Bug 4865: even without STUN/TURN, as long as the peer is on the open internet, the connectivity should work. This is actually a regression even for hangouts.
We need to issue the 0.0.0.0 candidate into Port::candidates_ and filter it out later. The reason is that when we create connection, we need a local candidate to match the remote candidate.
The same connection later will be updated with the prflx local candidate once the STUN ping response is received.
BUG=webrtc:4865
R=juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1274013002 .
Cr-Commit-Position: refs/heads/master@{#9708}
DtlsIdentityStoreImpl is updated to take KeyType into account, something which will be relevant after this CL lands:
https://codereview.webrtc.org/1189583002
The DtlsIdentityService[Interface] classes are about to be removed (to be removed when Chromium no longer implements and uses the interface). This was an unnecessary layer of complexity. The FakeIdentityService is now instead a FakeDtlsIdentityStore.
Where a service was previously passed around, a store is now passed around.
Identity generation is now commonly performed using DtlsIdentityStoreInterface. Previously, if a service was not specified, WebRtcSessionDescriptionFactory could fall back on its own generation code. Now, a store has to be provided for generation to occur.
For more information about the steps being taken to land this without breaking Chromium, see referenced bug.
BUG=webrtc:4899
R=magjed@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1176383004 .
Cr-Commit-Position: refs/heads/master@{#9696}
New PeerConnectionFactoryInterface::CreatePeerConnection taking both service and store added (old CreatePC signature still exists).
This is CL is part of an effort to land https://codereview.webrtc.org/1176383004 without breaking Chromium.
See bug for more information.
BUG=webrtc:4899
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1268363002 .
Cr-Commit-Position: refs/heads/master@{#9680}
Sometimes the port still try to send stun packet when the connection is disconnected,
causing an assertion error.
BUG=4859
Review URL: https://codereview.webrtc.org/1247573002
Cr-Commit-Position: refs/heads/master@{#9671}
Using explicit atomic operations permits TSan to understand them and
prevents false positives.
Downgrading the atomic Load to acquire semantics. This reduces the
number of memory barriers inserted from two down to one at most.
Also renaming Load/Store to AcquireLoad/ReleaseStore.
BUG=chromium:512382
R=dvyukov@chromium.org, glider@chromium.orgTBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1246073002
Cr-Commit-Position: refs/heads/master@{#9613}
This was already working in most cases, but not for some corner cases:
* If the PORTALLOCATOR_ENABLE_SHARED_SOCKET flag is not set
* If both a STUN server and TURN server are configured
I added unit tests for these cases, and centralized the code that gets
STUN server addresses in order to fix these and any related issues.
BUG=webrtc:4215
Review URL: https://codereview.webrtc.org/1215713003
Cr-Commit-Position: refs/heads/master@{#9596}
Fixes
..\..\third_party\webrtc/base/stringutils.h(295,49) : warning(clang): extra qualification on member "empty_str" [-Wmicrosoft]
No behavior change, but makes the code more standards-conformant.
BUG=chromium:505296
Review URL: https://codereview.webrtc.org/1228193002
Cr-Commit-Position: refs/heads/master@{#9562}
Note: Regarding the ICMP6_CLOSE_FUNC variable in winping.cc,
Icmp6CloseHandle does not exist, and IcmpCloseHandle is the correct way
to close an IPv6 handle. Therefore the existing code is correct to use
close_ on both types of connections and this variable is unnecessary.
BUG=505319
Review URL: https://codereview.webrtc.org/1231653003
Cr-Commit-Position: refs/heads/master@{#9555}
Clang warns if there are missing braces around a subobject
initializer. The most common idiom that triggers this is:
STRUCT s = {0};
if the first field of STRUCT is itself a struct. This can
be more simply written as:
STRUCT s = {};
which also prevents the warning from firing.
Other instances of the warning have been fixed by adding
braces where appropriate.
BUG=505297
TBR=jiayl@webrtc.org
Review URL: https://codereview.webrtc.org/1216353002
Cr-Commit-Position: refs/heads/master@{#9529}
The old OpenSSL threading hooks were removed in favor of the library knowing
about threads internally. Instead of CRYPTO_add, use FOO_up_ref wrappers that
don't require reaching into the type.
BUG=none
R=jiayl@webrtc.org, juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/54579004
Cr-Commit-Position: refs/heads/master@{#9324}
If srtp_create fails while adding streams, it deallocates the session
but doesn't clear the passed pointer which then could lead to a
double-free in the SrtpSession dtor.
The CL also adds locking for libsrtp initialization / shutdown.
BUG=4042
R=jiayl@webrtc.org, juberti@google.com, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47319004
Cr-Commit-Position: refs/heads/master@{#9300}
Add pylintrc file based on
https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc
bit tightened up quite a bit (the one in depot_tools is far
more relaxed).
Remove a few excluded directories from pylint check and fixed/
suppressed all warnings generated.
Add GN format check + formatted all GN files using 'gn format'.
Cleanup redundant rules in tools/PRESUBMIT.py
TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations.
Ran it again with a modification in webrtc/build/webrtc.gni, formatted
all the GN files and ran it again.
R=henrika@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50069004
Cr-Commit-Position: refs/heads/master@{#9274}
This is being done in preparation of moving base/logging.* to rtc_base_approved. base/stream.* has libjingle dependencies that webrtc can't use, so logging.* can't depend on streams. It does look like stream.* isn't used much, so cleaning that up as well as cleaning up usage of the actual stream support (now LogStream) in the logging code, is in order, but I'll leave that to another cl.
BUG=
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54529004
Cr-Commit-Position: refs/heads/master@{#9269}
I wanted to use Config::Get in Chromium code, but it triggered the following
warning:
../../third_party/webrtc/common.h:89:20: error: declaration requires an exit-time destructor [-Werror,-Wexit-time-destructors]
static const T def;
^
../../third_party/webrtc/common.h:110:10: note: in instantiation of function template specialization requested here
return default_value<T>();
^
I assume we don't hit this in webrtc because the warning is disabled.
This also switches to the RTC_ prefix from the deprecated LIBJINGLE_.
Needed due to this Chromium CL:
https://codereview.chromium.org/1148843004/R=andresp@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/53459004
Cr-Commit-Position: refs/heads/master@{#9268}
The current use of rtc::FifoBuffer can lead to reading across DTLS packet
boundaries which could cause packets to not being processed correctly.
This CL introduces the new class rtc::BufferQueue and changes the
StreamInterfaceChannel to use it instead of the rtc::FifoBuffer.
BUG=chromium:447431
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/52509004
Cr-Commit-Position: refs/heads/master@{#9254}
Previously SHA1Transform() kept a static buffer. As result SHA1 was not
always computed correctly when running that code in parallel on multiple
threads. That was causing spurious messages about invalid Message
Integrity attribute when running some tests in chromoting.
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/57379004
Cr-Commit-Position: refs/heads/master@{#9238}
This CL adds an API to SSL stream adapters to set the maximum allowed
protocol version and with that implements support for DTLS 1.2.
With DTLS 1.2 the default cipher changes in the unittests as follows.
BoringSSL
TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA -> TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
NSS
TLS_RSA_WITH_AES_128_CBC_SHA -> TLS_RSA_WITH_AES_128_GCM_SHA256
BUG=chromium:428343
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/50989004
Cr-Commit-Position: refs/heads/master@{#9232}
