[webtrans-overview] Editorial fixes ietf-wg-webtrans/draft-ietf-webtrans-overview@0097229

Author: triple-underscore

webtrans-overview-ja.html

@@ -101,7 +101,7 @@
 
 ●●options
 
-spec_date:2025-10-20
+spec_date:2026-03-01
 trans_update:2025-10-23
 source_checked:230906
 original_url:https://ietf-wg-webtrans.github.io/draft-ietf-webtrans-overview/draft-ietf-webtrans-overview.html
@@ -388,7 +388,7 @@
 ●●ref_informative
 
 [CSP]
-    W3C, ＜Content Security Policy Level 3＞, October 2025
+    W3C, ＜Content Security Policy Level 3＞, March 2026
     URL: https://www.w3.org/TR/CSP/
 [RFC5681]
     Allman, M., Paxson, V., and E. Blanton, ＜TCP Congestion Control＞, ~RFC5681, September 2009
@@ -524,7 +524,7 @@ <h2 title="Status of This Memo">このメモの位置付け</h2>
 <h2 title="Copyright Notice">著作権の告知</h2>
 
 <p lang="en">
-Copyright (c) 2025 IETF Trust and the persons identified as the document authors. All rights reserved.
+Copyright (c) 2026 IETF Trust and the persons identified as the document authors. All rights reserved.
 </p>
 
 <p class="trans-note">【
@@ -599,7 +599,7 @@ <h3 title="Background">1.1. 背景0</h3>
 次に挙げるものへの依存性に因り，
 `browser-to-browser^en 設定群の外側では広い採用が見られない：
 ◎
-One existing option available to Web developers are WebRTC data channels [RFC8831], which provide a WebSocket-like API for a peer-to-peer SCTP channel protected by DTLS. In theory, it is possible to use it for the use cases addressed by this specification. However, in practice, it has not seen wide adoption outside of browser-to-browser settings due to its dependency on＼
+One existing option available to Web developers is WebRTC data channels [RFC8831], which provide a WebSocket-like API for a peer-to-peer SCTP channel protected by DTLS. In theory, it is possible to use it for the use cases addressed by this specification. However, in practice, it has not seen wide adoption outside of browser-to-browser settings due to its dependency on＼
 </p>
 <ul>
 	<li>
@@ -1022,7 +1022,7 @@ <h2 title="Common Transport Requirements">2. 共通な~transport要件</h2>
 ~IANAの`~ALPN~protocol~ID~registry＠~IANA-a/tls-extensiontype-values/tls-extensiontype-values.xhtml#alpn-protocol-ids$cite
 には登録されないことに注意。
 ◎
-All WebTransport protocols MUST provide a way for the session initiator to negotiate a subprotocol with the peer when establishing a WebTransport session. The session initiator provides an optional list of subprotocols to the peer. The peer selects one and responds indicating the selected subprotocol or rejects the session establishment request if none of the subprotocols are supported. Note that the semantics of individual subprotocol token values is determined by the WebTransport resource in question and are not registered in IANA's "ALPN Protocol IDs" registry.
+All WebTransport protocols MUST provide a way for the session initiator to negotiate a subprotocol with the peer when establishing a WebTransport session. The session initiator provides an optional list of subprotocols to the peer. The peer selects one and responds indicating the selected subprotocol or rejects the session establishment request if none of the subprotocols are supported. Note that the semantics of individual subprotocol token values are determined by the WebTransport resource in question and are not registered in IANA's "ALPN Protocol IDs" registry.
 </p>
 
 	</section>
@@ -1451,7 +1451,7 @@ <h3 title="Streams">4.3. ~stream</h3>
 この演算は、
 流入~一方向~stream用には適用-可能でない。
 ◎
-Add bytes into the stream send buffer. The sender can also indicate a FIN, signalling the fact that no new data will be send on the stream. Not applicable for incoming unidirectional streams.
+Add bytes into the stream send buffer. The sender can also indicate a FIN, signalling the fact that no new data will be sent on the stream. Not applicable for incoming unidirectional streams.
 </dd>
 
 	<dt id="receive-bytes">