--- 1/draft-ietf-rtcweb-data-channel-08.txt 2014-05-15 12:14:17.143439516 -0700 +++ 2/draft-ietf-rtcweb-data-channel-09.txt 2014-05-15 12:14:17.175440295 -0700 @@ -1,49 +1,49 @@ Network Working Group R. Jesup Internet-Draft Mozilla Intended status: Standards Track S. Loreto -Expires: October 11, 2014 Ericsson +Expires: November 16, 2014 Ericsson M. Tuexen Muenster Univ. of Appl. Sciences - April 9, 2014 + May 15, 2014 WebRTC Data Channels - draft-ietf-rtcweb-data-channel-08.txt + draft-ietf-rtcweb-data-channel-09.txt Abstract The Real-Time Communication in WEB-browsers 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 specifies the non-(S)RTP media data transport aspects of the + document specifies the non-SRTP 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 October 11, 2014. + This Internet-Draft will expire on November 16, 2014. Copyright Notice Copyright (c) 2014 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 publication of this document. Please review these documents @@ -73,87 +73,87 @@ 7. Security Considerations . . . . . . . . . . . . . . . . . . . 11 8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 11 9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 12 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 12 10.1. Normative References . . . . . . . . . . . . . . . . . . 12 10.2. Informative References . . . . . . . . . . . . . . . . . 14 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 14 1. Introduction - Non-(S)RTP media data types in the context of WebRTC are handled by + Non-SRTP media data types in the context of WebRTC 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 (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 (S)RTP media - transports, and all of them can eventually share a single transport- - layer port number. + transport service operates in parallel to the SRTP media transports, + and all of them can eventually share a single transport-layer 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-(S)RTP media data between WEB-browsers. + should be used by the WebRTC protocol framework for transporting non- + SRTP 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. For general use cases see [I-D.ietf-rtcweb-use-cases-and-requirements]. 3.1. 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 (S)RTP media channels, or all - (S)RTP media channels may be inactive, and that there may - also be reliable data channels in use. + 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 (S)RTP media channels, or they may be inactive at any + 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. @@ -163,39 +163,39 @@ 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. Req. 1: Multiple simultaneous data channels MUST be supported. - Note that there may be 0 or more (S)RTP media streams in + 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 (S)RTP + 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. Req. 3: Data channels of a PeerConnection MUST be congestion controlled; either individually, as a class, or in - conjunction with the (S)RTP media streams of the + conjunction with the SRTP media streams of the PeerConnection, to ensure that data channels don't cause - congestion problems for these (S)RTP media streams, and - that the WebRTC PeerConnection as a whole is fair with - competing traffic such as TCP. + congestion problems for these SRTP media streams, and that + the WebRTC PeerConnection as a whole is fair with competing + traffic such as TCP. 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 (S)RTP media streams. This will + 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 @@ -221,22 +221,22 @@ Req. 10: It MUST be possible to implement the protocol stack in the user application space. 5. 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 - (S)RTP media stream 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. @@ -247,21 +247,21 @@ 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. 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-(S)RTP media data between + following features for transporting non-SRTP 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 @@ -292,29 +292,29 @@ | UDP1 | UDP2 | ... | +----------------------------------+ 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 (S)RTP media channels of the + 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 over UDP is 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. (S)RTP vs. DTLS is done as described in + 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-land, the SCTP stack also needs to be a user-land stack. When using DTLS as the lower layer, only single homed SCTP associations are supported, 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 interaction with the DTLS and SCTP layers. However, SCTP SHOULD be notified when an @@ -351,21 +351,21 @@ 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. Using a congestion control different from than the standard one might improve the impact on the - parallel (S)RTP media streams. + parallel SRTP media streams. 6. The Usage of SCTP for Data Channels 6.1. SCTP Protocol Considerations The DTLS encapsulation of SCTP packets as described in [I-D.ietf-tsvwg-sctp-dtls-encaps] MUST be used. The following SCTP protocol extensions are required: @@ -384,21 +384,21 @@ The support for message interleaving as defined in [I-D.ietf-tsvwg-sctp-ndata] SHOULD be used. 6.2. 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]. 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 (S)RTP + 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 negotiated during the association setup. 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 @@ -585,26 +585,26 @@ [I-D.ietf-tsvwg-sctp-ndata] Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, "A New Data Chunk for Stream Control Transmission Protocol", draft-ietf-tsvwg-sctp-ndata-00 (work in progress), February 2014. [I-D.ietf-rtcweb-data-protocol] Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data Channel Establishment Protocol", draft-ietf-rtcweb-data- - protocol-03 (work in progress), February 2014. + protocol-04 (work in progress), April 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-03 (work in progress), February 2014. + dtls-encaps-04 (work in progress), May 2014. [I-D.ietf-rtcweb-security] Rescorla, E., "Security Considerations for WebRTC", draft- ietf-rtcweb-security-06 (work in progress), January 2014. [I-D.ietf-rtcweb-security-arch] Rescorla, E., "WebRTC Security Architecture", draft-ietf- rtcweb-security-arch-09 (work in progress), February 2014. [I-D.ietf-rtcweb-jsep]