Network Working Group                                           R. JesupInternet-Draft                                                   MozillaIntended status: Standards Track                               S. LoretoExpires: July 8, 2015                                           Ericsson                                                               M. Tuexen                                        Muenster Univ. of Appl. Sciences                                                         January 4, 2015                          WebRTC Data Channels                 draft-ietf-rtcweb-data-channel-13.txtAbstract   The WebRTC framework specifies protocol support for direct   interactive rich communication using audio, video, and data between   two peers' web-browsers.  This document specifies 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-browsers 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 July 8, 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 ofJesup, et al.             Expires July 8, 2015                  [Page 1]Internet-Draft            WebRTC Data Channels              January 2015   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.  Conventions . . . . . . . . . . . . . . . . . . . . . . . . .   3   3.  Use Cases . . . . . . . . . . . . . . . . . . . . . . . . . .   3     3.1.  Use Cases for Unreliable Data Channels  . . . . . . . . .   4     3.2.  Use Cases for Reliable Data Channels  . . . . . . . . . .   4   4.  Requirements  . . . . . . . . . . . . . . . . . . . . . . . .   4   5.  SCTP over DTLS over UDP Considerations  . . . . . . . . . . .   6   6.  The Usage of SCTP for Data Channels . . . . . . . . . . . . .   8     6.1.  SCTP Protocol Considerations  . . . . . . . . . . . . . .   8     6.2.  SCTP Association Management . . . . . . . . . . . . . . .   9     6.3.  SCTP Streams  . . . . . . . . . . . . . . . . . . . . . .   9     6.4.  Data Channel Definition . . . . . . . . . . . . . . . . .  10     6.5.  Opening a Data Channel  . . . . . . . . . . . . . . . . .  10     6.6.  Transferring User Data on a Data Channel  . . . . . . . .  11     6.7.  Closing a Data Channel  . . . . . . . . . . . . . . . . .  12   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  13   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  13   9.  Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .  14   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  14     10.1.  Normative References . . . . . . . . . . . . . . . . . .  14     10.2.  Informative References . . . . . . . . . . . . . . . . .  15   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  161.  Introduction   In the WebRTC framework, communication between the parties consists   of media (for example audio and video) and non-media data.  Media is   sent using SRTP, and is not specified further here.  Non-media data   is handled by using SCTP [RFC4960] encapsulated in DTLS.  DTLS 1.0 is   defined in [RFC4347] and the present latest version, DTLS 1.2, is   defined in [RFC6347].Jesup, et al.             Expires July 8, 2015                  [Page 2]Internet-Draft            WebRTC Data Channels              January 2015                               +----------+                               |   SCTP   |                               +----------+                               |   DTLS   |                               +----------+                               | ICE/UDP  |                               +----------+                       Figure 1: Basic stack diagram   The encapsulation of SCTP over DTLS (see   [I-D.ietf-tsvwg-sctp-dtls-encaps]) over ICE/UDP (see [RFC5245])   provides a NAT traversal solution together with confidentiality,   source authentication, and integrity protected transfers.  This data   transport service operates in parallel to the SRTP media transports,   and all of them can eventually share a single UDP port number.   SCTP as specified in [RFC4960] with the partial reliability extension   defined in [RFC3758] and the additional policies defined in   [I-D.ietf-tsvwg-sctp-prpolicies] provides multiple streams natively   with reliable, and the relevant partially-reliable delivery modes for   user messages.  Using the reconfiguration extension defined in   [RFC6525] allows to increase the number of streams during the   lifetime of an SCTP association and to reset individual SCTP streams.   Using [I-D.ietf-tsvwg-sctp-ndata] allows to interleave large messages   to avoid the monopolization and adds the support of prioritizing of   SCTP streams.   The remainder of this document is organized as follows: Section 3 and   Section 4 provide use cases and requirements for both unreliable and   reliable peer to peer data channels; Section 5 discusses SCTP over   DTLS over UDP; Section 6 provides the specification of how SCTP   should be used by the WebRTC protocol framework for transporting non-   media data between WEB-browsers.2.  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.  Use Cases   This section defines use cases specific to data channels.  Please   note that this section is informational only.Jesup, et al.             Expires July 8, 2015                  [Page 3]Internet-Draft            WebRTC Data Channels              January 20153.1.  Use Cases for Unreliable Data 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 SRTP media channels, or all SRTP           media channels may be inactive, and that there may also be           reliable data channels in use.   U-C 2:  Providing non-critical information to a user about the reason           for a state update in a video chat or conference, such as           mute state.3.2.  Use Cases for Reliable Data Channels   U-C 3:  A real-time game where critical state information needs to be           transferred, such as control information.  Such a game may           have no SRTP 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.  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 during an audio and/or video call with an           individual or with multiple people in a conference.   U-C 6:  Renegotiation of the configuration of the PeerConnection.   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.4.  Requirements   This section lists the requirements for P2P data channels between two   browsers.  Please note that this section is informational only.   Req. 1:   Multiple simultaneous data channels must be supported.             Note that there may be 0 or more SRTP media streams in             parallel with the data channels in the same PeerConnection,             and the number and state (active/inactive) of these SRTP             media streams may change at any time.   Req. 2:   Both reliable and unreliable data channels must be             supported.Jesup, et al.             Expires July 8, 2015                  [Page 4]Internet-Draft            WebRTC Data Channels              January 2015   Req. 3:   Data channels of a PeerConnection must be congestion             controlled; either individually, as a class, or in             conjunction with the SRTP media streams of the             PeerConnection, to ensure that data channels don't cause             congestion problems for these SRTP media streams, and that             the WebRTC PeerConnection does not cause excessive problems             when run in parallel with TCP connections.   Req. 4:   The application should be able to provide guidance as to             the relative priority of each data channel relative to each             other, and relative to the SRTP media streams.  This will             interact with the congestion control algorithms.   Req. 5:   Data channels must be secured; 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:   Data channels must provide message fragmentation support             such that IP-layer fragmentation can be avoided no matter             how large a message the JavaScript application passes to be             sent.  It also must ensure that large data channel             transfers don't unduly delay traffic on other data             channels.   Req. 7:   The data channel 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. 8:   The data channel transport protocol should support             unbounded-length "messages" (i.e., a virtual socket stream)             at the application layer, for such things as image-file-             transfer; Implementations might enforce a reasonable             message size limit.   Req. 9:   The data channel transport protocol should avoid IP             fragmentation.  It must support PMTU (Path MTU) discovery             and must not rely on ICMP or ICMPv6 being generated or             being passed back, especially for PMTU discovery.   Req. 10:  It must be possible to implement the protocol stack in the             user application space.Jesup, et al.             Expires July 8, 2015                  [Page 5]Internet-Draft            WebRTC Data Channels              January 20155.  SCTP over DTLS over UDP Considerations   The important features of SCTP in the WebRTC context are:   o  Usage of a TCP-friendly congestion control.   o  The congestion control is modifiable for integration with the SRTP      media stream congestion control.   o  Support of multiple unidirectional streams, each providing its own      notion of ordered message delivery.   o  Support of ordered and out-of-order message delivery.   o  Supporting arbitrary large user messages by providing      fragmentation and reassembly.   o  Support of PMTU-discovery.   o  Support of reliable or partially reliable message transport.   The WebRTC Data Channel mechanism does not support SCTP multihoming.   The SCTP layer will simply act as if it were running on a single-   homed host, since that is the abstraction that the DTLS layer (a   connection oriented, unreliable datagram service) exposes.   The encapsulation of SCTP over DTLS defined in   [I-D.ietf-tsvwg-sctp-dtls-encaps] provides confidentiality, source   authenticated, and integrity protected transfers.  Using DTLS over   UDP in combination with ICE enables middlebox traversal in IPv4 and   IPv6 based networks.  SCTP as specified in [RFC4960] MUST be used in   combination with the extension defined in [RFC3758] and provides the   following features for transporting non-media data between browsers:   o  Support of multiple unidirectional streams.   o  Ordered and unordered delivery of user messages.   o  Reliable and partial-reliable transport of user messages.   Each SCTP user message contains a Payload Protocol Identifier (PPID)   that is passed to SCTP by its upper layer on the sending side and   provided to its upper layer on the receiving side.  The PPID can be   used to multiplex/demultiplex multiple upper layers over a single   SCTP association.  In the WebRTC context, the PPID is used to   distinguish between UTF-8 encoded user data, binary encoded userdata   and the Data Channel Establishment Protocol defined inJesup, et al.             Expires July 8, 2015                  [Page 6]Internet-Draft            WebRTC Data Channels              January 2015   [I-D.ietf-rtcweb-data-protocol].  Please note that the PPID is not   accessible via the Javascript API.   The encapsulation of SCTP over DTLS, together with the SCTP features   listed above satisfies all the requirements listed in Section 4.   The layering of protocols for WebRTC is shown in the following   Figure 2.                                 +------+------+------+                                 | DCEP | UTF-8|Binary|                                 |      | data | data |                                 +------+------+------+                                 |        SCTP        |                   +----------------------------------+                   | STUN | SRTP |        DTLS        |                   +----------------------------------+                   |                ICE               |                   +----------------------------------+                   | UDP1 | UDP2 | UDP3 | ...         |                   +----------------------------------+                     Figure 2: WebRTC protocol layers   This stack (especially in contrast to DTLS over SCTP [RFC6083] in   combination with SCTP over UDP [RFC6951]) has been chosen because it   o  supports the transmission of arbitrary large user messages.   o  shares the DTLS connection with the SRTP media channels of the      PeerConnection.   o  provides privacy for the SCTP control information.   Considering the protocol stack of Figure 2 the usage of DTLS 1.0 over   UDP is specified in [RFC4347] and the usage of DTLS 1.2 over UDP in   specified in [RFC6347], while the usage of SCTP on top of DTLS is   specified in [I-D.ietf-tsvwg-sctp-dtls-encaps].  Please note that the   demultiplexing STUN vs. SRTP vs. DTLS is done as described in   Section 5.1.2 of [RFC5764] and SCTP is the only payload of DTLS.   Since DTLS is typically implemented in user application space, the   SCTP stack also needs to be a user application space stack.   The ICE/UDP layer can handle IP address changes during a session   without needing interaction with the DTLS and SCTP layers.  However,   SCTP SHOULD be notified when an address changes has happened.  In   this case SCTP SHOULD retest the Path MTU and reset the congestionJesup, et al.             Expires July 8, 2015                  [Page 7]Internet-Draft            WebRTC Data Channels              January 2015   state to the initial state.  In case of a window based congestion   control like the one specified in [RFC4960], this means setting the   congestion window and slow start threshold to its initial values.   Incoming ICMP or ICMPv6 messages can't be processed by the SCTP   layer, since there is no way to identify the corresponding   association.  Therefore SCTP MUST support performing Path MTU   discovery without relying on ICMP or ICMPv6 as specified in [RFC4821]   using probing messages specified in [RFC4820].  The initial Path MTU   at the IP layer SHOULD NOT exceed 1200 bytes for IPv4 and 1280 for   IPv6.   In general, the lower layer interface of an SCTP implementation   should be adapted to address the differences between IPv4 and IPv6   (being connection-less) or DTLS (being connection-oriented).   When the 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.   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.  Using a congestion control   different from than the standard one might improve the impact on the   parallel SRTP media streams.   SCTP uses the same port number concept as TCP and UDP do.  Therefore   an SCTP association uses two port numbers, one at each SCTP end-   point.6.  The Usage of SCTP for Data Channels6.1.  SCTP Protocol Considerations   The DTLS encapsulation of SCTP packets as described in   [I-D.ietf-tsvwg-sctp-dtls-encaps] MUST be used.   This SCTP stack and its upper layer MUST support the usage of   multiple SCTP streams.  A user message can be sent ordered or   unordered and with partial or full reliability.   The following SCTP protocol extensions are required:   o  The stream reconfiguration extension defined in [RFC6525] MUST be      supported.  It is used for closing channels.Jesup, et al.             Expires July 8, 2015                  [Page 8]Internet-Draft            WebRTC Data Channels              January 2015   o  The dynamic address reconfiguration extension defined in [RFC5061]      MUST be used to signal the support of the stream reset extension      defined in [RFC6525].  Other features of [RFC5061] are OPTIONAL.   o  The partial reliability extension defined in [RFC3758] MUST be      supported.  In addition to the timed reliability PR-SCTP policy      defined in [RFC3758], the limited retransmission policy defined in      [I-D.ietf-tsvwg-sctp-prpolicies] MUST be supported.  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.   The support for message interleaving as defined in   [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used.6.2.  SCTP Association Management   In the WebRTC context, 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].  It will use the DTLS connection selected via   ICE; typically this will be shared via BUNDLE or equivalent with DTLS   connections used to key the SRTP media streams.   The number of streams negotiated during SCTP association setup SHOULD   be 65535, which is the maximum number of streams that can be   negotiated during the association setup.   SCTP supports two ways of terminating an SCTP association.  A   graceful one, using a procedure which ensures that no messages are   lost during the shutdown of the association.  The second method is a   non-graceful one, where one side can just abort the association.   Each SCTP end-point supervises continuously the reachability of its   peer by monitoring the number of retransmissions of user messages and   test messages.  In case of excessive retransmissions, the association   is terminated in a non-graceful way.   If an SCTP association is closed in a graceful way, all of its data   channels are closed.  In case of a non-graceful teardown, all data   channels are also closed, but an error indication SHOULD be provided   if possible.6.3.  SCTP Streams   SCTP defines a stream as a unidirectional logical channel existing   within an SCTP association to another SCTP endpoint.  The streams are   used to provide the notion of in-sequence delivery and forJesup, et al.             Expires July 8, 2015                  [Page 9]Internet-Draft            WebRTC Data Channels              January 2015   multiplexing.  Each user message is sent on a particular stream,   either ordered or unordered.  Ordering is preserved only for ordered   messages sent on the same stream.6.4.  Data Channel Definition   Data channels are defined such that their accompanying application-   level API can closely mirror the API for WebSockets, which implies   bidirectional streams of data and a textual field called 'label' used   to identify the meaning of the data channel.   The realization of a data channel is a pair of one incoming stream   and one outgoing SCTP stream having the same SCTP stream identifier.   How these SCTP stream identifiers are selected is protocol and   implementation dependent.  This allows a bidirectional communication.   Additionally, each data channel has the following properties in each   direction:   o  reliable or unreliable message transmission.  In case of      unreliable transmissions, the same level of unreliability is used.      Please note that in SCTP this is a property of an SCTP user      message and not of an SCTP stream.   o  in-order or out-of-order message delivery for message sent.      Please note that in SCTP this is a property of an SCTP user      message and not of an SCTP stream.   o  A priority, which is a 2 byte unsigned integer.  These priorities      MUST be interpreted as weighted-fair-queuing scheduling priorities      per the definition of the corresponding stream scheduler      supporting interleaving in [I-D.ietf-tsvwg-sctp-ndata].  For use      in WebRTC, the values used SHOULD be one of 128 ("below normal"),      256 ("normal"), 512 ("high") or 1024 ("extra high").   o  an optional label.   o  an optional protocol.   Please note that for a data channel being negotiated with the   protocol specified in [I-D.ietf-rtcweb-data-protocol] all of the   above properties are the same in both directions.6.5.  Opening a Data Channel   Data channels can be opened by using negotiation within the SCTP   association, called in-band negotiation, or out-of-band negotiation.   Out-of-band negotiation is defined as any method which results in anJesup, et al.             Expires July 8, 2015                 [Page 10]Internet-Draft            WebRTC Data Channels              January 2015   agreement as to the parameters of a channel and the creation thereof.   The details are out of scope of this document.  Applications using   data channels need to use the negotiation methods consistently on   both end-points.   A simple protocol for in-band negotiation is specified in   [I-D.ietf-rtcweb-data-protocol].   When one side wants to open a channel using out-of-band negotiation,   it picks a stream.  Unless otherwise defined or negotiated, the   streams are picked based on the DTLS role (the client picks even   stream identifiers, the server odd stream identifiers).  However, the   application is responsible for avoiding collisions with existing   streams.  If it attempts to re-use a stream which is part of an   existing data channel, the addition MUST fail.  In addition to   choosing a stream, the application SHOULD also determine the options   to use for sending messages.  The application MUST ensure in an   application-specific manner that the application at the peer will   also know the selected stream to be used, and the options for sending   data from that side.6.6.  Transferring User Data on a Data Channel   All data sent on a data channel in both directions MUST be sent over   the underlying stream using the reliability defined when the data   channel was opened unless the options are changed, or per-message   options are specified by a higher level.   The message-orientation of SCTP is used to preserve the message   boundaries of user messages.  Therefore, senders MUST NOT put more   than one application message into an SCTP user message.  Unless the   deprecated PPID-based fragmentation and reassembly is used, the   sender MUST include exactly one application message in each SCTP user   message.   The SCTP Payload Protocol Identifiers (PPIDs) are used to signal the   interpretation of the "Payload data".  The following PPIDs MUST be   used (see Section 8):   WebRTC String:  to identify a non-empty JavaScript string encoded in      UTF-8.   WebRTC String Empty:  to identify an empty JavaScript string encoded      in UTF-8.   WebRTC Binary:  to identify a non-empty JavaScript binary data      (ArrayBuffer, ArrayBufferView or Blob).Jesup, et al.             Expires July 8, 2015                 [Page 11]Internet-Draft            WebRTC Data Channels              January 2015   WebRTC Binary Empty:  to identify an empty JavaScript binary data      (ArrayBuffer, ArrayBufferView or Blob).   SCTP does not support the sending of empty user messages.  Therefore,   if an empty message has to be sent, the appropriate PPID (WebRTC   String Empty or WebRTC Binary Empty) is used and the SCTP user   message of one zero byte is sent.  When receiving an SCTP user   message with one of these PPIDs, the receiver MUST ignore the SCTP   user message and process it as an empty message.   The usage of the PPIDs "WebRTC String Partial" and "WebRTC Binary   Partial" is deprecated.  They were used for a PPID-based   fragmentation and reassembly of user messages belonging to reliable   and ordered data channels.   If a message with an unsupported PPID is received or some error   condition related to the received message is detected by the receiver   (for example, illegal ordering), the receiver SHOULD close the   corresponding data channel.  This implies in particular that   extensions using additional PPIDs can't be used without prior   negotiation.   The SCTP base protocol specified in [RFC4960] does not support the   interleaving of user messages.  Therefore sending a large user   message can monopolize the SCTP association.  To overcome this   limitation, [I-D.ietf-tsvwg-sctp-ndata] defines an extension to   support message interleaving, which SHOULD be used.  As long as   message interleaving is not supported, the sender SHOULD limit the   maximum message size to 16 KB to avoid monopolization.   It is recommended that the message size be kept within certain size   bounds as applications will not be able to support arbitrarily-large   single messages.  This limit has to be negotiated, for example by   using [I-D.ietf-mmusic-sctp-sdp].   The sender SHOULD disable the Nagle algorithm (see [RFC1122]) to   minimize the latency.6.7.  Closing a Data Channel   Closing of a data channel MUST be signaled by resetting the   corresponding outgoing streams [RFC6525].  This means that if one   side decides to close the data channel, it resets the corresponding   outgoing stream.  When the peer sees that an incoming stream was   reset, it also resets its corresponding outgoing stream.  Once this   is completed, the data channel is closed.  Resetting a stream sets   the Stream Sequence Numbers (SSNs) of the stream back to 'zero' with   a corresponding notification to the application layer that the resetJesup, et al.             Expires July 8, 2015                 [Page 12]Internet-Draft            WebRTC Data Channels              January 2015   has been performed.  Streams are available for reuse after a reset   has been performed.   [RFC6525] also guarantees that all the messages are delivered (or   abandoned) before the stream is reset.7.  Security Considerations   This document does not add any additional considerations to the ones   given in [I-D.ietf-rtcweb-security] and   [I-D.ietf-rtcweb-security-arch].   It should be noted that a receiver must be prepared that the sender   tries to send arbitrary large messages.8.  IANA Considerations   [NOTE to RFC-Editor:      "RFCXXXX" is to be replaced by the RFC number you assign this      document.   ]   This document uses six already registered SCTP Payload Protocol   Identifiers (PPIDs): "DOMString Last", "Binary Data Partial", "Binary   Data Last", "DOMString Partial", "WebRTC String Empty", and "WebRTC   Binary Empty".  [RFC4960] creates the registry "SCTP Payload Protocol   Identifiers" from which these identifiers were assigned.  IANA is   requested to update the reference of these six assignments to point   to this document and change the names of the first four PPIDs.  The   corresponding dates should be kept.   Therefore these six assignments should be updated to read:   +-------------------------------+----------+-----------+------------+   | Value                         | SCTP     | Reference | Date       |   |                               | PPID     |           |            |   +-------------------------------+----------+-----------+------------+   | WebRTC String                 | 51       | [RFCXXXX] | 2013-09-20 |   | WebRTC Binary Partial         | 52       | [RFCXXXX] | 2013-09-20 |   | (Deprecated)                  |          |           |            |   | WebRTC Binary                 | 53       | [RFCXXXX] | 2013-09-20 |   | WebRTC String Partial         | 54       | [RFCXXXX] | 2013-09-20 |   | (Deprecated)                  |          |           |            |   | WebRTC String Empty           | 56       | [RFCXXXX] | 2014-08-22 |   | WebRTC Binary Empty           | 57       | [RFCXXXX] | 2014-08-22 |   +-------------------------------+----------+-----------+------------+Jesup, et al.             Expires July 8, 2015                 [Page 13]Internet-Draft            WebRTC Data Channels              January 20159.  Acknowledgments   Many thanks for comments, ideas, and text from Harald Alvestrand,   Richard Barnes, Adam Bergkvist, Alissa Cooper, Benoit Claise, Spencer   Dawkins, Gunnar Hellstrom, Christer Holmberg, Cullen Jennings, Paul   Kyzivat, Eric Rescorla, Adam Roach, Irene Ruengeler, Randall Stewart,   Martin Stiemerling, Justin Uberti, and Magnus Westerlund.10.  References10.1.  Normative 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.   [RFC4347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer              Security", RFC 4347, April 2006.   [RFC4820]  Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and              Parameter for the Stream Control Transmission Protocol              (SCTP)", RFC 4820, March 2007.   [RFC4821]  Mathis, M. and J. Heffner, "Packetization Layer Path MTU              Discovery", RFC 4821, March 2007.   [RFC4960]  Stewart, R., "Stream Control Transmission Protocol", RFC              4960, September 2007.   [RFC5061]  Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M.              Kozuka, "Stream Control Transmission Protocol (SCTP)              Dynamic Address Reconfiguration", RFC 5061, 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.   [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.Jesup, et al.             Expires July 8, 2015                 [Page 14]Internet-Draft            WebRTC Data Channels              January 2015   [I-D.ietf-tsvwg-sctp-ndata]              Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann,              "Stream Schedulers and a New Data Chunk for the Stream              Control Transmission Protocol", draft-ietf-tsvwg-sctp-              ndata-01 (work in progress), July 2014.   [I-D.ietf-rtcweb-data-protocol]              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel              Establishment Protocol", draft-ietf-rtcweb-data-              protocol-08 (work in progress), September 2014.   [I-D.ietf-tsvwg-sctp-dtls-encaps]              Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "DTLS              Encapsulation of SCTP Packets", draft-ietf-tsvwg-sctp-              dtls-encaps-07 (work in progress), December 2014.   [I-D.ietf-rtcweb-security]              Rescorla, E., "Security Considerations for WebRTC", draft-              ietf-rtcweb-security-07 (work in progress), July 2014.   [I-D.ietf-rtcweb-security-arch]              Rescorla, E., "WebRTC Security Architecture", draft-ietf-              rtcweb-security-arch-10 (work in progress), July 2014.   [I-D.ietf-rtcweb-jsep]              Uberti, J., Jennings, C., and E. Rescorla, "Javascript              Session Establishment Protocol", draft-ietf-rtcweb-jsep-08              (work in progress), October 2014.   [I-D.ietf-tsvwg-sctp-prpolicies]              Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto,              "Additional Policies for the Partial Reliability Extension              of the Stream Control Transmission Protocol", draft-ietf-              tsvwg-sctp-prpolicies-06 (work in progress), December              2014.   [I-D.ietf-mmusic-sctp-sdp]              Holmberg, C., Loreto, S., and G. Camarillo, "Stream              Control Transmission Protocol (SCTP)-Based Media Transport              in the Session Description Protocol (SDP)", draft-ietf-              mmusic-sctp-sdp-11 (work in progress), December 2014.10.2.  Informative References   [RFC1122]  Braden, R., "Requirements for Internet Hosts -              Communication Layers", STD 3, RFC 1122, October 1989.Jesup, et al.             Expires July 8, 2015                 [Page 15]Internet-Draft            WebRTC Data Channels              January 2015   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer              Security (DTLS) Extension to Establish Keys for the Secure              Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.   [RFC6083]  Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram              Transport Layer Security (DTLS) for Stream Control              Transmission Protocol (SCTP)", RFC 6083, January 2011.   [RFC6951]  Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream              Control Transmission Protocol (SCTP) Packets for End-Host              to End-Host Communication", RFC 6951, May 2013.Authors' Addresses   Randell Jesup   Mozilla   US   Email: randell-ietf@jesup.org   Salvatore Loreto   Ericsson   Hirsalantie 11   Jorvas  02420   FI   Email: salvatore.loreto@ericsson.com   Michael Tuexen   Muenster University of Applied Sciences   Stegerwaldstrasse 39   Steinfurt  48565   DE   Email: tuexen@fh-muenster.deJesup, et al.             Expires July 8, 2015                 [Page 16]