Moving MediaStreamSignaling logic into PeerConnection.

talk/app/webrtc/sctputils_unittest.cc

Index: talk/app/webrtc/sctputils_unittest.cc
diff --git a/talk/app/webrtc/sctputils_unittest.cc b/talk/app/webrtc/sctputils_unittest.cc
index e5f323aa226285cb8a010aed6e927a022784ca67..e0e203f5cd314c86742fdec53dd7de4abb4a29da 100644
--- a/talk/app/webrtc/sctputils_unittest.cc
+++ b/talk/app/webrtc/sctputils_unittest.cc
@@ -1,6 +1,6 @@
 /*
  * libjingle
- * Copyright 2013 Google Inc
+ * Copyright 2013 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
@@ -159,3 +159,20 @@ TEST_F(SctpUtilsTest, WriteParseAckMessage) {
 
   EXPECT_TRUE(webrtc::ParseDataChannelOpenAckMessage(packet));
 }
+
+TEST_F(SctpUtilsTest, TestIsOpenMessage) {
+  rtc::ByteBuffer open;
+  open.WriteUInt8(0x03);
+  EXPECT_TRUE(webrtc::IsOpenMessage(open));
+
+  rtc::ByteBuffer openAck;
+  openAck.WriteUInt8(0x02);
+  EXPECT_FALSE(webrtc::IsOpenMessage(open));
+
+  rtc::ByteBuffer invalid;
+  openAck.WriteUInt8(0x01);
+  EXPECT_FALSE(webrtc::IsOpenMessage(invalid));
+
+  rtc::ByteBuffer empty;
+  EXPECT_FALSE(webrtc::IsOpenMessage(empty));
+}