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Versions: (draft-jesup-rtcweb-data) 00 01 02 03 04 05 06 07 09 10 11

RTCWeb Working Group                                            R. Jesup
Internet-Draft                                                   Mozilla
Intended status: Informational                                 S. Loreto
Expires: September 7, 2012                                      Ericsson
                                                               M. Tuexen
                                        Muenster Univ. of Appl. Sciences
                                                           March 6, 2012


                       RTCWeb Datagram Connection
                 draft-ietf-rtcweb-data-channel-00.txt

Abstract

   The Web Real-Time Communication (WebRTC) working group is charged to
   provide protocol support for direct interactive rich communication
   using audio, video, and data between two peers' web-browsers.  This
   document describes the non-media data transport aspects of the WebRTC
   framework.  It provides an architectural overview of how the Stream
   Control Transmission Protocol (SCTP) is used in the WebRTC context as
   a generic transport service allowing Web Browser to exchange generic
   data from peer to 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 September 7, 2012.

Copyright Notice

   Copyright (c) 2012 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 . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Requirements . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Use Cases  . . . . . . . . . . . . . . . . . . . . . . . . . .  5
     3.1.  Use Cases for Unreliable Datagram Based Channels . . . . .  5
     3.2.  Use Cases for Reliable Channels (Datagram or Stream
           Based) . . . . . . . . . . . . . . . . . . . . . . . . . .  5
   4.  Datagrams over SCTP over DTLS over UDP . . . . . . . . . . . .  6
   5.  The Envisioned Usage of SCTP in the RTCWeb Context . . . . . .  8
     5.1.  Association Setup  . . . . . . . . . . . . . . . . . . . .  8
     5.2.  SCTP Streams . . . . . . . . . . . . . . . . . . . . . . .  9
     5.3.  Channel Definition . . . . . . . . . . . . . . . . . . . .  9
     5.4.  Usage of Payload Protocol Identifier . . . . . . . . . . .  9
   6.  Minor Protocol and Message Format  . . . . . . . . . . . . . . 10
   7.  Security Considerations  . . . . . . . . . . . . . . . . . . . 10
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 10
   9.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 10
   10. Informative References . . . . . . . . . . . . . . . . . . . . 10
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 11























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1.  Introduction

   The issue of how best to handle non-media data types in the context
   of RTCWEB has reached a general consensus on the usage of SCTP
   [RFC4960] encapsulated on DTLS [RFC6347]:

                                +----------+
                                |   SCTP   |
                                +----------+
                                |   DTLS   |
                                +----------+
                                | ICE/UDP  |
                                +----------+

                       Figure 1: Basic stack diagram

   The encapsulation of SCTP over DTLS over ICE/UDP provides a NAT
   traversal solution together with confidentiality, source
   authenticated, integrity protected transfers.  This data transport
   service operates in parallel to the media transports, and all of them
   can eventually share a single transport-layer port number.

   SCTP provides multiple streams natively with reliable, unreliable and
   partially-reliable delivery modes.

   The remainder of this document is organized as follows: Section 2 and
   Section 3 provide requirements and use cases for both unreliable and
   reliable peer to peer datagram base channel; Section 4 arguments SCTP
   over DTLS over UDP; Section 5 provides an overview of how SCTP should
   be used by the RTCWeb protocol framework for transporting non-media
   data between browsers.


2.  Requirements

   This section lists the requirements for P2P data connections between
   two browsers.

   Req. 1   Multiple simultaneous datagram streams must be supported.
            Note that there may 0 or more media streams in parallel with
            the data streams, and the number and state (active/inactive)
            of the media streams may change at any time.

   Req. 2   Both reliable and unreliable datagram streams must be
            supported.






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   Req. 3   Data streams must be congestion controlled; either
            individually, as a class, or in conjunction with the media
            streams, to ensure that datagram exchanges don't cause
            congestion problems for the media streams, and that the
            rtcweb PeerConnection as a whole is fair with competing
            streams such as TCP.

   Req. 4   The application should be able to provide guidance as to the
            relative priority of each datagram stream relative to each
            other, and relative to the media streams. [ TBD: how this is
            encoded and what the impact of this is. ] This will interact
            with the congestion control algorithms.

   Req. 5   Datagram streams must be encrypted; allowing for
            confidentiality, integrity and source authentication.  See
            [I-D.ietf-rtcweb-security] and
            [I-D.ietf-rtcweb-security-arch] for detailed info.

   Req. 6   Consent and NAT traversal mechanism: These are handled by
            the PeerConnection's ICE [RFC5245] connectivity checks and
            optional TURN servers.

   Req. 7   Data streams must provide message fragmentation support such
            that IP-layer fragmentation does not occur no matter how
            large a message the Javascript application passes to be
            sent.

   Req. 8   The data stream transport protocol must not encode local IP
            addresses inside its protocol fields; doing so reveals
            potentially private information, and leads to failure if the
            address is depended upon.

   Req. 9   The data stream protocol should support unbounded-length
            "messages" (i.e., a virtual socket stream) at the
            application layer, for such things as image-file-transfer;
            or else it must support at least a maximum application-layer
            message size of 4GB.

   Req. 10  The data stream packet format/encoding must be such that it
            is impossible for a malicious Javascript to generate an
            application message crafted such that it could be
            interpreted as a native protocol over UDP - such as UPnP,
            RTP, SNMP, STUN, etc.

   Req. 11  The data stream transport protocol must start with the
            assumption of a PMTU of 1280 [ *** need justification ***]
            bytes until measured otherwise.




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   Req. 12  The data stream transport protocol must not rely on ICMP or
            ICMPv6 being generated or being passed back, such as for
            PMTU discovery.

   Req. 13  It must be possible to implement the protocol stack in the
            user application space.


3.  Use Cases

3.1.  Use Cases for Unreliable Datagram Based Channels

   U-C 1  A real-time game where position and object state information
          is sent via one or more unreliable data channels.  Note that
          at any time there may be no media channels, or all media
          channels may be inactive, and that there may also be reliable
          data channels in use.

   U-C 2  Non-critical state updates about a user in a video chat or
          conference, such as Mute state.

3.2.  Use Cases for Reliable Channels (Datagram or Stream Based)

   Note that either reliable datagrams or streams are possible; reliable
   streams would be fairly simple to layer on top of SCTP reliable
   datagrams with in-order delivery.

   U-C 3  A real-time game where critical state information needs to be
          transferred, such as control information.  Typically this
          would be datagrams.  Such a game may have no media channels,
          or they may be inactive at any given time, or may only be
          added due to in-game actions.

   U-C 4  Non-realtime file transfers between people chatting.  This
          could be datagrams or streaming.  Note that this may involve a
          large number of files to transfer sequentially or in parallel,
          such as when sharing a folder of images or a directory of
          files.

   U-C 5  Realtime text chat while talking with an individual or with
          multiple people in a conference.  Typically this would be
          datagrams.

   U-C 6  Renegotiation of the set of media streams in the
          PeerConnection.  Typically this would be datagrams.






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   U-C 7  Proxy browsing, where a browser uses data channels of a
          PeerConnection to send and receive HTTP/HTTPS requests and
          data, for example to avoid local internet filtering or
          monitoring.  Typically this would be streams.


4.  Datagrams over SCTP over DTLS over UDP

   The encapsulation of SCTP over DTLS as defined in
   [I-D.tuexen-tsvwg-sctp-dtls-encaps] provides a NAT traversal solution
   together with confidentiality, source authenticated, integrity
   protected transfers.  SCTP provides also natively several interesting
   features for transporting non-media data between browsers:

   o  Support of multiple streams.

   o  Ordered and unordered delivery of user messages.

   o  Reliable and partial-reliable transport of user messages.

   Each SCTP user message contains a so called Payload Protocol
   Identifier (PPID) that is passed to SCTP by its upper layer and sent
   to its peer.  This value represents an application (or upper layer)
   specified protocol identifier and be used to transport multiple
   protocols over a single SCTP association.  The sender provides for
   each protocol a specific PPID and the receiver demultiplexes the
   messages based on the received PPID.

   The encapsulation of SCTP over DTLS, together with the SCTP features
   listed above satisfies all the requirements listed in in Section 2.

   The layering of protocols for WebRTC is shown in the following
   Figure 2.

                                        +------+
                                        |WEBAPP|
                                        +------+
                                        | SCTP |
                          +--------------------+
                          | STUN | SRTP | DTLS |
                          +--------------------+
                          |         ICE        |
                          +--------------------+
                          | UDP1 | UDP2 | ...  |
                          +--------------------+

                     Figure 2: WebRTC protocol layers




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   This stack (especially in contrast to DTLS over SCTP [RFC6083]) has
   been chosen because it

   o  supports the transmission of arbitrary large user messages.

   o  shares the DTLS connection with the media channels.

   o  provides privacy for the SCTP control information.

   Considering the protocol stack of Figure 2 the usage of DTLS over UDP
   is specified in [RFC6347], while the usage of SCTP on top of DTLS is
   specified in [I-D.tuexen-tsvwg-sctp-dtls-encaps].

   Since DTLS is typically implemented in user-land, an SCTP user-land
   implementation must also be used.

   When using DTLS as the lower layer, only single homed SCTP
   associations can be used, since DTLS does not expose any address
   management to its upper layer.  The ICE/UDP layer can handle IP
   address changes during a session without needing to notify the DTLS
   and SCTP layers, though it would be advantageous to retest path MTU
   on an IP address change.

   DTLS implementations used for this stack must support controlling
   fields of the IP layer like the Don't fragment (DF)-bit in case of
   IPv4 and the Differentiated Services Code Point (DSCP) field.  This
   is required for performing path MTU discovery.  The DTLS
   implementation must also support sending user messages exceeding the
   path MTU.

   When supporting multiple SCTP associations over a single DTLS
   connection, incoming ICMP or ICMPv6 messages can't be processed by
   the SCTP layer, since there is no way to identify the corresponding
   association.  Therefore the number of SCTP associations should be
   limited to one or ICMP and ICMPv6 messages should be ignored.  In
   general, the lower layer interface of an SCTP implementation has to
   be adapted to address the differences between IPv4 or IPv6 (being
   connection-less) or DTLS (being connection-oriented).

   When protocol stack of Figure 2 is used, DTLS protects the complete
   SCTP packet, so it provides confidentiality, integrity and source
   authentication of the complete SCTP packet.

   This protocol stack supports the usage of multiple SCTP streams.  A
   user message can be sent ordered or unordered and, if the SCTP
   implementations support [RFC3758], with partial reliability.  When
   using partial reliability, it might make sense to use a policy
   limiting the number of retransmissions.  Limiting the number of



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   retransmissions to zero provides a UDP like service where each user
   messages is sent exactly once.

   SCTP provides congestion control on a per-association base.  This
   means that all SCTP streams within a single SCTP association share
   the same congestion window.  Traffic not being sent over SCTP is not
   covered by the SCTP congestion control.  Due to the typical parallel
   SRTP media streams, it will be advantageous to select a delay-
   sensitive congestion control algorithm or to at least coordinate
   congestion control between the data channels and the media streams to
   avoid a data channel transfer ending up with most or all the channel
   bandwidth.


5.  The Envisioned Usage of SCTP in the RTCWeb Context

   The appealing features of SCTP in the RTCWeb context are:

   o  TCP-friendly congestion control.

   o  The congestion control is modifiable for integration with media
      stream congestion control.

   o  Support for multiple channels with different characteristics.

   o  Support for out-of-order delivery.

   o  Support for large datagrams and PMTU-discovery and fragmentation.

   o  Reliable or partial reliability support.

   o  Support of multiple streams.

   Multihoming will not be used in this scenario.  The SCTP layer would
   simply act as if it were running on a single-homed host, since that
   is the abstraction that the lower layer (a connection oriented,
   unreliable datagram service) would expose.

5.1.  Association Setup

   The SCTP association would be set up when the two endpoints of the
   WebRTC PeerConnection agree on opening it, as negotiated by JSEP
   (typically an exchange of SDP) [I-D.ietf-rtcweb-jsep].  It would use
   the DTLS connection created at the start of the PeerConnection
   connection.

   The application should indicate the number of simultaneous streams
   required when opening the association, and if no value is supplied,



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   the implementation should provide a default, with a suggested value
   of 16.  If more simultaneous streams are needed, [RFC6525] allows
   adding additional (but not removing) streams to an existing
   association.  There can be up to 65536 SCTP streams per SCTP
   association in each direction.

5.2.  SCTP Streams

   SCTP defines a stream as an unidirectional logical channel existing
   within an SCTP association one to another SCTP endpoint.  The streams
   are used to provide the notion of in-sequence delivery.  Each user
   message is sent on a particular stream, either order or unordered.
   Ordering is preserved only for all ordered messages sent on the same
   stream.

5.3.  Channel Definition

   The W3C seems to have consensus on defining the application API for
   WebRTC dataChannels to be bidirectional.  They also consider the
   notions of in-sequence, out-of-sequence, reliable and un-reliable as
   properties of Channels.

   A possible realization of a bidirectional Data Channel is a pair of
   one incoming stream and one outcoming SCTP stream.

   Closing of a Data Channel can be signalled resetting the
   corresponding streams [RFC6525].  Resetting a stream set the Stream
   Sequence Numbers (SSNs) of the stream back to 'zero' with a
   corresponding notification to the application layer that the reset
   has been performed.  Closed streams are available to reuse.

   [RFC6525] also guarantees that all the messages are delivered (or
   expired) before resetting the stream.

   It might be useful to use a specific pair of SCTP streams for
   transporting control information.

5.4.  Usage of Payload Protocol Identifier

   The SCTP Payload Protocol Identifiers (PPIDs) can be used to signal
   the interpretation of the "Payload data", like a string, ASCII or
   binary data.

   RFC 4960 [RFC4960] creates the registry from which these identifiers
   have been assigned.  Eventual PPIDs defined within the RTCWeb Context
   have to be registered with IANA.





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6.  Minor Protocol and Message Format

   A separate draft (draft-jesup-rtcweb-data-protocol) is being
   submitted to define the minor protocol to set up and manage the
   bidirectional data channels needed to satisify the requirements in
   this document for WebRTC.

   Masking of the protocol is not needed if the lower layer always
   encrypts with DTLS.


7.  Security Considerations

   To be done.


8.  IANA Considerations

   This document does not require any actions by the IANA.


9.  Acknowledgments

   Many thanks for comments, ideas, and text from Cullen Jennings, Eric
   Rescorla, Randall Stewart, Justin Uberti, and Harald Alvestrand.


10.  Informative References

   [RFC3758]  Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
              Conrad, "Stream Control Transmission Protocol (SCTP)
              Partial Reliability Extension", RFC 3758, May 2004.

   [RFC4960]  Stewart, R., "Stream Control Transmission Protocol",
              RFC 4960, September 2007.

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

   [RFC6083]  Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram
              Transport Layer Security (DTLS) for Stream Control
              Transmission Protocol (SCTP)", RFC 6083, January 2011.

   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6347, January 2012.




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   [RFC6525]  Stewart, R., Tuexen, M., and P. Lei, "Stream Control
              Transmission Protocol (SCTP) Stream Reconfiguration",
              RFC 6525, February 2012.

   [I-D.ietf-rtcweb-security]
              Rescorla, E., "Security Considerations for RTC-Web",
              draft-ietf-rtcweb-security-01 (work in progress),
              October 2011.

   [I-D.ietf-rtcweb-security-arch]
              Rescorla, E., "RTCWEB Security Architecture",
              draft-ietf-rtcweb-security-arch-00 (work in progress),
              January 2012.

   [I-D.ietf-rtcweb-jsep]
              Uberti, J. and C. Jennings, "Javascript Session
              Establishment Protocol", draft-ietf-rtcweb-jsep-00 (work
              in progress), March 2012.

   [I-D.ietf-tsvwg-sctp-udp-encaps]
              Tuexen, M. and R. Stewart, "UDP Encapsulation of SCTP
              Packets", draft-ietf-tsvwg-sctp-udp-encaps-02 (work in
              progress), December 2011.

   [I-D.tuexen-tsvwg-sctp-dtls-encaps]
              Jesup, R., Loreto, S., Stewart, R., and M. Tuexen, "DTLS
              Encapsulation of SCTP Packets for RTCWEB",
              draft-tuexen-tsvwg-sctp-dtls-encaps-00 (work in progress),
              March 2012.


Authors' Addresses

   Randell Jesup
   Mozilla
   USA

   Email: randell-ietf@jesup.org


   Salvatore Loreto
   Ericsson
   Hirsalantie 11
   Jorvas  02420
   Finland

   Email: salvatore.loreto@ericsson.com




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   Michael Tuexen
   Muenster University of Applied Sciences
   Stegerwaldstrasse 39
   Steinfurt  48565
   Germany

   Email: tuexen@fh-muenster.de












































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