diff --git a/.gn b/.gn
index db578de39d..7993d0d0f5 100644
--- a/.gn
+++ b/.gn
@@ -24,6 +24,7 @@ secondary_source = "//build/secondary/"
 check_targets = [
   "//webrtc/api/*",
   "//webrtc/audio/*",
+  "//webrtc/call/*",
   "//webrtc/modules/audio_coding/*",
   "//webrtc/modules/audio_conference_mixer/*",
   "//webrtc/modules/audio_device/*",
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 680ec6bd2f..95db035b48 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -19,6 +19,15 @@ rtc_source_set("call_interfaces") {
     "syncable.cc",
     "syncable.h",
   ]
+  deps = [
+    "..:webrtc_common",
+    "../api:audio_mixer_api",
+    "../api:transport_api",
+    "../api/audio_codecs:audio_codecs_api",
+    "../base:rtc_base",
+    "../base:rtc_base_approved",
+    "../modules/audio_coding:audio_encoder_interface",
+  ]
 }
 
 rtc_static_library("call") {
@@ -45,9 +54,13 @@ rtc_static_library("call") {
     "../api:transport_api",
     "../audio",
     "../base:rtc_task_queue",
+    "../logging:rtc_event_log_api",
     "../logging:rtc_event_log_impl",
+    "../modules/bitrate_controller",
     "../modules/congestion_controller",
+    "../modules/pacing",
     "../modules/rtp_rtcp",
+    "../modules/utility",
     "../system_wrappers",
     "../video",
   ]
@@ -65,9 +78,17 @@ if (rtc_include_tests) {
     deps = [
       ":call",
       "../base:rtc_base_approved",
+      "../logging:rtc_event_log_api",
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer",
+      "../modules/bitrate_controller",
+      "../modules/pacing",
+      "../modules/rtp_rtcp",
+      "../system_wrappers",
+      "../test:direct_transport",
       "../test:test_common",
+      "../test:test_support",
+      "../test:video_test_common",
       "//testing/gmock",
       "//testing/gtest",
     ]
@@ -85,6 +106,20 @@ if (rtc_include_tests) {
       "rampup_tests.h",
     ]
     deps = [
+      ":call_interfaces",
+      "..:webrtc_common",
+      "../base:rtc_base_approved",
+      "../logging:rtc_event_log_api",
+      "../modules/audio_coding",
+      "../modules/audio_mixer:audio_mixer_impl",
+      "../modules/rtp_rtcp",
+      "../system_wrappers",
+      "../system_wrappers:metrics_default",
+      "../test:direct_transport",
+      "../test:test_support",
+      "../test:video_test_common",
+      "../video",
+      "../voice_engine",
       "//testing/gtest",
       "//webrtc/test:test_common",
     ]