RTCWEB                                                    Muthu.                                                        M. Perumal
Internet-Draft                                                   D. Wing
Intended status: Standards Track                         R. Ravindranath
Expires: September 25, October 13, 2014                                       T. Reddy
                                                           Cisco Systems
                                                          March 24,
                                                              M. Thomson
                                                          April 11, 2014

                    STUN Usage for Consent Freshness


   Verification of peer consent before

   To prevent sending excessive 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 application may want an endpoint, periodic consent
   needs to detect
   connection failure and take appropriate action. be obtained from that remote endpoint.

   This document describes how a WebRTC browser can verify peer consent
   to continue sending traffic and detect mechanism using a new STUN usage.
   This same mechanism can also determine connection failure. loss ("liveness")
   with a remote peer.

Status of This Memo

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   This Internet-Draft will expire on September 25, October 13, 2014.

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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Design Considerations . . . . . . . . . . . . . . . . . . . .   3
   4.  Solution Overview . . . . . . . . . . . . . . . . . . . . . .   3
   5.  W3C API Implications  Connection Liveness . . . . . . . . . . . . . . . . . . . .   5 .   4
   6.  Interaction with Keepalives used  DiffServ Treatment for Refreshing NAT Bindings Consent packets  . . . . . . . . . . .   5
   7.  W3C API Implications  . . . . . . . . . . . . . . . . . . . .   5
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .   5
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   5
   9.   6
   10. Acknowledgement . . . . . . . . . . . . . . . . . . . . . . .   5
   10.   6
   11. References  . . . . . . . . . . . . . . . . . . . . . . . . .   6
     11.1.  Normative References . . . . . . . . . . . . . . . . . .   6
     11.2.  Informative References . .   5 . . . . . . . . . . . . . . .   6
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .   6   7

1.  Introduction

   To prevent attacks on WebRTC peers, WebRTC browsers 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 using ICE
   connectivity checks), checks, and periodically for the duration of the session (using
   using the procedure procedures defined in this document). 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

   Related to consent is loss of connectivity ("liveness").  WebRTC  Many
   applications want notification of connection failure loss to take appropriate
   actions (e.g., alert the user, try switching to a different

   This document describes a new STUN usage with a request and response
   messages which verifies the remote peer consents peer's consent to receive traffic,
   detects can also detect loss of liveness.  To meet the security needs of consent, the
   JavaScript application has no control over the consent requests or
   the requirement to receive a consent response.  However, the
   JavaScript does get notification of consent failure and loss of

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in [RFC2119].

   Consent:  It is the mechanism of obtaining permission from the peer to send traffic on
      to a candidate pair. certain transport address.  This is usually obtained via ICE.

   Consent Freshness:  It is the mechanism of obtaining permission from
      the peer  Permission to continue sending traffic on to a nominated candidate pair
      after ICE has concluded.

   Session Liveness:  It
      certain transport address.  This is performed by the mechanism procedure
      described in this document.

   Session Liveness:  Detecting loss of detecting connectivity on to a
      nominated candidate pair after ICE has concluded. certain
      transport address.  This is performed by the procedure described
      in this document.

   Transport Address:  The combination of an remote peer's IP address and port number
      (such as a UDP (UDP or TCP TCP)
      port number). number.

3.  Design Considerations

   Although ICE requires periodic keepalive traffic to be sent in order
   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 ensure connectivity. verify session liveness.  Thus, we
   need a request/response

   Though mechanism for consent freshness.  ICE specifies STUN Binding indications to can 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 and there is no impact on mechanism because ICE processing, already requires ICE agents
   continue listening for ICE messages, as described in section 10 of

   The above considerations suggest that STUN Binding request/response
   is most suitable for performing consent freshness.

4.  Solution Overview

   Consent freshness serves as a circuit breaker (so that if the path or
   remote peer fails the WebRTC browser stops sending all traffic on
   that remote peer), determining session liveness serves the purpose of
   notifying the application of connectivity failure so that the
   application can take appropriate action.

   The solution uses three values:

   1.  A consent timer, Tc, whose value is determined by the browser.
       This value MUST be 15 seconds.

   2.  A packet receipt timer, Tr, whose value is determined by the
       application, but MUST NOT be shorter than 1 second or longer than
       15 seconds, and SHOULD have a default value of 5 seconds.

   3.  A consent timeout, Tf, which is how many seconds elapse without a
       consent response before the browser ceases transmission of media.
       Its value MUST be 15 seconds or less, and the value 15 seconds is

   A WebRTC browser performs a combined consent freshness and session
   liveness test using STUN request/response as described below:

   Every Tc seconds, the

   An endpoint MUST NOT send application data (in WebRTC browser sends a STUN Binding Request this means RTP
   or SCTP data) on an ICE-initiated connection unless the receiving
   endpoint consents to receive the peer.  This request MUST use data.  After a new, cryptographically random
   Transaction ID [RFC4086], and is formatted as for an successful ICE
   connectivity check [RFC5245].  A valid STUN Binding Response 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 also formatted as
   for indicated by:

   1.  Sending an ICE connectivity check [RFC5245].  The STUN Binding Request binding request to the remote peer's Transport
       Address and STUN Binding Response are validated as for an receiving a matching and authenticated ICE connectivity
   check [RFC5245].  The difference binding
       response from the inverted remote peer's Transport Address.

       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 is that
   there is no exponential back off for retransmissions.

   If a valid (or response) could be lost, and re-
       transmissions MUST use the same STUN Binding Response is received, transaction-id, and re-
       transmissions MUST NOT be sent more frequently than every 500ms
       or the smoothed round-trip time (from previous consent timer freshness
       checks or RTP round-trip time), whichever is
   reset and fires again Tc seconds later.

   If less.  For the
       purposes of this document, receipt of an ICE response with the
       matching transaction-id of its request with a valid STUN Binding Response 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 500ms, 15 seconds, the
   STUN Binding Request application MUST cease
   transmission on that 5-tuple.

   Consent is retransmitted (with ended immediately by receipt of a an authenticated message
   that closes the same Transaction ID
   and all other fields).  As long as connection (for instance, a valid STUN Binding Response is TLS fatal alert).

   Receipt of an unauthenticated end-of-session message (e.g., TCP FIN)
   does not received, this retransmission is repeated every 500ms 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 Tf
   seconds have elapsed consent expires
   or a valid response it receives an authenticated message revoking consent.

   Although receiving authenticated packets is received.  If no valid
   response sufficient for consent,
   it is received after Tf seconds, still RECOMMENDED to send messages to keep NAT or firewall
   bindings alive (see Section 10 of [RFC5245] and [RFC6263]).

   To meet the WebRTC browser security needs of consent, an implementation MUST quit
   transmitting traffic ensure
   that an application (e.g., Javascript application) is not able to this remote peer.  Considering
   obtain or control STUN information relevant to consent, specifically
   the default
   value of Tf=15 seconds, this means transmission will stop after 30
   consent check packets have resulted in no response. ICE transaction-id MUST NOT be accessible to upper-level

5.  Connection Liveness timer: If

   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 have
   received for a certain period (configurable).

   Similarly, if packets haven't been received on the local port in
   Tr seconds, within a certain period,
   an application can request a consent check (heartbeat) be generated.

   These two time intervals might be controlled by the WebRTC browser same
   configuration item.

   Sending consent checks (heartbeats) at a high rate could allow a
   malicious application to generate congestion, so applications MUST inform the JavaScript
   NOT be able to send heartbeats faster than 1 per second.

6.  DiffServ Treatment for Consent packets

   It is RECOMMENDED that
   connectivity has been lost.  The JavaScript application will STUN consent checks use this
   notification however it desires (e.g., cease transmitting to the
   remote peer, provide a notification to same Diffserv
   Codepoint markings as the user, etc.).  Note media packets sent on that transport
   address.  This follows the
   definition recommendation of a received packet ICE connectivity check
   described in section of [RFC5245].

   Note: It is liberal, and includes an SRTP
   packet possible that fails authentication, a 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 Binding Request with an
   invalid USERNAME or PASSWORD, or any other packet.

5. 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.

6.  Interaction with Keepalives used for Refreshing NAT Bindings

   When not actively sending traffic on a nominated candidate pair,
   performing consent freshness does not serve any purpose from a
   security perspective.  If consent freshness is not performed during
   this period, the browser continues to performs the ICE keepalives
   [RFC5245] or RTP keepalive [RFC6263] to refresh NAT bindings.


8.  Security Considerations


   This document describes a security mechanism.

   The security considerations discussed in [RFC5245] are to should also be
   taken into account.

   The browser MUST use short-term credential to authenticate

   SRTP is encrypted and authenticated with symmetric keys; that is,
   both sender and receiver know the STUN
   messages used for Consent freshness, 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 liveness 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 perform
   message integrity check on STUN binding request/response.

8. EKT
   [I-D.ietf-avtcore-srtp-ekt].  Thus, in such shared keying
   distributions, receipt of an authenticated SRTP packet is not

9.  IANA Considerations

   This document does not require any action from IANA.


10.  Acknowledgement

   Thanks to Eric Rescorla, Harald Alvestrand, Martin Thomson, 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.

   [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

   [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

              McGrew, D. and D. Wing, "Encrypted Key Transport for
              Secure RTP", draft-ietf-avtcore-srtp-ekt-02 (work in
              progress), February 2014.

              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.

   [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

   Email: mperumal@cisco.com

   Dan Wing
   Cisco Systems
   821 Alder Drive
   Milpitas, California  95035

   Email: dwing@cisco.com

   Ram Mohan Ravindranath
   Cisco Systems
   Cessna Business Park
   Sarjapur-Marathahalli Outer Ring Road
   Bangalore, Karnataka  560103

   Email: rmohanr@cisco.com

   Tirumaleswar Reddy
   Cisco Systems
   Cessna Business Park, Varthur Hobli
   Sarjapur Marathalli Outer Ring Road
   Bangalore, Karnataka  560103

   Email: tireddy@cisco.com
   Martin Thomson
   Suite 300
   650 Castro Street
   Mountain View, California  94041

   Email: martin.thomson@gmail.com