Index: talk/app/webrtc/peerconnection_unittest.cc |
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc |
index 3193ffd898e71c1cdf23ccb3049c477c6ecd9811..3367484605e39a59e1df0e58109ccc0a035d1f8d 100644 |
--- a/talk/app/webrtc/peerconnection_unittest.cc |
+++ b/talk/app/webrtc/peerconnection_unittest.cc |
@@ -658,6 +658,10 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver, |
return pc()->ice_gathering_state(); |
} |
+ static void set_use_alternate_key(bool use_alternate_key) { |
+ use_alternate_key_ = use_alternate_key; |
pthatcher1
2015/11/18 20:42:43
Why is this (and use_alternate_key_) static?
guoweis_webrtc
2015/11/25 21:03:13
Done.
|
+ } |
+ |
private: |
class DummyDtmfObserver : public DtmfSenderObserverInterface { |
public: |
@@ -740,9 +744,15 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver, |
ice_server.uri = "stun:stun.l.google.com:19302"; |
ice_servers.push_back(ice_server); |
- rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store( |
+ rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store( |
rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeDtlsIdentityStore() |
: nullptr); |
+ |
+ if (use_alternate_key_) { |
+ dtls_identity_store->use_alternate_key(); |
+ } else { |
+ dtls_identity_store->use_original_key(); |
+ } |
return peer_connection_factory_->CreatePeerConnection( |
ice_servers, constraints, factory, dtls_identity_store.Pass(), this); |
} |
@@ -890,8 +900,14 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver, |
rtc::scoped_refptr<DataChannelInterface> data_channel_; |
rtc::scoped_ptr<MockDataChannelObserver> data_observer_; |
+ |
+ static bool use_alternate_key_; |
}; |
+// FakeDtlsIdentityStore supports 2 keys. We'll use the original key to start |
+// with. |
+bool PeerConnectionTestClient::use_alternate_key_ = false; |
+ |
// TODO(deadbeef): Rename this to P2PTestConductor once the Linux memcheck and |
// Windows DrMemory Full bots' blacklists are updated. |
class JsepPeerConnectionP2PTestClient : public testing::Test { |
@@ -968,6 +984,11 @@ class JsepPeerConnectionP2PTestClient : public testing::Test { |
nullptr); |
} |
+ void MutuallySetSignalingReceiver() { |
pthatcher1
2015/11/18 20:42:43
How about "SetSignalingReceivers" (an "s" on the e
guoweis_webrtc
2015/11/25 21:03:13
Done.
|
+ initiating_client_->set_signaling_message_receiver(receiving_client_.get()); |
+ receiving_client_->set_signaling_message_receiver(initiating_client_.get()); |
+ } |
+ |
bool CreateTestClients(MediaConstraintsInterface* init_constraints, |
PeerConnectionFactory::Options* init_options, |
MediaConstraintsInterface* recv_constraints, |
@@ -979,8 +1000,7 @@ class JsepPeerConnectionP2PTestClient : public testing::Test { |
if (!initiating_client_ || !receiving_client_) { |
return false; |
} |
- initiating_client_->set_signaling_message_receiver(receiving_client_.get()); |
- receiving_client_->set_signaling_message_receiver(initiating_client_.get()); |
+ MutuallySetSignalingReceiver(); |
return true; |
} |
@@ -1058,6 +1078,47 @@ class JsepPeerConnectionP2PTestClient : public testing::Test { |
kMaxWaitForFramesMs); |
} |
+ // This test sets up a successful call then do the call transfer to another |
pthatcher1
2015/11/18 20:42:42
do => does
the call transfer => a call transfer
guoweis_webrtc
2015/11/25 21:03:13
Done.
|
+ // caller or callee. It uses the alternate identity in FakeDtlsIdentityStore |
+ // to make sure a different certificate/fingerprint is used. |
+ void TransferCall(bool test_callee) { |
+ MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); |
+ FakeConstraints setup_constraints; |
+ setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, |
+ true); |
pthatcher1
2015/11/18 20:42:42
Isn't this the default by now?
guoweis_webrtc
2015/11/25 21:03:13
It's not very clear. might be the case from https:
|
+ ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints)); |
+ LocalP2PTest(); |
+ VerifyRenderedSize(640, 480); |
+ |
+ LOG(LS_INFO) << "Starting call transfer"; |
+ |
+ // Make sure the new client is using a different certificate. |
+ PeerConnectionTestClient::set_use_alternate_key(true); |
+ rtc::scoped_ptr<PeerConnectionTestClient> new_client( |
+ PeerConnectionTestClient::CreateClient("New Peer: ", &setup_constraints, |
+ nullptr)); |
+ |
+ // Keeping the original peer around which will still send packets to the |
+ // receiving client. These SRTP packets will be dropped. |
+ rtc::scoped_ptr<PeerConnectionTestClient> original_peer; |
+ if (test_callee) { |
+ original_peer.reset(swap_initializing_client(new_client.release())); |
pthatcher1
2015/11/18 20:42:43
Wouldn't this work?
if (test_callee) {
original
guoweis_webrtc
2015/11/25 21:03:13
Done.
|
+ } else { |
+ original_peer.reset(swap_receiving_client(new_client.release())); |
+ } |
+ |
+ // Restore the original certificate to the transfee. |
+ PeerConnectionTestClient::set_use_alternate_key(false); |
+ MutuallySetSignalingReceiver(); |
+ if (test_callee) { |
+ receiving_client()->SetExpectIceRestart(true); |
+ } else { |
+ initializing_client()->IceRestart(); |
+ } |
+ LocalP2PTest(); |
+ VerifyRenderedSize(640, 480); |
+ } |
+ |
void SendRtpData(webrtc::DataChannelInterface* dc, const std::string& data) { |
// Messages may get lost on the unreliable DataChannel, so we send multiple |
// times to avoid test flakiness. |
@@ -1071,10 +1132,29 @@ class JsepPeerConnectionP2PTestClient : public testing::Test { |
PeerConnectionTestClient* initializing_client() { |
return initiating_client_.get(); |
} |
+ |
+ // Reset the |initiating_client_| to the |client| passed in and return the |
+ // original |initiating_client_|. |
+ PeerConnectionTestClient* swap_initializing_client( |
+ PeerConnectionTestClient* client) { |
+ PeerConnectionTestClient* old = initiating_client_.release(); |
+ initiating_client_.reset(client); |
+ return old; |
+ } |
+ |
PeerConnectionTestClient* receiving_client() { |
return receiving_client_.get(); |
} |
+ // Reset the |receiving_client_| to the |client| passed in and return the |
+ // original |receiving_client_|. |
+ PeerConnectionTestClient* swap_receiving_client( |
+ PeerConnectionTestClient* client) { |
+ PeerConnectionTestClient* old = receiving_client_.release(); |
+ receiving_client_.reset(client); |
+ return old; |
+ } |
+ |
private: |
rtc::scoped_ptr<rtc::PhysicalSocketServer> pss_; |
rtc::scoped_ptr<rtc::VirtualSocketServer> ss_; |
@@ -1159,6 +1239,18 @@ TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsRenegotiate) { |
receiving_client()->Negotiate(); |
} |
+// This test sets up a call transfer to a new caller with a different DTLS |
+// fingerprint. |
+TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsTransferCallee) { |
+ TransferCall(true); |
pthatcher1
2015/11/18 20:42:43
I think there are enough "if(test_callee)" checks
guoweis_webrtc
2015/11/25 21:03:13
Done.
|
+} |
+ |
+// This test sets up a call transfer to a new callee with a different DTLS |
+// fingerprint. |
+TEST_F(JsepPeerConnectionP2PTestClient, LocalP2PTestDtlsTransferCaller) { |
+ TransferCall(false); |
+} |
+ |
// This test sets up a call between two endpoints that are configured to use |
// DTLS key agreement. The offerer don't support SDES. As a result, DTLS is |
// negotiated and used for transport. |