diff --git a/media/base/codec.h b/media/base/codec.h
index fd8a97c5e4..c3be2334ce 100644
--- a/media/base/codec.h
+++ b/media/base/codec.h
@@ -67,6 +67,8 @@ struct RTC_EXPORT Codec {
   int id;
   std::string name;
   int clockrate;
+  // Non key-value parameters such as the telephone-event "0‐15" are
+  // represented using an empty string as key, i.e. {"": "0-15"}.
   CodecParameterMap params;
   FeedbackParams feedback_params;
 
diff --git a/pc/webrtc_sdp.cc b/pc/webrtc_sdp.cc
index 3188e3a956..af584791be 100644
--- a/pc/webrtc_sdp.cc
+++ b/pc/webrtc_sdp.cc
@@ -1762,8 +1762,13 @@ void WriteRtcpFbHeader(int payload_type, rtc::StringBuilder* os) {
 void WriteFmtpParameter(const std::string& parameter_name,
                         const std::string& parameter_value,
                         rtc::StringBuilder* os) {
-  // fmtp parameters: |parameter_name|=|parameter_value|
-  *os << parameter_name << kSdpDelimiterEqual << parameter_value;
+  if (parameter_name == "") {
+    // RFC 2198 and RFC 4733 don't use key-value pairs.
+    *os << parameter_value;
+  } else {
+    // fmtp parameters: |parameter_name|=|parameter_value|
+    *os << parameter_name << kSdpDelimiterEqual << parameter_value;
+  }
 }
 
 bool IsFmtpParam(const std::string& name) {
@@ -3603,8 +3608,10 @@ bool ParseFmtpParam(const std::string& line,
                     std::string* value,
                     SdpParseError* error) {
   if (!rtc::tokenize_first(line, kSdpDelimiterEqualChar, parameter, value)) {
-    ParseFailed(line, "Unable to parse fmtp parameter. \'=\' missing.", error);
-    return false;
+    // Support for non-key-value lines like RFC 2198 or RFC 4733.
+    *parameter = "";
+    *value = line;
+    return true;
   }
   // a=fmtp:<payload_type> <param1>=<value1>; <param2>=<value2>; ...
   return true;
@@ -3622,7 +3629,7 @@ bool ParseFmtpAttributes(const std::string& line,
   std::string line_payload;
   std::string line_params;
 
-  // RFC 5576
+  // https://tools.ietf.org/html/rfc4566#section-6
   // a=fmtp:<format> <format specific parameters>
   // At least two fields, whereas the second one is any of the optional
   // parameters.
@@ -3651,17 +3658,15 @@ bool ParseFmtpAttributes(const std::string& line,
 
   cricket::CodecParameterMap codec_params;
   for (auto& iter : fields) {
-    if (iter.find(kSdpDelimiterEqual) == std::string::npos) {
-      // Only fmtps with equals are currently supported. Other fmtp types
-      // should be ignored. Unknown fmtps do not constitute an error.
-      continue;
-    }
-
     std::string name;
     std::string value;
     if (!ParseFmtpParam(rtc::string_trim(iter), &name, &value, error)) {
       return false;
     }
+    if (codec_params.find(name) != codec_params.end()) {
+      RTC_LOG(LS_INFO) << "Overwriting duplicate fmtp parameter with key \""
+                       << name << "\".";
+    }
     codec_params[name] = value;
   }
 
diff --git a/pc/webrtc_sdp_unittest.cc b/pc/webrtc_sdp_unittest.cc
index 49fc0063cd..7b83c86ab1 100644
--- a/pc/webrtc_sdp_unittest.cc
+++ b/pc/webrtc_sdp_unittest.cc
@@ -1293,8 +1293,7 @@ class WebRtcSdpTest : public ::testing::Test {
         "inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32",
         "dummy_session_params"));
     audio->set_protocol(cricket::kMediaProtocolSavpf);
-    AudioCodec opus(111, "opus", 48000, 0, 2);
-    audio->AddCodec(opus);
+    audio->AddCodec(AudioCodec(111, "opus", 48000, 0, 2));
     audio->AddCodec(AudioCodec(103, "ISAC", 16000, 0, 1));
     audio->AddCodec(AudioCodec(104, "ISAC", 32000, 0, 1));
     return audio;
@@ -1934,13 +1933,14 @@ class WebRtcSdpTest : public ::testing::Test {
         // description.
         "a=msid-semantic: WMS\r\n"
         // Pl type 111 preferred.
-        "m=audio 9 RTP/SAVPF 111 104 103\r\n"
+        "m=audio 9 RTP/SAVPF 111 104 103 105\r\n"
         // Pltype 111 listed before 103 and 104 in the map.
         "a=rtpmap:111 opus/48000/2\r\n"
         // Pltype 103 listed before 104.
         "a=rtpmap:103 ISAC/16000\r\n"
         "a=rtpmap:104 ISAC/32000\r\n"
-        "a=fmtp:111 0-15,66,70\r\n"
+        "a=rtpmap:105 telephone-event/8000\r\n"
+        "a=fmtp:105 0-15,66,70\r\n"
         "a=fmtp:111 ";
     std::ostringstream os;
     os << "minptime=" << params.min_ptime << "; stereo=" << params.stereo
@@ -1987,6 +1987,14 @@ class WebRtcSdpTest : public ::testing::Test {
       VerifyCodecParameter(codec.params, "maxptime", params.max_ptime);
     }
 
+    cricket::AudioCodec dtmf = acd->codecs()[3];
+    EXPECT_EQ("telephone-event", dtmf.name);
+    EXPECT_EQ(105, dtmf.id);
+    EXPECT_EQ(3u,
+              dtmf.params.size());  // ptime and max_ptime count as parameters.
+    EXPECT_EQ(dtmf.params.begin()->first, "");
+    EXPECT_EQ(dtmf.params.begin()->second, "0-15,66,70");
+
     const VideoContentDescription* vcd =
         GetFirstVideoContentDescription(jdesc_output->description());
     ASSERT_TRUE(vcd);
@@ -3592,6 +3600,28 @@ TEST_F(WebRtcSdpTest, SerializeAudioFmtpWithPTimeAndMaxPTime) {
   EXPECT_EQ(sdp_with_fmtp, message);
 }
 
+TEST_F(WebRtcSdpTest, SerializeAudioFmtpWithTelephoneEvent) {
+  AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_);
+
+  cricket::AudioCodecs codecs = acd->codecs();
+  cricket::AudioCodec dtmf(105, "telephone-event", 8000, 0, 1);
+  dtmf.params[""] = "0-15";
+  codecs.push_back(dtmf);
+  acd->set_codecs(codecs);
+
+  ASSERT_TRUE(jdesc_.Initialize(desc_.Clone(), jdesc_.session_id(),
+                                jdesc_.session_version()));
+  std::string message = webrtc::SdpSerialize(jdesc_);
+  std::string sdp_with_fmtp = kSdpFullString;
+  InjectAfter("m=audio 2345 RTP/SAVPF 111 103 104", " 105", &sdp_with_fmtp);
+  InjectAfter(
+      "a=rtpmap:104 ISAC/32000\r\n",
+      "a=rtpmap:105 telephone-event/8000\r\n"  // No comma here. String merging!
+      "a=fmtp:105 0-15\r\n",
+      &sdp_with_fmtp);
+  EXPECT_EQ(sdp_with_fmtp, message);
+}
+
 TEST_F(WebRtcSdpTest, SerializeVideoFmtp) {
   VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_);