398c3fd6c2
SrtpTransport currently just delegates everything to RtpTransport. Also makes BaseChannel::rtp_transport_ an RtpTransportInternal and constructs an SrtpTransport if srtp is required. BUG=webrtc:7013 Review-Url: https://codereview.webrtc.org/2981013002 Cr-Commit-Position: refs/heads/master@{#19095}
77 lines
2.3 KiB
C++
77 lines
2.3 KiB
C++
/*
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* Copyright 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/pc/srtptransport.h"
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#include "webrtc/pc/rtptransport.h"
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#include "webrtc/pc/rtptransporttestutil.h"
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#include "webrtc/rtc_base/asyncpacketsocket.h"
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#include "webrtc/rtc_base/gunit.h"
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#include "webrtc/rtc_base/ptr_util.h"
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#include "webrtc/test/gmock.h"
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namespace webrtc {
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using testing::_;
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using testing::Return;
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class MockRtpTransport : public RtpTransport {
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public:
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MockRtpTransport() : RtpTransport(true) {}
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MOCK_METHOD4(SendPacket,
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bool(bool rtcp,
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rtc::CopyOnWriteBuffer* packet,
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const rtc::PacketOptions& options,
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int flags));
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void PretendReceivedPacket() {
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bool rtcp = false;
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rtc::CopyOnWriteBuffer buffer;
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rtc::PacketTime time;
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SignalPacketReceived(rtcp, &buffer, time);
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}
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};
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TEST(SrtpTransportTest, SendPacket) {
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auto rtp_transport = rtc::MakeUnique<MockRtpTransport>();
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EXPECT_CALL(*rtp_transport, SendPacket(_, _, _, _)).WillOnce(Return(true));
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SrtpTransport srtp_transport(std::move(rtp_transport), "a");
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const bool rtcp = false;
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rtc::CopyOnWriteBuffer packet;
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rtc::PacketOptions options;
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int flags = 0;
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EXPECT_TRUE(srtp_transport.SendPacket(rtcp, &packet, options, flags));
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// TODO(zstein): Also verify that the packet received by RtpTransport has been
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// protected once SrtpTransport handles that.
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}
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// Test that SrtpTransport fires SignalPacketReceived when the underlying
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// RtpTransport fires SignalPacketReceived.
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TEST(SrtpTransportTest, SignalPacketReceived) {
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auto rtp_transport = rtc::MakeUnique<MockRtpTransport>();
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MockRtpTransport* rtp_transport_raw = rtp_transport.get();
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SrtpTransport srtp_transport(std::move(rtp_transport), "a");
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SignalPacketReceivedCounter counter(&srtp_transport);
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rtp_transport_raw->PretendReceivedPacket();
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EXPECT_EQ(1, counter.rtp_count());
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// TODO(zstein): Also verify that the packet is unprotected once SrtpTransport
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// handles that.
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}
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} // namespace webrtc
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