RTCWEB M. Perumal
Internet-Draft D. Wing
Intended status: Standards Track R. Ravindranath
Expires: October 13, 2014 T. Reddy
Cisco Systems
M. Thomson
Mozilla
April 11, 2014
STUN Usage for Consent Freshness
draft-ietf-rtcweb-stun-consent-freshness-02
Abstract
To prevent sending excessive traffic to an endpoint, periodic consent
needs to be obtained from that remote endpoint.
This document describes a consent mechanism using a new STUN usage.
This same mechanism can also determine connection loss ("liveness")
with a remote peer.
Status of This Memo
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This Internet-Draft will expire on October 13, 2014.
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document authors. All rights reserved.
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carefully, as they describe your rights and restrictions with respect
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Design Considerations . . . . . . . . . . . . . . . . . . . . 3
4. Solution Overview . . . . . . . . . . . . . . . . . . . . . . 3
5. Connection Liveness . . . . . . . . . . . . . . . . . . . . . 4
6. DiffServ Treatment for Consent packets . . . . . . . . . . . 5
7. W3C API Implications . . . . . . . . . . . . . . . . . . . . 5
8. Security Considerations . . . . . . . . . . . . . . . . . . . 5
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 6
10. Acknowledgement . . . . . . . . . . . . . . . . . . . . . . . 6
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 6
11.1. Normative References . . . . . . . . . . . . . . . . . . 6
11.2. Informative References . . . . . . . . . . . . . . . . . 6
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 7
1. Introduction
To prevent attacks on peers, RTP endpoints 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 for the duration of the session
using the procedures 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 and it is assumed that ICE is being used for that initial
consent.
Related to consent is loss of connectivity ("liveness"). Many
applications want notification of connection loss 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
messages which verifies the remote peer's consent to receive traffic,
and can also detect loss of liveness.
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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].
Consent: It is the mechanism of obtaining permission to send traffic
to a certain transport address. This is usually obtained via ICE.
Consent Freshness: Permission to continue sending traffic to a
certain transport address. This is performed by the procedure
described in this document.
Session Liveness: Detecting loss of connectivity to a certain
transport address. This is performed by the procedure described
in this document.
Transport Address: The remote peer's IP address and (UDP or TCP)
port number.
3. Design Considerations
Although ICE requires periodic keepalive traffic to keep NAT bindings
alive (Section 10 of [RFC5245], [RFC6263]), those keepalives are sent
as STUN Indications which 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 verify session liveness. Thus, we
need a request/response mechanism for consent freshness. ICE can be
used for that mechanism because ICE already requires ICE agents
continue listening for ICE messages, as described in section 10 of
[RFC5245].
4. Solution Overview
A WebRTC browser performs a combined consent freshness and session
liveness test using STUN request/response as described below:
An endpoint MUST NOT send application data (in WebRTC this means RTP
or SCTP data) on an ICE-initiated connection unless the receiving
endpoint consents to receive the data. After a successful ICE
connectivity check on a particular transport address, subsequent
consent MUST be obtained following the procedure described in this
document. The consent expires after a fixed amount of time.
Explicit consent to send is indicated by:
1. Sending an ICE binding request to the remote peer's Transport
Address and receiving a matching and authenticated ICE binding
response from the inverted remote peer's Transport Address.
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These ICE binding request/response are authenticated using the
same short- term credentials as the initial ICE exchange, but
using a new (fresh) transaction-id each time consent needs to be
refresh. Implementations MUST obtain fresh consent before their
existing consent expires. When obtaining fresh consent a STUN
connectivity check (or response) could be lost, and re-
transmissions MUST use the same STUN transaction-id, and re-
transmissions MUST NOT be sent more frequently than every 500ms
or the smoothed round-trip time (from previous consent freshness
checks or RTP round-trip time), whichever is less. For the
purposes of this document, receipt of an ICE response with the
matching transaction-id of its request with a valid MESSAGE-
INTEGRITY is considered an authenticated packet.
Consent expires after 15 seconds. That is, if an authenticated
packet (e.g., DTLS, SRTP, ICE) has not been received from the
inverted 5-tuple after 15 seconds, the application MUST cease
transmission on that 5-tuple.
Consent is ended immediately by receipt of a an authenticated message
that closes the connection (for instance, a TLS fatal alert).
Receipt of an unauthenticated end-of-session message (e.g., TCP FIN)
does not indicate loss of consent. Thus, an endpoint receiving an
unauthenticated end-of-session message SHOULD continue sending media
(over connectionless transport) or attempt to re-establish the
connection (over connection-oriented transport) until consent expires
or it receives an authenticated message revoking consent.
Although receiving authenticated packets is sufficient for consent,
it is still RECOMMENDED to send messages to keep NAT or firewall
bindings alive (see Section 10 of [RFC5245] and [RFC6263]).
To meet the security needs of consent, an implementation MUST ensure
that an application (e.g., Javascript application) is not able to
obtain or control STUN information relevant to consent, specifically
the ICE transaction-id MUST NOT be accessible to upper-level
applications.
5. Connection Liveness
A connection is considered "live" if packets are received from a
remote endpoint within an application-dependent period. An
application can request a notification when there are no packets
received for a certain period (configurable).
Similarly, if packets haven't been received within a certain period,
an application can request a consent check (heartbeat) be generated.
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These two time intervals might be controlled by the same
configuration item.
Sending consent checks (heartbeats) at a high rate could allow a
malicious application to generate congestion, so applications MUST
NOT be able to send heartbeats faster than 1 per second.
6. DiffServ Treatment for Consent packets
It is RECOMMENDED that STUN consent checks use the same Diffserv
Codepoint markings as the media packets sent on that transport
address. This follows the recommendation of ICE connectivity check
described in section 7.1.2.4 of [RFC5245].
Note: It is possible that different Diffserv Codepoints are used by
different media over the same transport address
[I-D.ietf-tsvwg-rtcweb-qos]. In that case, what should this document
recommend as the Codepoint for STUN consent packets ?
7. 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.
8. Security Considerations
This document describes a security mechanism.
The security considerations discussed in [RFC5245] should also be
taken into account.
SRTP is encrypted and authenticated with symmetric keys; that is,
both sender and receiver know the keys. With two party sessions,
receipt of an authenticated packet from the single remote party is a
strong assurance the packet came from that party. However, when a
session involves more than two parties, all of whom know each others
keys, any of those parties could have sent (or spoofed) the packet.
Such shared key distributions are possible with some MIKEY [RFC3830]
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modes, Security Descriptions [RFC4568], and EKT
[I-D.ietf-avtcore-srtp-ekt]. Thus, in such shared keying
distributions, receipt of an authenticated SRTP packet is not
sufficient.
9. IANA Considerations
This document does not require any action from IANA.
10. Acknowledgement
Thanks to Eric Rescorla, Harald Alvestrand, Bernard Aboba, Magnus
Westerland, Cullen Jennings and Simon Perreault for their valuable
inputs and comments.
11. References
11.1. 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.
11.2. Informative References
[I-D.ietf-avtcore-srtp-ekt]
McGrew, D. and D. Wing, "Encrypted Key Transport for
Secure RTP", draft-ietf-avtcore-srtp-ekt-02 (work in
progress), February 2014.
[I-D.ietf-tsvwg-rtcweb-qos]
Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
other packet markings for RTCWeb QoS", draft-ietf-tsvwg-
rtcweb-qos-00 (work in progress), April 2014.
[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
August 2004.
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[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
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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Martin Thomson
Mozilla
Suite 300
650 Castro Street
Mountain View, California 94041
US
Email: martin.thomson@gmail.com
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