Internet Engineering Task Force (IETF) M. Westerlund Request for Comments: 7201 Ericsson Category: Informational C. Perkins ISSN: 2070-1721 University of Glasgow April 2014 Options for Securing RTP Sessions
AbstractThe Real-time Transport Protocol (RTP) is used in a large number of different application domains and environments. This heterogeneity implies that different security mechanisms are needed to provide services such as confidentiality, integrity, and source authentication of RTP and RTP Control Protocol (RTCP) packets suitable for the various environments. The range of solutions makes it difficult for RTP-based application developers to pick the most suitable mechanism. This document provides an overview of a number of security solutions for RTP and gives guidance for developers on how to choose the appropriate security mechanism. Status of This Memo This document is not an Internet Standards Track specification; it is published for informational purposes. This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Not all documents approved by the IESG are a candidate for any level of Internet Standard; see Section 2 of RFC 5741. Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at http://www.rfc-editor.org/info/rfc7201.
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1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4 2. Background . . . . . . . . . . . . . . . . . . . . . . . . . 5 2.1. Point-to-Point Sessions . . . . . . . . . . . . . . . . . 5 2.2. Sessions Using an RTP Mixer . . . . . . . . . . . . . . . 5 2.3. Sessions Using an RTP Translator . . . . . . . . . . . . 6 2.3.1. Transport Translator (Relay) . . . . . . . . . . . . 6 2.3.2. Gateway . . . . . . . . . . . . . . . . . . . . . . . 7 2.3.3. Media Transcoder . . . . . . . . . . . . . . . . . . 8 2.4. Any Source Multicast . . . . . . . . . . . . . . . . . . 8 2.5. Source-Specific Multicast . . . . . . . . . . . . . . . . 8 3. Security Options . . . . . . . . . . . . . . . . . . . . . . 10 3.1. Secure RTP . . . . . . . . . . . . . . . . . . . . . . . 10 3.1.1. Key Management for SRTP: DTLS-SRTP . . . . . . . . . 12 3.1.2. Key Management for SRTP: MIKEY . . . . . . . . . . . 14 3.1.3. Key Management for SRTP: Security Descriptions . . . 15 3.1.4. Key Management for SRTP: Encrypted Key Transport . . 16 3.1.5. Key Management for SRTP: ZRTP and Other Solutions . . 17 3.2. RTP Legacy Confidentiality . . . . . . . . . . . . . . . 17 3.3. IPsec . . . . . . . . . . . . . . . . . . . . . . . . . . 17 3.4. RTP over TLS over TCP . . . . . . . . . . . . . . . . . . 18 3.5. RTP over Datagram TLS (DTLS) . . . . . . . . . . . . . . 18 3.6. Media Content Security/Digital Rights Management . . . . 19 3.6.1. ISMA Encryption and Authentication . . . . . . . . . 19 4. Securing RTP Applications . . . . . . . . . . . . . . . . . . 20 4.1. Application Requirements . . . . . . . . . . . . . . . . 20 4.1.1. Confidentiality . . . . . . . . . . . . . . . . . . . 20 4.1.2. Integrity . . . . . . . . . . . . . . . . . . . . . . 21 4.1.3. Source Authentication . . . . . . . . . . . . . . . . 22 4.1.4. Identifiers and Identity . . . . . . . . . . . . . . 23 4.1.5. Privacy . . . . . . . . . . . . . . . . . . . . . . . 24 4.2. Application Structure . . . . . . . . . . . . . . . . . . 25 4.3. Automatic Key Management . . . . . . . . . . . . . . . . 25 4.4. End-to-End Security vs. Tunnels . . . . . . . . . . . . . 25 4.5. Plaintext Keys . . . . . . . . . . . . . . . . . . . . . 26 4.6. Interoperability . . . . . . . . . . . . . . . . . . . . 26 5. Examples . . . . . . . . . . . . . . . . . . . . . . . . . . 26 5.1. Media Security for SIP-Established Sessions Using DTLS-SRTP . . . . . . . . . . . . . . . . . . . . . . . . 27 5.2. Media Security for WebRTC Sessions . . . . . . . . . . . 27 5.3. IP Multimedia Subsystem (IMS) Media Security . . . . . . 28 5.4. 3GPP Packet-Switched Streaming Service (PSS) . . . . . . 29 5.5. RTSP 2.0 . . . . . . . . . . . . . . . . . . . . . . . . 30 6. Security Considerations . . . . . . . . . . . . . . . . . . . 31 7. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 31 8. Informative References . . . . . . . . . . . . . . . . . . . 31
RFC3550] is widely used in a large variety of multimedia applications, including Voice over IP (VoIP), centralized multimedia conferencing, sensor data transport, and Internet television (IPTV) services. These applications can range from point-to-point phone calls, through centralized group teleconferences, to large-scale television distribution services. The types of media can vary significantly, as can the signaling methods used to establish the RTP sessions. So far, this multidimensional heterogeneity has prevented development of a single security solution that meets the needs of the different applications. Instead, a significant number of different solutions have been developed to meet different sets of security goals. This makes it difficult for application developers to know what solutions exist and whether their properties are appropriate. This memo gives an overview of the available RTP solutions and provides guidance on their applicability for different application domains. It also attempts to provide an indication of actual and intended usage at the time of writing as additional input to help with considerations such as interoperability, availability of implementations, etc. The guidance provided is not exhaustive, and this memo does not provide normative recommendations. It is important that application developers consider the security goals and requirements for their application. The IETF considers it important that protocols implement secure modes of operation and makes them available to users [RFC3365]. Because of the heterogeneity of RTP applications and use cases, however, a single security solution cannot be mandated [RFC7202]. Instead, application developers need to select mechanisms that provide appropriate security for their environment. It is strongly encouraged that common mechanisms be used by related applications in common environments. The IETF publishes guidelines for specific classes of applications, so it is worth searching for such guidelines. The remainder of this document is structured as follows. Section 2 provides additional background. Section 3 outlines the available security mechanisms at the time of this writing and lists their key security properties and constraints. Section 4 provides guidelines and important aspects to consider when securing an RTP application. Finally, in Section 5, we give some examples of application domains where guidelines for security exist.
Figure 1, where A has established an RTP session with B. In this case, the RTP security is primarily about ensuring that any third party be unable to compromise the confidentiality and integrity of the media communication. This requires confidentiality protection of the RTP session, integrity protection of the RTP/RTCP packets, and source authentication of all the packets to ensure no man-in-the- middle (MITM) attack is taking place. The source authentication can also be tied to a user or an endpoint's verifiable identity to ensure that the peer knows with whom they are communicating. Here, the combination of the security protocol protecting the RTP session (and, hence, the RTP and RTCP traffic) and the key management protocol becomes important to determine what security claims can be made. +---+ +---+ | A |<------->| B | +---+ +---+ Figure 1: Point-to-Point Topology
An RTP session using a mixer might have a topology like that in Figure 2. In this example, participants A through D each send unicast RTP traffic to the RTP mixer, and receive an RTP stream from the mixer, comprising a mixture of the streams from the other participants. +---+ +------------+ +---+ | A |<---->| |<---->| B | +---+ | | +---+ | Mixer | +---+ | | +---+ | C |<---->| |<---->| D | +---+ +------------+ +---+ Figure 2: Example RTP Mixer Topology A consequence of an RTP mixer having its own source identifier and acting as an active participant towards the other endpoints is that the RTP mixer needs to be a trusted device that has access to the security context(s) established. The RTP mixer can also become a security-enforcing entity. For example, a common approach to secure the topology in Figure 2 is to establish a security context between the mixer and each participant independently and have the mixer source authenticate each peer. The mixer then ensures that one participant cannot impersonate another. RFC5117] operates on a level below RTP and RTCP. It relays the RTP/RTCP traffic from one endpoint to one or more other addresses. This can be done based only on IP addresses and transport protocol ports, and each receive port on the translator can have a very basic list of where to forward traffic. Transport translators also need to implement ingress filtering to prevent random traffic from being forwarded that isn't coming from a participant in the conference. Figure 3 shows an example transport translator, where traffic from any one of the four participants will be forwarded to the other three
participants unchanged. The resulting topology is very similar to an Any Source Multicast (ASM) session (as discussed in Section 2.4) but is implemented at the application layer. +---+ +------------+ +---+ | A |<---->| |<---->| B | +---+ | Relay | +---+ | Translator | +---+ | | +---+ | C |<---->| |<---->| D | +---+ +------------+ +---+ Figure 3: RTP Relay Translator Topology A transport translator can often operate without needing access to the security context, as long as the security mechanism does not provide protection over the transport-layer information. A transport translator does, however, make the group communication visible and, thus, can complicate keying and source authentication mechanisms. This is further discussed in Section 2.4. Figure 4 shows an example topology. The functions a gateway provides can be diverse and range from transport-layer relaying between two domains not allowing direct communication, via transport or media protocol function initiation or termination, to protocol- or media- encoding translation. The supported security protocol might even be one of the reasons a gateway is needed. +---+ +-----------+ +---+ | A |<---->| Gateway |<---->| B | +---+ +-----------+ +---+ Figure 4: RTP Gateway Topology The choice of security protocol, and the details of the gateway function, will determine if the gateway needs to be trusted with access to the application security context. Many gateways need to be trusted by all peers to perform the translation; in other cases, some or all peers might not be aware of the presence of the gateway. The security protocols have different properties depending on the degree of trust and visibility needed. Ensuring communication is possible without trusting the gateway can be a strong incentive for accepting different security properties. Some security solutions will be able to detect the gateways as manipulating the media stream, unless the gateway is a trusted device.
Section 2.3.2 applies. A media transcoder alters the media data and, thus, needs to be trusted with access to the security context. RFC1112] is the original multicast model where any multicast group participant can send to the multicast group and get their packets delivered to all group members (see Figure 5). This form of communication has interesting security properties due to the many-to-many nature of the group. Source authentication is important, but all participants with access to the group security context will have the necessary secrets to decrypt and verify the integrity of the traffic. Thus, use of any group security context fails if the goal is to separate individual sources; alternate solutions are needed. +-----+ +---+ / \ +---+ | A |----/ \---| B | +---+ / \ +---+ + Multicast + +---+ \ Network / +---+ | C |----\ /---| D | +---+ \ / +---+ +-----+ Figure 5: Any Source Multicast (ASM) Group In addition, the potential large size of multicast groups creates some considerations for the scalability of the solution and how the key management is handled. RFC4607] allows only a specific endpoint to send traffic to the multicast group, irrespective of the number of RTP media sources. The endpoint is known as the media distribution source. For the RTP session to function correctly with RTCP over an SSM session, extensions have been defined in [RFC5760]. Figure 6 shows a sample SSM-based RTP session where several media sources, MS1...MSm, all send media to a distribution source, which then forwards the media data to the SSM group for delivery to the receivers, R1...Rn, and the feedback targets, FT1...FTn. RTCP reception quality feedback is sent unicast from each receiver to one
of the feedback targets. The feedback targets aggregate reception quality feedback and forward it upstream towards the distribution source. The distribution source forwards (possibly aggregated and summarized) reception feedback to the SSM group and back to the original media sources. The feedback targets are also members of the SSM group and receive the media data, so they can send unicast repair data to the receivers in response to feedback if appropriate. +-----+ +-----+ +-----+ | MS1 | | MS2 | .... | MSm | +-----+ +-----+ +-----+ ^ ^ ^ | | | V V V +---------------------------------+ | Distribution Source | +--------+ | | FT Agg | | +--------+------------------------+ ^ ^ | : . | : +...................+ : | . : / \ . +------+ / \ +-----+ | FT1 |<----+ +----->| FT2 | +------+ / \ +-----+ ^ ^ / \ ^ ^ : : / \ : : : : / \ : : : : / \ : : : ./\ /\. : : /. \ / .\ : : V . V V . V : +----+ +----+ +----+ +----+ | R1 | | R2 | ... |Rn-1| | Rn | +----+ +----+ +----+ +----+ Figure 6: Example SSM-Based RTP Session with Two Feedback Targets The use of SSM makes it more difficult to inject traffic into the multicast group, but not impossible. Source authentication requirements apply for SSM sessions, too; an individual verification of who sent the RTP and RTCP packets is needed. An RTP session using SSM will have a group security context that includes the media sources, distribution source, feedback targets, and the receivers. Each has a different role and will be trusted to perform different actions. For example, the distribution source will need to
authenticate the media sources to prevent unwanted traffic from being distributed via the SSM group. Similarly, the receivers need to authenticate both the distribution source and their feedback target to prevent injection attacks from malicious devices claiming to be feedback targets. An understanding of the trust relationships and group security context is needed between all components of the system. RFC4949]. RFC3711] is one of the most commonly used mechanisms to provide confidentiality, integrity protection, source authentication, and replay protection for RTP. SRTP was developed with RTP header compression and third- party monitors in mind. Thus, the RTP header is not encrypted in RTP data packets, and the first 8 bytes of the first RTCP packet header in each compound RTCP packet are not encrypted. The entirety of RTP packets and compound RTCP packets are integrity protected. This allows RTP header compression to work and lets third-party monitors determine what RTP traffic flows exist based on the synchronization source (SSRC) fields, but it protects the sensitive content. SRTP works with transforms where different combinations of encryption algorithm, authentication algorithm, and pseudorandom function can be used, and the authentication tag length can be set to any value. SRTP can also be easily extended with additional cryptographic transforms. This gives flexibility but requires more security knowledge by the application developer. To simplify things, Session Description Protocol (SDP) security descriptions (see Section 3.1.3) and Datagram Transport Layer Security Extension for SRTP (DTLS-SRTP) (see Section 3.1.1) use predefined combinations of transforms, known as SRTP crypto suites and SRTP protection profiles, that bundle together transforms and other parameters, making them easier to use but reducing flexibility. The Multimedia Internet Keying (MIKEY) protocol (see Section 3.1.2) provides flexibility to negotiate the full selection of transforms. At the time of this writing, the following transforms, SRTP crypto suites, and SRTP protection profiles are defined or under definition:
AES-CM and HMAC-SHA-1: AES Counter Mode encryption with 128-bit keys combined with 160-bit keyed HMAC-SHA-1 with an 80-bit authentication tag. This is the default cryptographic transform that needs to be supported. The transforms are defined in SRTP [RFC3711], with the corresponding SRTP crypto suite defined in [RFC4568] and SRTP protection profile defined in [RFC5764]. AES-f8 and HMAC-SHA-1: AES f8-mode encryption using 128-bit keys combined with keyed HMAC-SHA-1 using 80-bit authentication. The transforms are defined in [RFC3711], with the corresponding SRTP crypto suite defined in [RFC4568]. The corresponding SRTP protection profile is not defined. SEED: A Korean national standard cryptographic transform that is defined to be used with SRTP in [RFC5669]. Three options are defined: one using SHA-1 authentication, one using Counter Mode with Cipher Block Chaining Message Authentication Code (CBC-MAC), and one using Galois Counter Mode. ARIA: A Korean block cipher [ARIA-SRTP] that supports 128-, 192-, and 256-bit keys. It also defines three options: Counter Mode where combined with HMAC-SHA-1 with 80- or 32-bit authentication tags, Counter Mode with CBC-MAC, and Galois Counter Mode. It also defines a different key derivation function than the AES-based systems. AES-192-CM and AES-256-CM: Cryptographic transforms for SRTP based on AES-192 and AES-256 Counter Mode encryption and 160-bit keyed HMAC-SHA-1 with 80- and 32-bit authentication tags. These provide 192- and 256-bit encryption keys, but otherwise match the default 128-bit AES-CM transform. The transforms are defined in [RFC3711] and [RFC6188], and the SRTP crypto suites are defined in [RFC6188]. AES-GCM and AES-CCM: AES Galois Counter Mode and AES Counter Mode with CBC-MAC for AES-128 and AES-256. This authentication is included in the cipher text, which becomes expanded with the length of the authentication tag instead of using the SRTP authentication tag. This is defined in [AES-GCM]. NULL: SRTP [RFC3711] also provides a NULL cipher that can be used when no confidentiality for RTP/RTCP is requested. The corresponding SRTP protection profile is defined in [RFC5764]. The source authentication guarantees provided by SRTP depend on the cryptographic transform and key management used. Some transforms give strong source authentication even in multiparty sessions; others give weaker guarantees and can authenticate group membership but not
sources. Timed Efficient Stream Loss-Tolerant Authentication (TESLA) [RFC4383] offers a complement to the regular symmetric keyed authentication transforms, like HMAC-SHA-1, and can provide per-source authentication in some group communication scenarios. The downside is the need for buffering the packets for a while before authenticity can be verified. [RFC4771] defines a variant of the authentication tag that enables a receiver to obtain the Roll over Counter for the RTP sequence number that is part of the Initialization Vector (IV) for many cryptographic transforms. This enables quicker and easier options for joining a long-lived RTP group; for example, a broadcast session. RTP header extensions are normally carried in the clear and are only integrity protected in SRTP. This can be problematic in some cases, so [RFC6904] defines an extension to also encrypt selected header extensions. SRTP is specified and deployed in a number of RTP usage contexts; significant support is provided in SIP-established VoIP clients, including IP Multimedia Subsystems (IMS), and in the Real Time Streaming Protocol (RTSP) [RTSP] and RTP-based media streaming. Thus, SRTP in general is widely deployed. When it comes to cryptographic transforms, the default (AES-CM and HMAC-SHA-1) is the most commonly used, but it might be expected that AES-GCM, AES-192-CM, and AES-256-CM will gain usage in future, especially due to the AES- and GCM-specific instructions in new CPUs. SRTP does not contain an integrated key management solution; instead, it relies on an external key management protocol. There are several protocols that can be used. The following sections outline some popular schemes. RFC5763][RFC5764]. This extension provides secure key exchange between two peers, enabling Perfect Forward Secrecy (PFS) and binding strong identity verification to an endpoint. PFS is a property of the key agreement protocol that ensures that a session key derived from a set of long-term keys will not be compromised if one of the long-term keys is compromised in the future. The default key generation will generate a key that contains material contributed by both peers. The key exchange happens in the media plane directly between the peers. The common key exchange procedures will take two round trips assuming no losses. Transport Layer Security (TLS) resumption can be used when establishing additional media streams with the same peer, and it reduces the setup
time to one RTT for these streams (see [RFC5764] for a discussion of TLS resumption in this context). The actual security properties of an established SRTP session using DTLS will depend on the cipher suites offered and used, as well as the mechanism for identifying the endpoints of the handshake. For example, some cipher suites provide PFS, while others do not. When using DTLS, the application designer needs to select which cipher suites DTLS-SRTP can offer and accept so that the desired security properties are achieved. The next choice is how to verify the identity of the peer endpoint. One choice can be to rely on the certificates and use a PKI to verify them to make an identity assertion. However, this is not the most common way; instead, self- signed certificates are common to use to establish trust through signaling or other third-party solutions. DTLS-SRTP key management can use the signaling protocol in four ways: First, to agree on using DTLS-SRTP for media security. Second, to determine the network location (address and port) where each side is running a DTLS listener to let the parts perform the key management handshakes that generate the keys used by SRTP. Third, to exchange hashes of each side's certificates to bind these to the signaling and ensure there is no MITM attack. This assumes that one can trust the signaling solution to be resistant to modification and not be in collaboration with an attacker. Finally, to provide an asserted identity, e.g., [RFC4474], that can be used to prevent modification of the signaling and the exchange of certificate hashes. That way, it enables binding between the key exchange and the signaling. This usage is well defined for SIP/SDP in [RFC5763] and, in most cases, can be adopted for use with other bidirectional signaling solutions. It is to be noted that there is work underway to revisit the SIP Identity mechanism [RFC4474] in the IETF STIR working group. The main question regarding DTLS-SRTP's security properties is how one verifies any peer identity or at least prevents MITM attacks. This does require trust in some DTLS-SRTP external parties: either a PKI, a signaling system, or some identity provider. DTLS-SRTP usage is clearly on the rise. It is mandatory to support in Web Real-Time Communication (WebRTC). It has growing support among SIP endpoints. DTLS-SRTP was developed in IETF primarily to meet security requirements for RTP-based media established using SIP. The requirements considered can be reviewed in "Requirements and Analysis of Media Security Management Protocols" [RFC5479].
RFC3830] is a keying protocol that has several modes with different properties. MIKEY can be used in point-to-point applications using SIP and RTSP (e.g., VoIP calls) but is also suitable for use in broadcast and multicast applications and centralized group communications. MIKEY can establish multiple security contexts or cryptographic sessions with a single message. It is usable in scenarios where one entity generates the key and needs to distribute the key to a number of participants. The different modes and the resulting properties are highly dependent on the cryptographic method used to establish the session keys actually used by the security protocol, like SRTP. MIKEY has the following modes of operation: Pre-Shared Key: Uses a pre-shared secret for symmetric key crypto used to secure a keying message carrying the already-generated session key. This system is the most efficient from the perspective of having small messages and processing demands. The downside is scalability, where usually the effort for the provisioning of pre-shared keys is only manageable if the number of endpoints is small. Public Key Encryption: Uses a public key crypto to secure a keying message carrying the already-generated session key. This is more resource intensive but enables scalable systems. It does require a public key infrastructure to enable verification. Diffie-Hellman: Uses Diffie-Hellman key agreement to generate the session key, thus providing perfect forward secrecy. The downside is high resource consumption in bandwidth and processing during the MIKEY exchange. This method can't be used to establish group keys as each pair of peers performing the MIKEY exchange will establish different keys. HMAC-Authenticated Diffie-Hellman: [RFC4650] defines a variant of the Diffie-Hellman exchange that uses a pre-shared key in a keyed Hashed Message Authentication Code (HMAC) to verify authenticity of the keying material instead of a digital signature as in the previous method. This method is still restricted to point-to-point usage. RSA-R: MIKEY-RSA in Reverse mode [RFC4738] is a variant of the public key method, which doesn't rely on the initiator of the key exchange knowing the responder's certificate. This method lets both the initiator and the responder specify the session keying
material depending on the use case. Usage of this mode requires one round-trip time. TICKET: Ticket Payload (TICKET) [RFC6043] is a MIKEY extension using a trusted centralized key management service (KMS). The initiator and responder do not share any credentials; instead, they trust a third party, the KMS, with which they both have or can establish shared credentials. IBAKE: Identity-Based Authenticated Key Exchange (IBAKE) [RFC6267] uses a KMS infrastructure but with lower demand on the KMS. It claims to provide both perfect forward and backwards secrecy. SAKKE: [RFC6509] provides Sakai-Kasahara Key Encryption (SAKKE) in MIKEY. It is based on Identity-based Public Key Cryptography and a KMS infrastructure to establish a shared secret value and certificateless signatures to provide source authentication. Its features include simplex transmission, scalability, low-latency call setup, and support for secure deferred delivery. MIKEY messages have several different transports. [RFC4567] defines how MIKEY messages can be embedded in general SDP for usage with the signaling protocols SIP, Session Announcement Protocol (SAP), and RTSP. There also exists a usage of MIKEY defined by the Third Generation Partnership Project (3GPP) that sends MIKEY messages directly over UDP [T3GPP.33.246] to key the receivers of Multimedia Broadcast and Multicast Service (MBMS) [T3GPP.26.346]. [RFC3830] defines the application/mikey media type, allowing MIKEY to be used in, e.g., email and HTTP. Based on the many choices, it is important to consider the properties needed in one's solution and based on that evaluate which modes are candidates for use. More information on the applicability of the different MIKEY modes can be found in [RFC5197]. MIKEY with pre-shared keys is used by 3GPP MBMS [T3GPP.33.246], and IMS media security [T3GPP.33.328] specifies the use of the TICKET mode transported over SIP and HTTP. RTSP 2.0 [RTSP] specifies use of the RSA-R mode. There are some SIP endpoints that support MIKEY. The modes they use are unknown to the authors. RFC4568] provides a keying solution based on sending plaintext keys in SDP [RFC4566]. It is primarily used with SIP and the SDP Offer/ Answer model and is well defined in point-to-point sessions where each side declares its own unique key. Using security descriptions to establish group keys is less well defined and can have security
issues since it's difficult to guarantee unique SSRCs (as needed to avoid a "two-time pad" attack -- see Section 9 of [RFC3711]). Since keys are transported in plaintext in SDP, they can easily be intercepted unless the SDP carrying protocol provides strong end-to-end confidentiality and authentication guarantees. This is not normally the case; instead, hop-by-hop security is provided between signaling nodes using TLS. This leaves the keying material sensitive to capture by the traversed signaling nodes. Thus, in most cases, the security properties of security descriptions are weak. The usage of security descriptions usually requires additional security measures; for example, the signaling nodes are trusted and protected by strict access control. Usage of security descriptions requires careful design in order to ensure that the security goals can be met. Security descriptions are the most commonly deployed keying solution for SIP-based endpoints, where almost all endpoints that support SRTP also support security descriptions. It is also used for access protection in IMS Media Security [T3GPP.33.328]. EKT] is an SRTP extension that enables group keying despite using a keying mechanism like DTLS-SRTP that doesn't support group keys. It is designed for centralized conferencing, but it can also be used in sessions where endpoints connect to a conference bridge or a gateway and need to be provisioned with the keys each participant on the bridge or gateway uses to avoid decryption and encryption cycles. This can enable interworking between DTLS-SRTP and other keying systems where either party can set the key (e.g., interworking with security descriptions). The mechanism is based on establishing an additional EKT key, which everyone uses to protect their actual session key. The actual session key is sent in an expanded authentication tag to the other session participants. This key is only sent occasionally or periodically depending on use cases and depending on what requirements exist for timely delivery or notification. The only known deployment of EKT so far is in some Cisco video conferencing products.
RFC6189] key management system for SRTP was proposed as an alternative to DTLS-SRTP. ZRTP provides best effort encryption independent of the signaling protocol and utilizes key continuity, Short Authentication Strings, or a PKI for authentication. ZRTP wasn't adopted as an IETF Standards Track protocol, but was instead published as an Informational RFC in the IETF stream. Commercial implementations exist. Additional proprietary solutions are also known to exist. Section 9 of the RTP standard [RFC3550] defines a Data Encryption Standard (DES) or 3DES-based encryption of RTP and RTCP packets. This mechanism is keyed using plaintext keys in SDP [RFC4566] using the "k=" SDP field. This method can provide confidentiality but, as discussed in Section 9 of [RFC3550], it has extremely weak security properties and is not to be used. RFC4301] can be used in either tunnel or transport mode to protect RTP and RTCP packets in transit from one network interface to another. This can be sufficient when the network interfaces have a direct relation or in a secured environment where it can be controlled who can read the packets from those interfaces. The main concern with using IPsec to protect RTP traffic is that in most cases, using a VPN approach that terminates the security association at some node prior to the RTP endpoint leaves the traffic vulnerable to attack between the VPN termination node and the endpoint. Thus, usage of IPsec requires careful thought and design of its usage so that it meets the security goals. An important question is how one ensures the IPsec terminating peer and the ultimate destination are the same. Applications can have issues using existing APIs when determining if IPsec is being used or not and when determining who the authenticated peer entity is when IPsec is used. IPsec with RTP is more commonly used as a security solution between infrastructure nodes that exchange many RTP sessions and media streams. The establishment of a secure tunnel between such nodes minimizes the key management overhead.
RFC4571], it can also be sent over TLS over TCP [RFC4572], using TLS to provide point-to-point security services. The security properties TLS provides are confidentiality, integrity protection, and possible source authentication if the client or server certificates are verified and provide a usable identity. When used in multiparty scenarios using a central node for media distribution, the security provided is only between the central node and the peers, so the security properties for the whole session are dependent on what trust one can place in the central node. RTSP 1.0 [RFC2326] and 2.0 [RTSP] specify the usage of RTP over the same TLS/TCP connection that the RTSP messages are sent over. It appears that RTP over TLS/TCP is also used in some proprietary solutions that use TLS to bypass firewalls. RFC6347] is based on TLS [RFC5246] but designed to work over an unreliable datagram-oriented transport rather than requiring reliable byte stream semantics from the transport protocol. Accordingly, DTLS can provide point-to-point security for RTP flows analogous to that provided by TLS but over a datagram transport such as UDP. The two peers establish a DTLS association between each other, including the possibility to do certificate-based source authentication when establishing the association. All RTP and RTCP packets flowing will be protected by this DTLS association. Note that using DTLS for RTP flows is different from using DTLS-SRTP key management. DTLS-SRTP uses the same key management steps as DTLS, but uses SRTP for the per-packet security operations. Using DTLS for RTP flows uses the normal datagram TLS data protection, wrapping complete RTP packets. When using DTLS for RTP flows, the RTP and RTCP packets are completely encrypted with no headers in the clear; when using DTLS-SRTP, the RTP headers are in the clear and only the payload data is encrypted. DTLS can use similar techniques to those available for DTLS-SRTP to bind a signaling-side agreement to communicate to the certificates used by the endpoint when doing the DTLS handshake. This enables use without having a certificate-based trust chain to a trusted certificate root. There does not appear to be significant usage of DTLS for RTP.
Section 3.6.1) and the deprecated 3GPP Packet-switched Streaming Service solution; see Annex K of [T3GPP.26.234R8]. ISMACryp2]. This specification defines how one encrypts and packetizes the encrypted application data units (ADUs) in an RTP payload using the MPEG-4 generic payload format [RFC3640]. The ADU types that are allowed are those that can be stored as elementary streams in an ISO Media File format-based file. ISMACryp uses SRTP for packet-level integrity and source authentication from a streaming server to the receiver.
Key management for an ISMACryp-based system can be achieved through Open Mobile Alliance (OMA) Digital Rights Management 2.0 [OMADRMv2], for example.