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This document defines the security architecture for WebRTC, a protocol suite intended for use with real-time applications that can be deployed in browsers -- "real-time communication on the Web".

Status of This Memo This is an Internet Standards Track document. 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). Further information on Internet Standards is available in Section 2 of RFC 7841. Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at .

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Table of Contents

• .  Introduction
• .  Terminology
• .  Trust Model
  • .  Authenticated Entities
  • .  Unauthenticated Entities
• .  Overview
  • .  Initial Signaling
  • .  Media Consent Verification
  • .  DTLS Handshake
  • .  Communications and Consent Freshness
• .  SDP Identity Attribute
  • .  Offer/Answer Considerations
    • .  Generating the Initial SDP Offer
    • .  Generating an SDP Answer
    • .  Processing an SDP Offer or Answer
    • .  Modifying the Session
• .  Detailed Technical Description
  • .  Origin and Web Security Issues
  • .  Device Permissions Model
  • .  Communications Consent
  • .  IP Location Privacy
  • .  Communications Security
• .  Web-Based Peer Authentication
  • .  Trust Relationships: IdPs, APs, and RPs
  • .  Overview of Operation
  • .  Items for Standardization
  • .  Binding Identity Assertions to JSEP Offer/Answer Transactions
    • .  Carrying Identity Assertions
  • .  Determining the IdP URI
    • .  Authenticating Party
    • .  Relying Party
  • .  Requesting Assertions
  • .  Managing User Login
• .  Verifying Assertions
  • .  Identity Formats
• .  Security Considerations
  • .  Communications Security
  • .  Privacy
  • .  Denial of Service
  • .  IdP Authentication Mechanism
    • .  PeerConnection Origin Check
    • .  IdP Well-Known URI
    • .  Privacy of IdP-Generated Identities and the Hosting Site
    • .  Security of Third-Party IdPs
      • .  Confusable Characters
    • .  Web Security Feature Interactions
      • .  Popup Blocking
      • .  Third Party Cookies
• . IANA Considerations
• . References
  • .  Normative References
  • .  Informative References
• Acknowledgements
• Author's Address

Introduction The Real-Time Communications on the Web (RTCWEB) Working Group standardized protocols for real-time communications between Web browsers, generally called "WebRTC" . The major use cases for WebRTC technology are real-time audio and/or video calls, Web conferencing, and direct data transfer. Unlike most conventional real-time systems (e.g., SIP-based soft phones), WebRTC communications are directly controlled by some Web server, via a JavaScript (JS) API as shown in .

A Simple WebRTC System +----------------+ | | | Web Server | | | +----------------+ ^ ^ / \ HTTP / \ HTTP / \ / \ v v JS API JS API +-----------+ +-----------+ | | Media | | | Browser |<---------->| Browser | | | | | +-----------+ +-----------+

A more complicated system might allow for inter-domain calling, as shown in . The protocol to be used between the domains is not standardized by WebRTC, but given the installed base and the form of the WebRTC API is likely to be something SDP-based like SIP or something like the Extensible Messaging and Presence Protocol (XMPP) .

A Multidomain WebRTC System +--------------+ +--------------+ | | SIP, XMPP, ... | | | Web Server |<-------------->| Web Server | | | | | +--------------+ +--------------+ ^ ^ | | HTTP | | HTTP | | v v JS API JS API +-----------+ +-----------+ | | Media | | | Browser |<------------------->| Browser | | | | | +-----------+ +-----------+

This system presents a number of new security challenges, which are analyzed in . This document describes a security architecture for WebRTC which addresses the threats and requirements described in that document.

Terminology The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14 when, and only when, they appear in all capitals, as shown here.

Trust Model The basic assumption of this architecture is that network resources exist in a hierarchy of trust, rooted in the browser, which serves as the user's Trusted Computing Base (TCB). Any security property which the user wishes to have enforced must be ultimately guaranteed by the browser (or transitively by some property the browser verifies). Conversely, if the browser is compromised, then no security guarantees are possible. Note that there are cases (e.g., Internet kiosks) where the user can't really trust the browser that much. In these cases, the level of security provided is limited by how much they trust the browser. Optimally, we would not rely on trust in any entities other than the browser. However, this is unfortunately not possible if we wish to have a functional system. Other network elements fall into two categories: those which can be authenticated by the browser and thus can be granted permissions to access sensitive resources, and those which cannot be authenticated and thus are untrusted.

Authenticated Entities There are two major classes of authenticated entities in the system:

Calling services:

Web sites whose origin we can verify (optimally via HTTPS, but in some cases because we are on a topologically restricted network, such as behind a firewall, and can infer authentication from firewall behavior).

Other users:

WebRTC peers whose origin we can verify cryptographically (optimally via DTLS-SRTP).

Note that merely being authenticated does not make these entities trusted. For instance, just because we can verify that <https://www.example.org/> is owned by Dr. Evil does not mean that we can trust Dr. Evil to access our camera and microphone. However, it gives the user an opportunity to determine whether they wish to trust Dr. Evil or not; after all, if they desire to contact Dr. Evil (perhaps to arrange for ransom payment), it's safe to temporarily give them access to the camera and microphone for the purpose of the call, but they don't want Dr. Evil to be able to access their camera and microphone other than during the call. The point here is that we must first identify other elements before we can determine whether and how much to trust them. Additionally, sometimes we need to identify the communicating peer before we know what policies to apply.

Unauthenticated Entities Other than the above entities, we are not generally able to identify other network elements; thus, we cannot trust them. This does not mean that it is not possible to have any interaction with them, but it means that we must assume that they will behave maliciously and design a system which is secure even if they do so.

Overview This section describes a typical WebRTC session and shows how the various security elements interact and what guarantees are provided to the user. The example in this section is a "best case" scenario in which we provide the maximal amount of user authentication and media privacy with the minimal level of trust in the calling service. Simpler versions with lower levels of security are also possible and are noted in the text where applicable. It's also important to recognize the tension between security (or performance) and privacy. The example shown here is aimed towards settings where we are more concerned about secure calling than about privacy, but as we shall see, there are settings where one might wish to make different tradeoffs -- this architecture is still compatible with those settings. For the purposes of this example, we assume the topology shown in the figures below. This topology is derived from the topology shown in , but separates Alice's and Bob's identities from the process of signaling. Specifically, Alice and Bob have relationships with some Identity Provider (IdP) that supports a protocol (such as OpenID Connect) that can be used to demonstrate their identity to other parties. For instance, Alice might have an account with a social network which she can then use to authenticate to other Web sites without explicitly having an account with those sites; this is a fairly conventional pattern on the Web. provides an overview of IdPs and the relevant terminology. Alice and Bob might have relationships with different IdPs as well. Note: The IdP mechanism described here has not seen wide adoption. See for more on the status of IdP-based authentication. This separation of identity provision and signaling isn't particularly important in "closed world" cases where Alice and Bob are users on the same social network and have identities based on that domain (). However, there are important settings where that is not the case, such as federation (calls from one domain to another; see ) and calling on untrusted sites, such as where two users who have a relationship via a given social network want to call each other on another, untrusted, site, such as a poker site. Note that the servers themselves are also authenticated by an external identity service, the SSL/TLS certificate infrastructure (not shown). As is conventional in the Web, all identities are ultimately rooted in that system. For instance, when an IdP makes an identity assertion, the Relying Party consuming that assertion is able to verify because it is able to connect to the IdP via HTTPS.

A Call with IdP-Based Identity +----------------+ | | | Signaling | | Server | | | +----------------+ ^ ^ / \ HTTPS / \ HTTPS / \ / \ v v JS API JS API +-----------+ +-----------+ | | Media | | Alice | Browser |<---------->| Browser | Bob | | (DTLS+SRTP)| | +-----------+ +-----------+ ^ ^--+ +--^ ^ | | | | v | | v +-----------+ | | +-----------+ | |<--------+ | | | IdP1 | | | IdP2 | | | +------->| | +-----------+ +-----------+

shows essentially the same calling scenario but with a call between two separate domains (i.e., a federated case), as in . As mentioned above, the domains communicate by some unspecified protocol, and providing separate signaling and identity allows for calls to be authenticated regardless of the details of the inter-domain protocol.

A Federated Call with IdP-Based Identity +----------------+ Unspecified +----------------+ | | protocol | | | Signaling |<----------------->| Signaling | | Server | (SIP, XMPP, ...) | Server | | | | | +----------------+ +----------------+ ^ ^ | | HTTPS | | HTTPS | | | | v v JS API JS API +-----------+ +-----------+ | | Media | | Alice | Browser |<--------------------------->| Browser | Bob | | DTLS+SRTP | | +-----------+ +-----------+ ^ ^--+ +--^ ^ | | | | v | | v +-----------+ | | +-----------+ | |<-------------------------+ | | | IdP1 | | | IdP2 | | | +------------------------>| | +-----------+ +-----------+

Initial Signaling For simplicity, assume the topology in . Alice and Bob are both users of a common calling service; they both have approved the calling service to make calls (we defer the discussion of device access permissions until later). They are both connected to the calling service via HTTPS and so know the origin with some level of confidence. They also have accounts with some IdP. This sort of identity service is becoming increasingly common in the Web environment (with technologies such as Federated Google Login, Facebook Connect, OAuth, OpenID, WebFinger), and is often provided as a side effect service of a user's ordinary accounts with some service. In this example, we show Alice and Bob using a separate identity service, though the identity service may be the same entity as the calling service or there may be no identity service at all. Alice is logged onto the calling service and decides to call Bob. She can see from the calling service that he is online and the calling service presents a JS UI in the form of a button next to Bob's name which says "Call". Alice clicks the button, which initiates a JS callback that instantiates a PeerConnection object. This does not require a security check: JS from any origin is allowed to get this far. Once the PeerConnection is created, the calling service JS needs to set up some media. Because this is an audio/video call, it creates a MediaStream with two MediaStreamTracks, one connected to an audio input and one connected to a video input. At this point, the first security check is required: untrusted origins are not allowed to access the camera and microphone, so the browser prompts Alice for permission. In the current W3C API, once some streams have been added, Alice's browser + JS generates a signaling message containing:

• Media channel information
• Interactive Connectivity Establishment (ICE) candidates
• A "fingerprint" attribute binding the communication to a key pair . Note that this key may simply be ephemerally generated for this call or specific to this domain, and Alice may have a large number of such keys.

Prior to sending out the signaling message, the PeerConnection code contacts the identity service and obtains an assertion binding Alice's identity to her fingerprint. The exact details depend on the identity service (though as discussed in PeerConnection can be agnostic to them), but for now it's easiest to think of as an OAuth token. The assertion may bind other information to the identity besides the fingerprint, but at minimum it needs to bind the fingerprint. This message is sent to the signaling server, e.g., by fetch() or by WebSockets , over TLS . The signaling server processes the message from Alice's browser, determines that this is a call to Bob, and sends a signaling message to Bob's browser (again, the format is currently undefined). The JS on Bob's browser processes it, and alerts Bob to the incoming call and to Alice's identity. In this case, Alice has provided an identity assertion and so Bob's browser contacts Alice's IdP (again, this is done in a generic way so the browser has no specific knowledge of the IdP) to verify the assertion. It is also possible to have IdPs with which the browser has a specific trust relationship, as described in . This allows the browser to display a trusted element in the browser chrome indicating that a call is coming in from Alice. If Alice is in Bob's address book, then this interface might also include her real name, a picture, etc. The calling site will also provide some user interface element (e.g., a button) to allow Bob to answer the call, though this is most likely not part of the trusted UI. If Bob agrees, a PeerConnection is instantiated with the message from Alice's side. Then, a similar process occurs as on Alice's browser: Bob's browser prompts him for device permission, the media streams are created, and a return signaling message containing media information, ICE candidates, and a fingerprint is sent back to Alice via the signaling service. If Bob has a relationship with an IdP, the message will also come with an identity assertion. At this point, Alice and Bob each know that the other party wants to have a secure call with them. Based purely on the interface provided by the signaling server, they know that the signaling server claims that the call is from Alice to Bob. This level of security is provided merely by having the fingerprint in the message and having that message received securely from the signaling server. Because the far end sent an identity assertion along with their message, they know that this is verifiable from the IdP as well. Note that if the call is federated, as shown in , then Alice is able to verify Bob's identity in a way that is not mediated by either her signaling server or Bob's. Rather, she verifies it directly with Bob's IdP. Of course, the call works perfectly well if either Alice or Bob doesn't have a relationship with an IdP; they just get a lower level of assurance. I.e., they simply have whatever information their calling site claims about the caller/callee's identity. Moreover, Alice might wish to make an anonymous call through an anonymous calling site, in which case she would of course just not provide any identity assertion and the calling site would mask her identity from Bob.

Media Consent Verification As described in , media consent verification is provided via ICE. Thus, Alice and Bob perform ICE checks with each other. At the completion of these checks, they are ready to send non-ICE data. At this point, Alice knows that (a) Bob (assuming he is verified via his IdP) or someone else who the signaling service is claiming is Bob is willing to exchange traffic with her and (b) either Bob is at the IP address which she has verified via ICE or there is an attacker who is on-path to that IP address detouring the traffic. Note that it is not possible for an attacker who is on-path between Alice and Bob but not attached to the signaling service to spoof these checks because they do not have the ICE credentials. Bob has the same security guarantees with respect to Alice.

DTLS Handshake Once the requisite ICE checks have completed, Alice and Bob can set up a secure channel or channels. This is performed via DTLS and DTLS-SRTP keying for SRTP for the media channel and the Stream Control Transmission Protocol (SCTP) over DTLS for data channels. Specifically, Alice and Bob perform a DTLS handshake on every component which has been established by ICE. The total number of channels depends on the amount of muxing; in the most likely case, we are using both RTP/RTCP mux and muxing multiple media streams on the same channel, in which case there is only one DTLS handshake. Once the DTLS handshake has completed, the keys are exported and used to key SRTP for the media channels. At this point, Alice and Bob know that they share a set of secure data and/or media channels with keys which are not known to any third-party attacker. If Alice and Bob authenticated via their IdPs, then they also know that the signaling service is not mounting a man-in-the-middle attack on their traffic. Even if they do not use an IdP, as long as they have minimal trust in the signaling service not to perform a man-in-the-middle attack, they know that their communications are secure against the signaling service as well (i.e., that the signaling service cannot mount a passive attack on the communications).

Communications and Consent Freshness From a security perspective, everything from here on in is a little anticlimactic: Alice and Bob exchange data protected by the keys negotiated by DTLS. Because of the security guarantees discussed in the previous sections, they know that the communications are encrypted and authenticated. The one remaining security property we need to establish is "consent freshness", i.e., allowing Alice to verify that Bob is still prepared to receive her communications so that Alice does not continue to send large traffic volumes to entities which went abruptly offline. ICE specifies periodic Session Traversal Utilities for NAT (STUN) keepalives but only if media is not flowing. Because the consent issue is more difficult here, we require WebRTC implementations to periodically send keepalives using the consent freshness mechanism specified in . If a keepalive fails and no new ICE channels can be established, then the session is terminated.

SDP Identity Attribute The SDP "identity" attribute is a session-level attribute that is used by an endpoint to convey its identity assertion to its peer. The identity-assertion value is encoded as base64, as described in . The procedures in this section are based on the assumption that the identity assertion of an endpoint is bound to the fingerprints of the endpoint. This does not preclude the definition of alternative means of binding an assertion to the endpoint, but such means are outside the scope of this specification. The semantics of multiple "identity" attributes within an offer or answer are undefined. Implementations SHOULD only include a single "identity" attribute in an offer or answer, and Relying Parties MAY elect to ignore all but the first "identity" attribute.

Name:

identity

Value:

identity-assertion

Usage Level:

session

Charset Dependent:

no

Default Value:

N/A

Syntax: identity-assertion = identity-assertion-value *(SP identity-extension) identity-assertion-value = base64 identity-extension = extension-name [ "=" extension-value ] extension-name = token extension-value = 1*(%x01-09 / %x0b-0c / %x0e-3a / %x3c-ff) ; byte-string from [RFC4566] <ALPHA and DIGIT as defined in [RFC4566]> <base64 as defined in [RFC4566]> Example: a=identity:\ eyJpZHAiOnsiZG9tYWluIjoiZXhhbXBsZS5vcmciLCJwcm90b2NvbCI6ImJvZ3Vz\ In0sImFzc2VydGlvbiI6IntcImlkZW50aXR5XCI6XCJib2JAZXhhbXBsZS5vcmdc\ IixcImNvbnRlbnRzXCI6XCJhYmNkZWZnaGlqa2xtbm9wcXJzdHV2d3l6XCIsXCJz\ aWduYXR1cmVcIjpcIjAxMDIwMzA0MDUwNlwifSJ9 This specification does not define any extensions for the attribute. The identity-assertion value is a JSON encoded string . The JSON object contains two keys: "assertion" and "idp". The "assertion" key value contains an opaque string that is consumed by the IdP. The "idp" key value contains a dictionary with one or two further values that identify the IdP. See for more details.

Offer/Answer Considerations This section defines the SDP offer/answer considerations for the SDP "identity" attribute. Within this section, 'initial offer' refers to the first offer in the SDP session that contains an SDP "identity" attribute.

Generating the Initial SDP Offer When an offerer sends an offer, in order to provide its identity assertion to the peer, it includes an "identity" attribute in the offer. In addition, the offerer includes one or more SDP "fingerprint" attributes. The "identity" attribute MUST be bound to all the "fingerprint" attributes in the session description.

Generating an SDP Answer If the answerer elects to include an "identity" attribute, it follows the same steps as those in . The answerer can choose to include or omit an "identity" attribute independently, regardless of whether the offerer did so.

Processing an SDP Offer or Answer When an endpoint receives an offer or answer that contains an "identity" attribute, the answerer can use the attribute information to contact the IdP and verify the identity of the peer. If the identity requires a third-party IdP as described in , then that IdP will need to have been specifically configured. If the identity verification fails, the answerer MUST discard the offer or answer as malformed.

Modifying the Session When modifying a session, if the set of fingerprints is unchanged, then the sender MAY send the same "identity" attribute. In this case, the established identity MUST be applied to existing DTLS connections as well as new connections established using one of those fingerprints. Note that requires that each media section use the same set of fingerprints. If a new "identity" attribute is received, then the receiver MUST apply that identity to all existing connections. If the set of fingerprints changes, then the sender MUST either send a new "identity" attribute or none at all. Because a change in fingerprints also causes a new DTLS connection to be established, the receiver MUST discard all previously established identities.

Detailed Technical Description

Origin and Web Security Issues The basic unit of permissions for WebRTC is the origin . Because the security of the origin depends on being able to authenticate content from that origin, the origin can only be securely established if data is transferred over HTTPS . Thus, clients MUST treat HTTP and HTTPS origins as different permissions domains. Note: This follows directly from the origin security model and is stated here merely for clarity. Many Web browsers currently forbid by default any active mixed content on HTTPS pages. That is, when JavaScript is loaded from an HTTP origin onto an HTTPS page, an error is displayed and the HTTP content is not executed unless the user overrides the error. Any browser which enforces such a policy will also not permit access to WebRTC functionality from mixed content pages (because they never display mixed content). Browsers which allow active mixed content MUST nevertheless disable WebRTC functionality in mixed content settings. Note that it is possible for a page which was not mixed content to become mixed content during the duration of the call. The major risk here is that the newly arrived insecure JS might redirect media to a location controlled by the attacker. Implementations MUST either choose to terminate the call or display a warning at that point. Also note that the security architecture depends on the keying material not being available to move between origins. However, it is assumed that the identity assertion can be passed to anyone that the page cares to.

Device Permissions Model Implementations MUST obtain explicit user consent prior to providing access to the camera and/or microphone. Implementations MUST at minimum support the following two permissions models for HTTPS origins.

• Requests for one-time camera/microphone access.
• Requests for permanent access.

Because HTTP origins cannot be securely established against network attackers, implementations MUST refuse all permissions grants for HTTP origins. In addition, they SHOULD support requests for access that promise that media from this grant will be sent to a single communicating peer (obviously there could be other requests for other peers), e.g., "Call customerservice@example.org". The semantics of this request are that the media stream from the camera and microphone will only be routed through a connection which has been cryptographically verified (through the IdP mechanism or an X.509 certificate in the DTLS-SRTP handshake) as being associated with the stated identity. Note that it is unlikely that browsers would have X.509 certificates, but servers might. Browsers servicing such requests SHOULD clearly indicate that identity to the user when asking for permission. The idea behind this type of permissions is that a user might have a fairly narrow list of peers they are willing to communicate with, e.g., "my mother" rather than "anyone on Facebook". Narrow permissions grants allow the browser to do that enforcement.

API Requirement:

The API MUST provide a mechanism for the requesting JS to relinquish the ability to see or modify the media (e.g., via MediaStream.record()). Combined with secure authentication of the communicating peer, this allows a user to be sure that the calling site is not accessing or modifying their conversion.

UI Requirement:

The UI MUST clearly indicate when the user's camera and microphone are in use. This indication MUST NOT be suppressible by the JS and MUST clearly indicate how to terminate device access, and provide a UI means to immediately stop camera/microphone input without the JS being able to prevent it.

UI Requirement:

If the UI indication of camera/microphone use is displayed in the browser such that minimizing the browser window would hide the indication, or the JS creating an overlapping window would hide the indication, then the browser SHOULD stop camera and microphone input when the indication is hidden. (Note: This may not be necessary in systems that are non-windows-based but that have good notifications support, such as phones.)

• Browsers MUST NOT permit permanent screen or application sharing permissions to be installed as a response to a JS request for permissions. Instead, they must require some other user action such as a permissions setting or an application install experience to grant permission to a site.
• Browsers MUST provide a separate dialog request for screen/application sharing permissions even if the media request is made at the same time as the request for camera and microphone permissions.
• The browser MUST indicate any windows which are currently being shared in some unambiguous way. Windows which are not visible MUST NOT be shared even if the application is being shared. If the screen is being shared, then that MUST be indicated.

Browsers MAY permit the formation of data channels without any direct user approval. Because sites can always tunnel data through the server, further restrictions on the data channel do not provide any additional security. (See for a related issue.) Implementations which support some form of direct user authentication SHOULD also provide a policy by which a user can authorize calls only to specific communicating peers. Specifically, the implementation SHOULD provide the following interfaces/controls:

• Allow future calls to this verified user.
• Allow future calls to any verified user who is in my system address book (this only works with address book integration, of course).

Implementations SHOULD also provide a different user interface indication when calls are in progress to users whose identities are directly verifiable. provides more on this.

Communications Consent Browser client implementations of WebRTC MUST implement ICE. Server gateway implementations which operate only at public IP addresses MUST implement either full ICE or ICE-Lite . Browser implementations MUST verify reachability via ICE prior to sending any non-ICE packets to a given destination. Implementations MUST NOT provide the ICE transaction ID to JavaScript during the lifetime of the transaction (i.e., during the period when the ICE stack would accept a new response for that transaction). The JS MUST NOT be permitted to control the local ufrag and password, though it of course knows it. While continuing consent is required, the ICE keepalives use STUN Binding Indications, which are one-way and therefore not sufficient. The current WG consensus is to use ICE Binding Requests for continuing consent freshness. ICE already requires that implementations respond to such requests, so this approach is maximally compatible. A separate document will profile the ICE timers to be used; see .

IP Location Privacy A side effect of the default ICE behavior is that the peer learns one's IP address, which leaks large amounts of location information. This has negative privacy consequences in some circumstances. The API requirements in this section are intended to mitigate this issue. Note that these requirements are not intended to protect the user's IP address from a malicious site. In general, the site will learn at least a user's server-reflexive address from any HTTP transaction. Rather, these requirements are intended to allow a site to cooperate with the user to hide the user's IP address from the other side of the call. Hiding the user's IP address from the server requires some sort of explicit privacy-preserving mechanism on the client (e.g., Tor Browser ) and is out of scope for this specification.

API Requirement:

The API MUST provide a mechanism to allow the JS to suppress ICE negotiation (though perhaps to allow candidate gathering) until the user has decided to answer the call. (Note: Determining when the call has been answered is a question for the JS.) This enables a user to prevent a peer from learning their IP address if they elect not to answer a call and also from learning whether the user is online.

API Requirement:

The API MUST provide a mechanism for the calling application JS to indicate that only TURN candidates are to be used. This prevents the peer from learning one's IP address at all. This mechanism MUST also permit suppression of the related address field, since that leaks local addresses.

API Requirement:

The API MUST provide a mechanism for the calling application to reconfigure an existing call to add non-TURN candidates. Taken together, this and the previous requirement allow ICE negotiation to start immediately on incoming call notification, thus reducing post-dial delay, but also to avoid disclosing the user's IP address until they have decided to answer. They also allow users to completely hide their IP address for the duration of the call. Finally, they allow a mechanism for the user to optimize performance by reconfiguring to allow non-TURN candidates during an active call if the user decides they no longer need to hide their IP address.

Note that some enterprises may operate proxies and/or NATs designed to hide internal IP addresses from the outside world. WebRTC provides no explicit mechanism to allow this function. Either such enterprises need to proxy the HTTP/HTTPS and modify the SDP and/or the JS, or there needs to be browser support to set the "TURN-only" policy regardless of the site's preferences. Note: These requirements are intended to allow sites to conceal the user's IP address from the peer. For guidance on concealing the user's IP address from the calling site see .

Communications Security Implementations MUST support SRTP . Implementations MUST support DTLS and DTLS-SRTP for SRTP keying. Implementations MUST support SCTP over DTLS . All media channels MUST be secured via SRTP and the Secure Real-time Transport Control Protocol (SRTCP). Media traffic MUST NOT be sent over plain (unencrypted) RTP or RTCP; that is, implementations MUST NOT negotiate cipher suites with NULL encryption modes. DTLS-SRTP MUST be offered for every media channel. WebRTC implementations MUST NOT offer SDP security descriptions or select it if offered. An SRTP Master Key Identifier (MKI) MUST NOT be used. All data channels MUST be secured via DTLS. All implementations MUST support DTLS 1.2 with the TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite and the P-256 curve. Earlier drafts of this specification required DTLS 1.0 with the cipher suite TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA, and at the time of this writing some implementations do not support DTLS 1.2; endpoints which support only DTLS 1.2 might encounter interoperability issues. The DTLS-SRTP protection profile SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP. Implementations MUST favor cipher suites which support Forward Secrecy (FS) over non-FS cipher suites and SHOULD favor Authenticated Encryption with Associated Data (AEAD) over non-AEAD cipher suites. Note: the IETF is in the process of standardizing DTLS 1.3 . Implementations MUST NOT implement DTLS renegotiation and MUST reject it with a "no_renegotiation" alert if offered. Endpoints MUST NOT implement TLS False Start .

API Requirement:

The API MUST generate a new authentication key pair for every new call by default. This is intended to allow for unlinkability.

API Requirement:

The API MUST provide a means to reuse a key pair for calls. This can be used to enable key continuity-based authentication, and could be used to amortize key generation costs.

API Requirement:

Unless the user specifically configures an external key pair, different key pairs MUST be used for each origin. (This avoids creating a super-cookie.)

API Requirement:

When DTLS-SRTP is used, the API MUST NOT permit the JS to obtain the negotiated keying material. This requirement preserves the end-to-end security of the media.

UI Requirements:

A user-oriented client MUST provide an "inspector" interface which allows the user to determine the "security characteristics" of the media.

The following properties SHOULD be displayed "up-front" in the browser chrome, i.e., without requiring the user to ask for them:

• A client MUST provide a user interface through which a user may determine the "security characteristics" for currently displayed audio and video stream(s).
• A client MUST provide a user interface through which a user may determine the "security characteristics" for transmissions of their microphone audio and camera video.
• If the far endpoint was directly verified, either via a third-party verifiable X.509 certificate or via a Web IdP mechanism (see ), the "security characteristics" MUST include the verified information. X.509 identities and Web IdP identities have similar semantics and should be displayed in a similar way.

The following properties are more likely to require some "drill-down" from the user:

• The "security characteristics" MUST indicate the cryptographic algorithms in use (for example, "AES-CBC").
• The "security characteristics" MUST indicate whether FS is provided.
• The "security characteristics" MUST include some mechanism to allow an out-of-band verification of the peer, such as a certificate fingerprint or a Short Authentication String (SAS). These are compared by the peers to authenticate one another.

Web-Based Peer Authentication NOTE: The mechanism described in this section was designed relatively early in the RTCWEB process. In retrospect, the WG was too optimistic about the enthusiasm for this kind of mechanism. At the time of publication, it has not been widely adopted or implemented. It appears in this document as a description of the state of the art as of this writing. In a number of cases, it is desirable for the endpoint (i.e., the browser) to be able to directly identify the endpoint on the other side without trusting the signaling service to which they are connected. For instance, users may be making a call via a federated system where they wish to get direct authentication of the other side. Alternately, they may be making a call on a site which they minimally trust (such as a poker site) but to someone who has an identity on a site they do trust (such as a social network). Recently, a number of Web-based identity technologies (OAuth, Facebook Connect, etc.) have been developed. While the details vary, what these technologies share is that they have a Web-based (i.e., HTTP/HTTPS) IdP which attests to Alice's identity. For instance, if Alice has an account at example.org, Alice could use the example.org IdP to prove to others that Alice is alice@example.org. The development of these technologies allows us to separate calling from identity provision: Alice could call you on a poker site but identify herself as alice@example.org. Whatever the underlying technology, the general principle is that the party which is being authenticated is NOT the signaling site but rather the user (and their browser). Similarly, the Relying Party is the browser and not the signaling site. Thus, the browser MUST generate the input to the IdP assertion process and display the results of the verification process to the user in a way which cannot be imitated by the calling site. The mechanisms defined in this document do not require the browser to implement any particular identity protocol or to support any particular IdP. Instead, this document provides a generic interface which any IdP can implement. Thus, new IdPs and protocols can be introduced without change to either the browser or the calling service. This avoids the need to make a commitment to any particular identity protocol, although browsers may opt to directly implement some identity protocols in order to provide superior performance or UI properties.

Trust Relationships: IdPs, APs, and RPs Any federated identity protocol has three major participants:

Authenticating Party (AP):

The entity which is trying to establish its identity.

Identity Provider (IdP):

The entity which is vouching for the AP's identity.

Relying Party (RP):

The entity which is trying to verify the AP's identity.

The AP and the IdP have an account relationship of some kind: the AP registers with the IdP and is able to subsequently authenticate directly to the IdP (e.g., with a password). This means that the browser must somehow know which IdP(s) the user has an account relationship with. This can either be something that the user configures into the browser or that is configured at the calling site and then provided to the PeerConnection by the Web application at the calling site. The use case for having this information configured into the browser is that the user may "log into" the browser to bind it to some identity. This is becoming common in new browsers. However, it should also be possible for the IdP information to simply be provided by the calling application. At a high level, there are two kinds of IdPs:

Authoritative:

IdPs which have verifiable control of some section of the identity space. For instance, in the realm of email, the operator of "example.com" has complete control of the namespace ending in "@example.com". Thus, "alice@example.com" is whoever the operator says it is. Examples of systems with authoritative IdPs include DNSSEC, an identity system for SIP (see ), and Facebook Connect (Facebook identities only make sense within the context of the Facebook system).

Third-Party:

IdPs which don't have control of their section of the identity space but instead verify users' identities via some unspecified mechanism and then attest to it. Because the IdP doesn't actually control the namespace, RPs need to trust that the IdP is correctly verifying AP identities, and there can potentially be multiple IdPs attesting to the same section of the identity space. Probably the best-known example of a third-party IdP is SSL/TLS certificates, where there are a large number of certificate authorities (CAs) all of whom can attest to any domain name.

If an AP is authenticating via an authoritative IdP, then the RP does not need to explicitly configure trust in the IdP at all. The identity mechanism can directly verify that the IdP indeed made the relevant identity assertion (a function provided by the mechanisms in this document), and any assertion it makes about an identity for which it is authoritative is directly verifiable. Note that this does not mean that the IdP might not lie, but that is a trustworthiness judgement that the user can make at the time they look at the identity. By contrast, if an AP is authenticating via a third-party IdP, the RP needs to explicitly trust that IdP (hence the need for an explicit trust anchor list in PKI-based SSL/TLS clients). The list of trustable IdPs needs to be configured directly into the browser, either by the user or potentially by the browser manufacturer. This is a significant advantage of authoritative IdPs and implies that if third-party IdPs are to be supported, the potential number needs to be fairly small.

Overview of Operation In order to provide security without trusting the calling site, the PeerConnection component of the browser must interact directly with the IdP. The details of the mechanism are described in the W3C API specification, but the general idea is that the PeerConnection component downloads JS from a specific location on the IdP dictated by the IdP domain name. That JS (the "IdP proxy") runs in an isolated security context within the browser, and the PeerConnection talks to it via a secure message passing channel. Note that there are two logically separate functions here:

• Identity assertion generation.
• Identity assertion verification.

The same IdP JS "endpoint" is used for both functions, but of course a given IdP might behave differently and load new JS to perform one function or the other. +--------------------------------------+ | Browser | | | | +----------------------------------+ | | | https://calling-site.example.com | | | | | | | | Calling JS Code | | | | ^ | | | +---------------|------------------+ | | | API Calls | | v | | PeerConnection | | ^ | | | API Calls | | +-----------|-------------+ | +---------------+ | | v | | | | | | IdP Proxy |<-------->| Identity | | | | | | Provider | | | https://idp.example.org | | | | | +-------------------------+ | +---------------+ | | +--------------------------------------+ When the PeerConnection object wants to interact with the IdP, the sequence of events is as follows:

1. The browser (the PeerConnection component) instantiates an IdP proxy. This allows the IdP to load whatever JS is necessary into the proxy. The resulting code runs in the IdP's security context.
2. The IdP registers an object with the browser that conforms to the API defined in .
3. The browser invokes methods on the object registered by the IdP proxy to create or verify identity assertions.

This approach allows us to decouple the browser from any particular IdP; the browser need only know how to load the IdP's JavaScript -- the location of which is determined based on the IdP's identity -- and to call the generic API for requesting and verifying identity assertions. The IdP provides whatever logic is necessary to bridge the generic protocol to the IdP's specific requirements. Thus, a single browser can support any number of identity protocols, including being forward compatible with IdPs which did not exist at the time the browser was written.

Items for Standardization There are two parts to this work:

• The precise information from the signaling message that must be cryptographically bound to the user's identity and a mechanism for carrying assertions in JavaScript Session Establishment Protocol (JSEP) messages. This is specified in .
• The interface to the IdP, which is defined in the companion W3C WebRTC API specification .

The WebRTC API specification also defines JavaScript interfaces that the calling application can use to specify which IdP to use. That API also provides access to the assertion-generation capability and the status of the validation process.

Binding Identity Assertions to JSEP Offer/Answer Transactions An identity assertion binds the user's identity (as asserted by the IdP) to the SDP offer/answer exchange and specifically to the media. In order to achieve this, the PeerConnection must provide the DTLS-SRTP fingerprint to be bound to the identity. This is provided as a JavaScript object (also known as a dictionary or hash) with a single "fingerprint" key, as shown below: { "fingerprint": [ { "algorithm": "sha-256", "digest": "4A:AD:B9:B1:3F:...:E5:7C:AB" }, { "algorithm": "sha-1", "digest": "74:E9:76:C8:19:...:F4:45:6B" } ] } The "fingerprint" value is an array of objects. Each object in the array contains "algorithm" and "digest" values, which correspond directly to the algorithm and digest values in the "fingerprint" attribute of the SDP . This object is encoded in a JSON string for passing to the IdP. The identity assertion returned by the IdP, which is encoded in the "identity" attribute, is a JSON object that is encoded as described in . This structure does not need to be interpreted by the IdP or the IdP proxy. It is consumed solely by the RP's browser. The IdP merely treats it as an opaque value to be attested to. Thus, new parameters can be added to the assertion without modifying the IdP.

Carrying Identity Assertions Once an IdP has generated an assertion (see ), it is attached to the SDP offer/answer message. This is done by adding a new "identity" attribute to the SDP. The sole contents of this value is the identity assertion. The identity assertion produced by the IdP is encoded into a UTF-8 JSON text, then base64-encoded to produce this string. For example: v=0 o=- 1181923068 1181923196 IN IP4 ua1.example.com s=example1 c=IN IP4 ua1.example.com a=fingerprint:sha-1 \ 4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB a=identity:\ eyJpZHAiOnsiZG9tYWluIjoiZXhhbXBsZS5vcmciLCJwcm90b2NvbCI6ImJvZ3Vz\ In0sImFzc2VydGlvbiI6IntcImlkZW50aXR5XCI6XCJib2JAZXhhbXBsZS5vcmdc\ IixcImNvbnRlbnRzXCI6XCJhYmNkZWZnaGlqa2xtbm9wcXJzdHV2d3l6XCIsXCJz\ aWduYXR1cmVcIjpcIjAxMDIwMzA0MDUwNlwifSJ9 a=... t=0 0 m=audio 6056 RTP/SAVP 0 a=sendrecv ... The "identity" attribute attests to all "fingerprint" attributes in the session description. It is therefore a session-level attribute. Multiple "fingerprint" values can be used to offer alternative certificates for a peer. The "identity" attribute MUST include all "fingerprint" values that are included in "fingerprint" attributes of the session description. The RP browser MUST verify that the in-use certificate for a DTLS connection is in the set of fingerprints returned from the IdP when verifying an assertion.

Determining the IdP URI In order to ensure that the IdP is under control of the domain owner rather than someone who merely has an account on the domain owner's server (e.g., in shared hosting scenarios), the IdP JavaScript is hosted at a deterministic location based on the IdP's domain name. Each IdP proxy instance is associated with two values:

authority:

The authority at which the IdP's service is hosted.

protocol:

The specific IdP protocol which the IdP is using. This is a completely opaque IdP-specific string, but allows an IdP to implement two protocols in parallel. This value may be the empty string. If no value for protocol is provided, a value of "default" is used.

Each IdP MUST serve its initial entry page (i.e., the one loaded by the IdP proxy) from a well-known URI. The well-known URI for an IdP proxy is formed from the following URI components:

1. The scheme, "https:". An IdP MUST be loaded using HTTPS.
2. The authority. As noted above, the authority MAY contain a non-default port number or userinfo sub-component. Both are removed when determining if an asserted identity matches the name of the IdP.
3. The path, starting with "/.well-known/idp-proxy/" and appended with the IdP protocol. Note that the separator characters '/' (%2F) and '\' (%5C) MUST NOT be permitted in the protocol field, lest an attacker be able to direct requests outside of the controlled "/.well-known/" prefix. Query and fragment values MAY be used by including '?' or '#' characters.

For example, for the IdP "identity.example.com" and the protocol "example", the URL would be: https://identity.example.com/.well-known/idp-proxy/example The IdP MAY redirect requests to this URL, but they MUST retain the "https:" scheme. This changes the effective origin of the IdP, but not the domain of the identities that the IdP is permitted to assert and validate. I.e., the IdP is still regarded as authoritative for the original domain.

Authenticating Party How an AP determines the appropriate IdP domain is out of scope of this specification. In general, however, the AP has some actual account relationship with the IdP, as this identity is what the IdP is attesting to. Thus, the AP somehow supplies the IdP information to the browser. Some potential mechanisms include:

• Provided by the user directly.
• Selected from some set of IdPs known to the calling site (e.g., a button that shows "Authenticate via Facebook Connect").

Relying Party Unlike the AP, the RP need not have any particular relationship with the IdP. Rather, it needs to be able to process whatever assertion is provided by the AP. As the assertion contains the IdP's identity in the "idp" field of the JSON-encoded object (see ), the URI can be constructed directly from the assertion, and thus the RP can directly verify the technical validity of the assertion with no user interaction. Authoritative assertions need only be verifiable. Third-party assertions also MUST be verified against local policy, as described in .

Requesting Assertions The input to the identity assertion generation process is the JSON-encoded object described in that contains the set of certificate fingerprints the browser intends to use. This string is treated as opaque from the perspective of the IdP. The browser also identifies the origin that the PeerConnection is run in, which allows the IdP to make decisions based on who is requesting the assertion. An application can optionally provide a user identifier hint when specifying an IdP. This value is a hint that the IdP can use to select amongst multiple identities, or to avoid providing assertions for unwanted identities. The "username" is a string that has no meaning to any entity other than the IdP; it can contain any data the IdP needs in order to correctly generate an assertion. An identity assertion that is successfully provided by the IdP consists of the following information:

idp:

The domain name of an IdP and the protocol string. This MAY identify a different IdP or protocol from the one that generated the assertion.

assertion:

An opaque value containing the assertion itself. This is only interpretable by the identified IdP or the IdP code running in the client.

shows an example assertion formatted as JSON. In this case, the message has presumably been digitally signed/MACed in some way that the IdP can later verify it, but this is an implementation detail and out of scope of this document.

Example Assertion { "idp":{ "domain": "example.org", "protocol": "bogus" }, "assertion": "{\"identity\":\"bob@example.org\", \"contents\":\"abcdefghijklmnopqrstuvwyz\", \"signature\":\"010203040506\"}" }

For use in signaling, the assertion is serialized into JSON, base64-encoded, and used as the value of the "identity" attribute. IdPs SHOULD ensure that any assertions they generate cannot be interpreted in a different context. E.g., they should use a distinct format or have separate cryptographic keys for assertion generation and other purposes. Line breaks are inserted solely for readability.

Managing User Login In order to generate an identity assertion, the IdP needs proof of the user's identity. It is common practice to authenticate users (using passwords or multi-factor authentication), then use cookies or HTTP authentication for subsequent exchanges. The IdP proxy is able to access cookies, HTTP authentication data, or other persistent session data because it operates in the security context of the IdP origin. Therefore, if a user is logged in, the IdP could have all the information needed to generate an assertion. An IdP proxy is unable to generate an assertion if the user is not logged in, or the IdP wants to interact with the user to acquire more information before generating the assertion. If the IdP wants to interact with the user before generating an assertion, the IdP proxy can fail to generate an assertion and instead indicate a URL where login should proceed. The application can then load the provided URL to enable the user to enter credentials. The communication between the application and the IdP is described in .

Verifying Assertions The input to identity validation is the assertion string taken from a decoded "identity" attribute. The IdP proxy verifies the assertion. Depending on the identity protocol, the proxy might contact the IdP server or other servers. For instance, an OAuth-based protocol will likely require using the IdP as an oracle, whereas with a signature-based scheme it might be able to verify the assertion without contacting the IdP, provided that it has cached the relevant public key. Regardless of the mechanism, if verification succeeds, a successful response from the IdP proxy consists of the following information:

identity:

The identity of the AP from the IdP's perspective. Details of this are provided in .

contents:

The original unmodified string provided by the AP as input to the assertion generation process.

shows an example response, which is JSON-formatted.

Example Verification Result { "identity": "bob@example.org", "contents": "{\"fingerprint\":[ ... ]}" }

Identity Formats The identity provided from the IdP to the RP browser MUST consist of a string representing the user's identity. This string is in the form "<user>@<domain>", where "user" consists of any character, and domain is an internationalized domain name encoded as a sequence of U-labels. The PeerConnection API MUST check this string as follows:

1. If the "domain" portion of the string is equal to the domain name of the IdP proxy, then the assertion is valid, as the IdP is authoritative for this domain. Comparison of domain names is done using the label equivalence rule defined in .
2. If the "domain" portion of the string is not equal to the domain name of the IdP proxy, then the PeerConnection object MUST reject the assertion unless both:
   1. the IdP domain is trusted as an acceptable third-party IdP; and
   2. local policy is configured to trust this IdP domain for the domain portion of the identity string.

Any '@' or '%' characters in the "user" portion of the identity MUST be escaped according to the "percent-encoding" rules defined in . Characters other than '@' and '%' MUST NOT be percent-encoded. For example, with a "user" of "user@133" and a "domain" of "identity.example.com", the resulting string will be encoded as "user%40133@identity.example.com". Implementations are cautioned to take care when displaying user identities containing escaped '@' characters. If such characters are unescaped prior to display, implementations MUST distinguish between the domain of the IdP proxy and any domain that might be implied by the portion of the "<user>" portion that appears after the escaped "@" sign.

Security Considerations Much of the security analysis of RTCWEB is contained in or in the discussion of the particular issues above. In order to avoid repetition, this section focuses on (a) residual threats that are not addressed by this document and (b) threats produced by failure/misbehavior of one of the components in the system.

Communications Security If HTTPS is not used to secure communications to the signaling server, and the identity mechanism used in is not used, then any on-path attacker can replace the DTLS-SRTP fingerprints in the handshake and thus substitute its own identity for that of either endpoint. Even if HTTPS is used, the signaling server can potentially mount a man-in-the-middle attack unless implementations have some mechanism for independently verifying keys. The UI requirements in are designed to provide such a mechanism for motivated/security conscious users, but are not suitable for general use. The identity service mechanisms in are more suitable for general use. Note, however, that a malicious signaling service can strip off any such identity assertions, though it cannot forge new ones. Note that all of the third-party security mechanisms available (whether X.509 certificates or a third-party IdP) rely on the security of the third party -- this is of course also true of the user's connection to the Web site itself. Users who wish to assure themselves of security against a malicious IdP can only do so by verifying peer credentials directly, e.g., by checking the peer's fingerprint against a value delivered out of band. In order to protect against malicious content JavaScript, that JavaScript MUST NOT be allowed to have direct access to -- or perform computations with -- DTLS keys. For instance, if content JS were able to compute digital signatures, then it would be possible for content JS to get an identity assertion for a browser's generated key and then use that assertion plus a signature by the key to authenticate a call protected under an ephemeral Diffie-Hellman (DH) key controlled by the content JS, thus violating the security guarantees otherwise provided by the IdP mechanism. Note that it is not sufficient merely to deny the content JS direct access to the keys, as some have suggested doing with the WebCrypto API . The JS must also not be allowed to perform operations that would be valid for a DTLS endpoint. By far the safest approach is simply to deny the ability to perform any operations that depend on secret information associated with the key. Operations that depend on public information, such as exporting the public key, are of course safe.

Privacy The requirements in this document are intended to allow:

• Users to participate in calls without revealing their location.
• Potential callees to avoid revealing their location and even presence status prior to agreeing to answer a call.

However, these privacy protections come at a performance cost in terms of using TURN relays and, in the latter case, delaying ICE. Sites SHOULD make users aware of these tradeoffs. Note that the protections provided here assume a non-malicious calling service. As the calling service always knows the user's status and (absent the use of a technology like Tor) their IP address, they can violate the user's privacy at will. Users who wish privacy against the calling sites they are using must use separate privacy-enhancing technologies such as Tor. Combined WebRTC/Tor implementations SHOULD arrange to route the media as well as the signaling through Tor. Currently this will produce very suboptimal performance. Additionally, any identifier which persists across multiple calls is potentially a problem for privacy, especially for anonymous calling services. Such services SHOULD instruct the browser to use separate DTLS keys for each call and also to use TURN throughout the call. Otherwise, the other side will learn linkable information that would allow them to correlate the browser across multiple calls. Additionally, browsers SHOULD implement the privacy-preserving CNAME generation mode of .

Denial of Service The consent mechanisms described in this document are intended to mitigate denial-of-service (DoS) attacks in which an attacker uses clients to send large amounts of traffic to a victim without the consent of the victim. While these mechanisms are sufficient to protect victims who have not implemented WebRTC at all, WebRTC implementations need to be more careful. Consider the case of a call center which accepts calls via WebRTC. An attacker proxies the call center's front-end and arranges for multiple clients to initiate calls to the call center. Note that this requires user consent in many cases, but because the data channel does not need consent, they can use that directly. Since ICE will complete, browsers can then be induced to send large amounts of data to the victim call center if it supports the data channel at all. Preventing this attack requires that automated WebRTC implementations implement sensible flow control and have the ability to triage out (i.e., stop responding to ICE probes on) calls which are behaving badly, and especially to be prepared to remotely throttle the data channel in the absence of plausible audio and video (which the attacker cannot control). Another related attack is for the signaling service to swap the ICE candidates for the audio and video streams, thus forcing a browser to send video to the sink that the other victim expects will contain audio (perhaps it is only expecting audio!), potentially causing overload. Muxing multiple media flows over a single transport makes it harder to individually suppress a single flow by denying ICE keepalives. Either media-level (RTCP) mechanisms must be used or the implementation must deny responses entirely, thus terminating the call. Yet another attack, suggested by Magnus Westerlund, is for the attacker to cross-connect offers and answers as follows. It induces the victim to make a call and then uses its control of other users' browsers to get them to attempt a call to someone. It then translates their offers into apparent answers to the victim, which looks like large-scale parallel forking. The victim still responds to ICE responses, and now the browsers all try to send media to the victim. Implementations can defend themselves from this attack by only responding to ICE Binding Requests for a limited number of remote ufrags (this is the reason for the requirement that the JS not be able to control the ufrag and password). documents a number of potential RTCP-based DoS attacks and countermeasures. Note that attacks based on confusing one end or the other about consent are possible even in the face of the third-party identity mechanism as long as major parts of the signaling messages are not signed. On the other hand, signing the entire message severely restricts the capabilities of the calling application, so there are difficult tradeoffs here.

IdP Authentication Mechanism This mechanism relies for its security on the IdP and on the PeerConnection correctly enforcing the security invariants described above. At a high level, the IdP is attesting that the user identified in the assertion wishes to be associated with the assertion. Thus, it must not be possible for arbitrary third parties to get assertions tied to a user or to produce assertions that RPs will accept.

PeerConnection Origin Check Fundamentally, the IdP proxy is just a piece of HTML and JS loaded by the browser, so nothing stops a Web attacker from creating their own IFRAME, loading the IdP proxy HTML/JS, and requesting a signature over their own keys rather than those generated in the browser. However, that proxy would be in the attacker's origin, not the IdP's origin. Only the browser itself can instantiate a context that (a) is in the IdP's origin and (b) exposes the correct API surface. Thus, the IdP proxy on the sender's side MUST ensure that it is running in the IdP's origin prior to issuing assertions. Note that this check only asserts that the browser (or some other entity with access to the user's authentication data) attests to the request and hence to the fingerprint. It does not demonstrate that the browser has access to the associated private key, and therefore an attacker can attach their own identity to another party's keying material, thus making a call which comes from Alice appear to come from the attacker. See for defenses against this form of attack.

IdP Well-Known URI As described in , the IdP proxy HTML/JS landing page is located at a well-known URI based on the IdP's domain name. This requirement prevents an attacker who can write some resources at the IdP (e.g., on one's Facebook wall) from being able to impersonate the IdP.

Privacy of IdP-Generated Identities and the Hosting Site Depending on the structure of the IdP's assertions, the calling site may learn the user's identity from the perspective of the IdP. In many cases, this is not an issue because the user is authenticating to the site via the IdP in any case -- for instance, when the user has logged in with Facebook Connect and is then authenticating their call with a Facebook identity. However, in other cases, the user may not have already revealed their identity to the site. In general, IdPs SHOULD either verify that the user is willing to have their identity revealed to the site (e.g., through the usual IdP permissions dialog) or arrange that the identity information is only available to known RPs (e.g., social graph adjacencies) but not to the calling site. The "domain" field of the assertion request can be used to check that the user has agreed to disclose their identity to the calling site; because it is supplied by the PeerConnection it can be trusted to be correct.

Security of Third-Party IdPs As discussed above, each third-party IdP represents a new universal trust point and therefore the number of these IdPs needs to be quite limited. Most IdPs, even those which issue unqualified identities such as Facebook, can be recast as authoritative IdPs (e.g., 123456@facebook.com). However, in such cases, the user interface implications are not entirely desirable. One intermediate approach is to have special (potentially user configurable) UI for large authoritative IdPs, thus allowing the user to instantly grasp that the call is being authenticated by Facebook, Google, etc.

Confusable Characters Because a broad range of characters are permitted in identity strings, it may be possible for attackers to craft identities which are confusable with other identities (see for more on this topic). This is a problem with any identifier space of this type (e.g., email addresses). Those minting identifiers should avoid mixed scripts and similar confusable characters. Those presenting these identifiers to a user should consider highlighting cases of mixed script usage (see ). Other best practices are still in development.

Web Security Feature Interactions A number of optional Web security features have the potential to cause issues for this mechanism, as discussed below.

Popup Blocking When popup blocking is in use, the IdP proxy is unable to generate popup windows, dialogs, or any other form of user interactions. This prevents the IdP proxy from being used to circumvent user interaction. The "LOGINNEEDED" message allows the IdP proxy to inform the calling site of a need for user login, providing the information necessary to satisfy this requirement without resorting to direct user interaction from the IdP proxy itself.

Third Party Cookies Some browsers allow users to block third party cookies (cookies associated with origins other than the top-level page) for privacy reasons. Any IdP which uses cookies to persist logins will be broken by third-party cookie blocking. One option is to accept this as a limitation; another is to have the PeerConnection object disable third-party cookie blocking for the IdP proxy.

IANA Considerations This specification defines the "identity" SDP attribute per the procedures of . The required information for the registration is included here:

Contact Name:

IESG (iesg@ietf.org)

Attribute Name:

identity

Long Form:

identity

Type of Attribute:

session

Charset Considerations:

This attribute is not subject to the charset attribute.

Purpose:

This attribute carries an identity assertion, binding an identity to the transport-level security session.

Appropriate Values:

See of RFC 8827.

Mux Category:

NORMAL

This section registers the "idp-proxy" well-known URI from .

URI suffix:

idp-proxy

Change controller:

IETF

References Normative References National Institute of Standards and Technology (NIST) In many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. This memo describes how to use Transport Layer Security (TLS) to secure Hypertext Transfer Protocol (HTTP) connections over the Internet. This memo provides information for the Internet community. This document defines a mechanism by which two entities can make use of the Session Description Protocol (SDP) to arrive at a common view of a multimedia session between them. In the model, one participant offers the other a description of the desired session from their perspective, and the other participant answers with the desired session from their perspective. This offer/answer model is most useful in unicast sessions where information from both participants is needed for the complete view of the session. The offer/answer model is used by protocols like the Session Initiation Protocol (SIP). [STANDARDS-TRACK] This document describes the Secure Real-time Transport Protocol (SRTP), a profile of the Real-time Transport Protocol (RTP), which can provide confidentiality, message authentication, and replay protection to the RTP traffic and to the control traffic for RTP, the Real-time Transport Control Protocol (RTCP). [STANDARDS-TRACK] A Uniform Resource Identifier (URI) is a compact sequence of characters that identifies an abstract or physical resource. This specification defines the generic URI syntax and a process for resolving URI references that might be in relative form, along with guidelines and security considerations for the use of URIs on the Internet. The URI syntax defines a grammar that is a superset of all valid URIs, allowing an implementation to parse the common components of a URI reference without knowing the scheme-specific requirements of every possible identifier. This specification does not define a generative grammar for URIs; that task is performed by the individual specifications of each URI scheme. [STANDARDS-TRACK] This memo defines the Session Description Protocol (SDP). SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. [STANDARDS-TRACK] This document defines a Session Description Protocol (SDP) cryptographic attribute for unicast media streams. The attribute describes a cryptographic key and other parameters that serve to configure security for a unicast media stream in either a single message or a roundtrip exchange. The attribute can be used with a variety of SDP media transports, and this document defines how to use it for the Secure Real-time Transport Protocol (SRTP) unicast media streams. The SDP crypto attribute requires the services of a data security protocol to secure the SDP message. [STANDARDS-TRACK] This document describes the commonly used base 64, base 32, and base 16 encoding schemes. It also discusses the use of line-feeds in encoded data, use of padding in encoded data, use of non-alphabet characters in encoded data, use of different encoding alphabets, and canonical encodings. [STANDARDS-TRACK] This document specifies how to use the Session Initiation Protocol (SIP) to establish a 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. 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. [STANDARDS-TRACK] This document describes a Datagram Transport Layer Security (DTLS) extension to establish keys for Secure RTP (SRTP) and Secure RTP Control Protocol (SRTCP) flows. DTLS keying happens on the media path, independent of any out-of-band signalling channel present. [STANDARDS-TRACK] This document is one of a collection that, together, describe the protocol and usage context for a revision of Internationalized Domain Names for Applications (IDNA), superseding the earlier version. It describes the document collection and provides definitions and other material that are common to the set. [STANDARDS-TRACK] This document specifies version 1.2 of the Datagram Transport Layer Security (DTLS) protocol. The DTLS protocol provides communications privacy for datagram protocols. The protocol allows client/server applications to communicate in a way that is designed to prevent eavesdropping, tampering, or message forgery. The DTLS protocol is based on the Transport Layer Security (TLS) protocol and provides equivalent security guarantees. Datagram semantics of the underlying transport are preserved by the DTLS protocol. This document updates DTLS 1.0 to work with TLS version 1.2. [STANDARDS-TRACK] This document defines the concept of an "origin", which is often used as the scope of authority or privilege by user agents. Typically, user agents isolate content retrieved from different origins to prevent malicious web site operators from interfering with the operation of benign web sites. In addition to outlining the principles that underlie the concept of origin, this document details how to determine the origin of a URI and how to serialize an origin into a string. It also defines an HTTP header field, named "Origin", that indicates which origins are associated with an HTTP request. [STANDARDS-TRACK] The RTP Control Protocol (RTCP) Canonical Name (CNAME) is a persistent transport-level identifier for an RTP endpoint. While the Synchronization Source (SSRC) identifier of an RTP endpoint may change if a collision is detected or when the RTP application is restarted, its RTCP CNAME is meant to stay unchanged, so that RTP endpoints can be uniquely identified and associated with their RTP media streams. For proper functionality, RTCP CNAMEs should be unique within the participants of an RTP session. However, the existing guidelines for choosing the RTCP CNAME provided in the RTP standard (RFC 3550) are insufficient to achieve this uniqueness. RFC 6222 was published to update those guidelines to allow endpoints to choose unique RTCP CNAMEs. Unfortunately, later investigations showed that some parts of the new algorithms were unnecessarily complicated and/or ineffective. This document addresses these concerns and replaces RFC 6222. To prevent WebRTC applications, such as browsers, from launching attacks by sending traffic to unwilling victims, periodic consent to send needs to be obtained from remote endpoints. This document describes a consent mechanism using a new Session Traversal Utilities for NAT (STUN) usage. This document specifies an optional behavior of Transport Layer Security (TLS) client implementations, dubbed "False Start". It affects only protocol timing, not on-the-wire protocol data, and can be implemented unilaterally. A TLS False Start reduces handshake latency to one round trip. This document specifies how to establish secure connection-oriented media transport sessions over the Transport Layer Security (TLS) protocol using the Session Description Protocol (SDP). It defines the SDP protocol identifier, 'TCP/TLS'. It also defines the syntax and semantics for an SDP 'fingerprint' attribute that identifies the certificate that will be presented for the TLS session. This mechanism allows media transport over TLS connections to be established securely, so long as the integrity of session descriptions is assured. This document obsoletes RFC 4572 by clarifying the usage of multiple fingerprints. RFC 2119 specifies common key words that may be used in protocol specifications. This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the defined special meanings. JavaScript Object Notation (JSON) is a lightweight, text-based, language-independent data interchange format. It was derived from the ECMAScript Programming Language Standard. JSON defines a small set of formatting rules for the portable representation of structured data. This document removes inconsistencies with other specifications of JSON, repairs specification errors, and offers experience-based interoperability guidance. The Stream Control Transmission Protocol (SCTP) is a transport protocol originally defined to run on top of the network protocols IPv4 or IPv6. This document specifies how SCTP can be used on top of the Datagram Transport Layer Security (DTLS) protocol. Using the encapsulation method described in this document, SCTP is unaware of the protocols being used below DTLS; hence, explicit IP addresses cannot be used in the SCTP control chunks. As a consequence, the SCTP associations carried over DTLS can only be single-homed. This document describes a protocol for Network Address Translator (NAT) traversal for UDP-based communication. This protocol is called Interactive Connectivity Establishment (ICE). ICE makes use of the Session Traversal Utilities for NAT (STUN) protocol and its extension, Traversal Using Relay NAT (TURN). This document obsoletes RFC 5245. This document specifies version 1.3 of the Transport Layer Security (TLS) protocol. TLS allows client/server applications to communicate over the Internet in a way that is designed to prevent eavesdropping, tampering, and message forgery. This document updates RFCs 5705 and 6066, and obsoletes RFCs 5077, 5246, and 6961. This document also specifies new requirements for TLS 1.2 implementations. This memo defines a path prefix for "well-known locations", "/.well-known/", in selected Uniform Resource Identifier (URI) schemes. In doing so, it obsoletes RFC 5785 and updates the URI schemes defined in RFC 7230 to reserve that space. It also updates RFC 7595 to track URI schemes that support well-known URIs in their registry. W3C Recommendation W3C Proposed Recommendation Informative References This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences. [STANDARDS-TRACK] A number of protocols wish to leverage Transport Layer Security (TLS) to perform key establishment but then use some of the keying material for their own purposes. This document describes a general mechanism for allowing that. [STANDARDS-TRACK] The Extensible Messaging and Presence Protocol (XMPP) is an application profile of the Extensible Markup Language (XML) that enables the near-real-time exchange of structured yet extensible data between any two or more network entities. This document defines XMPP's core protocol methods: setup and teardown of XML streams, channel encryption, authentication, error handling, and communication primitives for messaging, network availability ("presence"), and request-response interactions. This document obsoletes RFC 3920. [STANDARDS-TRACK] This document defines the HTTP Cookie and Set-Cookie header fields. These header fields can be used by HTTP servers to store state (called cookies) at HTTP user agents, letting the servers maintain a stateful session over the mostly stateless HTTP protocol. Although cookies have many historical infelicities that degrade their security and privacy, the Cookie and Set-Cookie header fields are widely used on the Internet. This document obsoletes RFC 2965. [STANDARDS-TRACK] The WebSocket Protocol enables two-way communication between a client running untrusted code in a controlled environment to a remote host that has opted-in to communications from that code. The security model used for this is the origin-based security model commonly used by web browsers. The protocol consists of an opening handshake followed by basic message framing, layered over TCP. The goal of this technology is to provide a mechanism for browser-based applications that need two-way communication with servers that does not rely on opening multiple HTTP connections (e.g., using XMLHttpRequest or <iframe>s and long polling). [STANDARDS-TRACK] Identifiers such as hostnames, URIs, IP addresses, and email addresses are often used in security contexts to identify security principals and resources. In such contexts, an identifier presented via some protocol is often compared using some policy to make security decisions such as whether the security principal may access the resource, what level of authentication or encryption is required, etc. If the parties involved in a security decision use different algorithms to compare identifiers, then failure scenarios ranging from denial of service to elevation of privilege can result. This document provides a discussion of these issues that designers should consider when defining identifiers and protocols, and when constructing architectures that use multiple protocols. This document defines the "Basic" Hypertext Transfer Protocol (HTTP) authentication scheme, which transmits credentials as user-id/ password pairs, encoded using Base64. The baseline security mechanisms in the Session Initiation Protocol (SIP) are inadequate for cryptographically assuring the identity of the end users that originate SIP requests, especially in an interdomain context. This document defines a mechanism for securely identifying originators of SIP requests. It does so by defining a SIP header field for conveying a signature used for validating the identity and for conveying a reference to the credentials of the signer. This document obsoletes RFC 4474. RTFM, Inc. Arm Limited Google, Inc. This document specifies Version 1.3 of the Datagram Transport Layer Security (DTLS) protocol. DTLS 1.3 allows client/server applications to communicate over the Internet in a way that is designed to prevent eavesdropping, tampering, and message forgery. The DTLS 1.3 protocol is intentionally based on the Transport Layer Security (TLS) 1.3 protocol and provides equivalent security guarantees with the exception of order protection/non-replayability. Datagram semantics of the underlying transport are preserved by the DTLS protocol. Work in Progress

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Author's Address Mozilla

ekr@rtfm.com