Internet DRAFT - draft-martinsen-tram-turnbandwidthprobe

draft-martinsen-tram-turnbandwidthprobe







TRAM                                                        P. Martinsen
Internet-Draft                                               T. Andersen
Intended status: Informational                              G. Salgueiro
Expires: November 30, 2015                                         Cisco
                                                       M. Petit-Huguenin
                                                      Impedance Mismatch
                                                            May 29, 2015


        Traversal Using Relays around NAT (TURN) Bandwidth Probe
               draft-martinsen-tram-turnbandwidthprobe-00

Abstract

   Performing pre-call probing to discover a reasonable value for the
   available bandwidth, is useful information that can be utilized by
   bandwidth sensitive or bandwidth intensive network devices (e.g.,
   video encoders).  The method described herein is intended to produce
   an initial bandwidth value.  Applications using this mechanism should
   also employ appropriate rate adaptation techniques.  In addition to
   bandwidth, latency and bufferbloat can also be measured.  No
   modification is needed on the server side.

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 November 30, 2015.

Copyright Notice

   Copyright (c) 2015 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
   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.  Notational Conventions  . . . . . . . . . . . . . . . . . . .   3
   3.  Overview of Operation . . . . . . . . . . . . . . . . . . . .   3
   4.  Base Protocol Procedures  . . . . . . . . . . . . . . . . . .   4
     4.1.  UDP Procedures  . . . . . . . . . . . . . . . . . . . . .   5
     4.2.  TCP Procedures  . . . . . . . . . . . . . . . . . . . . .   5
     4.3.  Sending Data to Measure Available Bandwidth and
           Latency . . . . . . . . . . . . . . . . . . . . . . . . .   6
   5.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   6
   6.  New STUN Attribute  . . . . . . . . . . . . . . . . . . . . .   7
     6.1.  TIMESTAMP . . . . . . . . . . . . . . . . . . . . . . . .   7
   7.  Implementation Status . . . . . . . . . . . . . . . . . . . .   7
     7.1.  Cisco Collaboration Endpoint (CE) . . . . . . . . . . . .   8
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .   8
   9.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   9
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .   9
     10.1.  Normative References . . . . . . . . . . . . . . . . . .   9
     10.2.  Informative References . . . . . . . . . . . . . . . . .  10
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  10

1.  Introduction

   When Interactive Connectivity Establishment (ICE) [RFC5245] and
   Traversal Using Relays around NAT (TURN) [RFC5766] are used by an
   endpoint as a firewall/NAT traversal mechanism, the TURN relay can
   also be used to measure bandwidth and latency prior to call setup.

   In normal ICE behavior the client first sends a message (allocate
   request) to the TURN server to allocate a RELAY address.  This
   address can be used by the endpoint to receive media from other
   endpoints.  The media stream is then received by the TURN server and
   then relayed back to the endpoint behind the firewall/NAT.  For
   security reasons the endpoint must first set the correct permissions
   on the TURN server to only allow media from remote participants it
   wants to communicate with (i.e., addresses taken from the Session
   Initiation Protocol (SIP) [RFC3261] Session Description Protocol
   (SDP) [RFC4566] offer/answer exchange [RFC3264]) . The endpoint will
   also learn its reflexive address on the firewall/NAT when talking to
   the TURN server.



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   Combining this with a TCP transfer on the same TURN server can be
   used to also measure bufferbloat, an important metric for multimedia
   applications.

   Note that only the maximum bandwidth, maximum latency and maximum
   bufferbloat of the aggregation of both uplink and downlink can be
   measured.  It is not possible with this technique to get the metrics
   of only one.  For most multimedia applications using TURN that is not
   an issue as they are generally symmetrical, but some other use cases
   (like conferencing) may need other techniques to measure these
   metrics separately.

   No modification to the TURN server is necessary.

2.  Notational Conventions

   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.  Overview of Operation

   Prior to the call (upon registering with the call control server,
   receiving a configuration, loading application, or a similar event)
   the endpoint can measure bandwidth and latency between the endpoint
   and the TURN server.


                                          *Relay
                                    +------+
                          *Rflx  -<-| TURN |<-\
      /-------\     +-----+     /   +------+   |
      |       |--<--| NAT |-<---               |
      | Alice |--->-|-->--|->-->----->----->--/
      \-------/     +-----+



                                 Figure 1

   The agent allocates a TURN Relay port on its designated TURN server
   as described in TURN RFC [RFC5766].  In the process the agent will
   also learn the outermost NAT address.  This is called a reflexive
   address (Rflx).  For more information see Section 2.1 of the TURN RFC
   regarding candidate gathering in ICE.






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   The agent must set the permissions on the allocated RELAY port as
   described in Section 8 of the TURN RFC to allow traffic from the
   discovered reflexive address.

   When sending packets to the allocated RELAY port on the TURN server,
   the packets will be forwarded back to the agent in a data indication
   packet.  See Section 10 of the TURN RFC for details on how the TURN
   server can relay packets back to the allocating agent.  Available
   bandwidth can be measured by sending varying number of packets and
   detecting the amount of packet loss.  Each packet sent affects both
   upstream and downstream links.

   To make it easier to calculate the available bandwidth a TIMESTAMP
   attribute is defined in this document (see Section Section 6.1) and
   can be added to the Session Traversal Utilities for NAT (STUN)
   [RFC5389] probe packets.  The PADDING attribute from the NAT Behavior
   Discovery Using STUN RFC [RFC5780] can be used to vary the packet
   size.

   Discovering the MTU and network path (using the STUN-PMTUD
   [I-D.petithuguenin-tram-stun-pmtud] and STUN Traceroute
   [I-D.martinsen-tram-stuntrace] mechanisms) can also be performed when
   probing for the bandwidth available between the client and the TURN
   server.

4.  Base Protocol Procedures

   In order to perform the STUN bandwidth probing mechanism described in
   this document, the client MUST take the following general steps
   (explained in greater detail in the following subsections).

   o  Allocate TURN RELAY address

   o  Set correct permissions on the allocated TURN RELAY address

   o  Originating client sends data to itself through the TURN server
      and measures bandwidth throughput and latency

   When initiating a bandwidth probe it is important to not do so when a
   device powers up or some similar initiating events.  If a power
   failure has happened and all devices within an area are rebooted
   concurrently the bandwidth probing of all the devices can have a
   DDOS-like effect.  Measures should be taken to avoid such scenarios
   (e.g., random delays to initiate bandwidth probing, etc).

   Discovery of the TURN server as well as the determination of what
   TURN server to uses is entirely at the discretion of the client and
   outside the scope of this document.  A client MUST be prepared to be



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   redirected to another TURN server if it receives an ALTERNATE-SERVER
   response.

   While allocating the TURN RELAY port the client will learn its
   outermost NAT address or reflexive address.  This is the address the
   TURN server will receive the bandwidth probing packets from.

   The bandwidth mechanism can use either a UDP transport or a TCP.
   Secure transports (i.e.  TLS or DTLS) may be used to discover if an
   intermediary network element tries to process flows differently when
   they are secured.

4.1.  UDP Procedures

   The client allocates a TURN RELAY port as described in the TURN RFC.
   The client then use a CreatePermission request with the obtained
   reflexive address encoded in a XOR-PEER-ADDRESS attribute as
   described in Section 9.1 of the TURN RFC.

   It is recommended to create a TURN channel as soon as possible to
   lower the overhead of the packets exchanged.

   If the transport address used to send the UDP packets to the TURN
   relay is identical to the transport address used to create the TURN
   allocation, then a TURN Channel can be created immediately by using
   the reflexive transport address learned during the Allocate.

   If not, the TURN Channel can be created as soon the first Data
   indication is received.

   The client can then send UDP packets to the relay transport address
   and receive then over the TURN Channel.

   Immediately after this the client can send UDP packets over the TURN
   channel and receive them directly, as an additional way of averaging
   the impact of the difference of encapsulation for the packets.  Note
   that the client still need to periodically send packets over the TURN
   Channel to persist eventual NAT bindings.

   Note that the client cannot use a TCP transport to the server with a
   UDP allocation because there would be no way to retrieve the UDP
   reflexive address for the CreatePermission request.

4.2.  TCP Procedures

   The client allocates a TURN RELAY port as described in TURN
   Extensions for TCP Allocations [RFC6062].  The client then use a
   CreatePermission request with the obtained reflexive address encoded



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   in a XOR-PEER-ADDRESS attribute as described in Section 9.1 of the
   TURN RFC.

   The client then establishes a TCP connection to the relay transport
   address.  The client will receive a ConnectAttempt indication that
   will trigger a new TCP connection to the TURN server, and the sending
   of a ConnectBind.

   After completion of this procedure, data sent over the direct TCP
   connection will be received over the bound TURN connection, and vice-
   versa, although there is no difference of overhead in that case.

4.3.  Sending Data to Measure Available Bandwidth and Latency

   The specific calculation and measurement of the bandwidth is client
   dependent and implementation-specific and is thus outside the scope
   of this document.

   If the client want to use STUN packets as the basis for the probing
   packets, then a TIMESTAMP attribute is defined in this specification
   (see Section Section 6.1) to simplify measurement of round-trip time
   (RTT) and available bandwidth.  A PADDING attribute is already
   defined in RFC 5780 [RFC5780] that makes it easy to vary the size of
   the STUN probing packet.

   The probing packet will be sent upstream to the TURN server and later
   received downstream from the TURN server.  Available bandwidth would
   typically be determined to be the lowest of the bandwidth values
   calculated for the upstream and downstream directions.

   If the RTP [RFC3550] loop-back mechanism described in RFC 6849
   [RFC6849] is in use the method described here can extend the use-
   cases mentioned in RFC6849 Section 1.1 to enable the "loopback
   source" and "loopback mirror" to be running on the same device.
   Using RTP would permit to reuse the standards RTP tools for
   calculating latency, jitter and other metrics.  It may also permit to
   get better results if some intermediary network element has
   preferential treatment for media packets.

   The client should take care to reuse the same congestion control
   mechanisms it uses when sending media to avoid unnecessary strain on
   the network.

5.  IANA Considerations

   This specification defines a new STUN attribute.  IANA added this new
   attribute to the STUN Attributes sub-registry of the Session




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   Traversal Utilities for NAT (STUN) Parameters registry.  (This is
   still an ID draft so not assignment yet)

6.  New STUN Attribute

   This STUN extension defines the following new attribute:


         0xXXX0: TIMESTAMP


6.1.  TIMESTAMP

   The TIMESTAMP attribute has a length of 80 bits.  Padding is needed
   to hit the required 32 bit STUN attribute boundary.

       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                    seconds (32bit)                            |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                    microseconds (32bit)                       |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |      seq (16bit)              |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                       Figure 2: TIMESTAMP Attribute

   The seconds and microseconds fields reflect what would be returned in
   the struct timeval when calling getTimeofDay() function.  Note that
   the size of that struct may vary based on platform, but 32 bits is
   more than sufficient to obtain the required accuracy for the feature
   described in this document.  It is RECOMMENDED to initialize these
   fields with a random value that later can be subtracted to get the
   right timing.

   The seq field is a 16 bit sequence number.  It is increased by one
   for each bandwidth probe STUN packet sent.  It is RECOMMENDED to
   choose a random starting value.

7.  Implementation Status

   [[Note to RFC Editor: Please remove this section and the reference to
   [RFC6982] before publication.]]

   This section records the status of known implementations of the
   protocol defined by this specification at the time of posting of this
   Internet-Draft, and is based on a proposal described in RFC 6982



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   [RFC6982].  The description of implementations in this section is
   intended to assist the IETF in its decision processes in progressing
   drafts to RFCs.  Please note that the listing of any individual
   implementation here does not imply endorsement by the IETF.
   Furthermore, no effort has been spent to verify the information
   presented here that was supplied by IETF contributors.  This is not
   intended as, and must not be construed to be, a catalog of available
   implementations or their features.  Readers are advised to note that
   other implementations may exist.

   According to RFC 6982 [RFC6982], "this will allow reviewers and
   working groups to assign due consideration to documents that have the
   benefit of running code, which may serve as evidence of valuable
   experimentation and feedback that have made the implemented protocols
   more mature.  It is up to the individual working groups to use this
   information as they see fit"

7.1.  Cisco Collaboration Endpoint (CE)

   Organization:  Cisco

   Name: Cisco Collaboration Endpoints (CE) software

   Description:  Hard video endpoint part of the Cisco collaboration
         portfolio

   Level of maturity:  In released products

   Coverage  Implementation of base procedures of the functionality
         described in this specification

   Licensing:  Proprietary

   Implementation experience:  Straight forward, but implementation was
         don prior to writing up the spec

   Contact:  Paal-Erik Martinsen (palmarti@cisco.com)

8.  Security Considerations

   When setting permissions this is done on a per IP address basis.
   Port number is not part of the permission.  This is necessary
   limitation of the TURN protocol [RFC5766] and not something
   introduced by this specification.

   To prevent replay attacks or other attacks that rely on static
   sequence number initialization it is important to randomly initialize
   the seq number in the TIMESTAMP Attribute.  Likewise it is important



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   to hide the time information by assigning a random value to the
   seconds and microseconds fields.  That random value can be added and
   subtracted by the client when sending and receiving packets to get
   the correct value.  This prevents any information leakage regarding
   time from the client.

9.  Acknowledgements

   Thanks to Dan Wing for input.

10.  References

10.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245, April
              2010.

   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              October 2008.

   [RFC5766]  Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN): Relay Extensions to Session
              Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.

   [RFC5780]  MacDonald, D. and B. Lowekamp, "NAT Behavior Discovery
              Using Session Traversal Utilities for NAT (STUN)", RFC
              5780, May 2010.

   [RFC6062]  Perreault, S. and J. Rosenberg, "Traversal Using Relays
              around NAT (TURN) Extensions for TCP Allocations", RFC
              6062, November 2010.

   [RFC6849]  Kaplan, H., Hedayat, K., Venna, N., Jones, P., and N.
              Stratton, "An Extension to the Session Description
              Protocol (SDP) and Real-time Transport Protocol (RTP) for
              Media Loopback", RFC 6849, February 2013.





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   [RFC6982]  Sheffer, Y. and A. Farrel, "Improving Awareness of Running
              Code: The Implementation Status Section", RFC 6982, July
              2013.

10.2.  Informative References

   [I-D.martinsen-tram-stuntrace]
              Martinsen, P. and D. Wing, "STUN Traceroute", draft-
              martinsen-tram-stuntrace-00 (work in progress), February
              2015.

   [I-D.petithuguenin-tram-stun-pmtud]
              Petit-Huguenin, M., "Path MTU Discovery Using Session
              Traversal Utilities for NAT (STUN)", draft-petithuguenin-
              tram-stun-pmtud-00 (work in progress), January 2015.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264, June
              2002.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

Authors' Addresses

   Paal-Erik Martinsen
   Cisco Systems, Inc.
   Philip Pedersens Vei 22
   Lysaker, Akershus  1325
   Norway

   Email: palmarti@cisco.com


   Trond Andersen
   Cisco Systems, Inc.
   Philip Pedersens Vei 22
   Lysaker, Akershus  1325
   Norway

   Email: trondand@cisco.com





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   Gonzalo Salgueiro
   Cisco Systems, Inc.
   7200-12 Kit Creek Road
   Research Triangle Park, NC  27709
   US

   Email: gsalguei@cisco.com


   Marc Petit-Huguenin
   Impedance Mismatch

   Email: marc@petit-huguenin.org






































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