Create the SrtpTransportInterface.

webrtc/ortc/rtptransportcontrolleradapter.cc

Index: webrtc/ortc/rtptransportcontrolleradapter.cc
diff --git a/webrtc/ortc/rtptransportcontrolleradapter.cc b/webrtc/ortc/rtptransportcontrolleradapter.cc
index 08e943a200ad1249784dd9267c7e341b6ba7462d..cb4c4bce7e04bde0e8af8adcf1cdc43286da8db1 100644
--- a/webrtc/ortc/rtptransportcontrolleradapter.cc
+++ b/webrtc/ortc/rtptransportcontrolleradapter.cc
@@ -11,8 +11,8 @@
 #include "webrtc/ortc/rtptransportcontrolleradapter.h"
 
 #include <algorithm>  // For "remove", "find".
-#include <sstream>
 #include <set>
+#include <sstream>
 #include <unordered_map>
 #include <utility>  // For std::move.
 
@@ -138,6 +138,21 @@ RtpTransportControllerAdapter::CreateProxiedRtpTransport(
   return result;
 }
 
+RTCErrorOr<std::unique_ptr<SrtpTransportInterface>>
+RtpTransportControllerAdapter::CreateProxiedSrtpTransport(
+    const RtcpParameters& rtcp_parameters,
+    PacketTransportInterface* rtp,
+    PacketTransportInterface* rtcp) {
+  auto result =
+      RtpTransportAdapter::CreateSrtpProxied(rtcp_parameters, rtp, rtcp, this);
+  if (result.ok()) {
+    transport_proxies_.push_back(result.value().get());
+    transport_proxies_.back()->GetInternal()->SignalDestroyed.connect(
+        this, &RtpTransportControllerAdapter::OnRtpTransportDestroyed);
+  }
+  return result;
+}
+
 RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>>
 RtpTransportControllerAdapter::CreateProxiedRtpSender(
     cricket::MediaType kind,
@@ -605,6 +620,10 @@ RTCError RtpTransportControllerAdapter::AttachAudioSender(
                          "RtpSender and RtpReceiver is not currently "
                          "supported.");
   }
+  RTCError err = MaybeSetCryptosForAudio(inner_transport);
+  if (!err.ok()) {
+    return err;
+  }
   // If setting new transport, extract its RTCP parameters and create voice
   // channel.
   if (!inner_audio_transport_) {
@@ -635,6 +654,10 @@ RTCError RtpTransportControllerAdapter::AttachVideoSender(
                          "RtpSender and RtpReceiver is not currently "
                          "supported.");
   }
+  RTCError err = MaybeSetCryptosForVideo(inner_transport);
+  if (!err.ok()) {
+    return err;
+  }
   // If setting new transport, extract its RTCP parameters and create video
   // channel.
   if (!inner_video_transport_) {
@@ -665,6 +688,10 @@ RTCError RtpTransportControllerAdapter::AttachAudioReceiver(
                          "RtpReceiver and RtpReceiver is not currently "
                          "supported.");
   }
+  RTCError err = MaybeSetCryptosForAudio(inner_transport);
+  if (!err.ok()) {
+    return err;
+  }
   // If setting new transport, extract its RTCP parameters and create voice
   // channel.
   if (!inner_audio_transport_) {
@@ -695,6 +722,10 @@ RTCError RtpTransportControllerAdapter::AttachVideoReceiver(
                          "RtpReceiver and RtpReceiver is not currently "
                          "supported.");
   }
+  RTCError err = MaybeSetCryptosForVideo(inner_transport);
+  if (!err.ok()) {
+    return err;
+  }
   // If setting new transport, extract its RTCP parameters and create video
   // channel.
   if (!inner_video_transport_) {
@@ -896,4 +927,40 @@ RtpTransportControllerAdapter::MakeSendStreamParamsVec(
   return result;
 }
 
+RTCError RtpTransportControllerAdapter::MaybeSetCryptosForAudio(

[Taylor Brandstetter, 2017/03/01 19:18:08]

nit: Instead of two separate methods, there could be one method that takes a
pointer to two MediaContentDescriptions

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

Done.

+    RtpTransportInterface* rtp_transport) {
+  if (rtp_transport->GetInternal()->is_srtp_transport()) {
+    if (!rtp_transport->GetInternal()->key_set()) {
+      LOG_AND_RETURN_ERROR(webrtc::RTCErrorType::INVALID_STATE,
+                           "The SRTP send key or receive key is not set.")
+    }
+    std::vector<cricket::CryptoParams> cryptos;
+    cryptos.push_back(rtp_transport->GetInternal()->receive_key());
+    local_audio_description_.set_cryptos(cryptos);
+
+    cryptos.clear();
+    cryptos.push_back(rtp_transport->GetInternal()->send_key());
+    remote_audio_description_.set_cryptos(cryptos);
+  }
+  return RTCError::OK();
+}
+
+RTCError RtpTransportControllerAdapter::MaybeSetCryptosForVideo(
+    RtpTransportInterface* rtp_transport) {
+  if (rtp_transport->GetInternal()->is_srtp_transport()) {
+    if (!rtp_transport->GetInternal()->key_set()) {
+      LOG_AND_RETURN_ERROR(webrtc::RTCErrorType::INVALID_STATE,
+                           "The SRTP send key or receive key is not set.")
+    }
+    std::vector<cricket::CryptoParams> cryptos;
+    cryptos.push_back(rtp_transport->GetInternal()->receive_key());
+    local_video_description_.set_cryptos(cryptos);
+
+    cryptos.clear();
+    cryptos.push_back(rtp_transport->GetInternal()->send_key());
+    remote_video_description_.set_cryptos(cryptos);
+  }
+  return RTCError::OK();
+}
+
 }  // namespace webrtc