diff --git a/video_engine/new_include/call.h b/video_engine/new_include/call.h
new file mode 100644
index 0000000..6650216
--- /dev/null
+++ b/video_engine/new_include/call.h
@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
+#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/common_types.h"
+#include "webrtc/video_engine/new_include/video_receive_stream.h"
+#include "webrtc/video_engine/new_include/video_send_stream.h"
+
+namespace webrtc {
+
+class VoiceEngine;
+
+const char* Version();
+
+class PacketReceiver {
+ public:
+  virtual bool DeliverPacket(const uint8_t* packet, size_t length) = 0;
+
+ protected:
+  virtual ~PacketReceiver() {}
+};
+
+// A Call instance can contain several send and/or receive streams. All streams
+// are assumed to have the same remote endpoint and will share bitrate estimates
+// etc.
+class Call {
+ public:
+  struct Config {
+    explicit Config(newapi::Transport* send_transport)
+        : send_transport(send_transport),
+          overuse_detection(false),
+          voice_engine(NULL),
+          trace_callback(NULL),
+          trace_filter(kTraceNone) {}
+
+    newapi::Transport* send_transport;
+    bool overuse_detection;
+
+    // VoiceEngine used for audio/video synchronization for this Call.
+    VoiceEngine* voice_engine;
+
+    TraceCallback* trace_callback;
+    uint32_t trace_filter;
+  };
+
+  static Call* Create(const Call::Config& config);
+
+  virtual std::vector<VideoCodec> GetVideoCodecs() = 0;
+
+  virtual VideoSendStream::Config GetDefaultSendConfig() = 0;
+
+  virtual VideoSendStream* CreateSendStream(
+      const VideoSendStream::Config& config) = 0;
+
+  // Returns the internal state of the send stream, for resume sending with a
+  // new stream with different settings.
+  // Note: Only the last returned send-stream state is valid.
+  virtual SendStreamState* DestroySendStream(VideoSendStream* send_stream) = 0;
+
+  virtual VideoReceiveStream::Config GetDefaultReceiveConfig() = 0;
+
+  virtual VideoReceiveStream* CreateReceiveStream(
+      const VideoReceiveStream::Config& config) = 0;
+  virtual void DestroyReceiveStream(VideoReceiveStream* receive_stream) = 0;
+
+  // All received RTP and RTCP packets for the call should be inserted to this
+  // PacketReceiver. The PacketReceiver pointer is valid as long as the
+  // Call instance exists.
+  virtual PacketReceiver* Receiver() = 0;
+
+  // Returns the estimated total send bandwidth. Note: this can differ from the
+  // actual encoded bitrate.
+  virtual uint32_t SendBitrateEstimate() = 0;
+
+  // Returns the total estimated receive bandwidth for the call. Note: this can
+  // differ from the actual receive bitrate.
+  virtual uint32_t ReceiveBitrateEstimate() = 0;
+
+  virtual ~Call() {}
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CALL_H_