--- 1/draft-ietf-rtcweb-security-10.txt 2019-02-01 15:13:33.362010445 -0800 +++ 2/draft-ietf-rtcweb-security-11.txt 2019-02-01 15:13:33.422011901 -0800 @@ -1,51 +1,51 @@ RTC-Web E. Rescorla Internet-Draft RTFM, Inc. -Intended status: Standards Track January 22, 2018 -Expires: July 26, 2018 +Intended status: Standards Track February 1, 2019 +Expires: August 5, 2019 Security Considerations for WebRTC - draft-ietf-rtcweb-security-10 + draft-ietf-rtcweb-security-11 Abstract WebRTC is a protocol suite for use with real-time applications that can be deployed in browsers - "real time communication on the Web". This document defines the WebRTC threat model and analyzes the security threats of WebRTC in that model. Status of This Memo This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- - Drafts is at http://datatracker.ietf.org/drafts/current/. + Drafts is at https://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." - This Internet-Draft will expire on July 26, 2018. + This Internet-Draft will expire on August 5, 2019. Copyright Notice - Copyright (c) 2018 IETF Trust and the persons identified as the + Copyright (c) 2019 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents - (http://trustee.ietf.org/license-info) in effect on the date of + (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. This document may contain material from IETF Documents or IETF Contributions published or made publicly available before November 10, 2008. The person(s) controlling the copyright in some of this @@ -57,89 +57,89 @@ not be created outside the IETF Standards Process, except to format it for publication as an RFC or to translate it into languages other than English. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 3. The Browser Threat Model . . . . . . . . . . . . . . . . . . 4 3.1. Access to Local Resources . . . . . . . . . . . . . . . . 5 - 3.2. Same Origin Policy . . . . . . . . . . . . . . . . . . . 5 + 3.2. Same-Origin Policy . . . . . . . . . . . . . . . . . . . 5 3.3. Bypassing SOP: CORS, WebSockets, and consent to communicate . . . . . . . . . . . . . . . . . . . . . . . 6 4. Security for WebRTC Applications . . . . . . . . . . . . . . 7 4.1. Access to Local Devices . . . . . . . . . . . . . . . . . 7 4.1.1. Threats from Screen Sharing . . . . . . . . . . . . . 8 4.1.2. Calling Scenarios and User Expectations . . . . . . . 8 - 4.1.2.1. Dedicated Calling Services . . . . . . . . . . . 8 + 4.1.2.1. Dedicated Calling Services . . . . . . . . . . . 9 4.1.2.2. Calling the Site You're On . . . . . . . . . . . 9 - 4.1.3. Origin-Based Security . . . . . . . . . . . . . . . . 9 + 4.1.3. Origin-Based Security . . . . . . . . . . . . . . . . 10 4.1.4. Security Properties of the Calling Page . . . . . . . 11 4.2. Communications Consent Verification . . . . . . . . . . . 12 - 4.2.1. ICE . . . . . . . . . . . . . . . . . . . . . . . . . 12 + 4.2.1. ICE . . . . . . . . . . . . . . . . . . . . . . . . . 13 4.2.2. Masking . . . . . . . . . . . . . . . . . . . . . . . 13 - 4.2.3. Backward Compatibility . . . . . . . . . . . . . . . 13 - 4.2.4. IP Location Privacy . . . . . . . . . . . . . . . . . 14 + 4.2.3. Backward Compatibility . . . . . . . . . . . . . . . 14 + 4.2.4. IP Location Privacy . . . . . . . . . . . . . . . . . 15 4.3. Communications Security . . . . . . . . . . . . . . . . . 15 4.3.1. Protecting Against Retrospective Compromise . . . . . 16 4.3.2. Protecting Against During-Call Attack . . . . . . . . 17 4.3.2.1. Key Continuity . . . . . . . . . . . . . . . . . 17 4.3.2.2. Short Authentication Strings . . . . . . . . . . 18 - 4.3.2.3. Third Party Identity . . . . . . . . . . . . . . 18 + 4.3.2.3. Third Party Identity . . . . . . . . . . . . . . 19 4.3.2.4. Page Access to Media . . . . . . . . . . . . . . 19 4.3.3. Malicious Peers . . . . . . . . . . . . . . . . . . . 20 4.4. Privacy Considerations . . . . . . . . . . . . . . . . . 20 4.4.1. Correlation of Anonymous Calls . . . . . . . . . . . 20 - 4.4.2. Browser Fingerprinting . . . . . . . . . . . . . . . 20 - 5. Security Considerations . . . . . . . . . . . . . . . . . . . 20 + 4.4.2. Browser Fingerprinting . . . . . . . . . . . . . . . 21 + 5. Security Considerations . . . . . . . . . . . . . . . . . . . 21 6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 21 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 21 8. Changes Since -04 . . . . . . . . . . . . . . . . . . . . . . 21 - 9. References . . . . . . . . . . . . . . . . . . . . . . . . . 21 - 9.1. Normative References . . . . . . . . . . . . . . . . . . 21 - 9.2. Informative References . . . . . . . . . . . . . . . . . 21 - Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 24 + 9. References . . . . . . . . . . . . . . . . . . . . . . . . . 22 + 9.1. Normative References . . . . . . . . . . . . . . . . . . 22 + 9.2. Informative References . . . . . . . . . . . . . . . . . 22 + Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 25 1. Introduction - The Real-Time Communications on the Web (RTCWEB) working group is - tasked with standardizing protocols for real-time communications - between Web browsers, generally called "WebRTC" - [I-D.ietf-rtcweb-overview]. 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[RFC3261] soft phones) WebRTC communications - are directly controlled by some Web server. A simple case is shown - below. + The Real-Time Communications on the Web (RTCWEB) working group has + standardized protocols for real-time communications between Web + browsers, generally called "WebRTC" [I-D.ietf-rtcweb-overview]. 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 [RFC3261] soft + phones) WebRTC communications are directly controlled by some Web + server. A simple case is shown below. +----------------+ | | | Web Server | | | +----------------+ ^ ^ / \ HTTP / \ HTTP or / \ or WebSockets / \ WebSockets v v JS API JS API +-----------+ +-----------+ | | Media | | | Browser |<---------->| Browser | | | | | +-----------+ +-----------+ + Alice Bob Figure 1: A simple WebRTC system - In the system shown in Figure 1, Alice and Bob both have WebRTC + In the system shown in Figure 1, Alice and Bob both have WebRTC- enabled browsers and they visit some Web server which operates a calling service. Each of their browsers exposes standardized JavaScript calling APIs (implementated as browser built-ins) which are used by the Web server to set up a call between Alice and Bob. The Web server also serves as the signaling channel to transport control messages between the browsers. While this system is topologically similar to a conventional SIP-based system (with the Web server acting as the signaling service and browsers acting as softphones), control has moved to the central Web server; the browser simply provides API points that are used by the calling service. As @@ -159,22 +159,24 @@ denial of service attacks. Any successful system will need to be resistant to this and other attacks. A companion document [I-D.ietf-rtcweb-security-arch] describes a security architecture intended to address the issues raised in this document. 2. Terminology The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", - "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this - document are to be interpreted as described in RFC 2119 [RFC2119]. + "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and + "OPTIONAL" in this document are to be interpreted as described in BCP + 14 [RFC2119] [RFC8174] when, and only when, they appear in all + capitals, as shown here. 3. The Browser Threat Model The security requirements for WebRTC follow directly from the requirement that the browser's job is to protect the user. Huang et al. [huang-w2sp] summarize the core browser security guarantee as: Users can safely visit arbitrary web sites and execute scripts provided by those sites. @@ -203,49 +205,50 @@ NETWORK ATTACKERS, who are able to control your network. Network attackers correspond to the [RFC3552] "Internet Threat Model". Note that for non-HTTPS traffic, a network attacker is also a Web attacker, since it can inject traffic as if it were any non-HTTPS Web site. Thus, when analyzing HTTP connections, we must assume that traffic is going to the attacker. 3.1. Access to Local Resources While the browser has access to local resources such as keying - material, files, the camera and the microphone, it strictly limits or - forbids web servers from accessing those same resources. For + material, files, the camera, and the microphone, it strictly limits + or forbids web servers from accessing those same resources. For instance, while it is possible to produce an HTML form which will allow file upload, a script cannot do so without user consent and in fact cannot even suggest a specific file (e.g., /etc/passwd); the user must explicitly select the file and consent to its upload. [Note: in many cases browsers are explicitly designed to avoid - dialogs with the semantics of "click here to screw yourself", as - extensive research shows that users are prone to consent under such - circumstances.] + dialogs with the semantics of "click here to bypass security checks", + as extensive research shows that users are prone to consent under + such circumstances.] Similarly, while Flash programs (SWFs) [SWF] can access the camera and microphone, they explicitly require that the user consent to that access. In addition, some resources simply cannot be accessed from the browser at all. For instance, there is no real way to run specific executables directly from a script (though the user can of course be induced to download executable files and run them). -3.2. Same Origin Policy +3.2. Same-Origin Policy Many other resources are accessible but isolated. For instance, while scripts are allowed to make HTTP requests via the - XMLHttpRequest() API those requests are not allowed to be made to any - server, but rather solely to the same ORIGIN from whence the script - came [RFC6454] (although CORS [CORS] and WebSockets [RFC6455] provide - a escape hatch from this restriction, as described below.) This SAME - ORIGIN POLICY (SOP) prevents server A from mounting attacks on server - B via the user's browser, which protects both the user (e.g., from - misuse of his credentials) and the server B (e.g., from DoS attack). + XMLHttpRequest() API (see [XmlHttpRequest]) those requests are not + allowed to be made to any server, but rather solely to the same + ORIGIN from whence the script came [RFC6454] (although CORS [CORS] + and WebSockets [RFC6455] provide a escape hatch from this + restriction, as described below.) This SAME ORIGIN POLICY (SOP) + prevents server A from mounting attacks on server B via the user's + browser, which protects both the user (e.g., from misuse of his + credentials) and the server B (e.g., from DoS attack). More generally, SOP forces scripts from each site to run in their own, isolated, sandboxes. While there are techniques to allow them to interact, those interactions generally must be mutually consensual (by each site) and are limited to certain channels. For instance, multiple pages/browser panes from the same origin can read each other's JS variables, but pages from the different origins--or even iframes from different origins on the same page--cannot. 3.3. Bypassing SOP: CORS, WebSockets, and consent to communicate @@ -346,21 +349,21 @@ The most obvious threats are simply those of "oversharing". I.e., the user may believe they are sharing a window when in fact they are sharing an application, or may forget they are sharing their whole screen, icons, notifications, and all. This is already an issue with existing screen sharing technologies and is made somewhat worse if a partially trusted site is responsible for asking for the resource to be shared rather than having the user propose it. A less obvious threat involves the impact of screen sharing on the - Web security model. A key part of the Same Origin Policy is that + Web security model. A key part of the Same-Origin Policy is that HTML or JS from site A can reference content from site B and cause the browser to load it, but (unless explicitly permitted) cannot see the result. However, if a web application from a site is screen sharing the browser, then this violates that invariant, with serious security consequences. For example, an attacker site might request screen sharing and then briefly open up a new Window to the user's bank or webmail account, using screen sharing to read the resulting displayed content. A more sophisticated attack would be open up a source view window to a site and use the screen sharing result to view anti cross-site request forgery tokens. @@ -408,23 +411,25 @@ representative" windows that appear on many shopping sites. In this case, the user's expectation is that they are calling the site they're actually visiting. However, it is unlikely that they want to provide a general consent to such a site; just because I want some information on a car doesn't mean that I want the car manufacturer to be able to activate my microphone whenever they please. Thus, this suggests the need for a second consent mechanism where I only grant consent for the duration of a given call. As described in Section 3.1, great care must be taken in the design of this interface to avoid the users just clicking through. Note also that the user - interface chrome must clearly display elements showing that the call - is continuing in order to avoid attacks where the calling site just - leaves it up indefinitely but shows a Web UI that implies otherwise. + interface chrome, which is the representation through which the user + interacts with the user agent itself, must clearly display elements + showing that the call is continuing in order to avoid attacks where + the calling site just leaves it up indefinitely but shows a Web UI + that implies otherwise. 4.1.3. Origin-Based Security Now that we have seen another use case, we can start to reason about the security requirements. As discussed in Section 3.2, the basic unit of Web sandboxing is the origin, and so it is natural to scope consent to origin. Specifically, a script from origin A MUST only be allowed to initiate communications (and hence to access camera and microphone) if the @@ -460,24 +465,24 @@ cases. As discussed above, individual consent puts the user's approval in the UI flow for every call. Not only does this quickly become annoying but it can train the user to simply click "OK", at which point the consent becomes useless. Thus, while it may be necessary to have individual consent in some case, this is not a suitable solution for (for instance) the calling service case. Where necessary, in-flow user interfaces must be carefully designed to avoid the risk of the user blindly clicking through. The other two options are designed to restrict calls to a given - target. Callee-oriented consent provided by the calling site not - work well because a malicious site can claim that the user is calling - any user of his choice. One fix for this is to tie calls to a - cryptographically established identity. While not suitable for all + target. Callee-oriented consent provided by the calling site would + not work well because a malicious site can claim that the user is + calling any user of his choice. One fix for this is to tie calls to + a cryptographically-established identity. While not suitable for all cases, this approach may be useful for some. If we consider the case of advertising, it's not particularly convenient to require the advertiser to instantiate an iframe on the hosting site just to get permission; a more convenient approach is to cryptographically tie the advertiser's certificate to the communication directly. We're still tying permissions to origin here, but to the media origin (and- or destination) rather than to the Web origin. [I-D.ietf-rtcweb-security-arch] describes mechanisms which facilitate this sort of consent. @@ -512,29 +517,29 @@ Note that this attack does not depend on the media being insecure. Because the call is to the attacker, it is also encrypted to him. Moreover, it need not be executed immediately; the attacker can "infect" the origin semi-permanently (e.g., with a web worker or a popped-up window that is hidden under the main window.) and thus be able to bug me long after I have left the infected network. This risk is created by allowing calls at all from a page fetched over HTTP. - Even if calls are only possible from HTTPS [RFC2818] sites, if the - site embeds active content (e.g., JavaScript) that is fetched over - HTTP or from an untrusted site, because that JavaScript is executed - in the security context of the page [finer-grained]. Thus, it is - also dangerous to allow WebRTC functionality from HTTPS origins that - embed mixed content. Note: this issue is not restricted to PAGES - which contain mixed content. If a page from a given origin ever - loads mixed content then it is possible for a network attacker to - infect the browser's notion of that origin semi-permanently. + Even if calls are only possible from HTTPS [RFC2818] sites, if those + sites include active content (e.g., JavaScript) from an untrusted + site, that JavaScript is executed in the security context of the page + [finer-grained]. This could lead to compromise of a call even if the + parent page is safe. Note: this issue is not restricted to PAGES + which contain untrusted content. If a page from a given origin ever + loads JavaScript from an attacker, then it is possible for that + attacker to infect the browser's notion of that origin semi- + permanently. 4.2. Communications Consent Verification As discussed in Section 3.3, allowing web applications unrestricted network access via the browser introduces the risk of using the browser as an attack platform against machines which would not otherwise be accessible to the malicious site, for instance because they are topologically restricted (e.g., behind a firewall or NAT). In order to prevent this form of attack as well as cross-protocol attacks it is important to require that the target of traffic @@ -551,30 +556,29 @@ matter there are a large number of Web sites which can act as data sources, so an attacker can at least use downlink bandwidth with existing Web APIs. However, this potential DoS vector reinforces the need for adequate congestion control for WebRTC protocols to ensure that they play fair with other demands on the user's bandwidth. 4.2.1. ICE Verifying receiver consent requires some sort of explicit handshake, but conveniently we already need one in order to do NAT hole- - punching. ICE [RFC5245] includes a handshake designed to verify that + punching. ICE [RFC8445] includes a handshake designed to verify that the receiving element wishes to receive traffic from the sender. It is important to remember here that the site initiating ICE is presumed malicious; in order for the handshake to be secure the receiving element MUST demonstrate receipt/knowledge of some value not available to the site (thus preventing the site from forging responses). In order to achieve this objective with ICE, the STUN transaction IDs must be generated by the browser and MUST NOT be made available to the initiating script, even via a diagnostic interface. - Verifying receiver consent also requires verifying the receiver wants to receive traffic from a particular sender, and at this time; for example a malicious site may simply attempt ICE to known servers that are using ICE for other sessions. ICE provides this verification as well, by using the STUN credentials as a form of per-session shared secret. Those credentials are known to the Web application, but would need to also be known and used by the STUN-receiving element to be useful. There also needs to be some mechanism for the browser to verify that @@ -613,21 +617,21 @@ The message/reply pair must be generated in such a way that an attacker who controls the Web application cannot forge them, generally by having the message contain some secret value that must be incorporated (e.g., echoed, hashed into, etc.). Non-ICE candidates for this role (in cases where the legacy endpoint has a public address) include: o STUN checks without using ICE (i.e., the non-RTC-web endpoint sets up a STUN responder.) - o Use or RTCP as an implicit reachability check. + o Use of RTCP as an implicit reachability check. In the RTCP approach, the WebRTC endpoint is allowed to send a limited number of RTP packets prior to receiving consent. This allows a short window of attack. In addition, some legacy endpoints do not support RTCP, so this is a much more expensive solution for such endpoints, for which it would likely be easier to implement ICE. For these two reasons, an RTCP-based approach does not seem to address the security issue satisfactorily. In the STUN approach, the WebRTC endpoint is able to verify that the @@ -658,23 +662,22 @@ those for initial consent, though are perhaps weaker since the threats is less severe. 4.2.4. IP Location Privacy Note that as soon as the callee sends their ICE candidates, the caller learns the callee's IP addresses. The callee's server reflexive address reveals a lot of information about the callee's location. In order to avoid tracking, implementations may wish to suppress the start of ICE negotiation until the callee has answered. - - In addition, either side may wish to hide their location entirely by - forcing all traffic through a TURN server. + In addition, either side may wish to hide their location from the + other side entirely by forcing all traffic through a TURN server. In ordinary operation, the site learns the browser's IP address, though it may be hidden via mechanisms like Tor [http://www.torproject.org] or a VPN. However, because sites can cause the browser to provide IP addresses, this provides a mechanism for sites to learn about the user's network environment even if the user is behind a VPN that masks their IP address. Implementations may wish to provide settings which suppress all non-VPN candidates if the user is on certain kinds of VPN, especially privacy-oriented systems such as Tor. @@ -683,21 +686,21 @@ Finally, we consider a problem familiar from the SIP world: communications security. For obvious reasons, it MUST be possible for the communicating parties to establish a channel which is secure against both message recovery and message modification. (See [RFC5479] for more details.) This service must be provided for both data and voice/video. Ideally the same security mechanisms would be used for both types of content. Technology for providing this service (for instance, SRTP [RFC3711], DTLS [RFC6347] and DTLS-SRTP [RFC5763]) is well understood. However, we must examine this - technology to the WebRTC context, where the threat model is somewhat + technology in the WebRTC context, where the threat model is somewhat different. In general, it is important to understand that unlike a conventional SIP proxy, the calling service (i.e., the Web server) controls not only the channel between the communicating endpoints but also the application running on the user's browser. While in principle it is possible for the browser to cut the calling service out of the loop and directly present trusted information (and perhaps get consent), practice in modern browsers is to avoid this whenever possible. "In- flow" modal dialogs which require the user to consent to specific @@ -721,25 +724,25 @@ During-call attack by calling service. The calling service is compromised during the call it wishes to attack (often called an "active attack"). Providing security against the former type of attack is practical using the techniques discussed in Section 4.3.1. However, it is extremely difficult to prevent a trusted but malicious calling service from actively attacking a user's calls, either by mounting a - MITM attack or by diverting them entirely. (Note that this attack - applies equally to a network attacker if communications to the - calling service are not secured.) We discuss some potential - approaches and why they are likely to be impractical in - Section 4.3.2. + Man-in-the-Middle (MITM) attack or by diverting them entirely. (Note + that this attack applies equally to a network attacker if + communications to the calling service are not secured.) We discuss + some potential approaches and why they are likely to be impractical + in Section 4.3.2. 4.3.1. Protecting Against Retrospective Compromise In a retrospective attack, the calling service was uncompromised during the call, but that an attacker subsequently wants to recover the content of the call. We assume that the attacker has access to the protected media stream as well as having full control of the calling service. If the calling service has access to the traffic keying material (as @@ -800,21 +803,21 @@ [cranor-wolf], it seems extremely unlikely that any key continuity mechanism will be effective rather than simply annoying. Moreover, it is trivial to bypass even this kind of mechanism. Recall that unlike the case of SSH, the browser never directly gets the peer's identity from the user. Rather, it is provided by the calling service. Even enabling a mechanism of this type would require an API to allow the calling service to tell the browser "this is a call to user X". All the calling service needs to do to avoid triggering a key continuity warning is to tell the browser that "this - is a call to user Y" where Y is close to X. Even if the user + is a call to user Y" where Y is confusable with X. Even if the user actually checks the other side's name (which all available evidence indicates is unlikely), this would require (a) the browser to trusted UI to provide the name and (b) the user to not be fooled by similar appearing names. 4.3.2.2. Short Authentication Strings ZRTP [RFC6189] uses a "short authentication string" (SAS) which is derived from the key agreement protocol. This SAS is designed to be compared by the users (e.g., read aloud over the the voice channel or @@ -853,31 +856,31 @@ simply ignore such indicators even in the rather more dire case of mixed content warnings. 4.3.2.3. Third Party Identity The conventional approach to providing communications identity has of course been to have some third party identity system (e.g., PKI) to authenticate the endpoints. Such mechanisms have proven to be too cumbersome for use by typical users (and nearly too cumbersome for administrators). However, a new generation of Web-based identity - providers (BrowserID, Federated Google Login, Facebook Connect, - OAuth, OpenID, WebFinger), has recently been developed and use Web - technologies to provide lightweight (from the user's perspective) - third-party authenticated transactions. It is possible to use - systems of this type to authenticate WebRTC calls, linking them to - existing user notions of identity (e.g., Facebook adjacencies). - Specifically, the third-party identity system is used to bind the - user's identity to cryptographic keying material which is then used - to authenticate the calling endpoints. Calls which are authenticated - in this fashion are naturally resistant even to active MITM attack by - the calling site. + providers (BrowserID, Federated Google Login, Facebook Connect, OAuth + [RFC6749], OpenID [OpenID], WebFinger [RFC7033]), has recently been + developed and use Web technologies to provide lightweight (from the + user's perspective) third-party authenticated transactions. It is + possible to use systems of this type to authenticate WebRTC calls, + linking them to existing user notions of identity (e.g., Facebook + adjacencies). Specifically, the third-party identity system is used + to bind the user's identity to cryptographic keying material which is + then used to authenticate the calling endpoints. Calls which are + authenticated in this fashion are naturally resistant even to active + MITM attack by the calling site. Note that there is one special case in which PKI-style certificates do provide a practical solution: calls from end-users to large sites. For instance, if you are making a call to Amazon.com, then Amazon can easily get a certificate to authenticate their media traffic, just as they get one to authenticate their Web traffic. This does not provide additional security value in cases in which the calling site and the media peer are one in the same, but might be useful in cases in which third parties (e.g., ad networks or retailers) arrange for calls but do not participate in them. @@ -890,24 +893,25 @@ manipulated by the calling site. Obviously, if the site can modify or view the media, then the user is not getting the level of assurance they would expect from being able to authenticate their peer. In many cases, this is acceptable because the user values site-based special effects over complete security from the site. However, there are also cases where users wish to know that the site cannot interfere. In order to facilitate that, it will be necessary to provide features whereby the site can verifiably give up access to the media streams. This verification must be possible both from the local side and the remote side. I.e., I must be able to verify that - the person I am calling has engaged a secure media mode. In order to - achieve this it will be necessary to cryptographically bind an - indication of the local media access policy into the cryptographic - authentication procedures detailed in the previous sections. + the person I am calling has engaged a secure media mode (see + Section 4.3.3). In order to achieve this it will be necessary to + cryptographically bind an indication of the local media access policy + into the cryptographic authentication procedures detailed in the + previous sections. 4.3.3. Malicious Peers One class of attack that we do not generally try to prevent is malicious peers. For instance, no matter what confidentiality measures you employ the person you are talking to might record the call and publish it on the Internet. Similarly, we do not attempt to prevent them from using voice or video processing technology from hiding or changing their appearance. While technologies (DRM, etc.) do exist to attempt to address these issues, they are generally not @@ -917,39 +921,40 @@ unwanted calls. In general, this is in the scope of the calling site, though because WebRTC does offer some forms of strong authentication, that may be useful as part of a defense against such attacks. 4.4. Privacy Considerations 4.4.1. Correlation of Anonymous Calls While persistent endpoint identifiers can be a useful security - feature (see Section 4.3.2.1 they can also represent a privacy threat - in settings where the user wishes to be anonymous. WebRTC provides a - number of possible persistent identifiers such as DTLS certificates - (if they are reused between connections) and RTCP CNAMES (if - generated according to [RFC6222] rather than the privacy preserving - mode of [RFC7022]). In order to prevent this type of correlation, - browsers need to provide mechanisms to reset these identifiers (e.g., - with the same lifetime as cookies). Moreover, the API should provide - mechanisms to allow sites intended for anonymous calling to force the - minting of fresh identifiers. In addition, IP addresses can be a - source of call linkage [I-D.ietf-rtcweb-ip-handling] + feature (see Section 4.3.2.1) they can also represent a privacy + threat in settings where the user wishes to be anonymous. WebRTC + provides a number of possible persistent identifiers such as DTLS + certificates (if they are reused between connections) and RTCP CNAMES + (if generated according to [RFC6222] rather than the privacy + preserving mode of [RFC7022]). In order to prevent this type of + correlation, browsers need to provide mechanisms to reset these + identifiers (e.g., with the same lifetime as cookies). Moreover, the + API should provide mechanisms to allow sites intended for anonymous + calling to force the minting of fresh identifiers. In addition, IP + addresses can be a source of call linkage + [I-D.ietf-rtcweb-ip-handling] 4.4.2. Browser Fingerprinting Any new set of API features adds a risk of browser fingerprinting, and WebRTC is no exception. Specifically, sites can use the presence or absence of specific devices as a browser fingerprint. In general, the API needs to be balanced between functionality and the - incremental fingerprint risk. + incremental fingerprint risk. See [Fingerprinting] 5. Security Considerations This entire document is about security. 6. Acknowledgements Bernard Aboba, Harald Alvestrand, Dan Druta, Cullen Jennings, Alan Johnston, Hadriel Kaplan (S 4.2.1), Matthew Kaufman, Martin Thomson, Magnus Westerlund. @@ -981,22 +986,26 @@ o Added a section describing screen sharing threats. o Assorted editorial changes. 9. References 9.1. Normative References [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, - DOI 10.17487/RFC2119, March 1997, . + DOI 10.17487/RFC2119, March 1997, + . + + [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC + 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, + May 2017, . 9.2. Informative References [abarth-rtcweb] Barth, A., "Prompting the user is security failure", RTC- Web Workshop, September 2010. [CORS] van Kesteren, A., "Cross-Origin Resource Sharing", January 2014. @@ -1007,85 +1016,87 @@ Symposium, 2009, August 2009. [farus-conversion] Farrus, M., Erro, D., and J. Hernando, "Speaker Recognition Robustness to Voice Conversion", January 2008. [finer-grained] Barth, A. and C. Jackson, "Beware of Finer-Grained Origins", W2SP, 2008, July 2008. + [Fingerprinting] + W3C, "Fingerprinting Guidance for Web Specification + Authors (Draft)", November 2013. + [huang-w2sp] Huang, L-S., Chen, E., Barth, A., Rescorla, E., and C. Jackson, "Talking to Yourself for Fun and Profit", W2SP, 2011, May 2011. [I-D.ietf-rtcweb-ip-handling] - Uberti, J. and G. Shieh, "WebRTC IP Address Handling - Requirements", draft-ietf-rtcweb-ip-handling-04 (work in - progress), July 2017. + Uberti, J., "WebRTC IP Address Handling Requirements", + draft-ietf-rtcweb-ip-handling-11 (work in progress), + November 2018. [I-D.ietf-rtcweb-overview] Alvestrand, H., "Overview: Real Time Protocols for Browser-based Applications", draft-ietf-rtcweb-overview-19 (work in progress), November 2017. [I-D.ietf-rtcweb-security-arch] Rescorla, E., "WebRTC Security Architecture", draft-ietf- - rtcweb-security-arch-13 (work in progress), October 2017. + rtcweb-security-arch-17 (work in progress), November 2018. [kain-conversion] Kain, A. and M. Macon, "Design and Evaluation of a Voice Conversion Algorithm based on Spectral Envelope Mapping and Residual Prediction", Proceedings of ICASSP, May 2001, May 2001. + [OpenID] Sakimura, N., Bradley, J., Jones, M., de Medeiros, B., and + C. Mortimore, "Fingerprinting Guidance for Web + Specification Authors (Draft)", November 2014. + [RFC2818] Rescorla, E., "HTTP Over TLS", RFC 2818, - DOI 10.17487/RFC2818, May 2000, . + DOI 10.17487/RFC2818, May 2000, + . [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, - DOI 10.17487/RFC3261, June 2002, . + DOI 10.17487/RFC3261, June 2002, + . [RFC3552] Rescorla, E. and B. Korver, "Guidelines for Writing RFC Text on Security Considerations", BCP 72, RFC 3552, - DOI 10.17487/RFC3552, July 2003, . + DOI 10.17487/RFC3552, July 2003, + . [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, DOI 10.17487/RFC3711, March 2004, . [RFC3760] Gustafson, D., Just, M., and M. Nystrom, "Securely Available Credentials (SACRED) - Credential Server Framework", RFC 3760, DOI 10.17487/RFC3760, April 2004, . [RFC4251] Ylonen, T. and C. Lonvick, Ed., "The Secure Shell (SSH) Protocol Architecture", RFC 4251, DOI 10.17487/RFC4251, January 2006, . [RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session Description Protocol (SDP) Security Descriptions for Media Streams", RFC 4568, DOI 10.17487/RFC4568, July 2006, . - [RFC5245] Rosenberg, J., "Interactive Connectivity Establishment - (ICE): A Protocol for Network Address Translator (NAT) - Traversal for Offer/Answer Protocols", RFC 5245, - DOI 10.17487/RFC5245, April 2010, . - [RFC5479] Wing, D., Ed., Fries, S., Tschofenig, H., and F. Audet, "Requirements and Analysis of Media Security Management Protocols", RFC 5479, DOI 10.17487/RFC5479, April 2009, . [RFC5763] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework for Establishing a Secure Real-time Transport Protocol (SRTP) Security Context Using Datagram Transport Layer Security (DTLS)", RFC 5763, DOI 10.17487/RFC5763, May 2010, . @@ -1098,44 +1109,62 @@ [RFC6222] Begen, A., Perkins, C., and D. Wing, "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)", RFC 6222, DOI 10.17487/RFC6222, April 2011, . [RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347, January 2012, . [RFC6454] Barth, A., "The Web Origin Concept", RFC 6454, - DOI 10.17487/RFC6454, December 2011, . + DOI 10.17487/RFC6454, December 2011, + . [RFC6455] Fette, I. and A. Melnikov, "The WebSocket Protocol", RFC 6455, DOI 10.17487/RFC6455, December 2011, . + [RFC6749] Hardt, D., Ed., "The OAuth 2.0 Authorization Framework", + RFC 6749, DOI 10.17487/RFC6749, October 2012, + . + [RFC7022] Begen, A., Perkins, C., Wing, D., and E. Rescorla, "Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022, September 2013, . + [RFC7033] Jones, P., Salgueiro, G., Jones, M., and J. Smarr, + "WebFinger", RFC 7033, DOI 10.17487/RFC7033, September + 2013, . + [RFC7675] Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M. Thomson, "Session Traversal Utilities for NAT (STUN) Usage for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675, October 2015, . + [RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive + Connectivity Establishment (ICE): A Protocol for Network + Address Translator (NAT) Traversal", RFC 8445, + DOI 10.17487/RFC8445, July 2018, + . + [SWF] Adobe, "SWF File Format Specification Version 19", April 2013. [whitten-johnny] Whitten, A. and J. Tygar, "Why Johnny Can't Encrypt: A Usability Evaluation of PGP 5.0", Proceedings of the 8th USENIX Security Symposium, 1999, August 1999. + [XmlHttpRequest] + van Kesteren, A., "XMLHttpRequesti Level 2", January 2012. + Author's Address + Eric Rescorla RTFM, Inc. 2064 Edgewood Drive Palo Alto, CA 94303 USA Phone: +1 650 678 2350 Email: ekr@rtfm.com