webrtc/talk/app/webrtc/peerconnectionendtoend_unittest.cc
buildbot@webrtc.org da510c5de6 (Auto)update libjingle 66923202-> 66924241
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6132 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 22:30:56 +00:00

226 lines
7.6 KiB
C++

/*
* 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/test/peerconnectiontestwrapper.h"
#include "talk/base/gunit.h"
#include "talk/base/logging.h"
#include "talk/base/ssladapter.h"
#include "talk/base/sslstreamadapter.h"
#include "talk/base/stringencode.h"
#include "talk/base/stringutils.h"
using webrtc::FakeConstraints;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStreamInterface;
using webrtc::PeerConnectionInterface;
namespace {
const char kExternalGiceUfrag[] = "1234567890123456";
const char kExternalGicePwd[] = "123456789012345678901234";
void RemoveLinesFromSdp(const std::string& line_start,
std::string* sdp) {
const char kSdpLineEnd[] = "\r\n";
size_t ssrc_pos = 0;
while ((ssrc_pos = sdp->find(line_start, ssrc_pos)) !=
std::string::npos) {
size_t end_ssrc = sdp->find(kSdpLineEnd, ssrc_pos);
sdp->erase(ssrc_pos, end_ssrc - ssrc_pos + strlen(kSdpLineEnd));
}
}
// Add |newlines| to the |message| after |line|.
void InjectAfter(const std::string& line,
const std::string& newlines,
std::string* message) {
const std::string tmp = line + newlines;
talk_base::replace_substrs(line.c_str(), line.length(),
tmp.c_str(), tmp.length(), message);
}
void Replace(const std::string& line,
const std::string& newlines,
std::string* message) {
talk_base::replace_substrs(line.c_str(), line.length(),
newlines.c_str(), newlines.length(), message);
}
void UseExternalSdes(std::string* sdp) {
// Remove current crypto specification.
RemoveLinesFromSdp("a=crypto", sdp);
RemoveLinesFromSdp("a=fingerprint", sdp);
// Add external crypto.
const char kAudioSdes[] =
"a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
"inline:PS1uQCVeeCFCanVmcjkpPywjNWhcYD0mXXtxaVBR\r\n";
const char kVideoSdes[] =
"a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
"inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj\r\n";
const char kDataSdes[] =
"a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
"inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj\r\n";
InjectAfter("a=mid:audio\r\n", kAudioSdes, sdp);
InjectAfter("a=mid:video\r\n", kVideoSdes, sdp);
InjectAfter("a=mid:data\r\n", kDataSdes, sdp);
}
void UseGice(std::string* sdp) {
InjectAfter("t=0 0\r\n", "a=ice-options:google-ice\r\n", sdp);
std::string ufragline = "a=ice-ufrag:";
std::string pwdline = "a=ice-pwd:";
RemoveLinesFromSdp(ufragline, sdp);
RemoveLinesFromSdp(pwdline, sdp);
ufragline.append(kExternalGiceUfrag);
ufragline.append("\r\n");
pwdline.append(kExternalGicePwd);
pwdline.append("\r\n");
const std::string ufrag_pwd = ufragline + pwdline;
InjectAfter("a=mid:audio\r\n", ufrag_pwd, sdp);
InjectAfter("a=mid:video\r\n", ufrag_pwd, sdp);
InjectAfter("a=mid:data\r\n", ufrag_pwd, sdp);
}
void RemoveBundle(std::string* sdp) {
RemoveLinesFromSdp("a=group:BUNDLE", sdp);
}
} // namespace
class PeerConnectionEndToEndTest
: public sigslot::has_slots<>,
public testing::Test {
public:
PeerConnectionEndToEndTest()
: caller_(new talk_base::RefCountedObject<PeerConnectionTestWrapper>(
"caller")),
callee_(new talk_base::RefCountedObject<PeerConnectionTestWrapper>(
"callee")) {
talk_base::InitializeSSL(NULL);
}
void CreatePcs() {
CreatePcs(NULL);
}
void CreatePcs(const MediaConstraintsInterface* pc_constraints) {
EXPECT_TRUE(caller_->CreatePc(pc_constraints));
EXPECT_TRUE(callee_->CreatePc(pc_constraints));
PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get());
}
void GetAndAddUserMedia() {
FakeConstraints audio_constraints;
FakeConstraints video_constraints;
GetAndAddUserMedia(true, audio_constraints, true, video_constraints);
}
void GetAndAddUserMedia(bool audio, FakeConstraints audio_constraints,
bool video, FakeConstraints video_constraints) {
caller_->GetAndAddUserMedia(audio, audio_constraints,
video, video_constraints);
callee_->GetAndAddUserMedia(audio, audio_constraints,
video, video_constraints);
}
void Negotiate() {
caller_->CreateOffer(NULL);
}
void WaitForCallEstablished() {
caller_->WaitForCallEstablished();
callee_->WaitForCallEstablished();
}
void SetupLegacySdpConverter() {
caller_->SignalOnSdpCreated.connect(
this, &PeerConnectionEndToEndTest::ConvertToLegacySdp);
callee_->SignalOnSdpCreated.connect(
this, &PeerConnectionEndToEndTest::ConvertToLegacySdp);
}
void ConvertToLegacySdp(std::string* sdp) {
UseExternalSdes(sdp);
UseGice(sdp);
RemoveBundle(sdp);
LOG(LS_INFO) << "ConvertToLegacySdp: " << *sdp;
}
void SetupGiceConverter() {
caller_->SignalOnIceCandidateCreated.connect(
this, &PeerConnectionEndToEndTest::AddGiceCredsToCandidate);
callee_->SignalOnIceCandidateCreated.connect(
this, &PeerConnectionEndToEndTest::AddGiceCredsToCandidate);
}
void AddGiceCredsToCandidate(std::string* sdp) {
std::string gice_creds = " username ";
gice_creds.append(kExternalGiceUfrag);
gice_creds.append(" password ");
gice_creds.append(kExternalGicePwd);
gice_creds.append("\r\n");
Replace("\r\n", gice_creds, sdp);
LOG(LS_INFO) << "AddGiceCredsToCandidate: " << *sdp;
}
~PeerConnectionEndToEndTest() {
talk_base::CleanupSSL();
}
protected:
talk_base::scoped_refptr<PeerConnectionTestWrapper> caller_;
talk_base::scoped_refptr<PeerConnectionTestWrapper> callee_;
};
// Disable for TSan v2, see
// https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
#if !defined(THREAD_SANITIZER)
TEST_F(PeerConnectionEndToEndTest, Call) {
CreatePcs();
GetAndAddUserMedia();
Negotiate();
WaitForCallEstablished();
}
// Disabled per b/14899892
TEST_F(PeerConnectionEndToEndTest, DISABLED_CallWithLegacySdp) {
FakeConstraints pc_constraints;
pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
false);
CreatePcs(&pc_constraints);
SetupLegacySdpConverter();
SetupGiceConverter();
GetAndAddUserMedia();
Negotiate();
WaitForCallEstablished();
}
#endif // if !defined(THREAD_SANITIZER)