RTCWEB M. Perumal
Internet-Draft Ericsson
Intended status: Standards Track D. Wing
Expires: January 5, 2015 R. Ravindranath
T. Reddy
Cisco Systems
M. Thomson
Mozilla
July 4, 2014
STUN Usage for Consent Freshness
draft-ietf-rtcweb-stun-consent-freshness-05
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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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. Design Considerations . . . . . . . . . . . . . . . . . . . . 3
4. Solution . . . . . . . . . . . . . . . . . . . . . . . . . . 3
4.1. Expiration of Consent . . . . . . . . . . . . . . . . . . 4
4.2. Immediate Revocation of Consent . . . . . . . . . . . . . 5
5. Connection Liveness . . . . . . . . . . . . . . . . . . . . . 5
6. DiffServ Treatment for Consent packets . . . . . . . . . . . 6
7. W3C API Implications . . . . . . . . . . . . . . . . . . . . 6
8. Security Considerations . . . . . . . . . . . . . . . . . . . 6
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7
10. Acknowledgement . . . . . . . . . . . . . . . . . . . . . . . 7
11. References . . . . . . . . . . . . . . . . . . . . . . . . . 7
11.1. Normative References . . . . . . . . . . . . . . . . . . 7
11.2. Informative References . . . . . . . . . . . . . . . . . 7
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 8
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 exchange of request and
response messages to verify the remote peer's consent to receive
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traffic, and the absence of which for a period of time indicates a
loss of liveness.
WebRTC endpoints are required to support full ICE as specified in
section 3.4 of [I-D.ietf-rtcweb-transports]. However, when WebRTC
endpoints interwork with other endpoints that support only ICE-lite
(e.g. gateways) those endpoints will not generate consent checks, but
just respond to consent checks they receive.
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 the initial consent to
send traffic, which is obtained by ICE or a TCP handshake.
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
There are two ways consent to send traffic is revoked: expiration of
consent and immediate revocation of consent, which are discussed in
the following sections.
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4.1. Expiration of Consent
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 (e.g., RTP, RTCP, SCTP,
DTLS) 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 obtained by sending an ICE binding
request to the remote peer's Transport Address and receiving a
matching, authenticated, non-error ICE binding response from the
remote peer's Transport Address. These ICE binding requests and
responses are authenticated using the same short-term credentials as
the initial ICE exchange. Implementations MUST cease sending data if
their consent expires. To prevent expiry of consent, a STUN binding
request is sent every N milliseconds, where N SHOULD be 5000
milliseconds and MUST be randomized at least 20% above and 20% below
that value (to prevent prevent network synchronization). Using the
value 5000 milliseconds and that 20% randomization range, N would be
a value between 4000 and 6000. These STUN binding requests for
consent are not re-transmitted. Each STUN binding request for
consent re-calculates a new random value N and a new
cryptographically-random [RFC4086] STUN transaction ID.
The initial Consent to send traffic is obtained by ICE. Consent
expires after 30 seconds. That is, if a valid STUN binding response
corresponding to one of the STUN requests sent in the last 30 seconds
has not been received from the remote peer's Transport Address, the
endpoint MUST cease transmission on that 5-tuple.
To meet the security needs of consent, an untrusted application
(e.g., JavaScript) MUST NOT be able to obtain or control the STUN
transaction ID, because that enables spoofing STUN responses,
falsifying consent.
While TCP affords some protection from off-path attackers ([RFC5961],
[RFC4953]), there is still a risk an attacker could cause a TCP
sender to send packets forever by spoofing ACKs. To prevent such an
attack, consent checks MUST be performed over all WebRTC-initiated
transport connections, including TCP. In this way, an off-path
attacker spoofing TCP segments can not cause a TCP sender to send
packets longer than the consent timer (30 seconds).
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An endpoint that is not sending any application traffic does not need
to obtain consent which can slightly conserve its resources.
However, the endpoint needs to ensure its NAT or firewall mappings
persist which can be done using keepalive or other techniques (see
Section 10 of [RFC5245] and see [RFC6263]). If the endpoint wants to
send application traffic, it needs to first obtain consent if its
consent has expired.
4.2. Immediate Revocation of Consent
The previous section explained how consent expires due to a timeout.
In some cases it is useful to signal a connection is terminated,
rather than relying on a timeout. This is done by immediately
revoking consent.
Consent for sending traffic on the media or data channel is
immediately revoked by receipt of a an authenticated message that
closes the connection (e.g., a TLS fatal alert) or receipt of a valid
and authenticated STUN response with error code Forbidden (403).
Receipt of an unauthenticated message that closes a connection (e.g.,
TCP FIN) does not indicate revocation 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.
Note that an authenticated SRTCP BYE does not terminate consent; it
only indicates the associated SRTP source has quit.
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.
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 at an average rate of more than 1 per
second.
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6. DiffServ Treatment for Consent packets
It is RECOMMENDED that STUN consent checks use the same Diffserv
Codepoint markings as the ICE connectivity checks described in
section 7.1.2.4 of [RFC5245] for a given 5-tuple.
Note: It is possible that different Diffserv Codepoints are used by
different media over the same transport address
[I-D.ietf-tsvwg-rtcweb-qos]. Such a case is outside the scope of
this document.
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 consent
freshness has failed for a 5-tuple and the browser has stopped
transmitting on that 5-tuple.
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]
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 to verify consent.
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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, Christer Holmberg 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.
[RFC4086] Eastlake, D., Schiller, J., and S. Crocker, "Randomness
Requirements for Security", BCP 106, RFC 4086, June 2005.
[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-rtcweb-transports]
Alvestrand, H., "Transports for RTCWEB", draft-ietf-
rtcweb-transports-05 (work in progress), June 2014.
[I-D.ietf-tsvwg-rtcweb-qos]
Dhesikan, S., Jennings, C., Druta, D., Jones, P., and J.
Polk, "DSCP and other packet markings for RTCWeb QoS",
draft-ietf-tsvwg-rtcweb-qos-02 (work in progress), June
2014.
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[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
August 2004.
[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session
Description Protocol (SDP) Security Descriptions for Media
Streams", RFC 4568, July 2006.
[RFC4953] Touch, J., "Defending TCP Against Spoofing Attacks", RFC
4953, July 2007.
[RFC5961] Ramaiah, A., Stewart, R., and M. Dalal, "Improving TCP's
Robustness to Blind In-Window Attacks", RFC 5961, August
2010.
Authors' Addresses
Muthu Arul Mozhi Perumal
Ericsson
Mahadevapura
Bangalore, Karnataka 560048
India
Email: muthu.arul@gmail.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
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Tirumaleswar Reddy
Cisco Systems
Cessna Business Park, Varthur Hobli
Sarjapur Marathalli Outer Ring Road
Bangalore, Karnataka 560103
India
Email: tireddy@cisco.com
Martin Thomson
Mozilla
Suite 300
650 Castro Street
Mountain View, California 94041
US
Email: martin.thomson@gmail.com
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