SIP J. Fischl Internet-Draft CounterPath Corporation
Expires: August 26, 2008Intended status: Standards Track H. Tschofenig Expires: January 14, 2009 Nokia Siemens Networks E. Rescorla Network Resonance February 23,RTFM, Inc. July 13, 2008 Framework for Establishing an SRTP Security Context using DTLS draft-ietf-sip-dtls-srtp-framework-01.txtdraft-ietf-sip-dtls-srtp-framework-02.txt Status of this Memo By submitting this Internet-Draft, each author represents that any applicable patent or other IPR claims of which he or she is aware have been or will be disclosed, and any of which he or she becomes aware will be disclosed, in accordance with Section 6 of BCP 79. 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 to cite them other than as "work in progress." The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt. The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. This Internet-Draft will expire on August 26, 2008.January 14, 2009. Copyright Notice Copyright (C) The IETF Trust (2008). Abstract This document specifies how to use the Session Initiation Protocol (SIP) to establish an Secure Real-time Transport Protocol (SRTP) security context using the Datagram Transport Layer Security (DTLS) protocol. It describes a mechanism of transporting a fingerprint attribute in the Session Description Protocol (SDP) that identifies the key that will be presented during the DTLS handshake. It relies on the SIP identity mechanism to ensure the integrity of the fingerprint attribute.The key exchange travels along the media path as opposed to the signaling path. The SIP Identity mechanism can be used to protect the integrity of the fingerprint attribute from modification by intermediate proxies. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Overview . . . . . . . . . . . . . . . . . . . . . . . . . . . 5 3. Motivation . . . . . . . . . . . . . . . . . . . . . . . . . . 67 4. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 7 5. Exchanging Certificates . . . . . . . . . . . . . . . . . . . 78 6. Miscellaneous Considerations . . . . . . . . . . . . . . . . . 910 6.1. Anonymous Calls . . . . . . . . . . . . . . . . . . . . . 910 6.2. Early Media . . . . . . . . . . . . . . . . . . . . . . . 910 6.3. Forking . . . . . . . . . . . . . . . . . . . . . . . . . 10 6.4. Delayed Offer Calls . . . . . . . . . . . . . . . . . . . 1011 6.5. Session Modification . . . . . . . . . . . . . . . . . . . 1011 6.6. ICE Interaction . . . . . . . . . . . . . . . . . . . . . 1011 6.7. Rekeying . . . . . . . . . . . . . . . . . . . . . . . . . 1112 6.8. Conference Servers and Shared Encryptions Contexts . . . . 1112 6.9. Media over SRTP . . . . . . . . . . . . . . . . . . . . . 12 6.10. Best Effort Encryption . . . . . . . . . . . . . . . . . . 1213 7. Example Message Flow . . . . . . . . . . . . . . . . . . . . . 1213 8. Security Considerations . . . . . . . . . . . . . . . . . . . 1720 8.1. UPDATE . . . . . .Responder Identity . . . . . . . . . . . . . . . . . . . . 1820 8.2. SIPS . . . . . . . . . . . . . . . . . . . . . . . . . . . 1821 8.3. S/MIME . . . . . . . . . . . . . . . . . . . . . . . . . . 1921 8.4. Single-sided Verification .Continuity of Authentication . . . . . . . . . . . . . . . 1922 8.5. Continuity ofShort Authentication String . . . . . . . . . . . . . . . 1922 8.6. Short Authentication String .Limits of Identity Assertions . . . . . . . . . . . . . . 1922 8.7. Perfect Forward Secrecy . . . . . . . . . . . . . . . . . 2024 9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 2024 10. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 2024 11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 2124 11.1. Normative References . . . . . . . . . . . . . . . . . . . 2124 11.2. Informational References . . . . . . . . . . . . . . . . . 2225 Appendix A. Requirements Analysis . . . . . . . . . . . . . . . . 2428 A.1. Forking and retargeting (R-FORK-RETARGET, R-BEST-SECURE, R-DISTINCT) . . . . . . . . . . . . . . . . 2428 A.2. Distinct Cryptographic Contexts (R-DISTINCT) . . . . . . . 2428 A.3. Reusage of a Security Context (R-REUSE) . . . . . . . . . 2428 A.4. Clipping (R-AVOID-CLIPPING) . . . . . . . . . . . . . . . 2428 A.5. Passive Attacks on the Media Path (R-PASS-MEDIA) . . . . . 2428 A.6. Passive Attacks on the Signaling Path (R-PASS-SIG) . . . . 2428 A.7. (R-SIG-MEDIA, R-ACT-ACT) . . . . . . . . . . . . . . . . . 2529 A.8. Binding to Identifiers (R-ID-BINDING) . . . . . . . . . . 2529 A.9. Perfect Forward Secrecy (R-PFS) . . . . . . . . . . . . . 2529 A.10. Algorithm Negotiation (R-COMPUTE) . . . . . . . . . . . . 2529 A.11. RTP Validity Check (R-RTP-VALID) . . . . . . . . . . . . . 2529 A.12. 3rd Party Certificates (R-CERTS, R-EXISTING) . . . . . . . 2630 A.13. FIPS 140-2 (R-FIPS) . . . . . . . . . . . . . . . . . . . 2630 A.14. Linkage between Keying Exchange and SIP Signaling (R-ASSOC) . . . . . . . . . . . . . . . . . . . . . . . . 2630 A.15. Denial of Service Vulnerability (R-DOS) . . . . . . . . . 2630 A.16. Adversary Model (R-SIG-MEDIA) . . . . . . . . . . . . . . 2630 A.17. Crypto-Agility (R-AGILITY) . . . . . . . . . . . . . . . . 2630 A.18. Downgrading Protection (R-DOWNGRADE) . . . . . . . . . . . 2630 A.19. Media Security Negotation (R-NEGOTIATE) . . . . . . . . . 2630 A.20. Signaling Protocol Independence (R-OTHER-SIGNALING) . . . 2731 A.21. Media Recording (R-RECORDING) . . . . . . . . . . . . . . 2731 A.22. Interworking with Intermediaries (R-TRANSCODER) . . . . . 2731 A.23. PSTN Gateway Termination (R-PSTN) . . . . . . . . . . . . 2731 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 2831 Intellectual Property and Copyright Statements . . . . . . . . . . 2933 1. Introduction The Session Initiation Protocol (SIP) [RFC3261] and the Session Description Protocol (SDP) [RFC4566] are used to set up multimedia sessions or calls. SDP is also used to set up TCP [RFC4145] and additionally TCP/TLS connections for usage with media sessions [RFC4572]. The Real-time Transport Protocol (RTP) [RFC3550] is used to transmit real time media on top of UDP and TCP [RFC4571]. Datagram TLS [RFC4347] was introduced to allow TLS functionality to be applied to datagram transport protocols, such as UDP and DCCP. This draft provides guidelines on how to establish SRTP [RFC3711] security using extensionsan extension to DTLS (see [I-D.ietf-avt-dtls-srtp]). The goal of this work is to provide a key negotiation technique that allows encrypted communication between devices with no prior relationships. It also does not require the devices to trust every call signaling element that was involved in routing or session setup. This approach does not require any extra effort by end users and does not require deployment of certificates that are signed by a well- known certificate authority to all devices. The media is transported over a mutually authenticated DTLS session where both sides have certificates. It is very important to note that certificates are being used purely as a carrier for the public keys of the peers. This is required because DTLS does not have a mode for carrying bare keys, but it is purely an issue of formatting. The certificates can be self-signed and completely self-generated. All major TLS stacks have the capability to generate such certificates on demand. However, third party certificates MAY also be used for extra security. The certificate fingerprints are sent in SDP over SIP as part of the offer/answer exchange. The SIP Identityfingerprint mechanism [RFC4474] is usedallows one side of the connection to provideverify that the certificate presented in the DTLS handshake matches the certificate used by the party in the signalling. However, this requires some form of integrity forprotection on the fingerprints.signalling. S/MIME signatures, as described in RFC 3261, or SIP Identity, as described in [RFC4474] provides the highest level of security because they are not susceptible to modification by malicious intermediaries. However, even hop-by-hop security such as provided by SIPS provides some protection against modification by attackers who are not on the signalling path. This DTLS-SRTPapproach differs from previous attempts to secure media traffic where the authentication and key exchange protocol (e.g., MIKEY [RFC3830]) is piggybacked in the signaling message exchange. With DTLS-SRTP, establishing the protection of the media traffic between the endpoints is done by the media endpoints without involving the SIP/SDP communication. It allows RTP and SIP to be used in the usual manner when there is no encrypted media. In SIP, typically the caller sends an offer and the callee may subsequently send one-way media back to the caller before a SIP answer is received by the caller. The approach in this specification, where the media key negotiation is decoupled from the SIP signaling, allows the early media to be set up before the SIP answer is received while preserving the important security property of allowing the media sender to choose some of the keying material for the media. This also allows the media sessions to be changed, re-keyed, and otherwise modified after the initial SIP signaling without any additional SIP signaling. Design decisions that influence the applicability of this specification are discussed in Section 3. 2. Overview Endpoints wishing to set up an RTP media session do so by exchanging offers and answers in SDP messages over SIP. In a typical use case, two endpoints would negotiate to transmit audio data over RTP using the UDP protocol. Figure 1 shows a typical message exchange in the SIP Trapezoid. +-----------+ +-----------+ |SIP | SIP/SDP |SIP | +------>|Proxy |----------->|Proxy |-------+ | |Server X | (+finger- |Server Y | | | +-----------+ print, +-----------+ | | +auth.id.) | | SIP/SDP SIP/SDP | | (+fingerprint) (+fingerprint,| | +auth.id.) | | | | v +-----------+ Datagram TLS +-----------+ |SIP | <-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-> |SIP | |User Agent | Media |User Agent | |Alice@X | <=================================> |Bob@Y | +-----------+ +-----------+ Legend: ------>: Signaling Traffic <-+-+->: Key Management Traffic <=====>: Data Traffic Figure 1: DTLS Usage in the SIP Trapezoid Consider Alice wanting to set up an encrypted audio session with Bob. Both Bob and Alice could use public-key based authentication in order to establish a confidentiality protected channel using DTLS. Since providing mutual authentication between two arbitrary end points on the Internet using public key based cryptography tends to be problematic, we consider more deployment-friendly alternatives. This document uses one approach and several others are discussed in Section 8. Alice sends an SDP offer to Bob over SIP. If Alice uses only self- signed certificates for the communication with Bob, a fingerprint is included in the SDP offer/answer exchange. This fingerprint is integrity protected usingbinds the identity mechanism defined in Enhancements for Authenticated Identity ManagementDTLS key exchange in SIP [RFC4474]. When Bob receivesthe offer, Bob establishes a mutually authenticated DTLS connection with Alice. At this point Bob can beginmedia plan to the signaling plane. The fingerprint alone protects against active attacks on the media but not active attacks on the signalling. In order to prevent active attacks on the signalling, in Enhancements for Authenticated Identity Management in SIP [RFC4474] is used. When Bob receives the offer, Bob establishes a mutually authenticated DTLS connection with Alice. At this point Bob can begin sending media to Alice. Once Bob accepts Alice's offer and sends an SDP answer to Alice, Alice can begin sending confidential media to Bob. Alice and Bob will verify the fingerprints from the certificates received over the DTLS handshakes match with the fingerprints received in the SDP of the SIP signaling. This provides the security property that Alice knows that the media traffic is going to Bob and vice-versa without necessarily requiring global PKI certificates for Alice and Bob. 3. Motivation Although there is already prior work in this area (e.g., SecureSecurity Descriptions for SDP [RFC4568], Key Management Extensions [RFC4567] combined with MIKEY [RFC3830] for authentication and key exchange), this specification is motivated as follows: o TLS will be used to offer security for connection-oriented media. The design of TLS is well-known and implementations are widely available. o This approach deals with forking and early media without requiring support for PRACK [RFC3262] while preserving the important security property of allowing the offerer to choose keying material for encrypting the media. o The establishment of security protection for the media path is also provided along the media path and not over the signaling path. In many deployment scenarios, the signaling and media traffic travel along a different path through the network. o ThisWhen RFC 4474 Identity is used, this solution works even when the SIP proxies downstream of the identity service are not trusted. There is no need to reveal keys in the SIP signaling or in the SDP message exchange. In order for SDES and MIKEY to provide this security property, they require distribution of certificates to the endpoints that are signed by well known certificate authorities. SDES further requires that the endpoints employ S/MIME to encrypt the keying material. o In this method, SSRC collisions do not result in any extra SIP signaling. o Many SIP endpoints already implement TLS. The changes to existing SIP and RTP usage are minimal even when DTLS-SRTP [I-D.ietf-avt- dtls-srtp][I-D.ietf-avt-dtls-srtp] is used. 4. Terminology The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in [RFC2119]. DTLS/TLS uses the term "session" to refer to a long-lived set of keying material that spans associations. In this document, consistent with SIP/SDP usage, we use it to refer to a multimedia session and use the term "TLS session" to refer to the TLS construct. We use the term "association" to refer to a particular DTLS ciphersuite and keying material set which is associated with a single host/port quartet. The same DTLS/TLS session can be used to establish the keying material for multiple associations. For consistency with other SIP/SDP usage, we use the term "connection" when what's being referred to is a multimedia stream that is not specifically DTLS/TLS. In this document, the term "Mutual DTLS" indicates that both the DTLS client and server present certificates even if one or both certificates are self-signed. 5. Exchanging Certificates The two endpoints in the exchange present their identities as part of the DTLS handshake procedure using certificates. This document uses certificates in the same style as described in Comedia over TLS in SDP [RFC4572]. If self-signed certificates are used, the content of the subjectAltName attribute inside the certificate MAY use the uniform resource identifier (URI) of the user. This is useful for debugging purposes only and is not required to bind the certificate to one of the communication endpoints. The integrity of the certificate is ensured through the fingerprint attribute in the SDP. The subjectAltName is not an important component of the certificate verification. The generation of public/private key pairs is relatively expensive. Endpoints are not required to generate certificates for each session. The offer/answer model, defined in [RFC3264], is used by protocols like the Session Initiation Protocol (SIP) [RFC3261] to set up multimedia sessions. In addition to the usual contents of an SDP [RFC4566] message, each media description ('m' line and associated parameters) will also contain several attributes as specified in [I-D.ietf-avt-dtls-srtp], [RFC4145] and [RFC4572]. The endpoint MUST use the setup attribute defined in [RFC4145]. The endpoint which is the offerer MUST use the setup attribute value of setup:actpass and be prepared to receive a client_hello before it receives the answer. The answerer SHOULD use the setup attribute value of setup:active and will send the client_hello in the media path. The endpoint MUST NOT use the connection attribute defined in [RFC4145]. The endpoint MUST use the certificate fingerprint attribute as specified in [RFC4572]. The certificate presented during the DTLS handshake MUST match the fingerprint exchanged via the signaling path in the SDP. The security properties of this mechanism are described in Section 8. If the fingerprint does not match the hashed certificate then the endpoint MUST tear down the media session immediately.When an endpoint wishes to set up a secure media session with another endpoint it sends an offer in a SIP message to the other endpoint. This offer includes, as part of the SDP payload, the fingerprint of the certificate that the endpoint wants to use. The SIP message containing the offer isSHOULD be sent to the offerer's sip proxy over an integrity protected channel which willSHOULD add an identity header according to the procedures outlined in [RFC4474]. When the far endpoint receives the SIP message it can verify the identity of the sender using the identity header. Since the identity header is a digital signature across several SIP headers, in addition to the bodies of the SIP message, the receiver can also be certain that the message has not been tampered with after the digital signature was applied and added to the SIP message. The far endpoint (answerer) may now establish a mutually authenticated DTLS association to the offerer. After completing the DTLS handshake, information about the authenticated identities, including the certificates, are made available to the endpoint application. The answerer is then able to verify that the offerer's certificate used for authentication in the DTLS handshake can be associated to the certificate fingerprint contained in the offer in the SDP. At this point the answerer may indicate to the end user that the media is secured. The offerer may only tentatively accept the answerer's certificate since it may not yet have the answerer's certificate fingerprint. When the answerer accepts the offer, it provides an answer back to the offerer containing the answerer's certificate fingerprint. At this point the offerer can accept or reject the peer's certificate and the offerer can indicate to the end user that the media is secured. Note that the entire authentication and key exchange for securing the media traffic is handled in the media path through DTLS. The signaling path is only used to verify the peers' certificate fingerprints. 6. Miscellaneous Considerations 6.1. Anonymous Calls DTLS-SRTP does not provide anonymous calling. However, if care is not taken, DTLS-SRTP may allow deanonymizing an otherwise anonymous call.The following procedures shouldoffer and answer MUST be usedconform to prevent deanonymization. When making anonymous calls, a new self-signed certificate SHOULD be used for each call so thatthe calls can not befollowing requirements. o The endpoint MUST use the setup attribute defined in [RFC4145]. The endpoint which is the offerer MUST use the setup attribute value of setup:actpass and be prepared to receive a client_hello before it receives the answer. The answerer SHOULD use the setup attribute value of setup:active and will send the client_hello in the media path. o The endpoint MUST NOT use the connection attribute defined in [RFC4145]. o The endpoint MUST use the certificate fingerprint attribute as specified in [RFC4572]. o The certificate presented during the DTLS handshake MUST match the fingerprint exchanged via the signaling path in the SDP. The security properties of this mechanism are described in Section 8. o If the fingerprint does not match the hashed certificate then the endpoint MUST tear down the media session immediately. 6. Miscellaneous Considerations 6.1. Anonymous Calls DTLS-SRTP does not provide anonymous calling. However, if care is not taken, DTLS-SRTP may allow deanonymizing an otherwise anonymous call. When anonymous calls are being made, the following procedures SHOULD be used to prevent deanonymization. When making anonymous calls, a new self-signed certificate SHOULD be used for each call so that the calls can not be correlated as to being from the same caller. In situations where some degree of correlation is acceptable, the same certificate SHOULD be used for a number of calls in order to enable continuity of authentication, see Section 22.214.171.124. Additionally, it MUST be ensured that the Privacy header [RFC3325] is used in conjunction with the SIP identity mechanism to ensure that the identity of the user is not asserted when enabling anonymous calls. Furthermore, the content of the subjectAltName attribute inside the certificate MUST NOT contain information that either allows correlation or identification of the user that wishes to place an anonymous call. Note that following this recommendation is not sufficient to provide anonymization. 6.2. Early Media If an offer is received by an endpoint that wishes to provide early media, it MUST take the setup:active role and can immediately establish a DTLS association with the other endpoint and begin sending media. The setup:passive endpoint may not yet have validated the fingerprint of the active endpoint's certificate. The security aspects of media handling in this situation are discussed in Section 8. 6.3. Forking In SIP, it is possible for a request to fork to multiple endpoints. Each forked request can result in a different answer. Assuming that the requester provided an offer, each of the answerers' will provide a unique answer. Each answerer will create a DTLS association with the offerer. The offerer can then securely correlate the SDP answer received in the SIP message by comparing the fingerprint in the answer to the hashed certificate for each DTLS association. 6.4. Delayed Offer Calls An endpoint may send a SIP INVITE request with no offer in it. When this occurs, the receiver(s) of the INVITE will provide the offer in the response and the originator will provide the answer in the subsequent ACK request or in the PRACK request [RFC3262] if both endpoints support reliable provisional responses. In any event, the active endpoint still establishes the DTLS association with the passive endpoint as negotiated in the offer/answer exchange. 6.5. Session Modification Once an answer is provided to the offerer, either endpoint MAY request a session modification which MAY include an updated offer. This session modification can be carried in either an INVITE or UPDATE request. Once the answer is received, the active endpoint will either reuse the existing association or establish a new one, tearing down the existing association as soon as the offer/answer exchange is completed. 6.6. ICE Interaction Interactive Connectivity Establishment (ICE), as specified in [I-D.ietf-mmusic-ice], provides a methodology of allowing participants in multi-media sessions to verify mutual connectivity. When ICE is being used the ICE connectivity checks are performed before the DTLS handshake begins. Note that if aggressive nomination mode is used, multiple candidate pairs may be marked valid before ICE finally converges on a single candidate pair. Implementations MUST treat all ICE candidate pairs associated with a single component as part of the same DTLS association. Thus, there will be only one DTLS handshake even if there are multiple valid candidate pairs. Note that this may mean adjusting the endpoint IP addresses if the selected candidate pair shifts, just as if the DTLS packets were an ordinary media stream. Note that STUN packets are sent directly over UDP, not over DTLS. [I-D.ietf-avt-dtls-srtp] describes how to demultiplex STUN packets from DTLS packets and SRTP packets. If ICE is not being used, then there is potential for a bad interaction with SBCs via "latching", as described in [I-D.ietf- mmusic-media-path-middleboxes].[I-D.ietf-mmusic-media-path-middleboxes]. In order to avoid this issue, if ICE is not being used,used and the DTLS handshake has not completed, upon receiving the other side's then the passive side MUST do a single unauthenticadunauthenticated STUN [I-D.ietf-behave-rfc3489bis] connectivity check in order to open up the appropriate pinhole. All implementations MUST be prepared to answer this request during the handshake period even if they do not otherwise do ICE. 6.7. Rekeying As with TLS, DTLS endpoints can rekey at any time by redoing the DTLS handshake. While the rekey is under way, the endpoints continue to use the previously established keying material for usage with DTLS. Once the new session keys are established the session can switch to using these and abandon the old keys. This ensures that latency is not introduced during the rekeying process. Further considerations regarding rekeying in case the SRTP security context is established with DTLS can be found in Section 3.7 of [I-D.ietf-avt-dtls-srtp]. 6.8. Conference Servers and Shared Encryptions Contexts It has been proposed that conference servers might use the same encryption context for all of the participants in a conference. The advantage of this approach is that the conference server only needs to encrypt the output for all speakers instead of once per participant. This shared encryption context approach is not possible under this specification because each DTLS handshake establishes fresh keys which are not completely under the control of either side. However, it is argued that the effort to encrypt each RTP packet is small compared to the other tasks performed by the conference server such as the codec processing. Future extensions such as [I-D.mcgrew-srtp-ekt] or [I-D.wing-avt- dtls-srtp-key-transport][I-D.wing-avt-dtls-srtp-key-transport] could be used to provide this functionality in concert with the mechanisms described in this specification. 6.9. Media over SRTP Because DTLS's data transfer protocol is generic, it is less highly optimized for use with RTP than is SRTP [RFC3711], which has been specifically tuned for that purpose. DTLS-SRTP [I-D.ietf-avt-dtls- srtp],[I-D.ietf-avt-dtls-srtp], has been defined to provide for the negotiation of SRTP transport using a DTLS connection, thus allowing the performance benefits of SRTP with the easy key management of DTLS. The ability to reuse existing SRTP software and hardware implementations may in some environments provide another important motivation for using DTLS-SRTP instead of RTP over DTLS. Implementations of this specification SHOULD support DTLS-SRTP [I-D.ietf-avt-dtls-srtp]. 6.10. Best Effort Encryption [I-D.ietf-sip-media-security-requirements] describes a requirement for best effort encryption where SRTP is used where both endpoints support it and key negotiation succeeds otherwise RTP is used. [I-D.ietf-mmusic-sdp-capability-negotiation] describes a mechanism which can signal both RTP and SRTP as an alternative. RTP is the default and will be understood by endpoints that do not understand SRTP or this key exchange mechanism but SRTP is preferred. 7. Example Message Flow Prior to establishing the session, both Alice and Bob generate self- signed certificates which are used for a single session or, more likely, reused for multiple sessions. In this example, Alice calls Bob. In this example we assume that Alice and Bob share the same proxy. The example shows the SIP message flows where Alice acts as the passive endpoint and Bob acts as the active endpoint meaning that as soon as Bob receives the INVITE from Alice, with DTLS specified in the 'm' line of the offer, Bob will begin to negotiate a DTLS association with Alice for both RTP and RTCP streams. Early media (RTP and RTCP) starts to flow from Bob to Alice as soon as Bob sends the DTLS finished message to Alice. Bi-directional media (RTP and RTCP) can flow after Alice receives the SIP 200 response and once Alice has sent the DTLS finished message. The SIP signaling from Alice to her proxy is transported over TLS to ensure an integrity protected channel between Alice and her identity service. Note that all other signaling is transported over TCP in this example although it could be done over any supported transport. Alice Proxies Bob |(1) INVITE | | |---------------->| | | |(2) INVITE | | |----------------->| | |(3) conn-check | |<-----------------------------------| | |(4)hello | |<-----------------------------------| | |(5) conn-response | |----------------------------------->| |(6)|(4) hello | | |----------------------------------->| | |(7)|(5) finished | |<-----------------------------------| | |(8)|(6) media | |<-----------------------------------| |(9)|(7) finished | | |----------------------------------->| | |(10)|(8) 200 OK | |<-----------------------------------| | |(11)|(9) media | |----------------------------------->| |(12)|(10) ACK | | |----------------------------------->| Message (1): INVITE Alice -> Proxy This shows the initial INVITE from Alice to Bob carried over the TLS transport protocol to ensure an integrity protected channel between Alice and her proxy which acts as Alice's identity service. Note that Alice has requested to be either the active or passive endpoint by specifying a=setup:actpass. Bob chooses to act as the DTLS server and will initiate the session. Also note that there is a fingerprint attribute on the 'c' line of the SDP. This is computed from Bob's self-signed certificate. INVITE sip:firstname.lastname@example.org SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:5060;branch=z9hG4bK-0e53sadfkasldkfj Max-Forwards: 70 Contact: <sip:email@example.com:6937;transport=TLS> To: <sip:firstname.lastname@example.org> From: "Alice"<sip:email@example.com>;tag=843c7b0b Call-ID: 6076913b1c39c212@REVMTEpG CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Type: application/sdp Content-Length: xxxx v=0 o=- 1181923068 1181923196 IN IP4 192.168.1.103 s=example1 c=IN IP4 192.168.1.103 a=setup:actpass a=fingerprint: \ SHA-1 4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB t=0 0 m=audio 6056 RTP/AVP 0 a=sendrecv a=tcap:1 UDP/TLS/RTP/AVP RTP/AVP a=pcfg:1 t=1 Message (2): INVITE Proxy -> Bob This shows the INVITE being relayed to Bob from Alice (and Bob's) proxy. Note that Alice's proxy has inserted an Identity and Identity-Info header. This example only shows one element for both proxies for the purposes of simplification. Bob verifies the identity provided with the INVITE. Note that this offer includes a default m-line offering RTP in case the answerer does not support SRTP. However, the potential configuration utilizing a transport of SRTP is preferred. See [I-D.ietf-mmusic-sdp- capability-negotiation][I-D.ietf-mmusic-sdp-capability-negotiation] for more details on the details of SDP capability negotiation. INVITE sip:firstname.lastname@example.org SIP/2.0 Via: SIP/2.0/TLS 192.168.1.101:5060;branch=z9hG4bK-0e53sadfkasldkfj Via: SIP/2.0/TCP 192.168.1.100:5060;branch=z9hG4bK-0e53244234324234 Via: SIP/2.0/TCP 192.168.1.103:6937;branch=z9hG4bK-0e5b7d3edb2add32 Max-Forwards: 70 Contact: <sip:email@example.com:6937;transport=TLS> To: <sip:firstname.lastname@example.org> From: "Alice"<sip:email@example.com>;tag=843c7b0b Call-ID: 6076913b1c39c212@REVMTEpG CSeq: 1 INVITE Identity: CyI4+nAkHrH3ntmaxgr01TMxTmtjP7MASwliNRdupRI1vpkXRvZXx1ja9k 3W+v1PDsy32MaqZi0M5WfEkXxbgTnPYW0jIoK8HMyY1VT7egt0kk4XrKFC HYWGCl0nB2sNsM9CG4hq+YJZTMaSROoMUBhikVIjnQ8ykeD6UXNOyfI= Identity-Info: https://example.com/cert Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Type: application/sdp Content-Length: xxxx v=0 o=- 1181923068 1181923196 IN IP4 192.168.1.103 s=example1 c=IN IP4 192.168.1.103 a=setup:actpass a=fingerprint: \ SHA-1 4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB t=0 0 m=audio 6056 RTP/AVP 0 a=sendrecv a=tcap:1 UDP/TLS/RTP/SAVP RTP/AVP a=pcfg:1 t=1 Message (3): ICE connectivity-checkClientHello Bob -> Alice Section 6.6 describes an approach to avoid an SBC interaction issue where the endpoints do not support ICE. Bob (the active endpoint) sends a STUN connectivity check to Alice and may begin the DTLS negotiation immediately after sending the STUN check. Message (4): ClientHello Bob -> Alice Assuming that Alice's identity is valid, Message 3 shows Bob sending a DTLS ClientHello directlyAssuming that Alice's identity is valid, Message 3 shows Bob sending a DTLS ClientHello directly to Alice for each 'm' line in the SDP. In this case two DTLS ClientHello messages are sent to Alice. Bob sends a DTLS ClientHello to 192.168.1.103:6056 for RTP and another to port 6057 for RTCP. Message (5): ICE connectivity-check response Alice -> Bob Alice (the passive endpoint) sends a response to the STUN connectivity check (Message 3) to Bob. Message (6):(4): ServerHello+Certificate Alice -> Bob Alice sends back a ServerHello, Certificate, ServerHelloDone for both RTP and RTCP associations. Note that the same certificate is used for both the RTP and RTCP associations. If RTP/RTCP multiplexing [I-D.ietf-avt-rtp-and-rtcp-mux] were being used only a single association would be required. Message (7):(5): Certificate Bob -> Alice Bob sends a Certificate, ClientKeyExchange, CertificateVerify, change_cipher_spec and Finished for both RTP and RTCP associations. Again note that Bob uses the same server certificate for both associations. Message (8):(6): Early Media Bob -> Alice At this point, Bob can begin sending early media (RTP and RTCP) to Alice. Note that Alice can't yet trust the media since the fingerprint has not yet been received. This lack of trusted, secure media is indicated to Alice. Message (9):(7): Finished Alice -> Bob After Message 7 is received by Bob, Alice sends change_cipher_spec and Finished. Message (10):(8): 200 OK Bob -> Alice When Bob answers the call, Bob sends a 200 OK SIP message which contains the fingerprint for Bob's certificate. When Alice receives the message and validates the certificate presented in Message 7. The endpoint now shows Alice that the call as secured. SIP/2.0 200 OK To: <sip:firstname.lastname@example.org>;tag=6418913922105372816 From: "Alice" <sip:email@example.com>;tag=843c7b0b Via: SIP/2.0/TCP 192.168.1.103:6937;branch=z9hG4bK-0e5b7d3edb2add32 Call-ID: 6076913b1c39c212@REVMTEpG CSeq: 1 INVITE Contact: <sip:192.168.1.104:5060;transport=TCP> Content-Type: application/sdp Content-Length: xxxx v=0 o=- 6418913922105372816 2105372818 IN IP4 192.168.1.104 s=example2 c=IN IP4 192.168.1.104 a=setup:active a=fingerprint:\ SHA-1 FF:FF:FF:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB t=0 0 m=audio 12000 UDP/TLS/RTP/SAVP 0 a=acfg:1 t=1 Message (11):(9): RTP+RTCP Alice -> Bob At this point, Alice can also start sending RTP and RTCP to Bob. Note that in this case, Bob signals the actual transport protocol configuration of SRTP over DTLS in the acfg parameter. Message (12):(10): ACK Alice -> Bob Finally, Alice sends the SIP ACK to Bob. 8. Security Considerations DTLS or TLS media signalled with SIP requires a way to ensure thatIn this example, the communicating peers' certificates are correct. The standard TLS/DTLS strategy for authenticatingDTLS handshake has already completed by the communicating partiestime Alice receives Bob's 200 OK (8). Therefore, no STUN check is to give the server (and optionally the client)sent. However, if Alice had a PKIX [RFC3280] certificate. The clientNAT, then verifies the certificate and checksBob's ClientHello might get blocked by that the nameNAT, in the certificate matches the server's domain name. This works because there are a relatively small number of servers with well-defined names; a situationwhich does not usually occur incase Alice would send the VoIP context. The designthe STUN check described in this document is intended to leverage the authenticity ofSection 6.6 upon receiving the signaling channel (while not requiring confidentiality). As long200 OK, as each side of the connection can verify the integrity of the SDPshown below: Alice Proxies Bob |(1) INVITE then the DTLS handshake cannot be hijacked via a man-in-the-middle attack. This integrity protection| | |---------------->| | | |(2) INVITE | | |----------------->| | |(3) hello | | X<-----------------| | |(4) 200 OK | |<-----------------------------------| | (5) conn-check | | |----------------------------------->| | |(6) conn-response | |<-----------------------------------| | |(7) hello | |<-----------------------------------| |(8) hello (rtx) | | |----------------------------------->| | |(9) finished | |<-----------------------------------| | |(10) media | |<-----------------------------------| |(11) finished | | |----------------------------------->| | |(11) media | |----------------------------------->| |(12) ACK | | |----------------------------------->| The messages here are the same as in the previous example, with the following three new messages: Message (5): STUN connectivity-check Alice -> Bob Section 6.6 describes an approach to avoid an SBC interaction issue where the endpoints do not support ICE. Alice (the passive endpoint) sends a STUN connectivity check to Bob. This opens a pinhole in Alice's NAT/firewall. Message (6): STUN connectivity-check response Bob -> Alice Bob (the active endpoint) sends a response to the STUN connectivity check (Message 3) to Alice. This tells Alice that her connectivity check has succeeded and she can stop the retransmit state machine. Message (7): Hello (retransmit) Bob -> Alice Bob retransmits his DTLS ClientHello which now passes through the pinhole created in Alice's firewall. At this point, the DTLS handshake proceeds as before. 8. Security Considerations DTLS or TLS media signalled with SIP requires a way to ensure that the communicating peers' certificates are correct. The standard TLS/DTLS strategy for authenticating the communicating parties is to give the server (and optionally the client) a PKIX [RFC3280] certificate. The client then verifies the certificate and checks that the name in the certificate matches the server's domain name. This works because there are a relatively small number of servers with well-defined names; a situation which does not usually occur in the VoIP context. The design described in this document is intended to leverage the authenticity of the signaling channel (while not requiring confidentiality). As long each side of the connection can verify the integrity of the SDP received from the other side, then the DTLS handshake cannot be hijacked via a man-in-the-middle attack. This integrity protection is easily provided by the caller to the callee (seethe callee (see Alice to Bob in Section 7) via the SIP Identity [RFC4474] mechanism. Other mechanisms, such as the S/MIME mechanism described in RFC 3261, or the mechanisms described in [I-D.wing-sip-identity-media] or [I-D.fischer-sip-e2e-sec-media], could also serve this purpose. While this mechanism can still be used without such integrity mechanisms, the security provided is limited to defense against passive attack by intermediaries. An active attack on the signaling plus an active attack on the media plane can allow an attacker to attack the connection (R-SIG-MEDIA in the notation of [I-D.ietf-sip-media-security-requirements]). 8.1. Responder Identity SIP Identity does not support signatures in responses. Ideally Alice would want to know that Bob's SDP had not been tampered with and who it was from so that Alice's User Agent could indicate to Alice that there was a secure phone call to Bob. [RFC4916] defines an approach for a UA to supply its identity to its peer UA and for this identity to be signed by an authentication service. For example, using this approach, Bob in Section 7) viasends an answer, then immediately follows up with an UPDATE that includes the fingerprint and uses the SIP Identity [RFC4474] mechanism.mechanism to assert that the message is from Bob@example.com. The downside of this approach is that it requires the extra round trip of the UPDATE. However, it is less straightforwardsimple and secure even when not all of the proxies are trusted. In this example, Bob only needs to trust his proxy. Answerers SHOULD use this UPDATE mechanisms. In some cases, answerers will not send an UPDATE and in many calls, some media will be sent before the UPDATE is received. In these cases, no integrity is provided for the responder. Ideallyfingerprint from Bob to Alice. In this approach, an attacker that was on the signaling path could tamper with the fingerprint and insert themselves as a man-in- the-middle on the media. Alice would wantknow that she had a secure call with someone but would not know if it was with Bob or a man-in-the- middle. Bob would know that an attack was happening. The fact that one side can detect this attack means that in most cases where Alice and Bob both wish the communications to be encrypted there is not a problem. Keep in mind that in any of the possible approaches Bob could always reveal the media that was received to knowanyone. We are making the assumption that Bob's SDP hadBob also wants secure communications. In this do nothing case, Bob knows the media has not been tampered with or intercepted by a third party and whothat it wasis from so that Alice's User Agent could indicate toAlice@example.com. Alice knows that there was a secure phone callshe is talking to Bob. Thissomeone and that whoever that is known ashas probably checked that the media is not being intercepted or tampered with. This approach is certainly less than ideal but very usable for many situations. 8.2. SIPS If SIP connected party problem andIdentity is not used, but the signaling is protected by SIPS, the security guarantees are weaker, but some security is still provided as long as all proxies are trusted, this provides integrity for the fingerprint. It does not provide a topicstrong assertion of ongoing work in the SIP community. Inwho Alice is communicating with. However, as much as the meantime, there are several approaches thattarget domain can be used to mitigate this problem: Use UPDATE, Use SIPS, Use S/MIME, Single Sided Verification, or use human-read Short Authentication String (SAS)trusted to validate the certificates. Each one is discussed here followed bycorrectly populate the From header field value, Alice can use that. The security implications of that approach. 8.1. UPDATE [RFC4916] defines anissue with this approach for a UA to supply its identityis that if one of the Proxies wished to its peer UA and for this identitymount a man-in-the-middle attack, it could convince Alice that she was talking to be signed by an authentication service. For example, usingBob when really the media was flowing through a man in the middle media relay. However, this approach,attack could not convince Bob sends an answer, then immediately follows up with an UPDATEthat includes the fingerprint and uses the SIP Identityhe was taking to Alice. 8.3. S/MIME RFC 3261 [RFC3261] defines a S/MIME security mechanism for SIP that could be used to assertsign that the message isfingerprint was from Bob@example.com. The downsideBob. This would be secure. However, so far there have been no deployments of this approachS/MIME for SIP. 8.4. Continuity of Authentication One desirable property of a secure media system is to provide continuity of authentication: being able to ensure cryptographically that it requiresyou are talking to the extra round tripsame person as before. With DTLS, continuity of authentication is achieved by having each side use the UPDATE. However,same public key/self-signed certificate for each connection (at least with a given peer entity). It then becomes possible to cache the credential (or its hash) and verify that it is simple andunchanged. Thus, once a single secure connection has been established, an implementation can establish a future secure channel even when not all ofin the proxies are trusted.face of future insecure signalling. In this example, Bob only needsorder to trust his proxy. Answerersenable continuity of authentication, implementations SHOULD send use this UPDATE mechanisms. 8.2. SIPS In this approach,attempt to keep a constant long-term key. Verifying implementations SHOULD maintain a cache of the signalingkey used for each peer identity and alert the user if that key changes. 8.5. Short Authentication String An alternative available to Alice and Bob is to use human speech to verify each others' identity and then to verify each others' fingerprints also using human speech. Assuming that it is protected by TLS from hopdifficult to hop. As long as all proxies are trusted, this provides integrity forimpersonate another's speech and seamlessly modify the fingerprint.audio contents of a call, this approach is relatively safe. It doeswould not provide a strong assertionbe effective if other forms of who Alice is communicating with. However,communication were being used such as muchvideo or instant messaging. DTLS supports this mode of operation. The minimal secure fingerprint length is around 64 bits. ZRTP [I-D.zimmermann-avt-zrtp] includes Short Authentication String mode in which a unique per-connection bitstring is generated as part of the target domaincryptographic handshake. The SAS can be trusted to correctly populate the From header field value, Alice can use that. The security issue with this approachas short as 25 bits and so is that if one of the Proxies wishedsomewhat easier to mount a man-in-the-middle attack,read. DTLS does not natively support this mode, however it could convince Alice that she was talkingwould be straightforward to Bob when really the media was flowing throughadd one as a man inTLS extension [RFC3546]. 8.6. Limits of Identity Assertions When RFC 4474 is used to bind the middlemedia relay. However, this attack could not convince Bob that he was takingkeying material to Alice. 8.3. S/MIME RFC 3261 [RFC3261] defines a S/MIME security mechanism forthe SIP that could be used to sign thatsignalling, the fingerprint was from Bob. This would be secure. However, so far there have been no deploymentsassurances about the provenance and security of S/MIME for SIP. 8.4. Single-sided Verification In this approach, no integrity is providedthe media are only as good as those for the fingerprint from Bobsignalling. There are two important cases to Alice. In this approach, an attackernote here: o RFC 4474 assumes that was onthe signaling path could tamperproxy with the fingerprint and insert themselves as a man-in-the-middle oncertificate "example.com" controls the media. Alice would know that she had a secure call with someone but would not know if it was with Bob ornamespace "example.com". Therefore the RFC 4474 authentication service which is authoritative for a man-in-the-middle. Bob would know that an attack was happening. The fact that one sidegiven namespace can detect this attack means that in most cases wherecontrol which user is assigned each name. Thus, the authentication service can take an address formerly assigned to Alice and Bob both wish the communicationstransfer it to be encrypted thereBob. This is notan intentional design feature of RFC 4474 and a problem. Keep in mind that in anydirect consequence of the possible approaches Bob could always reveal the media that was received to anyone. WeSIP namespace architecture. o When phone number URIs (e.g., 'sip:+firstname.lastname@example.org') are making the assumptionused, there is no structural reason to trust that Bob also wants secure communications. In this do nothing case, Bob knowsthe media has not been tampered with or intercepted bydomain name is authoritative for a third partygiven phone number, although individual proxies and UAs may have private arrangements that itallow them to trust other domains. This is from Alice@example.com. Alice knowsa structural issue in that she is talkingPSTN elements are trusted to someoneassert their phone number correctly and that whoever that is has probably checked that the media is not being intercepted or tampered with. This approachthere is certainly less than ideal but very usableno real concept of a given entity being authoritative for many situations. 8.5. Continuitysome number space. In both of Authentication One desirable propertythese cases, the assurances of a secure media systemDTLS-SRTP provides in terms of data origin integrity and confidentiality are necessarily no better than SIP provides for signalling integrity when RFC 4474 is used. Implementors should therefore take care not to provide continuity of authentication: being ableindicate misleading peer identity information in the user interface. e.g. If the peer's identity is sip:+email@example.com, it is not sufficient to ensure cryptographicallydisplay that you are talking tothe same person as before. With DTLS, continuityidentity of authentication is achieved by having each side usethe same public key/self-signed certificate for each connection (at least with a givenpeer entity). It then becomes possibleas +17005551008, unless there is some policy that states that the domain "chicago.example.com" is trusted to cacheassert E.164 numbers. In cases where the credential (or its hash) and verifyUA can determine that itthe peer identity is unchanged. Thus, once a single secure connection has been established,clearly an implementation can establish a future secure channel even inE.164 number, it may be less confusing to simply identify the face of future insecure signalling. In ordercall as encrypted but to enable continuity of authentication, implementations SHOULD attemptan unknown peer. In addition, some middleboxes (B2BUAs and Session Border Controllers) are known to keep a constant long-term key. Verifying implementations SHOULD maintain a cachemodify portions of the key used for each peer identity and alertSIP message which are included in the user ifRFC 4474 signature computation, thus breaking the signature. This sort of man-in-the-middle operation is precisely the sort of message modification that key changes. 8.6. Short Authentication String An alternative available to Alice and Bob4474 is intended to use human speechdetect. In cases where the middlebox is itself permitted to verify each others' identity andgenerate valid RFC 4474 signatures (e.g., it is within the same administrative domain as the RFC 4474 authentication service), then it may generate a new signature on the modified message. Alternately, the middlebox may be able to verify each others' fingerprints also using human speech. Assumingsign with some other identity that it is difficultpermitted to impersonate another's speechassert. Otherwise, the recipient cannot rely on the RFC 4474 Identity assertion and seamlessly modifythe audio contents ofUA MUST not indicate to the user that a call,secure call has been established to the claimed identity. Implementations which are configured to only establish secure calls SHOULD terminate the call in this approachcase. If SIP Identity or an equivalent mechanism is relatively safe. It wouldnot be effective if other forms of communication were being used such as video or instant messaging. DTLS supports this mode of operation. The minimal secure fingerprint lengthused, then only protection against attackers who cannot actively change the signaling is around 64 bits. ZRTP [I-D.zimmermann-avt-zrtp] includes Short Authentication String mode in which a unique per-connection bitstringprovided. while this is generated as part ofstill superior to previous mechanisms, the cryptographic handshake. The SAS can be as short as 25 bits and sosecurity provided is somewhat easier to read. DTLS does not natively support this mode, however it would be straightforwardinferior to add one as a TLS extension [RFC3546].that provided if integrity is provided for the signaling. 8.7. Perfect Forward Secrecy One concern about the use of a long-term key is that compromise of that key may lead to compromise of past communications. In order to prevent this attack, DTLS supports modes with Perfect Forward Secrecy using Diffie-Hellman and Elliptic-Curve Diffie-Hellman cipher suites. When these modes are in use, the system is secure against such attacks. Note that compromise of a long-term key may still lead to future active attacks. If this is a concern, a backup authentication channel such as manual fingerprint establishment or a short authentication string should be used. 9. IANA Considerations This specification does not require any IANA actions. 10. Acknowledgments Cullen Jennings contributed substantial text and comments to this document. This document benefited from discussions with Francois Audet, Nagendra Modadugu, and Dan Wing. Thanks also for useful comments by Flemming Andreasen, Jonathan Rosenberg, Rohan Mahy, David McGrew, Miguel Garcia, Steffen Fries, Brian Stucker, Robert Gilman and David Oran. We would like to thank Thomas Belling, Guenther Horn, Steffen Fries, Brian Stucker, Francois Audet, Dan Wing, Jari Arkko, and Vesa Lehtovirta for their input regarding traversal of SBCs. 11. References 11.1. Normative References [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, June 2002. [RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, June 2002. [RFC3280] Housley, R., Polk, W., Ford, W., and D. Solo, "Internet X.509 Public Key Infrastructure Certificate and Certificate Revocation List (CRL) Profile", RFC 3280, April 2002. [RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks", RFC 3325, November 2002. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003. [RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in the Session Description Protocol (SDP)", RFC 4145, September 2005. [RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer Security", RFC 4347, April 2006. [RFC4474] Peterson, J. and C. Jennings, "Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)", RFC 4474, August 2006. [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, July 2006. [RFC4572] Lennox, J., "Connection-Oriented Media Transport over the Transport Layer Security (TLS) Protocol in the Session Description Protocol (SDP)", RFC 4572, July 2006. [I-D.ietf-behave-rfc3489bis] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, "Session Traversal Utilities for (NAT) (STUN)", draft-ietf-behave-rfc3489bis-15draft-ietf-behave-rfc3489bis-16 (work in progress), FebruaryJuly 2008. 11.2. Informational References [RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) Packets over Connection- Oriented Transport", RFC 4571, July 2006. [I-D.ietf-mmusic-ice] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols", draft-ietf-mmusic-ice-19 (work in progress), October 2007. [RFC4567] Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E. Carrara, "Key Management Extensions for Session Description Protocol (SDP) and Real Time Streaming Protocol (RTSP)", RFC 4567, July 2006. [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session Description Protocol (SDP) Security Descriptions for Media Streams", RFC 4568, July 2006. [I-D.zimmermann-avt-zrtp] Zimmermann, P., Johnston, A., and J. Callas, "ZRTP: Media Path Key Agreement for Secure RTP", draft-zimmermann-avt-zrtp-04draft-zimmermann-avt-zrtp-07 (work in progress), July 2007.June 2008. [I-D.mcgrew-srtp-ekt] McGrew, D., "Encrypted Key Transport for Secure RTP", draft-mcgrew-srtp-ekt-03 (work in progress), July 2007. [I-D.ietf-avt-dtls-srtp] McGrew, D. and E. Rescorla, "Datagram Transport Layer Security (DTLS) Extension to Establish Keys for Secure Real-time Transport Protocol (SRTP)", draft-ietf-avt-dtls-srtp-01draft-ietf-avt-dtls-srtp-02 (work in progress), November 2007.February 2008. [I-D.ietf-sip-media-security-requirements] Wing, D., Fries, S., Tschofenig, H., and F. Audet, "Requirements and Analysis of Media Security Management Protocols", draft-ietf-sip-media-security-requirements-03draft-ietf-sip-media-security-requirements-07 (work in progress), FebruaryJune 2008. [I-D.ietf-mmusic-sdp-capability-negotiation] Andreasen, F., "SDP Capability Negotiation", draft-ietf-mmusic-sdp-capability-negotiation-08 (work in progress), December 2007. [I-D.ietf-avt-rtp-and-rtcp-mux] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and Control Packets on a Single Port", draft-ietf-avt-rtp-and-rtcp-mux-07 (work in progress), August 2007. [RFC3262] Rosenberg, J. and H. Schulzrinne, "Reliability of Provisional Responses in Session Initiation Protocol (SIP)", RFC 3262, June 2002. [RFC3546] Blake-Wilson, S., Nystrom, M., Hopwood, D., Mikkelsen, J., and T. Wright, "Transport Layer Security (TLS) Extensions", RFC 3546, June 2003. [RFC4916] Elwell, J., "Connected Identity in the Session Initiation Protocol (SIP)", RFC 4916, June 2007. [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, March 2004. [RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K. Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830, August 2004. [I-D.wing-sipping-srtp-key] Wing, D., Audet, F., Fries, S., Tschofenig, H., and A. Johnston, "Secure Media Recording and Transcoding with the Session Initiation Protocol", draft-wing-sipping-srtp-key-03 (work in progress), February 2008. [I-D.wing-avt-dtls-srtp-key-transport] Wing, D., "Datagram TLS Secure RTP (DTLS-SRTP) Key Transport", draft-wing-avt-dtls-srtp-key-transport-01 (work in progress), February 2008. [I-D.ietf-mmusic-media-path-middleboxes] Stucker, B. and H. Tschofenig, "Analysis of Middlebox Interactions for Signaling Protocol Communication along theTschofenig, "Analysis of Middlebox Interactions for Signaling Protocol Communication along the Media Path", draft-ietf-mmusic-media-path-middleboxes-00 (work in progress), January 2008. [I-D.fischer-sip-e2e-sec-media] Fischer, K., "End-to-End Security for DTLS-SRTP", draft-fischer-sip-e2e-sec-media-00 (work in progress), January 2008. [I-D.wing-sip-identity-media] Wing, D. and H. Kaplan, "SIP Identity using Media Path", draft-ietf-mmusic-media-path-middleboxes-00draft-wing-sip-identity-media-02 (work in progress), JanuaryFebruary 2008. Appendix A. Requirements Analysis [I-D.ietf-sip-media-security-requirements] describes security requirements for media keying. This section evaluates this proposal with respect to each requirement. A.1. Forking and retargeting (R-FORK-RETARGET, R-BEST-SECURE, R-DISTINCT) In this draft, the SDP offer (in the INVITE) is simply an advertisement of the capability to do security. This advertisement does not depend on the identity of the communicating peer, so forking and retargeting work work when all the endpoints will do SRTP. When a mix of SRTP and non-SRTP endpoints are present, we use the SDP capabilities mechanism currently being defined [I-D.ietf-mmusic-sdp- capability-negotiation][I-D.ietf-mmusic-sdp-capability-negotiation] to transparently negotiate security where possible. Because DTLS establishes a new key for each session, only the entity with which the call is finally established gets the media encryption keys (R3). A.2. Distinct Cryptographic Contexts (R-DISTINCT) DTLS performs a new DTLS handshake with each endpoint, which establishes distinct keys and cryptographic contexts for each endpoint. A.3. Reusage of a Security Context (R-REUSE) DTLS allows sessions to be resumed with the 'TLS session resumption' functionality. This feature can be used to lower the amount of cryptographic computation that needs to be done when two peers re- initiates the communication. A.4. Clipping (R-AVOID-CLIPPING) Because the key establishment occurs in the media plane, media need not be clipped before the receipt of the SDP answer. A.5. Passive Attacks on the Media Path (R-PASS-MEDIA) The public key algorithms used by DTLS ciphersuites, such as RSA, Diffie-Hellman, and Elliptic Curve Diffie-Hellman, are secure against passive attacks. A.6. Passive Attacks on the Signaling Path (R-PASS-SIG) DTLS provides protection against passive attacks by adversaries on the signaling path since only a fingerprint is exchanged using SIP signaling. A.7. (R-SIG-MEDIA, R-ACT-ACT) An attacker who controls the media channel but not the signalling channel can perform a MITM attack on the DTLS handshake but this will change the certificates which will cause the fingerprint check to fail. Thus, any successful attack requires that the attacker modify the signalling messages to replace the fingerprints. AnIf RFC 4474 Identity or an equivalent mechanism is used, a attacker who controls the signalling channel at any point between the proxies performing the Identity signatures cannot modify the fingerprints without invalidating the Identitysignature. Thus, even an attacker who controls both signalling and media paths cannot successfully attack the media traffic. Note that an attacker who controls the authentication service can impersonate the UA using that authentication service. This is an intended feature of SIP Identity--the authentication service owns the namespace and therefore defines which user has which identity. A.8. Binding to Identifiers (R-ID-BINDING) This mechanism uses SIP-Identity [RFC4474] and SIP-Connected-Identity [RFC4916] to bind the endpoint's certificate fingerprints to the From: address in the signalling. The fingerprint is covered by the Identity signature. A.9. Perfect Forward Secrecy (R-PFS) DTLS supports Diffie-Hellman and Elliptic Curve Diffie-Hellman cipher suites which provide PFS. A.10. Algorithm Negotiation (R-COMPUTE) DTLS negotiates cipher suites before performing significant cryptographic computation and therefore supports algorithm negotiation and multiple cipher suites without additional computational expense. A.11. RTP Validity Check (R-RTP-VALID) DTLS packets do not pass the RTP validity check. The first byte of a DTLS packet is the content type and All current DTLS content types have the first two bits set to zero, resulting in a version of 0, thus failing the first validity check. A.12. 3rd Party Certificates (R-CERTS, R-EXISTING) Third party certificates are not required. However, if the parties share an authentication infrastructure that is compatible with TLS (3rd party certificates or shared keys) it can be used. A.13. FIPS 140-2 (R-FIPS) TLS implementations already may be FIPS 140-2 approved and the algorithms used here are consistent with the approval of DTLS and DTLS-SRTP. A.14. Linkage between Keying Exchange and SIP Signaling (R-ASSOC) The signaling exchange is linked to the key management exchange using the fingerprints carried in SIP and the certificates are exchanged in DTLS. A.15. Denial of Service Vulnerability (R-DOS) DTLS offers some degree of DoS protection particuarly as a built-in feature. A.16. Adversary Model (R-SIG-MEDIA) DTLS-SRTP requires that an adversary is at least able to intercept the fingerprint exchange along the SIP signaling path (i.e., active attack) and to intercept the DTLS handshake by acting as a man-in- the-middle adversary (i.e., active attack). A.17. Crypto-Agility (R-AGILITY) DTLS allows ciphersuites to be negotiated and hence new algorithms can be incrementally deployed. Work on replacing the fixed MD5/SHA-1 key derivation function is ongoing. A.18. Downgrading Protection (R-DOWNGRADE) DTLS provides protection against downgrading attacks since the selection of the offered ciphersuites is confirmed in a later stage of the handshake. This protection is efficient unless an adversary is able to break a ciphersuite in real-time. A.19. Media Security Negotation (R-NEGOTIATE) DTLS allows a User Agent to negotiate media security parameters for each individual session. A.20. Signaling Protocol Independence (R-OTHER-SIGNALING) The DTLS-SRTP framework does not rely on SIP; every protocol that is capable of exchanging a fingerprint and the media description can be secured. A.21. Media Recording (R-RECORDING) An extension, see [I-D.wing-sipping-srtp-key], has been specified to support media recording that does not require intermediaries to act as a MITM. When media recording is done by intermediaries then they need to act as a MITM. A.22. Interworking with Intermediaries (R-TRANSCODER) A description of the interworking with Session Border Controllers is described in this document. A.23. PSTN Gateway Termination (R-PSTN) The DTLS-SRTP framework allows the media security to terminate at a PSTN gateway. This does not provide end-to-end security, but is consistent with the security goals of this framework because the gateway is authorized to speak for the PSTN namespace. Authors' Addresses Jason Fischl CounterPath Corporation Suite 300, One Bentall Centre, 505 Burrard Street Vancouver, BC V7X 1M3 Canada Phone: +1 604 320-3340 Email: firstname.lastname@example.org Hannes Tschofenig Nokia Siemens Networks Otto-Hahn-Ring 6 Munich, Bavaria 81739 Germany Email: Hannes.Tschofenig@nsn.com URI: http://www.tschofenig.com Eric Rescorla Network Resonance 2483 E. Bayshore #212RTFM, Inc. 2064 Edgewood Drive Palo Alto, CA 94303 USA Email: email@example.com@rtfm.com Full Copyright Statement Copyright (C) The IETF Trust (2008). This document is subject to the rights, licenses and restrictions contained in BCP 78, and except as set forth therein, the authors retain all their rights. 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