Network Working Group                                           R. Jesup
Internet-Draft                                                   Mozilla
Intended status: Standards Track                               S. Loreto
Expires: April 26, 2013                                         Ericsson
                                                               M. Tuexen
                                        Muenster Univ. of Appl. Sciences
                                                        October 23, 2012

                       RTCWeb Datagram Connection


   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

   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 April 26, 2013.

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
   ( 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.  Conventions  . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Requirements . . . . . . . . . . . . . . . . . . . . . . . . .  3
   4.  Use Cases  . . . . . . . . . . . . . . . . . . . . . . . . . .  5
     4.1.  Use Cases for Unreliable Datagram Based Channels . . . . .  5
     4.2.  Use Cases for Reliable Channels (Datagram or Stream
           Based) . . . . . . . . . . . . . . . . . . . . . . . . . .  5
   5.  SCTP over DTLS over UDP Considerations . . . . . . . . . . . .  6
   6.  The Usage of SCTP in the RTCWeb Context  . . . . . . . . . . .  8
     6.1.  Association Setup  . . . . . . . . . . . . . . . . . . . .  9
     6.2.  SCTP Streams . . . . . . . . . . . . . . . . . . . . . . .  9
     6.3.  Channel Definition . . . . . . . . . . . . . . . . . . . .  9
     6.4.  Usage of Payload Protocol Identifier . . . . . . . . . . . 10
   7.  Security Considerations  . . . . . . . . . . . . . . . . . . . 10
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 10
   9.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 10
   10. Informative References . . . . . . . . . . . . . . . . . . . . 11
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 12

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1.  Introduction

   Non-media data types in the context of RTCWEB are handled by using
   SCTP [RFC4960] encapsulated in 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 as specified in [RFC4960] with the extension defined in
   [RFC3758] provides multiple streams natively with reliable, and
   partially-reliable delivery modes.

   The remainder of this document is organized as follows: Section 3 and
   Section 4 provide requirements and use cases for both unreliable and
   reliable peer to peer datagram base channel; Section 5 arguments SCTP
   over DTLS over UDP; Section 6 provides an overview of how SCTP should
   be used by the RTCWeb protocol framework for transporting non-media
   data between browsers.

2.  Conventions

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

3.  Requirements

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

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   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

   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

   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 it MUST support a maximum application-layer message size
            of at least 2GB.

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   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.

   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.

4.  Use Cases

4.1.  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.

4.2.  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

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

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

   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.

5.  SCTP over DTLS over UDP Considerations

   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 as specified in [RFC4960] MUST be used in
   combination with the extension defined in [RFC3758] and provides the
   following 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 MAY demultiplex 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 Section 3.

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

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                                         | DATA |
                                         | SCTP |
                           | STUN | SRTP | DTLS |
                           |         ICE        |
                           | UDP1 | UDP2 | ...  |

                     Figure 2: WebRTC protocol layers

   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, the SCTP stack also
   needs to be a user-land stack.

   When using DTLS as the lower layer, only single homed SCTP
   associations SHOULD 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 SHOULD 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 required
   for supporting [I-D.ietf-rtcweb-qos].  Being able to set the (DF)-bit
   in case of IPv4 is required for performing path MTU discovery.  The
   DTLS implementation SHOULD also support sending user messages
   exceeding the path MTU.

   Incoming ICMP or ICMPv6 messages can't be processed by the SCTP
   layer, since there is no way to identify the corresponding

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   association.  Therefore SCTP MUST support performing Path MTU
   discovery without relying on ICMP or ICMPv6.  In general, the lower
   layer interface of an SCTP implementation SHOULD 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 MUST support the usage of multiple SCTP streams.
   A user message can be sent ordered or unordered and with partial or
   full reliability.  The partial reliability extension MUST support
   policies to limit

   o  the transmission and retransmission by time.

   o  the number of retransmissions.

   Limiting the number of retransmissions to zero combined with
   unordered delivery provides a UDP-like service where each user
   message is sent exactly once and delivered in the order received.

   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, a delay-sensitive congestion control algorithm
   MUST be supported and the congestion control MAY be coordinated
   between the data channels and the media streams to avoid a data
   channel transfer ending up with most or all the channel bandwidth.

   Since SCTP does not support the negotiation of a congestion control
   algorithm, the algorithm either MUST be negotiated before
   establishment of the SCTP association or MUST not require any
   negotiation because it only requires sender side behavior using
   existing information carried in the association.

6.  The Usage of SCTP in the RTCWeb Context

   The important 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.

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   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.

   SCTP multihoming will not be used in RTCWeb.  The SCTP layer will
   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) exposes.

6.1.  Association Setup

   The SCTP association will 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].  Additionally,
   the negotiation SHOULD include some type of congestion control
   selection.  It will use the DTLS connection selected via SDP;
   typically this will be shared via BUNDLE with DTLS connections used
   to key the DTLS-SRTP media streams.

   The application SHOULD indicate the initial number of streams
   required when opening the association, and if no value is supplied,
   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.  Note there can be up to 65536 SCTP streams per SCTP
   association in each direction.

6.2.  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 and for
   multiplexing.  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.

6.3.  Channel Definition

   The W3C has 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.  One strong wish is for the application-level API to be

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   close to the API for WebSockets, which implies bidirectional streams
   of data and waiting for onopen to fire before sending, a textual
   label used to identify the meaning of the stream, among other things.

   The realization of a bidirectional Data Channel is a pair of one
   incoming stream and one outgoing SCTP stream.

   The simple protocol specified in [I-D.jesup-rtcweb-data-protocol]
   MUST be used to set up and manage the bidirectional data channels.

   Note that there's no requirement for the SCTP streams used to create
   a bidirectional channel have the same number in each direction.  How
   stream values are selected is protocol and implementation dependent.

   Closing of a Data Channel MUST be signalled by 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.

6.4.  Usage of Payload Protocol Identifier

   The SCTP Payload Protocol Identifiers (PPIDs) MUST used to signal the
   interpretation of the "Payload data", like the protocol specified in
   [I-D.jesup-rtcweb-data-protocol] uses them to identify a Javascript
   string, a Javascript array or a a Javascript blob.

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 Harald Alvestrand,
   Adam Bergkvist, Cullen Jennings, Eric Rescorla, Randall Stewart, and
   Justin Uberti.

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10.  Informative References

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

   [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.

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

              Rescorla, E., "Security Considerations for RTC-Web",
              draft-ietf-rtcweb-security-03 (work in progress),
              June 2012.

              Rescorla, E., "RTCWEB Security Architecture",
              draft-ietf-rtcweb-security-arch-05 (work in progress),
              October 2012.

              Uberti, J. and C. Jennings, "Javascript Session
              Establishment Protocol", draft-ietf-rtcweb-jsep-01 (work
              in progress), June 2012.

              Dhesikan, S., Druta, D., Jones, P., and J. Polk, "DSCP and
              other packet markings for RTCWeb QoS",
              draft-ietf-rtcweb-qos-00 (work in progress), October 2012.

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              Tuexen, M. and R. Stewart, "UDP Encapsulation of SCTP
              Packets", draft-ietf-tsvwg-sctp-udp-encaps-06 (work in
              progress), October 2012.

              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel
              Protocol", draft-jesup-rtcweb-data-protocol-03 (work in
              progress), September 2012.

              Jesup, R., Loreto, S., Stewart, R., and M. Tuexen, "DTLS
              Encapsulation of SCTP Packets for RTCWEB",
              draft-tuexen-tsvwg-sctp-dtls-encaps-01 (work in progress),
              July 2012.

Authors' Addresses

   Randell Jesup


   Salvatore Loreto
   Hirsalantie 11
   Jorvas  02420


   Michael Tuexen
   Muenster University of Applied Sciences
   Stegerwaldstrasse 39
   Steinfurt  48565


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