diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index a3c5832fd9..9f7ea58d90 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -371,12 +371,6 @@ class WebRtcVoiceCodecs final {
           // Reset G722 sample rate to 16000 to match WebRTC.
           MaybeFixupG722(&voe_codec, 16000);
 
-          // Apply codec-specific settings.
-          if (IsCodec(codec, kIsacCodecName)) {
-            // If ISAC and an explicit bitrate is not specified,
-            // enable auto bitrate adjustment.
-            voe_codec.rate = (in.bitrate > 0) ? in.bitrate : -1;
-          }
           *out = voe_codec;
         }
         return true;
diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
index 60389342a8..eb6f606cf2 100644
--- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc
+++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc
@@ -1888,7 +1888,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) {
     const auto& gcodec = GetSendStreamConfig(kSsrc1).send_codec_spec.codec_inst;
     EXPECT_EQ(103, gcodec.pltype);
     EXPECT_STREQ("ISAC", gcodec.plname);
-    EXPECT_EQ(-1, gcodec.rate);
+    EXPECT_EQ(32000, gcodec.rate);
   }
   parameters.codecs[0].bitrate = 28000;     // bitrate == 28000
   SetSendParameters(parameters);
diff --git a/webrtc/pc/webrtcsdp.cc b/webrtc/pc/webrtcsdp.cc
index d4003c2a02..ced85cb323 100644
--- a/webrtc/pc/webrtcsdp.cc
+++ b/webrtc/pc/webrtcsdp.cc
@@ -205,11 +205,6 @@ static const char kApplicationSpecificMaximum[] = "AS";
 
 static const int kDefaultVideoClockrate = 90000;
 
-// ISAC special-case.
-static const char kIsacCodecName[] = "ISAC";  // From webrtcvoiceengine.cc
-static const int kIsacWbDefaultRate = 32000;  // From acm_common_defs.h
-static const int kIsacSwbDefaultRate = 56000;  // From acm_common_defs.h
-
 static const char kDefaultSctpmapProtocol[] = "webrtc-datachannel";
 
 // RTP payload type is in the 0-127 range. Use -1 to indicate "all" payload
@@ -3098,21 +3093,9 @@ bool ParseRtpmapAttribute(const std::string& line,
         return false;
       }
     }
-    int bitrate = 0;
-    // The default behavior for ISAC (bitrate == 0) in webrtcvoiceengine.cc
-    // (specifically FindWebRtcCodec) is bandwidth-adaptive variable bitrate.
-    // The bandwidth adaptation doesn't always work well, so this code
-    // sets a fixed target bitrate instead.
-    if (_stricmp(encoding_name.c_str(), kIsacCodecName) == 0) {
-      if (clock_rate <= 16000) {
-        bitrate = kIsacWbDefaultRate;
-      } else {
-        bitrate = kIsacSwbDefaultRate;
-      }
-    }
     AudioContentDescription* audio_desc =
         static_cast<AudioContentDescription*>(media_desc);
-    UpdateCodec(payload_type, encoding_name, clock_rate, bitrate, channels,
+    UpdateCodec(payload_type, encoding_name, clock_rate, 0, channels,
                 audio_desc);
   } else if (media_type == cricket::MEDIA_TYPE_DATA) {
     DataContentDescription* data_desc =
diff --git a/webrtc/pc/webrtcsdp_unittest.cc b/webrtc/pc/webrtcsdp_unittest.cc
index 54eb22e11a..43774abf7d 100644
--- a/webrtc/pc/webrtcsdp_unittest.cc
+++ b/webrtc/pc/webrtcsdp_unittest.cc
@@ -1128,8 +1128,8 @@ class WebRtcSdpTest : public testing::Test {
     audio->set_protocol(cricket::kMediaProtocolSavpf);
     AudioCodec opus(111, "opus", 48000, 0, 2);
     audio->AddCodec(opus);
-    audio->AddCodec(AudioCodec(103, "ISAC", 16000, 32000, 1));
-    audio->AddCodec(AudioCodec(104, "ISAC", 32000, 56000, 1));
+    audio->AddCodec(AudioCodec(103, "ISAC", 16000, 0, 1));
+    audio->AddCodec(AudioCodec(104, "ISAC", 32000, 0, 1));
     return audio;
   }
 
@@ -1664,13 +1664,6 @@ class WebRtcSdpTest : public testing::Test {
       cricket::AudioCodec codec = acd->codecs()[i];
       VerifyCodecParameter(codec.params, "ptime", params.ptime);
       VerifyCodecParameter(codec.params, "maxptime", params.max_ptime);
-      if (codec.name == "ISAC") {
-        if (codec.clockrate == 16000) {
-          EXPECT_EQ(32000, codec.bitrate);
-        } else {
-          EXPECT_EQ(56000, codec.bitrate);
-        }
-      }
     }
 
     const ContentInfo* vc = GetFirstVideoContent(jdesc_output->description());
@@ -2275,7 +2268,7 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutRtpmap) {
   // the payload types (<fmt>s) on the m= line.
   ref_codecs.push_back(AudioCodec(0, "PCMU", 8000, 0, 1));
   ref_codecs.push_back(AudioCodec(18, "G729", 16000, 0, 1));
-  ref_codecs.push_back(AudioCodec(103, "ISAC", 16000, 32000, 1));
+  ref_codecs.push_back(AudioCodec(103, "ISAC", 16000, 0, 1));
   EXPECT_EQ(ref_codecs, audio->codecs());
 }