RTCWeb Working Group | R. Jesup |
Internet-Draft | Mozilla |
Intended status: Informational | S. Loreto |
Expires: September 05, 2012 | Ericsson |
M. Tuexen | |
Muenster Univ. of Appl. Sciences | |
March 06, 2012 |
RTCWeb Datagram Connection
draft-ietf-rtcweb-data-channel-00.txt
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.
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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 | +----------+
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.
This section lists the requirements for P2P data connections between two browsers.
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.
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:
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 | ... | +--------------------+
This stack (especially in contrast to DTLS over SCTP [RFC6083]) has been chosen because it
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 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.
The appealing features of SCTP in the RTCWeb context are:
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.
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, 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.
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.
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.
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.
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.
To be done.
This document does not require any actions by the IANA.
Many thanks for comments, ideas, and text from Cullen Jennings, Eric Rescorla, Randall Stewart, Justin Uberti, and Harald Alvestrand.