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Versions: 00 01 02 03 04 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
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   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
   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  . . . . . . . . . . . . . . . . . . . . . . . . . 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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