1. 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 the Secure Real-time Transport Protocol (SRTP) and is not specified further here. Non-media data is handled by using the Stream Control Transmission Protocol (SCTP) [RFC4960] encapsulated in DTLS. DTLS 1.0 is defined in [RFC4347]; the present latest version, DTLS 1.2, is defined in [RFC6347]; and an upcoming version, DTLS 1.3, is defined in [TLS-DTLS13].

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

Figure 1: Basic Stack Diagram

The encapsulation of SCTP over DTLS (see [RFC8261]) over ICE/UDP (see [RFC8445]) 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 (PR-SCTP) defined in [RFC3758] and the additional policies defined in [RFC7496], provides multiple streams natively with reliable, and the relevant partially reliable, delivery modes for user messages. Using the reconfiguration extension defined in [RFC6525] allows an increase in the number of streams during the lifetime of an SCTP association and allows individual SCTP streams to be reset. Using [RFC8260] allows the interleave of large messages to avoid monopolization and adds support for prioritizing SCTP streams.

The remainder of this document is organized as follows: Sections 3 and 4 provide use cases and requirements for both unreliable and reliable peer-to-peer data channels; Section 5 discusses SCTP over DTLS over UDP; and Section 6 specifies how SCTP should be used by the WebRTC protocol framework for transporting non-media data between web browsers.