Internet DRAFT - draft-muthu-behave-consent-freshness
draft-muthu-behave-consent-freshness
Behave Muthu. Perumal
Internet-Draft D. Wing
Intended status: Standards Track R. Ravindranath
Expires: January 16, 2014 T. Reddy
Cisco Systems
July 15, 2013
STUN Usage for Consent Freshness
draft-muthu-behave-consent-freshness-04
Abstract
Verification of peer consent before sending traffic is necessary in
WebRTC deployments to ensure that a malicious JavaScript cannot use
the browser as a platform for launching attacks. A related problem
is session liveness. WebRTC application may want to detect
connection failure and take appropriate action.
This document describes how a WebRTC browser can verify peer consent
to continue sending traffic and detect connection failure.
Status of This Memo
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This Internet-Draft will expire on January 16, 2014.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 2
3. Design Considerations . . . . . . . . . . . . . . . . . . . . 3
4. Solution Overview . . . . . . . . . . . . . . . . . . . . . . 4
5. W3C API Implications . . . . . . . . . . . . . . . . . . . . 5
6. Interaction with Keepalives used for Refreshing NAT Bindings 5
7. Security Considerations . . . . . . . . . . . . . . . . . . . 5
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 6
9. Acknowledgement . . . . . . . . . . . . . . . . . . . . . . . 6
10. Normative References . . . . . . . . . . . . . . . . . . . . 6
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 6
1. Introduction
To prevent attacks on WebRTC peers, WebRTC browsers have to ensure
the remote peer wants to receive traffic. This is performed both
when the session is first established to the remote peer (using ICE
connectivity checks), and periodically when for the duration of the
session (using the procedure defined in this document).
When a session is first established, WebRTC implementations are
required to perform STUN connectivity checks as part of ICE
[RFC5245]. That initial consent is not described further in this
document.
Related to consent is loss of connectivity ("liveness"). WebRTC
applications want notification of connection failure to take
appropriate actions (e.g., alert the user, try switching to a
different interface).
This document describes a new STUN usage with a request and response
which verifies the remote peer consents to receive traffic, and
detects loss of liveness. To meet the security needs of consent, the
JavaScript application has no control over the consent requests or
the requirement to receive a consent response. However, the
JavaScript does get notification of consent failure and loss of
connectivity.
2. Terminology
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The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119].
Consent: It is the mechanism of obtaining permission from the peer
to send traffic on a candidate pair.
Consent Freshness: It is the mechanism of obtaining permission from
the peer to continue sending traffic on a nominated candidate pair
after ICE has concluded.
Session Liveness: It is the mechanism of detecting connectivity on a
nominated candidate pair after ICE has concluded.
Transport Address: The combination of an IP address and port number
(such as a UDP or TCP port number).
3. Design Considerations
Although ICE requires periodic keepalive traffic to be sent in order
to keep NAT bindings alive (Section 10 of [RFC5245], [RFC6263]),
those keepalives are send-and-forget, and do not evoke a response. A
response is necessary both for consent to continue sending traffic,
as well as to ensure connectivity. Thus, we need a request/response
mechanism.
Though ICE specifies STUN Binding indications to be used for
keepalives, it requires that an agent be prepared to receive
connectivity check as well. If a connectivity check is received, a
response is generated, but there is no impact on ICE processing, as
described in section 10 of [RFC5245].
While a WebRTC browser could verify whether the peer continues to
send SRTCP reports before sending traffic to the peer, the usage of
SRTCP together with Security Descriptions [RFC4568] requires exposing
the media keys to the JavaScript and renders SRTCP unsuitable for
consent freshness.
For consent, calculating the SHA1 HMAC is necessary for MESSAGE-
INTEGRITY which is computationally expensive. Security analysis
concluded that the STUN 96 bits transaction ID is sufficient for
consent, without needing MESSAGE-INTEGRITY. However, omitting the
MESSAGE-INTEGRITY attribute from STUN Binding request/response to
avoid the cost of computing SHA1 would make browsers incapable of
verifying consent freshness with legacy ICE/ICE-lite implementations,
resulting in backward compatibility issues.
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The above considerations suggest that STUN Binding request/response
is most suitable for performing consent freshness.
4. Solution Overview
Consent freshness serves as a circuit breaker (so that if the path or
remote peer fails the WebRTC browser stops sending all traffic on
that remote peer), determining session liveness serves the purpose of
notifying the application of connectivity failure so that the
application can take appropriate action.
The solution uses three values:
1. A consent timer, Tc, whose value is determined by the browser.
This value MUST be 15 seconds.
2. A packet receipt timer, Tr, whose value is determined by the
application, but MUST NOT be shorter than 1 second or longer than
15 seconds, and SHOULD have a default value of 5 seconds.
3. A consent timeout, Tf, which is how many seconds elapse without a
consent response before the browser ceases transmission of media.
Its value MUST be 15 seconds or less, and the value 15 seconds is
RECOMMENDED.
A WebRTC browser performs a combined consent freshness and session
liveness test using STUN request/response as described below:
Every Tc seconds, the WebRTC browser sends a STUN Binding Request to
the peer. This request MUST use a new, cryptographically random
Transaction ID [RFC4086], and is formatted as for an ICE connectivity
check [RFC5245]. A valid STUN Binding Response is also formatted as
for an ICE connectivity check [RFC5245]. The STUN Binding Request
and STUN Binding Response are validated as for an ICE connectivity
check [RFC5245].
If a valid STUN Binding Response is received, the consent timer is
reset and fires again Tc seconds later.
If a valid STUN Binding Response is not received after 500ms, the
STUN Binding Request is retransmitted (with the same Transaction ID
and all other fields). As long as a valid STUN Binding Response is
not received, this retransmission is repeated every 500ms until Tf
seconds have elapsed or a valid response is received. If no valid
response is received after Tf seconds, the WebRTC browser MUST quit
transmitting traffic to this remote peer. Considering the default
value of Tf=15 seconds, this means transmission will stop after 30
consent check packets have resulted in no response.
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Liveness timer: If no packets have been received on the local port in
Tr seconds, the WebRTC browser MUST inform the JavaScript that
connectivity has been lost. The JavaScript application will use this
notification however it desires (e.g., cease transmitting to the
remote peer, provide a notification to the user, etc.). Note the
definition of a received packet is liberal, and includes an SRTP
packet that fails authentication, a STUN Binding Request with an
invalid USERNAME or PASSWORD, or any other packet.
5. W3C API Implications
For the consent freshness and liveness test the W3C specification
should provide APIs as described below:
1. Ability for the browser to notify the JavaScript that a consent
freshness transaction has failed for a media stream and the
browser has stopped transmitting for that stream.
2. Ability for the JavaScript to start and stop liveness test and
set the liveness test interval.
3. Ability for the browser to notify the JavaScript that a liveness
test has failed for a media stream.
6. Interaction with Keepalives used for Refreshing NAT Bindings
When not actively sending traffic on a nominated candidate pair,
performing consent freshness does not serve any purpose from a
security perspective. If consent freshness is not performed during
this period, the browser continues to performs the ICE keepalives
[RFC5245] or RTP keepalive [RFC6263] to refresh NAT bindings.
7. Security Considerations
Security considerations discussed in [RFC5245] are to be taken into
account.
In ICE [RFC5245] the STUN request/response are protected with
MESSAGE-INTEGRITY, using an ephemeral username and password exchanged
in the SDP ice-ufrag and ice-pwd attributes. This prevents ICE from
accidentally connecting to an in-intended peer, in that ICE will only
connect to a peer that also knows the same username and password
(exchanged in call signaling). Once that connection to the remote
peer has been established with ICE, the consent to continue sending
traffic does not benefit from re-asserting that same username and
password, so long as the senders and receiver's IP addresses remain
the same (as they usually do).
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8. IANA Considerations
This document does not require any action from IANA.
9. Acknowledgement
Thanks to Eric Rescorla, Harald Alvestrand, Martin Thomson, Bernard
Aboba, Cullen Jennings and Simon Perreault for their valuable inputs
and comments.
10. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC4086] Eastlake, D., Schiller, J., and S. Crocker, "Randomness
Requirements for Security", BCP 106, RFC 4086, June 2005.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, April
2010.
[RFC6263] Marjou, X. and A. Sollaud, "Application Mechanism for
Keeping Alive the NAT Mappings Associated with RTP / RTP
Control Protocol (RTCP) Flows", RFC 6263, June 2011.
Authors' Addresses
Muthu Arul Mozhi Perumal
Cisco Systems
Cessna Business Park
Sarjapur-Marathahalli Outer Ring Road
Bangalore, Karnataka 560103
India
Email: mperumal@cisco.com
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Dan Wing
Cisco Systems
821 Alder Drive
Milpitas, California 95035
USA
Email: dwing@cisco.com
Ram Mohan Ravindranath
Cisco Systems
Cessna Business Park
Sarjapur-Marathahalli Outer Ring Road
Bangalore, Karnataka 560103
India
Email: rmohanr@cisco.com
Tirumaleswar Reddy
Cisco Systems
Cessna Business Park, Varthur Hobli
Sarjapur Marathalli Outer Ring Road
Bangalore, Karnataka 560103
India
Email: tireddy@cisco.com
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