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Versions: (draft-thomson-rtcweb-consent) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 RFC 7675

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

   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
   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 October 13, 2014.

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
   publication of this document.  Please review these documents



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   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

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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