diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index 9e78e82d7f..5d611432f4 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -94,6 +94,8 @@ void CallTest::CreateSendConfig(size_t num_streams) {
   send_config_.encoder_settings.encoder = &fake_encoder_;
   send_config_.encoder_settings.payload_name = "FAKE";
   send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
+  send_config_.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
   encoder_config_.streams = test::CreateVideoStreams(num_streams);
   for (size_t i = 0; i < num_streams; ++i)
     send_config_.rtp.ssrcs.push_back(kSendSsrcs[i]);
@@ -107,6 +109,8 @@ void CallTest::CreateMatchingReceiveConfigs() {
   assert(allocated_decoders_.empty());
   VideoReceiveStream::Config config;
   config.rtp.local_ssrc = kReceiverLocalSsrc;
+  for (const RtpExtension& extension : send_config_.rtp.extensions)
+    config.rtp.extensions.push_back(extension);
   for (size_t i = 0; i < send_config_.rtp.ssrcs.size(); ++i) {
     VideoReceiveStream::Decoder decoder =
         test::CreateMatchingDecoder(send_config_.encoder_settings);
@@ -162,6 +166,7 @@ const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE,
 const uint32_t CallTest::kSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE, 0xC0FFEF};
 const uint32_t CallTest::kReceiverLocalSsrc = 0x123456;
 const int CallTest::kNackRtpHistoryMs = 1000;
+const int CallTest::kAbsSendTimeExtensionId = 7;
 
 BaseTest::BaseTest(unsigned int timeout_ms) : RtpRtcpObserver(timeout_ms) {
 }
diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h
index 4771dee59d..b8d13c4fd9 100644
--- a/webrtc/test/call_test.h
+++ b/webrtc/test/call_test.h
@@ -42,6 +42,7 @@ class CallTest : public ::testing::Test {
   static const uint32_t kSendSsrcs[kNumSsrcs];
   static const uint32_t kReceiverLocalSsrc;
   static const int kNackRtpHistoryMs;
+  static const int kAbsSendTimeExtensionId;
 
  protected:
   void RunBaseTest(BaseTest* test);
diff --git a/webrtc/video/rampup_tests.cc b/webrtc/video/rampup_tests.cc
index 35d3297f39..7d8f98143b 100644
--- a/webrtc/video/rampup_tests.cc
+++ b/webrtc/video/rampup_tests.cc
@@ -380,6 +380,7 @@ void RampUpTest::RunRampUpTest(bool rtx,
   }
 
   CreateSendConfig(num_streams);
+  send_config_.rtp.extensions.clear();
 
   rtc::scoped_ptr<RemoteBitrateEstimatorFactory> rbe_factory;
   RateControlType control_type;
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 61e3d5b795..5624e45f80 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -155,6 +155,7 @@ TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
     void ModifyConfigs(VideoSendStream::Config* send_config,
                        std::vector<VideoReceiveStream::Config>* receive_configs,
                        VideoEncoderConfig* encoder_config) override {
+      send_config->rtp.extensions.clear();
       send_config->rtp.extensions.push_back(
           RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
     }
@@ -593,11 +594,9 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
       send_config->rtp.max_packet_size = kMaxPacketSize;
       send_config->post_encode_callback = this;
 
-      // Add an extension header, to make the RTP header larger than the base
-      // length of 12 bytes.
-      static const uint8_t kAbsSendTimeExtensionId = 13;
-      send_config->rtp.extensions.push_back(
-          RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
+      // Make sure there is at least one extension header, to make the RTP
+      // header larger than the base length of 12 bytes.
+      EXPECT_FALSE(send_config->rtp.extensions.empty());
     }
 
     void PerformTest() override {