diff --git a/webrtc/call.cc b/webrtc/call.cc
index dff913b04..68d484376 100644
--- a/webrtc/call.cc
+++ b/webrtc/call.cc
@@ -38,14 +38,14 @@ class Call : public webrtc::Call, public PacketReceiver {
 
   virtual VideoSendStream::Config GetDefaultSendConfig() OVERRIDE;
 
-  virtual VideoSendStream* CreateSendStream(
+  virtual VideoSendStream* CreateVideoSendStream(
       const VideoSendStream::Config& config) OVERRIDE;
 
   virtual void DestroySendStream(webrtc::VideoSendStream* send_stream) OVERRIDE;
 
   virtual VideoReceiveStream::Config GetDefaultReceiveConfig() OVERRIDE;
 
-  virtual VideoReceiveStream* CreateReceiveStream(
+  virtual VideoReceiveStream* CreateVideoReceiveStream(
       const VideoReceiveStream::Config& config) OVERRIDE;
 
   virtual void DestroyReceiveStream(
@@ -211,7 +211,8 @@ VideoSendStream::Config Call::GetDefaultSendConfig() {
   return config;
 }
 
-VideoSendStream* Call::CreateSendStream(const VideoSendStream::Config& config) {
+VideoSendStream* Call::CreateVideoSendStream(
+    const VideoSendStream::Config& config) {
   assert(config.rtp.ssrcs.size() > 0);
   assert(config.codec.numberOfSimulcastStreams == 0 ||
          config.codec.numberOfSimulcastStreams == config.rtp.ssrcs.size());
@@ -253,7 +254,7 @@ VideoReceiveStream::Config Call::GetDefaultReceiveConfig() {
   return VideoReceiveStream::Config();
 }
 
-VideoReceiveStream* Call::CreateReceiveStream(
+VideoReceiveStream* Call::CreateVideoReceiveStream(
     const VideoReceiveStream::Config& config) {
   VideoReceiveStream* receive_stream = new VideoReceiveStream(
       video_engine_, config, config_.send_transport, config_.voice_engine);
diff --git a/webrtc/call.h b/webrtc/call.h
index 5f4c7b614..7b3de93f1 100644
--- a/webrtc/call.h
+++ b/webrtc/call.h
@@ -60,7 +60,7 @@ class Call {
 
   virtual VideoSendStream::Config GetDefaultSendConfig() = 0;
 
-  virtual VideoSendStream* CreateSendStream(
+  virtual VideoSendStream* CreateVideoSendStream(
       const VideoSendStream::Config& config) = 0;
 
   // Returns the internal state of the send stream, for resume sending with a
@@ -70,7 +70,7 @@ class Call {
 
   virtual VideoReceiveStream::Config GetDefaultReceiveConfig() = 0;
 
-  virtual VideoReceiveStream* CreateReceiveStream(
+  virtual VideoReceiveStream* CreateVideoReceiveStream(
       const VideoReceiveStream::Config& config) = 0;
   virtual void DestroyReceiveStream(VideoReceiveStream* receive_stream) = 0;
 
diff --git a/webrtc/call_tests.cc b/webrtc/call_tests.cc
index 3c8d78eb0..e8637bc85 100644
--- a/webrtc/call_tests.cc
+++ b/webrtc/call_tests.cc
@@ -88,8 +88,8 @@ class CallTest : public ::testing::Test {
     assert(send_stream_ == NULL);
     assert(receive_stream_ == NULL);
 
-    send_stream_ = sender_call_->CreateSendStream(send_config_);
-    receive_stream_ = receiver_call_->CreateReceiveStream(receive_config_);
+    send_stream_ = sender_call_->CreateVideoSendStream(send_config_);
+    receive_stream_ = receiver_call_->CreateVideoReceiveStream(receive_config_);
   }
 
   void CreateFrameGenerator() {
@@ -787,14 +787,15 @@ TEST_F(CallTest, SendsAndReceivesMultipleStreams) {
         receiver_call->GetDefaultReceiveConfig();
     receive_config.renderer = observers[i];
     receive_config.rtp.ssrc = ssrc;
-    receive_streams[i] = receiver_call->CreateReceiveStream(receive_config);
+    receive_streams[i] =
+        receiver_call->CreateVideoReceiveStream(receive_config);
     receive_streams[i]->StartReceive();
 
     VideoSendStream::Config send_config = sender_call->GetDefaultSendConfig();
     send_config.rtp.ssrcs.push_back(ssrc);
     send_config.codec.width = width;
     send_config.codec.height = height;
-    send_streams[i] = sender_call->CreateSendStream(send_config);
+    send_streams[i] = sender_call->CreateVideoSendStream(send_config);
     send_streams[i]->StartSend();
 
     frame_generators[i] = test::FrameGeneratorCapturer::Create(
diff --git a/webrtc/loopback.cc b/webrtc/loopback.cc
index 5c5dbce9b..52cdcaa51 100644
--- a/webrtc/loopback.cc
+++ b/webrtc/loopback.cc
@@ -63,7 +63,7 @@ TEST_F(LoopbackTest, Test) {
   send_config.codec.maxBitrate =
       static_cast<unsigned int>(test::flags::MaxBitrate());
 
-  VideoSendStream* send_stream = call->CreateSendStream(send_config);
+  VideoSendStream* send_stream = call->CreateVideoSendStream(send_config);
 
   Clock* test_clock = Clock::GetRealTimeClock();
 
@@ -79,7 +79,7 @@ TEST_F(LoopbackTest, Test) {
   receive_config.renderer = loopback_video.get();
 
   VideoReceiveStream* receive_stream =
-      call->CreateReceiveStream(receive_config);
+      call->CreateVideoReceiveStream(receive_config);
 
   receive_stream->StartReceive();
   send_stream->StartSend();
diff --git a/webrtc/video/full_stack.cc b/webrtc/video/full_stack.cc
index b154df301..317f126ad 100644
--- a/webrtc/video/full_stack.cc
+++ b/webrtc/video/full_stack.cc
@@ -402,7 +402,7 @@ TEST_P(FullStackTest, NoPacketLoss) {
   send_config.codec.startBitrate = params.bitrate;
   send_config.codec.maxBitrate = params.bitrate;
 
-  VideoSendStream* send_stream = call->CreateSendStream(send_config);
+  VideoSendStream* send_stream = call->CreateVideoSendStream(send_config);
   analyzer.input_ = send_stream->Input();
 
   scoped_ptr<test::FrameGeneratorCapturer> file_capturer(
@@ -422,7 +422,7 @@ TEST_P(FullStackTest, NoPacketLoss) {
   receive_config.renderer = &analyzer;
 
   VideoReceiveStream* receive_stream =
-      call->CreateReceiveStream(receive_config);
+      call->CreateVideoReceiveStream(receive_config);
 
   receive_stream->StartReceive();
   send_stream->StartSend();
diff --git a/webrtc/video/rampup_tests.cc b/webrtc/video/rampup_tests.cc
index aa164b5a4..dababaf2b 100644
--- a/webrtc/video/rampup_tests.cc
+++ b/webrtc/video/rampup_tests.cc
@@ -159,13 +159,13 @@ TEST_P(RampUpTest, RampUpWithPadding) {
 
   test::GenerateRandomSsrcs(&send_config, &reserved_ssrcs_);
 
-  VideoSendStream* send_stream = call->CreateSendStream(send_config);
+  VideoSendStream* send_stream = call->CreateVideoSendStream(send_config);
 
   VideoReceiveStream::Config receive_config;
   receive_config.rtp.ssrc = send_config.rtp.ssrcs[0];
   receive_config.rtp.nack.rtp_history_ms = send_config.rtp.nack.rtp_history_ms;
   VideoReceiveStream* receive_stream =
-      call->CreateReceiveStream(receive_config);
+      call->CreateVideoReceiveStream(receive_config);
 
   scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
       test::FrameGeneratorCapturer::Create(send_stream->Input(),
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index 7e12713a2..dd9bc5575 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -40,7 +40,7 @@ class VideoSendStreamTest : public ::testing::Test {
   void RunSendTest(Call* call,
                    const VideoSendStream::Config& config,
                    test::RtpRtcpObserver* observer) {
-    send_stream_ = call->CreateSendStream(config);
+    send_stream_ = call->CreateVideoSendStream(config);
     scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer(
         test::FrameGeneratorCapturer::Create(
             send_stream_->Input(), 320, 240, 30, Clock::GetRealTimeClock()));