diff --git a/DEPS b/DEPS
index 663ef809a0..33b653177a 100644
--- a/DEPS
+++ b/DEPS
@@ -2457,7 +2457,6 @@ include_rules = [
   "+absl/strings/match.h",
   "+absl/strings/str_replace.h",
   "+absl/strings/string_view.h",
-  "+absl/types/optional.h",
   "+absl/types/variant.h",
 
   # Abseil flags are allowed in tests and tools.
diff --git a/api/voip/DEPS b/api/voip/DEPS
index 3845dffab0..a36df41a9c 100644
--- a/api/voip/DEPS
+++ b/api/voip/DEPS
@@ -1,8 +1,4 @@
 specific_include_rules = {
-  ".*\.h": [
-    "+third_party/absl/types/optional.h",
-  ],
-
   "voip_engine_factory.h": [
     "+modules/audio_device/include/audio_device.h",
     "+modules/audio_processing/include/audio_processing.h",
diff --git a/g3doc/abseil-in-webrtc.md b/g3doc/abseil-in-webrtc.md
index f6c7f2efe3..7798f24b21 100644
--- a/g3doc/abseil-in-webrtc.md
+++ b/g3doc/abseil-in-webrtc.md
@@ -1,5 +1,5 @@
 <!-- go/cmark -->
-<!--* freshness: {owner: 'danilchap' reviewed: '2021-05-12'} *-->
+<!--* freshness: {owner: 'danilchap' reviewed: '2024-09-02'} *-->
 
 # Using Abseil in WebRTC
 
@@ -22,8 +22,6 @@ The GN templates will take care of generating the proper dependency when
 used within Chromium or standalone. In that build mode, WebRTC will depend
 on a monolithic Abseil build target that will generate a shared library.
 
-Previously, `absl_deps` was used but been deprecated in May 2024.
-
 ## **Allowed**
 
 * `absl::AnyInvocable`
@@ -47,6 +45,8 @@ Previously, `absl_deps` was used but been deprecated in May 2024.
   `absl/base/macros.h`.
 * `absl/numeric/bits.h`
 
+* ABSL_FLAG is allowed in tests and tools, but disallowed in in non-test code.
+
 
 ## **Disallowed**
 
@@ -58,12 +58,6 @@ Previously, `absl_deps` was used but been deprecated in May 2024.
 
 *Use `webrtc::Mutex` instead.*
 
-Chromium has a ban on new static initializers, and `absl::Mutex` uses
-one. To make `absl::Mutex` available, we would need to nicely ask the
-Abseil team to remove that initializer (like they already did for a
-spinlock initializer). Additionally, `absl::Mutex` handles time in a
-way that may not be compatible with the rest of WebRTC.
-
 ### `absl::optional`
 
 *Use `std::optional` instead.*
@@ -73,12 +67,9 @@ way that may not be compatible with the rest of WebRTC.
 *Use `rtc::ArrayView` instead.*
 
 `absl::Span` differs from `rtc::ArrayView` on several points, and both
-of them differ from the `std::span` that was voted into
-C++20&mdash;and `std::span` is likely to undergo further changes
-before C++20 is finalized. We should just keep using `rtc::ArrayView`
-and avoid `absl::Span` until C++20 is finalized and the Abseil team
-has decided if they will change `absl::Span` to match.
-[Bug](https://bugs.webrtc.org/9214).
+of them differ from the `std::span` introduced in C++20. We should just keep
+using `rtc::ArrayView` and avoid `absl::Span`. When WebRTC switches to C++20,
+we will consider replacing `rtc::ArrayView` with `std::span`.
 
 ### `absl::StrCat`, `absl::StrAppend`, `absl::StrJoin`, `absl::StrSplit`