Create the SrtpTransportInterface.

webrtc/api/ortc/srtptransportinterface.h

Index: webrtc/api/ortc/srtptransportinterface.h
diff --git a/webrtc/api/ortc/srtptransportinterface.h b/webrtc/api/ortc/srtptransportinterface.h
new file mode 100644
index 0000000000000000000000000000000000000000..729cad3efc9b19135834b307773e10480f54a33d
--- /dev/null
+++ b/webrtc/api/ortc/srtptransportinterface.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright 2017 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_API_ORTC_SRTPTRANSPORTINTERFACE_H_
+#define WEBRTC_API_ORTC_SRTPTRANSPORTINTERFACE_H_
+
+#include "webrtc/api/ortc/rtptransportinterface.h"
+#include "webrtc/api/rtcerror.h"
+#include "webrtc/media/base/cryptoparams.h"
+
+namespace webrtc {
+
+// The subclass of the RtpTransport which uses SRTP. The keying information
+// is explicitly passed in from the application.
+//
+// After applying the answer, the negotiated keying information from the offer

[Taylor Brandstetter, 2017/03/01 19:18:07]

I'd say "If using SDP and SDES (RFC4568) for signaling, then after an applying
an answer..."

Just to make it clear that any signaling protocol can be used on top of this
API.

[Zhi Huang, 2017/03/02 23:35:39]

Done.

+// and answer would be set and the SRTP would be active.
+//
+// Note that Edge's implementation of ORTC provides a similar API point, called
+// RTCSrtpSdesTransport:
+// https://msdn.microsoft.com/en-us/library/mt502527(v=vs.85).aspx
+class SrtpTransportInterface : public RtpTransportInterface {
+ public:
+  virtual ~SrtpTransportInterface() {}
+
+  // Set the SRTP keying material for outgoing stream from the application.

[Taylor Brandstetter, 2017/03/01 19:18:07]

nit: "streams" or "stream(s)" instead of "stream", since there may both an audio
and video stream using the same SRTP session. Also, saying "from the
application" may not be necessary.

[Zhi Huang, 2017/03/02 23:35:39]

Done.

+  virtual RTCError SetSrtpSendKey(const cricket::CryptoParams& params) = 0;

[Taylor Brandstetter, 2017/03/01 19:18:07]

We should document the current limitations of the implementation. Which, as I
understand it, are:

* Send and receive keys must use the same crypto suite
* Can't change keys after initially set.
* Must set the keys before creating a sender/receiver using the transport.

[Zhi Huang, 2017/03/02 23:35:39]

Done.

+
+  // Set the SRTP keying material for incoming stream from the application.
+  virtual RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_API_ORTC_SRTPTRANSPORTINTERFACE_H_