 1321 Mobile Broadcast Services [OMABCAST]. 1323 Worth noting is that this solution doesn't provide any integrity 1324 verification method for the RTP header and payload header 1325 information, only the encoded media AU is protected. 3GPP has not 1326 defined any requirement for supporting any solution that could 1327 provide that service. Thus, replay or insertion attacks are 1328 possible. Another property is that the media content can be 1329 protected by the ones providing the media, so that the operators of 1330 the RTSP server has no access to unprotected content. Instead all 1331 that want to access the media is supposed to contact the DRM keying 1332 server and if the device is acceptable they will be given the key to 1333 decrypt the media. 1335 To protect the signalling, RTSP 1.0 supports the usage of TLS. This 1336 is, however, not explicitly discussed in the PSS specification. 1337 Usage of TLS can prevent both modification of the session description 1338 information and help maintain some privacy of what content the user 1339 is watching as all URLs would then be confidentiality protected. 1341 5.5. RTSP 2.0 1343 Real-time Streaming Protocol 2.0 [I-D.ietf-mmusic-rfc2326bis] offers 1344 an interesting comparison to the PSS service (Section 5.4) that is 1345 based on RTSP 1.0 and service requirements perceived by mobile 1346 operators. A major difference between RTSP 1.0 and RTSP 2.0 is that 1347 2.0 is fully defined under the requirement to have mandatory to 1348 implement security mechanism. As it specifies how one transport 1349 media over RTP it is also defining security mechanisms for the RTP 1350 transported media streams. 1352 The security goals for RTP in RTSP 2.0 is to ensure that there is 1353 confidentiality, integrity and source authentication between the RTSP 1354 server and the client. This to prevent eavesdropping on what the 1355 user is watching for privacy reasons and to prevent replay or 1356 injection attacks on the media stream. To reach these goals, the 1357 signalling also has to be protected, requiring the use of TLS between 1358 the client and server. 1360 Using TLS-protected signalling the client and server agree on the 1361 media transport method when doing the SETUP request and response. 1362 The secured media transport is SRTP (SAVP/RTP) normally over UDP. 1363 The key management for SRTP is MIKEY using RSA-R mode. The RSA-R 1364 mode is selected as it allows the RTSP Server to select the key 1365 despite having the RTSP Client initiate the MIKEY exchange. It also 1366 enables the reuse of the RTSP servers TLS certificate when creating 1367 the MIKEY messages thus ensuring a binding between the RTSP server 1368 and the key exchange. Assuming the SETUP process works, this will 1369 establish a SRTP crypto context to be used between the RTSP Server 1370 and the Client for the RTP transported media streams. 1372 6. IANA Considerations 1374 This document makes no request of IANA. 1376 Note to RFC Editor: this section can be removed on publication as an 1377 RFC. 1379 7. Security Considerations 1381 This entire document is about security. Please read it. 1383 8. Acknowledgements 1385 We thank the IESG for their careful review of 1386 [I-D.ietf-avt-srtp-not-mandatory] which led to the writing of this 1387 memo. John Mattsson has contributed the IMS Media Security example 1388 (Section 5.3). 1390 The authors wished to thank Christian Correll, Dan Wing, Kevin Gross, 1391 Alan Johnston, Michael Peck, Ole Jacobsen, and John Mattsson for 1392 review and proposals for improvements of the text. 1394 9. Informative References 1396 [I-D.ietf-avt-srtp-not-mandatory] 1397 Perkins, C. and M. Westerlund, "Securing the RTP Protocol 1398 Framework: Why RTP Does Not Mandate a Single Media 1399 Security Solution", draft-ietf-avt-srtp-not-mandatory-14 1400 (work in progress), October 2013. 1402 [I-D.ietf-avtcore-aria-srtp] 1403 Kim, W., Lee, J., Kim, D., Park, J., and D. Kwon, "The 1404 ARIA Algorithm and Its Use with the Secure Real-time 1405 Transport Protocol(SRTP)", draft-ietf-avtcore-aria-srtp-05 1406 (work in progress), September 2013. 1408 [I-D.ietf-avtcore-srtp-aes-gcm] 1409 McGrew, D. and K. Igoe, "AES-GCM and AES-CCM Authenticated 1410 Encryption in Secure RTP (SRTP)", draft-ietf-avtcore-srtp- 1411 aes-gcm-10 (work in progress), September 2013. 1413 [I-D.ietf-avtcore-srtp-ekt] 1414 McGrew, D. and D. Wing, "Encrypted Key Transport for 1415 Secure RTP", draft-ietf-avtcore-srtp-ekt-01 (work in 1416 progress), October 2013. 1418 [I-D.ietf-mmusic-rfc2326bis] 1419 Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M., 1420 and M. Stiemerling, "Real Time Streaming Protocol 2.0 1421 (RTSP)", draft-ietf-mmusic-rfc2326bis-38 (work in 1422 progress), October 2013. 1424 [I-D.ietf-rtcweb-overview] 1425 Alvestrand, H., "Overview: Real Time Protocols for Brower- 1426 based Applications", draft-ietf-rtcweb-overview-08 (work 1427 in progress), September 2013. 1429 [I-D.ietf-rtcweb-security-arch] 1430 Rescorla, E., "WebRTC Security Architecture", draft-ietf- 1431 rtcweb-security-arch-07 (work in progress), July 2013. 1433 [ISMACryp2] 1434 Internet Streaming Media Alliance (ISMA), "ISMA Encryption 1435 and Authentication, Version 2.0 release version", November 1436 2007. 1438 [OMABCAST] 1439 Open Mobile Alliance, "OMA Mobile Broadcast Services 1440 V1.0", February 2009. 1442 [OMADRMv2] 1443 Open Mobile Alliance, "OMA Digital Rights Management 1444 V2.0", July 2008. 1446 [RFC1112] Deering, S., "Host extensions for IP multicasting", STD 5, 1447 RFC 1112, August 1989. 1449 [RFC2326] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time 1450 Streaming Protocol (RTSP)", RFC 2326, April 1998. 1452 [RFC3365] Schiller, J., "Strong Security Requirements for Internet 1453 Engineering Task Force Standard Protocols", BCP 61, RFC 1454 3365, August 2002. 1456 [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. 1457 Jacobson, "RTP: A Transport Protocol for Real-Time 1458 Applications", STD 64, RFC 3550, July 2003. 1460 [RFC3640] van der Meer, J., Mackie, D., Swaminathan, V., Singer, D., 1461 and P. Gentric, "RTP Payload Format for Transport of 1462 MPEG-4 Elementary Streams", RFC 3640, November 2003. 1464 [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. 1465 Norrman, "The Secure Real-time Transport Protocol (SRTP)", 1466 RFC 3711, March 2004. 1468 [RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K. 1469 Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830, 1470 August 2004. 1472 [RFC4107] Bellovin, S. and R. Housley, "Guidelines for Cryptographic 1473 Key Management", BCP 107, RFC 4107, June 2005. 1475 [RFC4301] Kent, S. and K. Seo, "Security Architecture for the 1476 Internet Protocol", RFC 4301, December 2005. 1478 [RFC4383] Baugher, M. and E. Carrara, "The Use of Timed Efficient 1479 Stream Loss-Tolerant Authentication (TESLA) in the Secure 1480 Real-time Transport Protocol (SRTP)", RFC 4383, February 1481 2006. 1483 [RFC4474] Peterson, J. and C. Jennings, "Enhancements for 1484 Authenticated Identity Management in the Session 1485 Initiation Protocol (SIP)", RFC 4474, August 2006. 1487 [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session 1488 Description Protocol", RFC 4566, July 2006. 1490 [RFC4567] Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E. 1491 Carrara, "Key Management Extensions for Session 1492 Description Protocol (SDP) and Real Time Streaming 1493 Protocol (RTSP)", RFC 4567, July 2006. 1495 [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session 1496 Description Protocol (SDP) Security Descriptions for Media 1497 Streams", RFC 4568, July 2006. 1499 [RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP) 1500 and RTP Control Protocol (RTCP) Packets over Connection- 1501 Oriented Transport", RFC 4571, July 2006. 1503 [RFC4572] Lennox, J., "Connection-Oriented Media Transport over the 1504 Transport Layer Security (TLS) Protocol in the Session 1505 Description Protocol (SDP)", RFC 4572, July 2006. 1507 [RFC4607] Holbrook, H. and B. Cain, "Source-Specific Multicast for 1508 IP", RFC 4607, August 2006. 1510 [RFC4650] Euchner, M., "HMAC-Authenticated Diffie-Hellman for 1511 Multimedia Internet KEYing (MIKEY)", RFC 4650, September 1512 2006. 1514 [RFC4738] Ignjatic, D., Dondeti, L., Audet, F., and P. Lin, "MIKEY- 1515 RSA-R: An Additional Mode of Key Distribution in 1516 Multimedia Internet KEYing (MIKEY)", RFC 4738, November 1517 2006. 1519 [RFC4771] Lehtovirta, V., Naslund, M., and K. Norrman, "Integrity 1520 Transform Carrying Roll-Over Counter for the Secure Real- 1521 time Transport Protocol (SRTP)", RFC 4771, January 2007. 1523 [RFC4949] Shirey, R., "Internet Security Glossary, Version 2", RFC 1524 4949, August 2007. 1526 [RFC5117] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, 1527 January 2008. 1529 [RFC5197] Fries, S. and D. Ignjatic, "On the Applicability of 1530 Various Multimedia Internet KEYing (MIKEY) Modes and 1531 Extensions", RFC 5197, June 2008. 1533 [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment 1534 (ICE): A Protocol for Network Address Translator (NAT) 1535 Traversal for Offer/Answer Protocols", RFC 5245, April 1536 2010. 1538 [RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security 1539 (TLS) Protocol Version 1.2", RFC 5246, August 2008. 1541 [RFC5479] Wing, D., Fries, S., Tschofenig, H., and F. Audet, 1542 "Requirements and Analysis of Media Security Management 1543 Protocols", RFC 5479, April 2009. 1545 [RFC5669] Yoon, S., Kim, J., Park, H., Jeong, H., and Y. Won, "The 1546 SEED Cipher Algorithm and Its Use with the Secure Real- 1547 Time Transport Protocol (SRTP)", RFC 5669, August 2010. 1549 [RFC5760] Ott, J., Chesterfield, J., and E. Schooler, "RTP Control 1550 Protocol (RTCP) Extensions for Single-Source Multicast 1551 Sessions with Unicast Feedback", RFC 5760, February 2010. 1553 [RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework 1554 for Establishing a Secure Real-time Transport Protocol 1555 (SRTP) Security Context Using Datagram Transport Layer 1556 Security (DTLS)", RFC 5763, May 2010. 1558 [RFC5764] McGrew, D. and E. Rescorla, "Datagram Transport Layer 1559 Security (DTLS) Extension to Establish Keys for the Secure 1560 Real-time Transport Protocol (SRTP)", RFC 5764, May 2010. 1562 [RFC5766] Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using 1563 Relays around NAT (TURN): Relay Extensions to Session 1564 Traversal Utilities for NAT (STUN)", RFC 5766, April 2010. 1566 [RFC6043] Mattsson, J. and T. Tian, "MIKEY-TICKET: Ticket-Based 1567 Modes of Key Distribution in Multimedia Internet KEYing 1568 (MIKEY)", RFC 6043, March 2011. 1570 [RFC6188] McGrew, D., "The Use of AES-192 and AES-256 in Secure 1571 RTP", RFC 6188, March 2011. 1573 [RFC6189] Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media 1574 Path Key Agreement for Unicast Secure RTP", RFC 6189, 1575 April 2011. 1577 [RFC6267] Cakulev, V. and G. Sundaram, "MIKEY-IBAKE: Identity-Based 1578 Authenticated Key Exchange (IBAKE) Mode of Key 1579 Distribution in Multimedia Internet KEYing (MIKEY)", RFC 1580 6267, June 2011. 1582 [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer 1583 Security Version 1.2", RFC 6347, January 2012. 1585 [RFC6509] Groves, M., "MIKEY-SAKKE: Sakai-Kasahara Key Encryption in 1586 Multimedia Internet KEYing (MIKEY)", RFC 6509, February 1587 2012. 1589 [RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of 1590 Variable Bit Rate Audio with Secure RTP", RFC 6562, March 1591 2012. 1593 [RFC6904] Lennox, J., "Encryption of Header Extensions in the Secure 1594 Real-time Transport Protocol (SRTP)", RFC 6904, April 1595 2013. 1597 [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, 1598 "Guidelines for Choosing RTP Control Protocol (RTCP) 1599 Canonical Names (CNAMEs)", RFC 7022, September 2013. 1601 [T3GPP.26.234R11] 1602 3GPP, "Technical Specification Group Services and System 1603 Aspects; Transparent end-to-end Packet-switched Streaming 1604 Service (PSS); Protocols and codecs", 3GPP TS 26.234 1605 11.1.0, September 2012. 1607 [T3GPP.26.234R8] 1608 3GPP, "Technical Specification Group Services and System 1609 Aspects; Transparent end-to-end Packet-switched Streaming 1610 Service (PSS); Protocols and codecs", 3GPP TS 26.234 1611 8.4.0, September 2009. 1613 [T3GPP.26.346] 1614 3GPP, "Multimedia Broadcast/Multicast Service (MBMS); 1615 Protocols and codecs", 3GPP TS 26.346 10.7.0, March 2013. 1617 [T3GPP.33.246] 1618 3GPP, "3G Security; Security of Multimedia Broadcast/ 1619 Multicast Service (MBMS)", 3GPP TS 33.246 12.1.0, December 1620 2012. 1622 [T3GPP.33.328] 1623 3GPP, "IP Multimedia Subsystem (IMS) media plane 1624 security", 3GPP TS 33.328 12.1.0, December 2012. 1626 Authors' Addresses 1628 Magnus Westerlund 1629 Ericsson 1630 Farogatan 6 1631 SE-164 80 Kista 1632 Sweden 1634 Phone: +46 10 714 82 87 1635 Email: magnus.westerlund@ericsson.com 1637 Colin Perkins 1638 University of Glasgow 1639 School of Computing Science 1640 Glasgow G12 8QQ 1641 United Kingdom 1643 Email: csp@csperkins.org 1644 URI: http://csperkins.org/