| /* |
| * libjingle |
| * 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: |
| * |
| * 1. Redistributions of source code must retain the above copyright notice, |
| * this list of conditions and the following disclaimer. |
| * 2. Redistributions in binary form must reproduce the above copyright notice, |
| * this list of conditions and the following disclaimer in the documentation |
| * and/or other materials provided with the distribution. |
| * 3. The name of the author may not be used to endorse or promote products |
| * derived from this software without specific prior written permission. |
| * |
| * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED |
| * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF |
| * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO |
| * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, |
| * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, |
| * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; |
| * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, |
| * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR |
| * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF |
| * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. |
| */ |
| |
| #include "talk/app/webrtc/datachannel.h" |
| #include "talk/app/webrtc/sctputils.h" |
| #include "talk/app/webrtc/test/fakedatachannelprovider.h" |
| #include "webrtc/base/gunit.h" |
| |
| using webrtc::DataChannel; |
| using webrtc::SctpSidAllocator; |
| |
| class FakeDataChannelObserver : public webrtc::DataChannelObserver { |
| public: |
| FakeDataChannelObserver() |
| : messages_received_(0), |
| on_state_change_count_(0), |
| on_buffered_amount_change_count_(0) {} |
| |
| void OnStateChange() { |
| ++on_state_change_count_; |
| } |
| |
| void OnBufferedAmountChange(uint64_t previous_amount) { |
| ++on_buffered_amount_change_count_; |
| } |
| |
| void OnMessage(const webrtc::DataBuffer& buffer) { |
| ++messages_received_; |
| } |
| |
| size_t messages_received() const { |
| return messages_received_; |
| } |
| |
| void ResetOnStateChangeCount() { |
| on_state_change_count_ = 0; |
| } |
| |
| void ResetOnBufferedAmountChangeCount() { |
| on_buffered_amount_change_count_ = 0; |
| } |
| |
| size_t on_state_change_count() const { |
| return on_state_change_count_; |
| } |
| |
| size_t on_buffered_amount_change_count() const { |
| return on_buffered_amount_change_count_; |
| } |
| |
| private: |
| size_t messages_received_; |
| size_t on_state_change_count_; |
| size_t on_buffered_amount_change_count_; |
| }; |
| |
| class SctpDataChannelTest : public testing::Test { |
| protected: |
| SctpDataChannelTest() |
| : webrtc_data_channel_( |
| DataChannel::Create( |
| &provider_, cricket::DCT_SCTP, "test", init_)) { |
| } |
| |
| void SetChannelReady() { |
| provider_.set_transport_available(true); |
| webrtc_data_channel_->OnTransportChannelCreated(); |
| if (webrtc_data_channel_->id() < 0) { |
| webrtc_data_channel_->SetSctpSid(0); |
| } |
| provider_.set_ready_to_send(true); |
| } |
| |
| void AddObserver() { |
| observer_.reset(new FakeDataChannelObserver()); |
| webrtc_data_channel_->RegisterObserver(observer_.get()); |
| } |
| |
| webrtc::InternalDataChannelInit init_; |
| FakeDataChannelProvider provider_; |
| rtc::scoped_ptr<FakeDataChannelObserver> observer_; |
| rtc::scoped_refptr<DataChannel> webrtc_data_channel_; |
| }; |
| |
| // Verifies that the data channel is connected to the transport after creation. |
| TEST_F(SctpDataChannelTest, ConnectedToTransportOnCreated) { |
| provider_.set_transport_available(true); |
| rtc::scoped_refptr<DataChannel> dc = DataChannel::Create( |
| &provider_, cricket::DCT_SCTP, "test1", init_); |
| |
| EXPECT_TRUE(provider_.IsConnected(dc.get())); |
| // The sid is not set yet, so it should not have added the streams. |
| EXPECT_FALSE(provider_.IsSendStreamAdded(dc->id())); |
| EXPECT_FALSE(provider_.IsRecvStreamAdded(dc->id())); |
| |
| dc->SetSctpSid(0); |
| EXPECT_TRUE(provider_.IsSendStreamAdded(dc->id())); |
| EXPECT_TRUE(provider_.IsRecvStreamAdded(dc->id())); |
| } |
| |
| // Verifies that the data channel is connected to the transport if the transport |
| // is not available initially and becomes available later. |
| TEST_F(SctpDataChannelTest, ConnectedAfterTransportBecomesAvailable) { |
| EXPECT_FALSE(provider_.IsConnected(webrtc_data_channel_.get())); |
| |
| provider_.set_transport_available(true); |
| webrtc_data_channel_->OnTransportChannelCreated(); |
| EXPECT_TRUE(provider_.IsConnected(webrtc_data_channel_.get())); |
| } |
| |
| // Tests the state of the data channel. |
| TEST_F(SctpDataChannelTest, StateTransition) { |
| EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, |
| webrtc_data_channel_->state()); |
| SetChannelReady(); |
| |
| EXPECT_EQ(webrtc::DataChannelInterface::kOpen, webrtc_data_channel_->state()); |
| webrtc_data_channel_->Close(); |
| EXPECT_EQ(webrtc::DataChannelInterface::kClosed, |
| webrtc_data_channel_->state()); |
| // Verifies that it's disconnected from the transport. |
| EXPECT_FALSE(provider_.IsConnected(webrtc_data_channel_.get())); |
| } |
| |
| // Tests that DataChannel::buffered_amount() is correct after the channel is |
| // blocked. |
| TEST_F(SctpDataChannelTest, BufferedAmountWhenBlocked) { |
| AddObserver(); |
| SetChannelReady(); |
| webrtc::DataBuffer buffer("abcd"); |
| EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); |
| |
| EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount()); |
| EXPECT_EQ(0U, observer_->on_buffered_amount_change_count()); |
| |
| provider_.set_send_blocked(true); |
| |
| const int number_of_packets = 3; |
| for (int i = 0; i < number_of_packets; ++i) { |
| EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); |
| } |
| EXPECT_EQ(buffer.data.size() * number_of_packets, |
| webrtc_data_channel_->buffered_amount()); |
| EXPECT_EQ(number_of_packets, observer_->on_buffered_amount_change_count()); |
| } |
| |
| // Tests that the queued data are sent when the channel transitions from blocked |
| // to unblocked. |
| TEST_F(SctpDataChannelTest, QueuedDataSentWhenUnblocked) { |
| AddObserver(); |
| SetChannelReady(); |
| webrtc::DataBuffer buffer("abcd"); |
| provider_.set_send_blocked(true); |
| EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); |
| |
| EXPECT_EQ(1U, observer_->on_buffered_amount_change_count()); |
| |
| provider_.set_send_blocked(false); |
| SetChannelReady(); |
| EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount()); |
| EXPECT_EQ(2U, observer_->on_buffered_amount_change_count()); |
| } |
| |
| // Tests that no crash when the channel is blocked right away while trying to |
| // send queued data. |
| TEST_F(SctpDataChannelTest, BlockedWhenSendQueuedDataNoCrash) { |
| AddObserver(); |
| SetChannelReady(); |
| webrtc::DataBuffer buffer("abcd"); |
| provider_.set_send_blocked(true); |
| EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); |
| EXPECT_EQ(1U, observer_->on_buffered_amount_change_count()); |
| |
| // Set channel ready while it is still blocked. |
| SetChannelReady(); |
| EXPECT_EQ(buffer.size(), webrtc_data_channel_->buffered_amount()); |
| EXPECT_EQ(1U, observer_->on_buffered_amount_change_count()); |
| |
| // Unblock the channel to send queued data again, there should be no crash. |
| provider_.set_send_blocked(false); |
| SetChannelReady(); |
| EXPECT_EQ(0U, webrtc_data_channel_->buffered_amount()); |
| EXPECT_EQ(2U, observer_->on_buffered_amount_change_count()); |
| } |
| |
| // Tests that the queued control message is sent when channel is ready. |
| TEST_F(SctpDataChannelTest, OpenMessageSent) { |
| // Initially the id is unassigned. |
| EXPECT_EQ(-1, webrtc_data_channel_->id()); |
| |
| SetChannelReady(); |
| EXPECT_GE(webrtc_data_channel_->id(), 0); |
| EXPECT_EQ(cricket::DMT_CONTROL, provider_.last_send_data_params().type); |
| EXPECT_EQ(provider_.last_send_data_params().ssrc, |
| static_cast<uint32_t>(webrtc_data_channel_->id())); |
| } |
| |
| TEST_F(SctpDataChannelTest, QueuedOpenMessageSent) { |
| provider_.set_send_blocked(true); |
| SetChannelReady(); |
| provider_.set_send_blocked(false); |
| |
| EXPECT_EQ(cricket::DMT_CONTROL, provider_.last_send_data_params().type); |
| EXPECT_EQ(provider_.last_send_data_params().ssrc, |
| static_cast<uint32_t>(webrtc_data_channel_->id())); |
| } |
| |
| // Tests that the DataChannel created after transport gets ready can enter OPEN |
| // state. |
| TEST_F(SctpDataChannelTest, LateCreatedChannelTransitionToOpen) { |
| SetChannelReady(); |
| webrtc::InternalDataChannelInit init; |
| init.id = 1; |
| rtc::scoped_refptr<DataChannel> dc = DataChannel::Create( |
| &provider_, cricket::DCT_SCTP, "test1", init); |
| EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, dc->state()); |
| EXPECT_TRUE_WAIT(webrtc::DataChannelInterface::kOpen == dc->state(), |
| 1000); |
| } |
| |
| // Tests that an unordered DataChannel sends data as ordered until the OPEN_ACK |
| // message is received. |
| TEST_F(SctpDataChannelTest, SendUnorderedAfterReceivesOpenAck) { |
| SetChannelReady(); |
| webrtc::InternalDataChannelInit init; |
| init.id = 1; |
| init.ordered = false; |
| rtc::scoped_refptr<DataChannel> dc = DataChannel::Create( |
| &provider_, cricket::DCT_SCTP, "test1", init); |
| |
| EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, dc->state(), 1000); |
| |
| // Sends a message and verifies it's ordered. |
| webrtc::DataBuffer buffer("some data"); |
| ASSERT_TRUE(dc->Send(buffer)); |
| EXPECT_TRUE(provider_.last_send_data_params().ordered); |
| |
| // Emulates receiving an OPEN_ACK message. |
| cricket::ReceiveDataParams params; |
| params.ssrc = init.id; |
| params.type = cricket::DMT_CONTROL; |
| rtc::Buffer payload; |
| webrtc::WriteDataChannelOpenAckMessage(&payload); |
| dc->OnDataReceived(NULL, params, payload); |
| |
| // Sends another message and verifies it's unordered. |
| ASSERT_TRUE(dc->Send(buffer)); |
| EXPECT_FALSE(provider_.last_send_data_params().ordered); |
| } |
| |
| // Tests that an unordered DataChannel sends unordered data after any DATA |
| // message is received. |
| TEST_F(SctpDataChannelTest, SendUnorderedAfterReceiveData) { |
| SetChannelReady(); |
| webrtc::InternalDataChannelInit init; |
| init.id = 1; |
| init.ordered = false; |
| rtc::scoped_refptr<DataChannel> dc = DataChannel::Create( |
| &provider_, cricket::DCT_SCTP, "test1", init); |
| |
| EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, dc->state(), 1000); |
| |
| // Emulates receiving a DATA message. |
| cricket::ReceiveDataParams params; |
| params.ssrc = init.id; |
| params.type = cricket::DMT_TEXT; |
| webrtc::DataBuffer buffer("data"); |
| dc->OnDataReceived(NULL, params, buffer.data); |
| |
| // Sends a message and verifies it's unordered. |
| ASSERT_TRUE(dc->Send(buffer)); |
| EXPECT_FALSE(provider_.last_send_data_params().ordered); |
| } |
| |
| // Tests that the channel can't open until it's successfully sent the OPEN |
| // message. |
| TEST_F(SctpDataChannelTest, OpenWaitsForOpenMesssage) { |
| webrtc::DataBuffer buffer("foo"); |
| |
| provider_.set_send_blocked(true); |
| SetChannelReady(); |
| EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, |
| webrtc_data_channel_->state()); |
| provider_.set_send_blocked(false); |
| EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, |
| webrtc_data_channel_->state(), 1000); |
| EXPECT_EQ(cricket::DMT_CONTROL, provider_.last_send_data_params().type); |
| } |
| |
| // Tests that close first makes sure all queued data gets sent. |
| TEST_F(SctpDataChannelTest, QueuedCloseFlushes) { |
| webrtc::DataBuffer buffer("foo"); |
| |
| provider_.set_send_blocked(true); |
| SetChannelReady(); |
| EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, |
| webrtc_data_channel_->state()); |
| provider_.set_send_blocked(false); |
| EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, |
| webrtc_data_channel_->state(), 1000); |
| provider_.set_send_blocked(true); |
| webrtc_data_channel_->Send(buffer); |
| webrtc_data_channel_->Close(); |
| provider_.set_send_blocked(false); |
| EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kClosed, |
| webrtc_data_channel_->state(), 1000); |
| EXPECT_EQ(cricket::DMT_TEXT, provider_.last_send_data_params().type); |
| } |
| |
| // Tests that messages are sent with the right ssrc. |
| TEST_F(SctpDataChannelTest, SendDataSsrc) { |
| webrtc_data_channel_->SetSctpSid(1); |
| SetChannelReady(); |
| webrtc::DataBuffer buffer("data"); |
| EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); |
| EXPECT_EQ(1U, provider_.last_send_data_params().ssrc); |
| } |
| |
| // Tests that the incoming messages with wrong ssrcs are rejected. |
| TEST_F(SctpDataChannelTest, ReceiveDataWithInvalidSsrc) { |
| webrtc_data_channel_->SetSctpSid(1); |
| SetChannelReady(); |
| |
| AddObserver(); |
| |
| cricket::ReceiveDataParams params; |
| params.ssrc = 0; |
| webrtc::DataBuffer buffer("abcd"); |
| webrtc_data_channel_->OnDataReceived(NULL, params, buffer.data); |
| |
| EXPECT_EQ(0U, observer_->messages_received()); |
| } |
| |
| // Tests that the incoming messages with right ssrcs are acceted. |
| TEST_F(SctpDataChannelTest, ReceiveDataWithValidSsrc) { |
| webrtc_data_channel_->SetSctpSid(1); |
| SetChannelReady(); |
| |
| AddObserver(); |
| |
| cricket::ReceiveDataParams params; |
| params.ssrc = 1; |
| webrtc::DataBuffer buffer("abcd"); |
| |
| webrtc_data_channel_->OnDataReceived(NULL, params, buffer.data); |
| EXPECT_EQ(1U, observer_->messages_received()); |
| } |
| |
| // Tests that no CONTROL message is sent if the datachannel is negotiated and |
| // not created from an OPEN message. |
| TEST_F(SctpDataChannelTest, NoMsgSentIfNegotiatedAndNotFromOpenMsg) { |
| webrtc::InternalDataChannelInit config; |
| config.id = 1; |
| config.negotiated = true; |
| config.open_handshake_role = webrtc::InternalDataChannelInit::kNone; |
| |
| SetChannelReady(); |
| rtc::scoped_refptr<DataChannel> dc = DataChannel::Create( |
| &provider_, cricket::DCT_SCTP, "test1", config); |
| |
| EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, dc->state(), 1000); |
| EXPECT_EQ(0U, provider_.last_send_data_params().ssrc); |
| } |
| |
| // Tests that OPEN_ACK message is sent if the datachannel is created from an |
| // OPEN message. |
| TEST_F(SctpDataChannelTest, OpenAckSentIfCreatedFromOpenMessage) { |
| webrtc::InternalDataChannelInit config; |
| config.id = 1; |
| config.negotiated = true; |
| config.open_handshake_role = webrtc::InternalDataChannelInit::kAcker; |
| |
| SetChannelReady(); |
| rtc::scoped_refptr<DataChannel> dc = DataChannel::Create( |
| &provider_, cricket::DCT_SCTP, "test1", config); |
| |
| EXPECT_EQ_WAIT(webrtc::DataChannelInterface::kOpen, dc->state(), 1000); |
| |
| EXPECT_EQ(static_cast<unsigned int>(config.id), |
| provider_.last_send_data_params().ssrc); |
| EXPECT_EQ(cricket::DMT_CONTROL, provider_.last_send_data_params().type); |
| } |
| |
| // Tests the OPEN_ACK role assigned by InternalDataChannelInit. |
| TEST_F(SctpDataChannelTest, OpenAckRoleInitialization) { |
| webrtc::InternalDataChannelInit init; |
| EXPECT_EQ(webrtc::InternalDataChannelInit::kOpener, init.open_handshake_role); |
| EXPECT_FALSE(init.negotiated); |
| |
| webrtc::DataChannelInit base; |
| base.negotiated = true; |
| webrtc::InternalDataChannelInit init2(base); |
| EXPECT_EQ(webrtc::InternalDataChannelInit::kNone, init2.open_handshake_role); |
| } |
| |
| // Tests that the DataChannel is closed if the sending buffer is full. |
| TEST_F(SctpDataChannelTest, ClosedWhenSendBufferFull) { |
| SetChannelReady(); |
| |
| rtc::Buffer buffer(1024); |
| memset(buffer.data(), 0, buffer.size()); |
| |
| webrtc::DataBuffer packet(buffer, true); |
| provider_.set_send_blocked(true); |
| |
| for (size_t i = 0; i < 16 * 1024 + 1; ++i) { |
| EXPECT_TRUE(webrtc_data_channel_->Send(packet)); |
| } |
| |
| EXPECT_TRUE( |
| webrtc::DataChannelInterface::kClosed == webrtc_data_channel_->state() || |
| webrtc::DataChannelInterface::kClosing == webrtc_data_channel_->state()); |
| } |
| |
| // Tests that the DataChannel is closed on transport errors. |
| TEST_F(SctpDataChannelTest, ClosedOnTransportError) { |
| SetChannelReady(); |
| webrtc::DataBuffer buffer("abcd"); |
| provider_.set_transport_error(); |
| |
| EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); |
| |
| EXPECT_EQ(webrtc::DataChannelInterface::kClosed, |
| webrtc_data_channel_->state()); |
| } |
| |
| // Tests that a already closed DataChannel does not fire onStateChange again. |
| TEST_F(SctpDataChannelTest, ClosedDataChannelDoesNotFireOnStateChange) { |
| AddObserver(); |
| webrtc_data_channel_->Close(); |
| // OnStateChange called for kClosing and kClosed. |
| EXPECT_EQ(2U, observer_->on_state_change_count()); |
| |
| observer_->ResetOnStateChangeCount(); |
| webrtc_data_channel_->RemotePeerRequestClose(); |
| EXPECT_EQ(0U, observer_->on_state_change_count()); |
| } |
| |
| // Tests that RemotePeerRequestClose closes the local DataChannel. |
| TEST_F(SctpDataChannelTest, RemotePeerRequestClose) { |
| AddObserver(); |
| webrtc_data_channel_->RemotePeerRequestClose(); |
| |
| // OnStateChange called for kClosing and kClosed. |
| EXPECT_EQ(2U, observer_->on_state_change_count()); |
| EXPECT_EQ(webrtc::DataChannelInterface::kClosed, |
| webrtc_data_channel_->state()); |
| } |
| |
| // Tests that the DataChannel is closed if the received buffer is full. |
| TEST_F(SctpDataChannelTest, ClosedWhenReceivedBufferFull) { |
| SetChannelReady(); |
| rtc::Buffer buffer(1024); |
| memset(buffer.data(), 0, buffer.size()); |
| |
| cricket::ReceiveDataParams params; |
| params.ssrc = 0; |
| |
| // Receiving data without having an observer will overflow the buffer. |
| for (size_t i = 0; i < 16 * 1024 + 1; ++i) { |
| webrtc_data_channel_->OnDataReceived(NULL, params, buffer); |
| } |
| EXPECT_EQ(webrtc::DataChannelInterface::kClosed, |
| webrtc_data_channel_->state()); |
| } |
| |
| // Tests that sending empty data returns no error and keeps the channel open. |
| TEST_F(SctpDataChannelTest, SendEmptyData) { |
| webrtc_data_channel_->SetSctpSid(1); |
| SetChannelReady(); |
| EXPECT_EQ(webrtc::DataChannelInterface::kOpen, |
| webrtc_data_channel_->state()); |
| |
| webrtc::DataBuffer buffer(""); |
| EXPECT_TRUE(webrtc_data_channel_->Send(buffer)); |
| EXPECT_EQ(webrtc::DataChannelInterface::kOpen, |
| webrtc_data_channel_->state()); |
| } |
| |
| // Tests that a channel can be closed without being opened or assigned an sid. |
| TEST_F(SctpDataChannelTest, NeverOpened) { |
| provider_.set_transport_available(true); |
| webrtc_data_channel_->OnTransportChannelCreated(); |
| webrtc_data_channel_->Close(); |
| } |
| |
| class SctpSidAllocatorTest : public testing::Test { |
| protected: |
| SctpSidAllocator allocator_; |
| }; |
| |
| // Verifies that an even SCTP id is allocated for SSL_CLIENT and an odd id for |
| // SSL_SERVER. |
| TEST_F(SctpSidAllocatorTest, SctpIdAllocationBasedOnRole) { |
| int id; |
| EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_SERVER, &id)); |
| EXPECT_EQ(1, id); |
| EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_CLIENT, &id)); |
| EXPECT_EQ(0, id); |
| EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_SERVER, &id)); |
| EXPECT_EQ(3, id); |
| EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_CLIENT, &id)); |
| EXPECT_EQ(2, id); |
| } |
| |
| // Verifies that SCTP ids of existing DataChannels are not reused. |
| TEST_F(SctpSidAllocatorTest, SctpIdAllocationNoReuse) { |
| int old_id = 1; |
| EXPECT_TRUE(allocator_.ReserveSid(old_id)); |
| |
| int new_id; |
| EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_SERVER, &new_id)); |
| EXPECT_NE(old_id, new_id); |
| |
| old_id = 0; |
| EXPECT_TRUE(allocator_.ReserveSid(old_id)); |
| EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_CLIENT, &new_id)); |
| EXPECT_NE(old_id, new_id); |
| } |
| |
| // Verifies that SCTP ids of removed DataChannels can be reused. |
| TEST_F(SctpSidAllocatorTest, SctpIdReusedForRemovedDataChannel) { |
| int odd_id = 1; |
| int even_id = 0; |
| EXPECT_TRUE(allocator_.ReserveSid(odd_id)); |
| EXPECT_TRUE(allocator_.ReserveSid(even_id)); |
| |
| int allocated_id = -1; |
| EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_SERVER, &allocated_id)); |
| EXPECT_EQ(odd_id + 2, allocated_id); |
| |
| EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_CLIENT, &allocated_id)); |
| EXPECT_EQ(even_id + 2, allocated_id); |
| |
| EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_SERVER, &allocated_id)); |
| EXPECT_EQ(odd_id + 4, allocated_id); |
| |
| EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_CLIENT, &allocated_id)); |
| EXPECT_EQ(even_id + 4, allocated_id); |
| |
| allocator_.ReleaseSid(odd_id); |
| allocator_.ReleaseSid(even_id); |
| |
| // Verifies that removed ids are reused. |
| EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_SERVER, &allocated_id)); |
| EXPECT_EQ(odd_id, allocated_id); |
| |
| EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_CLIENT, &allocated_id)); |
| EXPECT_EQ(even_id, allocated_id); |
| |
| // Verifies that used higher ids are not reused. |
| EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_SERVER, &allocated_id)); |
| EXPECT_EQ(odd_id + 6, allocated_id); |
| |
| EXPECT_TRUE(allocator_.AllocateSid(rtc::SSL_CLIENT, &allocated_id)); |
| EXPECT_EQ(even_id + 6, allocated_id); |
| } |