Internet Engineering Task Force (IETF)                          R. Jesup
Request for Comments: 8831                                       Mozilla
Category: Standards Track                                      S. Loreto
ISSN: 2070-1721                                                 Ericsson
                                                                M. Tüxen
                                         Münster Univ. of Appl. Sciences
                                                            January 2021

WebRTC Data Channels




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 that allows web browsers to exchange generic data from peer to peer.

WebRTCフレームワークは、2つのピアのWebブラウザー間でオーディオ、ビデオ、およびデータを使用した、直接、インタラクティブ、リッチな通信のプロトコルサポートを指定します。このドキュメントでは、WebRTCフレームワークのメディア以外のデータ転送の側面について説明します。これは、Webブラウザがピアツーピアで汎用データを交換できるようにする汎用トランスポートサービスとして、WebRTCコンテキストでStream Control Transmission Protocol(SCTP)がどのように使用されるかについてのアーキテクチャの概要を提供します。

Status of This Memo


This is an Internet Standards Track document.


This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Further information on Internet Standards is available in Section 2 of RFC 7841.

このドキュメントは、インターネット技術特別調査委員会(IETF)の製品です。これは、IETFコミュニティのコンセンサスを表しています。パブリックレビューを受け、Internet Engineering Steering Group(IESG)による公開が承認されました。インターネット標準の詳細については、RFC7841のセクション2をご覧ください。

Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at


Copyright Notice


Copyright (c) 2021 IETF Trust and the persons identified as the document authors. All rights reserved.

Copyright(c)2021 IETFTrustおよびドキュメントの作成者として識別された人物。全著作権所有。

This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents ( in effect on the date of 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.

このドキュメントは、このドキュメントの発行日に有効なBCP 78およびIETFドキュメントに関連するIETFトラストの法的規定(の対象となります。これらのドキュメントは、このドキュメントに関するお客様の権利と制限について説明しているため、注意深く確認してください。このドキュメントから抽出されたコードコンポーネントには、Trust LegalProvisionsのセクション4.eで説明されているSimplifiedBSD Licenseテキストが含まれている必要があり、Simplified BSDLicenseで説明されているように保証なしで提供されます。

Table of Contents


   1.  Introduction
   2.  Conventions
   3.  Use Cases
     3.1.  Use Cases for Unreliable Data Channels
     3.2.  Use Cases for Reliable Data Channels
   4.  Requirements
   5.  SCTP over DTLS over UDP Considerations
   6.  The Usage of SCTP for Data Channels
     6.1.  SCTP Protocol Considerations
     6.2.  SCTP Association Management
     6.3.  SCTP Streams
     6.4.  Data Channel Definition
     6.5.  Opening a Data Channel
     6.6.  Transferring User Data on a Data Channel
     6.7.  Closing a Data Channel
   7.  Security Considerations
   8.  IANA Considerations
   9.  References
     9.1.  Normative References
     9.2.  Informative References
   Authors' Addresses
1. Introduction
1. はじめに

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].

WebRTCフレームワークでは、当事者間の通信はメディア(オーディオやビデオなど)と非メディアデータで構成されます。メディアはSecureReal-time Transport Protocol(SRTP)を使用して送信されるため、ここでは特に指定しません。非メディアデータは、DTLSにカプセル化されたStream Control Transmission Protocol(SCTP)[RFC4960]を使用して処理されます。DTLS1.0は[RFC4347]で定義されています。現在の最新バージョンであるDTLS1.2は、[RFC6347]で定義されています。今後のバージョンであるDTLS1.3は、[TLS-DTLS13]で定義されています。

                               |   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.

ICE / UDP([RFC8445]を参照)を介したDTLS([RFC8261]を参照)を介したSCTPのカプセル化は、機密性、ソース認証、および整合性保護された転送とともにNATトラバーサルソリューションを提供します。このデータトランスポートサービスはSRTPメディアトランスポートと並行して動作し、最終的にはすべてが単一のUDPポート番号を共有できます。

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.

このドキュメントの残りの部分は次のように構成されています。セクション3と4は、信頼性の低いピアツーピアデータチャネルの使用例と要件を示しています。セクション5では、SCTP over DTLS overUDPについて説明します。セクション6は、Webブラウザ間で非メディアデータを転送するためにWebRTCプロトコルフレームワークがSCTPをどのように使用するかを指定します。

2. Conventions
2. 規約

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.

キーワード「MUST」、「MUST NOT」、「REQUIRED」、「SHALL」、「SHALL NOT」、「SHOULD」、「SHOULD NOT」、「RECOMMENDED」、「NOT RECOMMENDED」、「MAY」、「OPTIONAL」「このドキュメントでは、BCP 14 [RFC2119] [RFC8174]で説明されているように、ここに示すように、すべて大文字で表示される場合にのみ解釈されます。

3. Use Cases
3. ユースケース

This section defines use cases specific to data channels. Please note that this section is informational only.


3.1. Use Cases for Unreliable Data Channels
3.1. 信頼性の低いデータチャネルのユースケース

U-C 1: A real-time game where position and object state information are sent via one or more unreliable data channels. Note that at any time, there may not be any SRTP media channels or all SRTP media channels may be inactive, and there may also be reliable data channels in use.

U-C 1:位置とオブジェクトの状態情報が1つ以上の信頼できないデータチャネルを介して送信されるリアルタイムゲーム。いつでも、SRTPメディアチャネルがないか、すべてのSRTPメディアチャネルが非アクティブである可能性があり、信頼できるデータチャネルが使用されている可能性があることに注意してください。

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.

U-C 2:ミュート状態など、ビデオチャットまたは会議での状態更新の理由に関する重要ではない情報をユーザーに提供します。

3.2. Use Cases for Reliable Data Channels
3.2. 信頼性の高いデータチャネルのユースケース

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 3:制御情報などの重要な状態情報を転送する必要があるリアルタイムゲーム。このようなゲームにはSRTPメディアチャネルがない場合や、常に非アクティブである場合、またはゲーム内のアクションによってのみ追加される場合があります。

U-C 4: Non-real-time 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 4:チャットしている人々の間の非リアルタイムのファイル転送。これには、画像のフォルダやファイルのディレクトリを共有する場合など、順次または並行して転送する多数のファイルが含まれる場合があることに注意してください。

U-C 5: Real-time text chat during an audio and/or video call with an individual or with multiple people in a conference.

U-C 5:個人または会議の複数の人との音声通話および/またはビデオ通話中のリアルタイムテキストチャット。

U-C 6: Renegotiation of the configuration of the PeerConnection.

U-C 6:PeerConnectionの構成の再ネゴシエーション。

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.

U-C 7:プロキシブラウジング。ブラウザはPeerConnectionのデータチャネルを使用してHTTP / HTTPS要求とデータを送受信し、たとえば、ローカルインターネットのフィルタリングや監視を回避します。

4. Requirements
4. 要件

This section lists the requirements for Peer-to-Peer (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 zero 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.


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. This ensures 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, which allows for confidentiality, integrity, and source authentication. See [RFC8826] and [RFC8827] for detailed information.


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 Path MTU (PMTU) 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.


5. SCTP over DTLS over UDP Considerations
5. SCTP over DTLS overUDPの考慮事項

The important features of SCTP in the WebRTC context are the following:


* Usage of TCP-friendly congestion control.

* TCP対応の輻輳制御の使用。

* modifiable congestion control for integration with the SRTP media stream congestion control.

* SRTPメディアストリーム輻輳制御と統合するための変更可能な輻輳制御。

* Support of multiple unidirectional streams, each providing its own notion of ordered message delivery.

* 複数の単方向ストリームのサポート。それぞれが順序付けられたメッセージ配信の独自の概念を提供します。

* Support of ordered and out-of-order message delivery.

* 順序付きおよび順序外のメッセージ配信のサポート。

* Support of arbitrarily large user messages by providing fragmentation and reassembly.

* 断片化と再構築を提供することにより、任意に大きなユーザーメッセージをサポートします。

* Support of PMTU discovery.

* PMTUディスカバリーのサポート。

* Support of reliable or partially reliable message transport.

* 信頼できるまたは部分的に信頼できるメッセージ転送のサポート。

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 [RFC8261] provides confidentiality, source authentication, and integrity-protected transfers. Using DTLS over UDP in combination with Interactive Connectivity Establishment (ICE) [RFC8445] 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:

[RFC8261]で定義されているSCTPover DTLSのカプセル化は、機密性、ソース認証、および整合性保護された転送を提供します。DTLS over UDPをInteractiveConnectivity Establishment(ICE)[RFC8445]と組み合わせて使用すると、IPv4およびIPv6ベースのネットワークでミドルボックストラバーサルが可能になります。[RFC4960]で指定されているSCTPは、[RFC3758]で定義されている拡張機能と組み合わせて使用する必要があり、ブラウザ間で非メディアデータを転送するための次の機能を提供します。

* Support of multiple unidirectional streams.

* 複数の単方向ストリームのサポート。

* Ordered and unordered delivery of user messages.

* ユーザーメッセージの順序付きおよび順序なしの配信。

* Reliable and partially reliable transport of user messages.

* ユーザーメッセージの信頼性と部分的に信頼性の高い転送。

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 user data, and the Data Channel Establishment Protocol (DCEP) defined in [RFC8832]. 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 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] and in combination with SCTP over UDP [RFC6951]) has been chosen for the following reasons:

このスタック(特にDTLS over SCTP [RFC6083]とは対照的に、SCTP over UDP [RFC6951]との組み合わせ)は、次の理由で選択されました。

* supports the transmission of arbitrarily large user messages;

* 任意の大きなユーザーメッセージの送信をサポートします。

* shares the DTLS connection with the SRTP media channels of the PeerConnection; and

* PeerConnectionのSRTPメディアチャネルとDTLS接続を共有します。そして

* provides privacy for the SCTP control information.

* SCTP制御情報のプライバシーを提供します。

Referring to the protocol stack shown in Figure 2:


* the usage of DTLS 1.0 over UDP is specified in [RFC4347];

* UDPを介したDTLS1.0の使用法は、[RFC4347]で指定されています。

* the usage of DTLS 1.2 over UDP in specified in [RFC6347];

* [RFC6347]で指定されているUDPを介したDTLS1.2の使用。

* the usage of DTLS 1.3 over UDP is specified in an upcoming document [TLS-DTLS13]; and

* UDPを介したDTLS1.3の使用法は、今後のドキュメント[TLS-DTLS13]で指定されています。そして

* the usage of SCTP on top of DTLS is specified in [RFC8261].

* DTLS上でのSCTPの使用法は、[RFC8261]で指定されています。

Please note that the demultiplexing Session Traversal Utilities for NAT (STUN) [RFC5389] 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 change has happened. In this case, SCTP SHOULD retest the Path MTU and reset the congestion state to the initial state. In the case of window-based congestion control like the one specified in [RFC4960], this means setting the congestion window and slow-start threshold to its initial values.

ICE / UDPレイヤーは、DTLSおよびSCTPレイヤーとの対話を必要とせずに、セッション中のIPアドレスの変更を処理できます。ただし、アドレス変更が発生した場合は、SCTPに通知する必要があります。この場合、SCTPはパスMTUを再テストし、輻輳状態を初期状態にリセットする必要があります。[RFC4960]で指定されているようなウィンドウベースの輻輳制御の場合、これは輻輳ウィンドウとスロースタートしきい値を初期値に設定することを意味します。

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] by using probing messages specified in [RFC4820]. The initial Path MTU at the IP layer SHOULD NOT exceed 1200 bytes for IPv4 and 1280 bytes for IPv6.


In general, the lower-layer interface of an SCTP implementation should be adapted to address the differences between IPv4 and IPv6 (being connectionless) or DTLS (being connection oriented).


When the protocol stack shown in 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 basis. 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 SCTP congestion control. Using a congestion control different from the standard one might improve the impact on the parallel SRTP media streams.


SCTP uses the same port number concept as TCP and UDP. Therefore, an SCTP association uses two port numbers, one at each SCTP endpoint.


6. The Usage of SCTP for Data Channels
6. データチャネルでのSCTPの使用
6.1. SCTP Protocol Considerations
6.1. SCTPプロトコルに関する考慮事項

The DTLS encapsulation of SCTP packets as described in [RFC8261] 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:


* The stream reconfiguration extension defined in [RFC6525] MUST be supported. It is used for closing channels.

* [RFC6525]で定義されているストリーム再構成拡張機能をサポートする必要があります。チャネルを閉じるために使用されます。

* 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.

* [RFC5061]で定義されている動的アドレス再構成拡張機能は、[RFC6525]で定義されているストリームリセット拡張機能のサポートを通知するために使用する必要があります。[RFC5061]の他の機能はオプションです。

* 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 [RFC7496] 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.

* [RFC3758]で定義されている部分的な信頼性拡張をサポートする必要があります。[RFC3758]で定義されている時限信頼性PR-SCTPポリシーに加えて、[RFC7496]で定義されている制限付き再送信ポリシーをサポートする必要があります。再送信の数をゼロに制限し、順序付けられていない配信と組み合わせると、UDPのようなサービスが提供され、各ユーザーメッセージが1回だけ送信され、受信した順序で配信されます。

The support for message interleaving as defined in [RFC8260] SHOULD be used.


6.2. SCTP Association Management
6.2. SCTPアソシエーション管理

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 the JavaScript Session Establishment Protocol (JSEP), which is typically an exchange of the Session Description Protocol (SDP) [RFC8829]. It will use the DTLS connection selected via ICE, and 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. The first method is a graceful one, where a procedure that ensures no messages are lost during the shutdown of the association is used. The second method is a non-graceful one, where one side can just abort the association.


Each SCTP endpoint continuously supervises 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
6.3. SCTPストリーム

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 for 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
6.4. データチャネルの定義

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:


* reliable or unreliable message transmission: In case of unreliable transmissions, the same level of unreliability is used. Note that, in SCTP, this is a property of an SCTP user message and not of an SCTP stream.

* 信頼性または信頼性の低いメッセージ送信:信頼性の低い送信の場合、同じレベルの信頼性が使用されません。SCTPでは、これはSCTPユーザーメッセージのプロパティであり、SCTPストリームのプロパティではないことに注意してください。

* in-order or out-of-order message delivery for message sent: Note that, in SCTP, this is a property of an SCTP user message and not of an SCTP stream.

* 送信されたメッセージの順不同または順不同のメッセージ配信:SCTPでは、これはSCTPユーザーメッセージのプロパティであり、SCTPストリームのプロパティではないことに注意してください。

* 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 [RFC8260]. For use in WebRTC, the values used SHOULD be one of 128 ("below normal"), 256 ("normal"), 512 ("high"), or 1024 ("extra high").

* 2バイトの符号なし整数である優先度:これらの優先度は、[RFC8260]でインターリーブをサポートする対応するストリームスケジューラの定義に従って、重み付き公平キューイングのスケジューリング優先度として解釈する必要があります。WebRTCで使用する場合、使用する値は128(「通常より低い」)、256(「通常」)、512(「高」)、または1024(「超高」)のいずれかである必要があります。

* an optional label.

* オプションのラベル。

* an optional protocol.

* オプションのプロトコル。

Note that for a data channel being negotiated with the protocol specified in [RFC8832], all of the above properties are the same in both directions.


6.5. Opening a Data Channel
6.5. データチャネルを開く

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 that results in an 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 endpoints.


A simple protocol for in-band negotiation is specified in [RFC8832].


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, and the server picks odd stream identifiers). However, the application is responsible for avoiding collisions with existing streams. If it attempts to reuse a stream that 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 be used 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, as well as the options for sending data from that side.


6.6. Transferring User Data on a Data Channel
6.6. データチャネルでのユーザーデータの転送

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 String Empty:UTF-8でエンコードされた空のJavaScript文字列を識別します。

WebRTC Binary: to identify non-empty JavaScript binary data (ArrayBuffer, ArrayBufferView, or Blob).


WebRTC Binary Empty: to identify empty JavaScript binary data (ArrayBuffer, ArrayBufferView, or Blob).

WebRTC Binary Empty:空のJavaScriptバイナリデータ(ArrayBuffer、ArrayBufferView、またはBlob)を識別します。

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.

SCTPは、空のユーザーメッセージの送信をサポートしていません。したがって、空のメッセージを送信する必要がある場合は、適切なPPID(WebRTC StringEmptyまたはWebRTCBinary Empty)が使用され、1つのゼロバイトのSCTPユーザーメッセージが送信されます。これらのPPIDのいずれかを含むSCTPユーザーメッセージを受信する場合、受信者はSCTPユーザーメッセージを無視し、それを空のメッセージとして処理する必要があります。

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, [RFC8260] 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 [RFC8841].


The sender SHOULD disable the Nagle algorithm (see [RFC1122]) to minimize the latency.


6.7. Closing a Data Channel
6.7. データチャネルを閉じる

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 reset 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
7. セキュリティに関する考慮事項

This document does not add any additional considerations to the ones given in [RFC8826] and [RFC8827].


It should be noted that a receiver must be prepared for a sender that tries to send arbitrarily large messages.


8. IANA Considerations
8. IANAの考慮事項

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 "SCTP Payload Protocol Identifiers" registry from which these identifiers were assigned. IANA has updated the reference of these six assignments to point to this document and changed the names of the first four PPIDs. The corresponding dates remain unchanged.

このドキュメントでは、すでに登録されている6つのSCTPペイロードプロトコル識別子(PPID)を使用しています。「DOMStringLast」、「Binary Data Partial」、「Binary Data Last」、「DOMString Partial」、「WebRTC String Empty」、「WebRTCBinaryEmpty」です。[RFC4960]は、これらの識別子が割り当てられた「SCTPペイロードプロトコル識別子」レジストリを作成します。IANAは、このドキュメントを指すようにこれら6つの割り当ての参照を更新し、最初の4つのPPIDの名前を変更しました。対応する日付は変更されません。

The six assignments have been updated to read:


       | Value                | SCTP PPID | Reference | Date       |
       | WebRTC String        | 51        | RFC 8831  | 2013-09-20 |
       | WebRTC Binary        | 52        | RFC 8831  | 2013-09-20 |
       | Partial (deprecated) |           |           |            |
       | WebRTC Binary        | 53        | RFC 8831  | 2013-09-20 |
       | WebRTC String        | 54        | RFC 8831  | 2013-09-20 |
       | Partial (deprecated) |           |           |            |
       | WebRTC String Empty  | 56        | RFC 8831  | 2014-08-22 |
       | WebRTC Binary Empty  | 57        | RFC 8831  | 2014-08-22 |

Table 1


9. References
9. 参考文献
9.1. Normative References
9.1. 引用文献

[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997, <>.

[RFC2119] Bradner、S。、「要件レベルを示すためにRFCで使用するキーワード」、BCP 14、RFC 2119、DOI 10.17487 / RFC2119、1997年3月、<>。

[RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P. Conrad, "Stream Control Transmission Protocol (SCTP) Partial Reliability Extension", RFC 3758, DOI 10.17487/RFC3758, May 2004, <>.

[RFC3758] Stewart、R.、Ramalho、M.、Xie、Q.、Tuexen、M。、およびP. Conrad、「Stream Control Transmission Protocol(SCTP)Partial Reliability Extension」、RFC 3758、DOI 10.17487 / RFC3758、5月2004年、<>。

[RFC4820] Tuexen, M., Stewart, R., and P. Lei, "Padding Chunk and Parameter for the Stream Control Transmission Protocol (SCTP)", RFC 4820, DOI 10.17487/RFC4820, March 2007, <>.

[RFC4820] Tuexen、M.、Stewart、R。、およびP. Lei、「ストリーム制御伝送プロトコル(SCTP)のパディングチャンクとパラメーター」、RFC 4820、DOI 10.17487 / RFC4820、2007年3月、<>。

[RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007, <>.

[RFC4821] Mathis、M。およびJ. Heffner、「Packetization Layer Path MTU Discovery」、RFC 4821、DOI 10.17487 / RFC4821、2007年3月、<>。

[RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol", RFC 4960, DOI 10.17487/RFC4960, September 2007, <>.

[RFC4960] Stewart、R.、Ed。、 "Stream Control Transmission Protocol"、RFC 4960、DOI 10.17487 / RFC4960、2007年9月、<>。

[RFC5061] Stewart, R., Xie, Q., Tuexen, M., Maruyama, S., and M. Kozuka, "Stream Control Transmission Protocol (SCTP) Dynamic Address Reconfiguration", RFC 5061, DOI 10.17487/RFC5061, September 2007, <>.

[RFC5061] Stewart、R.、Xie、Q.、Tuexen、M.、Maruyama、S。、およびM. Kozuka、「Stream Control Transmission Protocol(SCTP)Dynamic Address Reconfiguration」、RFC 5061、DOI 10.17487 / RFC5061、9月2007、<>。

[RFC6525] Stewart, R., Tuexen, M., and P. Lei, "Stream Control Transmission Protocol (SCTP) Stream Reconfiguration", RFC 6525, DOI 10.17487/RFC6525, February 2012, <>.

[RFC6525] Stewart、R.、Tuexen、M。、およびP. Lei、「Stream Control Transmission Protocol(SCTP)Stream Reconfiguration」、RFC 6525、DOI 10.17487 / RFC6525、2012年2月、<>。

[RFC7496] Tuexen, M., Seggelmann, R., Stewart, R., and S. Loreto, "Additional Policies for the Partially Reliable Stream Control Transmission Protocol Extension", RFC 7496, DOI 10.17487/RFC7496, April 2015, <>.

[RFC7496] Tuexen、M.、Seggelmann、R.、Stewart、R。、およびS. Loreto、「部分的に信頼できるStream Control Transmission Protocol Extensionの追加ポリシー」、RFC 7496、DOI 10.17487 / RFC7496、2015年4月、<>。

[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174, May 2017, <>.

[RFC8174] Leiba、B。、「RFC 2119キーワードにおける大文字と小文字のあいまいさ」、BCP 14、RFC 8174、DOI 10.17487 / RFC8174、2017年5月、<>。

[RFC8260] Stewart, R., Tuexen, M., Loreto, S., and R. Seggelmann, "Stream Schedulers and User Message Interleaving for the Stream Control Transmission Protocol", RFC 8260, DOI 10.17487/RFC8260, November 2017, <>.

[RFC8260] Stewart、R.、Tuexen、M.、Loreto、S。、およびR. Seggelmann、「Stream Control Transmission Protocolのストリームスケジューラとユーザーメッセージインターリーブ」、RFC 8260、DOI 10.17487 / RFC8260、2017年11月、<>。

[RFC8261] Tuexen, M., Stewart, R., Jesup, R., and S. Loreto, "Datagram Transport Layer Security (DTLS) Encapsulation of SCTP Packets", RFC 8261, DOI 10.17487/RFC8261, November 2017, <>.

[RFC8261] Tuexen、M.、Stewart、R.、Jesup、R。、およびS. Loreto、「SCTPパケットのデータグラムトランスポート層セキュリティ(DTLS)カプセル化」、RFC 8261、DOI 10.17487 / RFC8261、2017年11月、<>。

[RFC8445] Keranen, A., Holmberg, C., and J. Rosenberg, "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal", RFC 8445, DOI 10.17487/RFC8445, July 2018, <>.

[RFC8445] Keranen、A.、Holmberg、C。、およびJ. Rosenberg、「Interactive Connectivity Establishment(ICE):A Protocol for Network Address Translator(NAT)Traversal」、RFC 8445、DOI 10.17487 / RFC8445、2018年7月、<>。

[RFC8826] Rescorla, E., "Security Considerations for WebRTC", RFC 8826, DOI 10.17487/RFC8826, January 2021, <>.

[RFC8826] Rescorla、E。、「WebRTCのセキュリティに関する考慮事項」、RFC 8826、DOI 10.17487 / RFC8826、2021年1月、<>。

[RFC8827] Rescorla, E., "WebRTC Security Architecture", RFC 8827, DOI 10.17487/RFC8827, January 2021, <>.

[RFC8827] Rescorla、E。、「WebRTC Security Architecture」、RFC 8827、DOI 10.17487 / RFC8827、2021年1月、<>。

[RFC8829] Uberti, J., Jennings, C., and E. Rescorla, Ed., "JavaScript Session Establishment Protocol (JSEP)", RFC 8829, DOI 10.17487/RFC8829, January 2021, <>.

[RFC8829] Uberti、J.、Jennings、C。、およびE. Rescorla、Ed。、「JavaScript Session Establishment Protocol(JSEP)」、RFC 8829、DOI 10.17487 / RFC8829、2021年1月、<>。

[RFC8832] Jesup, R., Loreto, S., and M. Tüxen, "WebRTC Data Channel Establishment Protocol", RFC 8832, DOI 10.17487/RFC8832, January 2021, <>.

[RFC8832] Jesup、R.、Loreto、S。、およびM.Tüxen、「WebRTC Data Channel Establishment Protocol」、RFC 8832、DOI 10.17487 / RFC8832、2021年1月、< / rfc8832>。

[RFC8841] Holmberg, C., Shpount, R., Loreto, S., and G. Camarillo, "Session Description Protocol (SDP) Offer/Answer Procedures for Stream Control Transmission Protocol (SCTP) over Datagram Transport Layer Security (DTLS) Transport", RFC 8841, DOI 10.17487/RFC8841, January 2021, <>.

[RFC8841] Holmberg、C.、Shpount、R.、Loreto、S。、およびG. Camarillo、「Datagram Transport Layer Security(DTLS)を介したStream Control Transmission Protocol(SCTP)のセッション記述プロトコル(SDP)オファー/アンサー手順トランスポート」、RFC 8841、DOI 10.17487 / RFC8841、2021年1月、<>。

9.2. Informative References
9.2. 参考引用

[RFC1122] Braden, R., Ed., "Requirements for Internet Hosts - Communication Layers", STD 3, RFC 1122, DOI 10.17487/RFC1122, October 1989, <>.

[RFC1122] Braden、R.、Ed。、 "Requirements for Internet Hosts-Communication Layers"、STD 3、RFC 1122、DOI 10.17487 / RFC1122、October 1989、<>。

[RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer Security", RFC 4347, DOI 10.17487/RFC4347, April 2006, <>.

[RFC4347] Rescorla、E。およびN. Modadugu、「Datagram Transport Layer Security」、RFC 4347、DOI 10.17487 / RFC4347、2006年4月、<>。

[RFC5389] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, "Session Traversal Utilities for NAT (STUN)", RFC 5389, DOI 10.17487/RFC5389, October 2008, <>.

[RFC5389] Rosenberg、J.、Mahy、R.、Matthews、P。、およびD. Wing、「Session Traversal Utilities for NAT(STUN)」、RFC 5389、DOI 10.17487 / RFC5389、2008年10月、<>。

[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, DOI 10.17487/RFC5764, May 2010, <>.

[RFC5764] McGrew、D。およびE. Rescorla、「Secure Real-time Transport Protocol(SRTP)のキーを確立するためのDatagram Transport Layer Security(DTLS)Extension」、RFC 5764、DOI 10.17487 / RFC5764、2010年5月、<>。

[RFC6083] Tuexen, M., Seggelmann, R., and E. Rescorla, "Datagram Transport Layer Security (DTLS) for Stream Control Transmission Protocol (SCTP)", RFC 6083, DOI 10.17487/RFC6083, January 2011, <>.

[RFC6083] Tuexen、M.、Seggelmann、R。、およびE. Rescorla、「Stream Control Transmission Protocol(SCTP)のデータグラムトランスポート層セキュリティ(DTLS)」、RFC 6083、DOI 10.17487 / RFC6083、2011年1月、<>。

[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347, January 2012, <>.

[RFC6347] Rescorla、E。およびN. Modadugu、「Datagram Transport Layer Security Version 1.2」、RFC 6347、DOI 10.17487 / RFC6347、2012年1月、<>。

[RFC6951] Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream Control Transmission Protocol (SCTP) Packets for End-Host to End-Host Communication", RFC 6951, DOI 10.17487/RFC6951, May 2013, <>.

[RFC6951] Tuexen、M。and R. Stewart、 "UDP Encapsulation of Stream Control Transmission Protocol(SCTP)Packets for End-Host to End-Host Communication"、RFC 6951、DOI 10.17487 / RFC6951、May 2013、<>。

[TLS-DTLS13] Rescorla, E., Tschofenig, H., and N. Modadugu, "The Datagram Transport Layer Security (DTLS) Protocol Version 1.3", Work in Progress, Internet-Draft, draft-ietf-tls-dtls13-39, 2 November 2020, <>.

[TLS-DTLS13] Rescorla、E.、Tschofenig、H。、およびN. Modadugu、「データグラムトランスポート層セキュリティ(DTLS)プロトコルバージョン1.3」、進行中の作業、インターネットドラフト、draft-ietf-tls-dtls13-39、2020年11月2日、<>。



Many thanks for comments, ideas, and text from Harald Alvestrand, Richard Barnes, Adam Bergkvist, Alissa Cooper, Benoit Claise, Spencer Dawkins, Gunnar Hellström, Christer Holmberg, Cullen Jennings, Paul Kyzivat, Eric Rescorla, Adam Roach, Irene Rüngeler, Randall Stewart, Martin Stiemerling, Justin Uberti, and Magnus Westerlund.

Harald Alvestrand、Richard Barnes、Adam Bergkvist、Alissa Cooper、Benoit Claise、Spencer Dawkins、GunnarHellström、Christer Holmberg、Cullen Jennings、Paul Kyzivat、Eric Rescorla、Adam Roach、IreneRüngeler、Randallからのコメント、アイデア、テキストに感謝します。スチュワート、マーティンスティーマーリング、ジャスティンウベルティ、マグナスウェスターランド。

Authors' Addresses


Randell Jesup Mozilla United States of America

Randell JesupMozillaアメリカ合衆国


Salvatore Loreto Ericsson Hirsalantie 11 FI-02420 Jorvas Finland



Michael Tüxen Münster University of Applied Sciences Stegerwaldstrasse 39 48565 Steinfurt Germany