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draft-ietf-rtcweb-stun-consent-freshness
Behave Muthu A M. Perumal
Internet-Draft D. Wing
Intended status: Standards Track Ram Mohan. Ravindranath
Expires: August 29, 2013 Cisco Systems
H. Kaplan
Acme Packet
February 25, 2013
STUN Usage for Consent Freshness
draft-muthu-behave-consent-freshness-03
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 applications may want to detect
connection failure and take appropriate actions. This document
describes a STUN usage that enables a WebRTC browser to perform the
following on a candidate pair ICE is using for a media component
after session establishment:
1. Verify the peer consent for continuing to send traffic.
2. Dectect connection failure and notify the JavaScript.
This also serves the purpose of refreshing NAT bindings.
Status of this Memo
This Internet-Draft is submitted to IETF 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/.
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 August 29, 2013.
Copyright Notice
Copyright (c) 2013 IETF Trust and the persons identified as the
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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
publication of this document. Please review these documents
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described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . . 3
4. Design Considerations . . . . . . . . . . . . . . . . . . . . . 4
5. Solution Overview . . . . . . . . . . . . . . . . . . . . . . . 4
6. W3C API Implications . . . . . . . . . . . . . . . . . . . . . 5
7. Interaction with Keepalives used for Refreshing NAT
Bindings . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
8. Security Considerations . . . . . . . . . . . . . . . . . . . . 6
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 6
10. Acknowledgement . . . . . . . . . . . . . . . . . . . . . . . . 6
11. Normative References . . . . . . . . . . . . . . . . . . . . . 6
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 6
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1. Introduction
WebRTC implementations obtain the peer consent before sending traffic
on candidate media transport addresses. This has two parts:
1. Obtaining peer consent for sending traffic at session
establishment.
2. Obtaining peer consent for continuing to send traffic after
session establishment.
WebRTC implements are required to perform STUN connectivity checks at
session establishment as part of ICE procedures [RFC5245]. This
takes care of the first part of the consent verification described
above.
After session establishment ICE requires STUN Binding indications to
be used for refreshing NAT bindings for a candidate pair ICE is using
for a media component. Since a STUN Binding indication does not
evoke a response, it cannot be used for the second part of the
consent verification described above.
A related problem is session liveness. WebRTC applications may want
to detect connection failure on candidate media transport addresses
after session establishment and take appropriate actions. Again, the
STUN Binding indications in ICE sent after session establishment
cannot be used for determining session liveness.
This document describes a STUN usage based on STUN request/response
that enables a WebRTC browser to perform the following on a candidate
pair ICE is using for a media component after session establishment:
1. Verify the peer consent for continuing to send traffic.
2. Dectect connection failure and notify the JavaScript.
This also serves the purpose of refreshing NAT bindings.
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 [RFC2119].
3. Definitions
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Consent: It is the mechanism of obtaining permission to send traffic
on a candidate pair.
Consent Freshness: It is the mechanism of obtaining permission to
continue sending traffic on a candidate pair ICE is using for a
media component after ICE has concluded.
Session Liveness: It is the mechanism of detecting connectivity on a
candidate pair ICE is using for a media component after ICE has
concluded.
Transport Address: The combination of an IP address and port number
(such as a UDP or TCP port number).
4. Design Considerations
As described earlier, STUN indications are not suitable for
performing consent freshness. Hence, performing consent freshness
requires the use of STUN request/response.
STUN requires the 96 bits transaction ID to be uniformly and randomly
chosen from the interval 0 .. 2**96-1, and be cryptographically
random. This is deemed sufficient for consent freshness from a
security perspective. However, omitting the MESSAGE-INTEGRITY
attribute from STUN Binding request/response to avoid the cost of
computing SHA1 would break backward compatibility with ICE/ICE-lite
agents.
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 SDESC [RFC4568] exposes the media keys to the
JavaScript and renders SRTCP unsuitable for consent freshness.
The above considerations suggest that STUN Binding request/response
is most suitable for performing consent freshness.
5. Solution Overview
The solution uses two timers:
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1. A consent timer Tc whose value is determined by the browser.
2. A packet receipt timer Tr whose value is determined by the
application.
A WebRTC browser performs a combined consent freshness and session
liveness test using STUN resuest/respose as described below:
o Starts a consent timer Tc (no less than 15 sec).
o Starts a packet receipt timer Tr (no less than 1 sec); application
configurable.
o When either timer expires it starts a STUN transaction.
o When the STUN transaction succeeds, it re-starts both timers.
o When the STUN transaction fails
* If the transaction was started by timer Tc, it stops sending
traffic on that candidate pair.
* Else it notifies the application of the failure and continues.
o It resets timer Tr on receiving any packet from the other side.
While consent freshness serves as a circuit breaker (if there is a
failure the WebRTC browser stops sending all traffic on that
candidate pair), determining session liveness serves the purpose of
notifying the application of connectivity failure so that the
application can take appropriate action.
6. 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 steam and the browser has
stopped transmitting for that stream.
2. Ability for the JavaScript to set the liveness test interval.
3. Ability for the browser to notify the JavaScript that a liveness
test has failed for a steam.
7. Interaction with Keepalives used for Refreshing NAT Bindings
An implementation that performs the procedures described in this
document has no need to also perform the keepalives described in ICE
[RFC5245] or RTP keepalive [RFC6263], as they both force recurring
messages to be sent over the UDP port used by RTP. Thus, an
implementation that performs the procedures described in this
document SHOULD NOT also do the keepalives described in ICE [RFC5245]
or RTP keepalives [RFC6263] for the same UDP port.
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8. Security Considerations
TBD
9. IANA Considerations
TBD
10. Acknowledgement
Thanks to Eric Rescorla, Harald Alvestrand, Martin Thomson, Bernard
Aboba, Cullen Jennings and Simon Perreault for their valuable inputs
and comments
11. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[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.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006.
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
Hadriel Kaplan
Acme Packet
Email: hkaplan@acmepacket.com
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