STUN Usage for Consent Freshness
draft-ietf-rtcweb-stun-consent-freshness-05

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

   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
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   This Internet-Draft will expire on January 5, 2015.

Copyright Notice

   Copyright (c) 2014 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

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   publication of this document.  Please review these documents
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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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