JSEP implementations are required to always generate offers with
"actpass", but remote endpoints are not. So we should accept remote
offers with the current negotiated DTLS role.
This was recently clarified in dtls-sdp; it was somewhat ambiguous
before.
Also doing a bit of refactoring of JsepTransport (making a method
private that should have been private, fixing unit tests that were
directly calling said method).
BUG=webrtc:7072
Review-Url: https://codereview.webrtc.org/2770903003
Cr-Commit-Position: refs/heads/master@{#17396}
Extract the remote addresses from SDP c= line on both session level and
media level. The media level address will overwrite the session level one if
exists.
WebRTC is not using c= and this is used for new SDP parsing API.
BUG=webrtc:7311
Review-Url: https://codereview.webrtc.org/2742903002
Cr-Commit-Position: refs/heads/master@{#17326}
To simplify things, the candidate pool is only used in the first
offer/answer.
After setting a local description, the size is frozen, and changing ICE
servers won't refresh the pool.
After setting an answer, the pooled candidates are discarded.
BUG=webrtc:5180
Review-Url: https://codereview.webrtc.org/2717893003
Cr-Commit-Position: refs/heads/master@{#17178}
Add an attribute to the RTCConfiguration which can be used by specific
mobile devices so that the IPv6 ICE candidates on WiFi will not be collected.
BUG=b/35725283
Review-Url: https://codereview.webrtc.org/2731813002
Cr-Commit-Position: refs/heads/master@{#17100}
vector<unique_ptr<StunAttribute>> as suggested in a TODO.
BUG=NONE
Review-Url: https://codereview.webrtc.org/2735523002
Cr-Commit-Position: refs/heads/master@{#17076}
This isn't used any more so there's no point in maintaining it.
BUG=None
Review-Url: https://codereview.webrtc.org/2731673002
Cr-Commit-Position: refs/heads/master@{#17016}
Introduing IceTransportInternal2 is a temporary fix to switch the base
class of P2PTransportChannel to IceTransportInternal without breaking
Chromium. It is removed in this CL.
Reland this CL after Chromium doesn't depend on this.
BUG=webrtc:6951
Review-Url: https://codereview.webrtc.org/2632563002
Cr-Original-Commit-Position: refs/heads/master@{#16131}
Committed: c7953fa716
Review-Url: https://codereview.webrtc.org/2632563002
Cr-Commit-Position: refs/heads/master@{#16960}
Collected in accordance with the spec:
https://w3c.github.io/webrtc-stats/#candidatepair-dict*
totalRoundTripTime is collected as the sum of rtt measurements, it was
previously not collected.
currentRoundTripTime is collected as the latest rtt measurement, it
was previously collected as a smoothed value, which was incorrect.
Connection is updated to collect these values which are surfaced
through ConnectionInfo.
BUG=webrtc:7062, webrtc:7204
Review-Url: https://codereview.webrtc.org/2719523002
Cr-Commit-Position: refs/heads/master@{#16905}
Connection::nominated() is updated to mean
(remote_nomination_ || acked_nomination_), which means both a
controlling and controlled agent can be said to be "nominated".
Previously this was (remote_nomination_ > 0) which only applies to the
controlling agent.
PortTest.TestNomination added to test nomination values and nomination
stat.
This value is surfaced through cricket::ConnectionInfo::nominated.
RTCStatsCollector uses this value in its collection of
RTCIceCandidatePairStats.
RTCStatsCollectorTest.CollectRTCIceCandidatePairStats updated to test
that ConnectionInfo::nominated is surfaced using mocks.
rtcstats_integrationtest.cc updated to expect nomination set without
using mocks.
Spec: https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-nominated
BUG=webrtc:7062, webrtc:7204
Review-Url: https://codereview.webrtc.org/2709293004
Cr-Commit-Position: refs/heads/master@{#16855}
This CL adds the following interfaces:
* RtpTransportController
* RtpTransport
* RtpSender
* RtpReceiver
They're implemented on top of the "BaseChannel" object, which is normally used
in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result
of this, there are several limitations:
* You can only have one of each type of sender and receiver (audio/video) on top
of the same transport controller.
* The sender/receiver with the same media type must use the same RTP transport.
* You can't change the transport after creating the sender or receiver.
* Some of the parameters aren't supported.
Later, these "adapter" objects will be gradually replaced by real objects that don't
have these limitations, as "BaseChannel", "MediaChannel" and related code is
restructured. In this CL, we essentially have:
ORTC adapter objects -> BaseChannel -> Media engine
PeerConnection -> BaseChannel -> Media engine
And later we hope to have simply:
PeerConnection -> "Real" ORTC objects -> Media engine
See the linked bug for more context.
BUG=webrtc:7013
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2675173003
Cr-Commit-Position: refs/heads/master@{#16842}
Where "TRANSPORT attributes" refers to:
https://tools.ietf.org/html/draft-ietf-mmusic-sdp-mux-attributes-16
The BUNDLE draft now says that these attributes can
(in fact, MUST) be omitted when m= sections are bundled
(they only need to go in one of the bundled m= sections),
so we should start accepting that SDP.
This CL doesn't fix "a=rtcp-mux", unfortunately. That will be easier
to fix once we've split apart an "RtpTransport" object from
BaseChannel.
BUG=webrtc:6351
Review-Url: https://codereview.webrtc.org/2647593003
Cr-Commit-Position: refs/heads/master@{#16782}
This reduces binary size considerably and solves some other problems.
Also rewrote using variadic templates.
Initial patch contributed by andrey.semashev@gmail.com.
BUG=webrtc:2305
Review-Url: https://codereview.webrtc.org/2509733003
Cr-Commit-Position: refs/heads/master@{#16703}
downstream application depends on it.
Mark the old Port::AddAddress deprecated and will be removed after the
applications stop replying on it.
BUG=None.
R=deadbeef@webrtc.org
Review-Url: https://codereview.webrtc.org/2694103003 .
Cr-Commit-Position: refs/heads/master@{#16598}
The URL of the ICE server will be reconstructed by the Port and the URL
attribute is added to the cricket::Candidate struct so that we can tell
which ICE server the candidate was gathered from.
This CL only changes the native C++ code. The Java and Objc wrapper will
be created in separate CLs.
BUG=webrtc::7128
Review-Url: https://codereview.webrtc.org/2685053004
Cr-Commit-Position: refs/heads/master@{#16591}
This is the naming scheme we've been using for internal interfaces.
Also, this CL will introduce a PacketTransportInterface in the webrtc namespace,
which would get too easily confused with the rtc:: one:
https://codereview.webrtc.org/2675173003/
BUG=None
Review-Url: https://codereview.webrtc.org/2679103006
Cr-Commit-Position: refs/heads/master@{#16539}
Replaced by assigning value to a local variable, followed by a DCHECK.
Also deletes dead test code under the always false TEST_DIGEST define.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2623473004
Cr-Commit-Position: refs/heads/master@{#16476}
This CL replaces the previously-hardcoded DTLS timeout value of 50 ms with one that is picked dynamically as 2x ICE RTT (clamped between 50 and 3000 ms to keep it reasonable).
No tests yet - I'm working on these, but sending out this CL early to get feedback. I've tested it manually and on slow networks it results in successful DTLS setup in a single attempt - instead of clogging up the slow link with multiple retransmissions as done previosly.
BUG=webrtc:7099
Review-Url: https://codereview.webrtc.org/2670113002
Cr-Commit-Position: refs/heads/master@{#16426}
If an application sets a non-null value in RTCConfiguration.iceCheckMinInterval, we do not sent STUN pings more often than that. This is useful for bandwidth constrained scenarios.
This CL also increases the maximum STUN ping timeout to 60 seconds up from its previous value of 5 (which meant that a ping response received 5 seconds later would not be counted), and allows the RTT estimate to go up to 60 seconds from its previous limit of 3. RTTs above 3 seconds are possible on mobile links. (webrtc:7109)
This CL was originally written by pthatcher@, I am just submitting it after a minor cleanup.
BUG=webrtc:7082, webrtc:7109
Review-Url: https://codereview.webrtc.org/2670053002
Cr-Commit-Position: refs/heads/master@{#16421}
Proper error handling was missing, using VERIFY to crash in debug
builds, while release builds would ignore the error and leak the
attribute memory. The check of attribute type consistency was changed
to a RTC_DCHECK.
Also removes a large number of uses of the deprecated VERIFY macro.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2665343002
Cr-Commit-Position: refs/heads/master@{#16413}
Bulk of the changes were done using
git grep -l '#include "webrtc/base/common.h"' | \
xargs sed -i '\,^#include.*webrtc/base/common\.h,d'
followed by adding back the include in the few places where it is
still needed, and in one case (pseudotcp.cc) instead deleting its use
of RTC_UNUSED.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2644103002
Cr-Commit-Position: refs/heads/master@{#16263}
... As opposed to DtlsTransportInternal.
The code is suboptimal right now, storing two pointers to the different
interfaces. This will all be cleaned up when we have an "RtpTransport"
abstraction that BaseChannel can use.
This CL also cleans up the "fake transport" classes a bit, and gives
them their own header files.
BUG=None
Review-Url: https://codereview.webrtc.org/2648233003
Cr-Commit-Position: refs/heads/master@{#16258}
These defines don't work any more, so they only cause confusion:
FEATURE_ENABLE_SSL
HAVE_OPENSSL_SSL_H
SSL_USE_OPENSSL
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2640513002
Cr-Commit-Position: refs/heads/master@{#16224}
Create a new target //webrtc/api:libjingle_peerconnection_api and start moving
things into it. Move remaining parts of //webrtc/api:libjingle_peerconnection
to //webrtc/pc:libjingle_peerconnection.
Moved the RTCStatsCollectorCallback into its own header file, so that
PeerConnectionInterface can include that instead of pulling in
RTCStatsCollector and PeerConnection and everything.
Separated cricket::MediaType into its own header/source set, so that it
can be used in the api.
BUG=webrtc:5883
Review-Url: https://codereview.webrtc.org/2514883002
Cr-Commit-Position: refs/heads/master@{#16210}
This is used for fallback from DTLS to SDES encryption, which we probably still
want to support. Setting a certificate puts the DTLS transport in a "DTLS
enabled" mode, so it should be delayed until SDP with "a=fingerprint" is set.
BUG=webrtc:6972
Review-Url: https://codereview.webrtc.org/2641633002
Cr-Commit-Position: refs/heads/master@{#16199}
Reason for revert:
Broke chromium build, due to a config being removed. Will add it back and remove the dependency in a chromium CL.
Original issue's description:
> Removing #defines previously used for building without BoringSSL/OpenSSL.
>
> These defines don't work any more, so they only cause confusion:
>
> FEATURE_ENABLE_SSL
> HAVE_OPENSSL_SSL_H
> SSL_USE_OPENSSL
>
> BUG=webrtc:7025
>
> Review-Url: https://codereview.webrtc.org/2640513002
> Cr-Commit-Position: refs/heads/master@{#16196}
> Committed: eaa826c2eeTBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2648003003
Cr-Commit-Position: refs/heads/master@{#16197}
These defines don't work any more, so they only cause confusion:
FEATURE_ENABLE_SSL
HAVE_OPENSSL_SSL_H
SSL_USE_OPENSSL
BUG=webrtc:7025
Review-Url: https://codereview.webrtc.org/2640513002
Cr-Commit-Position: refs/heads/master@{#16196}
DtlsTransportChannelWrapper is renamed to be DtlsTransport which inherits from
DtlsTransportInternal. There will be no concept of "channel" in p2p level.
Both P2PTransportChannel and DtlsTransport don't depend on TransportChannel
and TransportChannelImpl any more and they are removed in this CL.
BUG=none
Review-Url: https://codereview.webrtc.org/2606123002
Cr-Commit-Position: refs/heads/master@{#16173}
Previously PseudoTcp::process() didn't handle the case when the peer
sends a packet that's outside of the receive window, which was causing
DCHECK failures in the fuzzer.
BUG=681849
Review-Url: https://codereview.webrtc.org/2640173002
Cr-Commit-Position: refs/heads/master@{#16169}
Reason for revert:
Failed the memory check.
May need to fix the memory leak.
Original issue's description:
> make the DtlsTransportWrapper inherit form DtlsTransportInternal
>
> BUG=none
>
> Review-Url: https://codereview.webrtc.org/2606123002
> Cr-Commit-Position: refs/heads/master@{#16160}
> Committed: 5aed06c8d3TBR=deadbeef@webrtc.org,pthatcher@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=none
Review-Url: https://codereview.webrtc.org/2639203004
Cr-Commit-Position: refs/heads/master@{#16162}
The factory follows the same principles as PeerConnectionFactory;
various modules can be passed into its constructor but default
implementations are provided. Currently the only object the factory can
create is a UdpTransport (need to start somewhere).
UdpTransportChannel (renamed to UdpTransport)
will now accept a socket passed into its constructor,
relying on the factory to create the socket. This allows some
simplifications to be made, such as getting rid of "State" since the
only states are now "has destination set or doesn't".
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/2632613002
Cr-Commit-Position: refs/heads/master@{#16154}
In top level test functions, replaced with gtest ASSERT_*. In helper
methods in main test files, replaced with EXPECT_* or RTC_DCHECK on a
case-by-case basis.
In separate mock/fake classes used by tests (which might be of some
use also in tests of third-party applications), ASSERT was replaced
with RTC_CHECK, using
git grep -l ' ASSERT(' | grep -v common.h | \
xargs sed -i 's/ ASSERT(/ RTC_CHECK(/'
followed by additional includes of base/checks.h in affected files,
and git cl format.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2622413005
Cr-Commit-Position: refs/heads/master@{#16150}
Introduing IceTransportInternal2 is a temporary fix to switch the base
class of P2PTransportChannel to IceTransportInternal without breaking
Chromium. It is removed in this CL.
Reland this CL after Chromium doesn't depend on this.
BUG=webrtc:6951
Review-Url: https://codereview.webrtc.org/2632563002
Cr-Commit-Position: refs/heads/master@{#16131}
Reason for revert:
Broke Chromium FYI bots. See for example https://build.chromium.org/p/chromium.webrtc.fyi/builders/Win%20Builder/builds/8631
FAILED: obj/remoting/protocol/protocol/ice_transport_channel.obj
ninja -t msvc -e environment.x86 -- E:\b\c\cipd\goma/gomacc.exe "E:\b\depot_tools\win_toolchain\vs_files\d3cb0e37bdd120ad0ac4650b674b09e81be45616\VC\bin\amd64_x86/cl.exe" /nologo /showIncludes /FC @obj/remoting/protocol/protocol/ice_transport_channel.obj.rsp /c ../../remoting/protocol/ice_transport_channel.cc /Foobj/remoting/protocol/protocol/ice_transport_channel.obj /Fd"obj/remoting/protocol/protocol_cc.pdb"
e:\b\c\b\win_builder\src\remoting\protocol\ice_transport_channel.cc(186): error C2039: 'IceTransportInternal2': is not a member of 'cricket'
e:\b\c\b\win_builder\src\third_party\webrtc\p2p\base\p2ptransportchannel.h(38): note: see declaration of 'cricket'
e:\b\c\b\win_builder\src\remoting\protocol\ice_transport_channel.cc(186): error C2065: 'IceTransportInternal2': undeclared identifier
e:\b\c\b\win_builder\src\remoting\protocol\ice_transport_channel.cc(186): error C2065: 'ice_transport': undeclared identifier
e:\b\c\b\win_builder\src\remoting\protocol\ice_transport_channel.cc(187): error C2059: syntax error: 'const'
e:\b\c\b\win_builder\src\remoting\protocol\ice_transport_channel.cc(187): error C2143: syntax error: missing ';' before '{'
e:\b\c\b\win_builder\src\remoting\protocol\ice_transport_channel.cc(187): error C2447: '{': missing function header (old-style formal list?)
Original issue's description:
> Remove the IceTransportInternal2.
>
> Introduing IceTransportInternal2 is a temporary fix to switch the base
> class of P2PTransportChannel to IceTransportInternal without breaking
> Chromium. It is removed in this CL.
>
> BUG=webrtc:6951
>
> Review-Url: https://codereview.webrtc.org/2632563002
> Cr-Commit-Position: refs/heads/master@{#16083}
> Committed: 8aaa51190eTBR=deadbeef@webrtc.org,zhihuang@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6951
Review-Url: https://codereview.webrtc.org/2630243002
Cr-Commit-Position: refs/heads/master@{#16086}
Introduing IceTransportInternal2 is a temporary fix to switch the base
class of P2PTransportChannel to IceTransportInternal without breaking
Chromium. It is removed in this CL.
BUG=webrtc:6951
Review-Url: https://codereview.webrtc.org/2632563002
Cr-Commit-Position: refs/heads/master@{#16083}
The BaseChannel can set the transport directly without depending on
TransportController.
When initializing the network of the BaseChannel, the ChannelManager will
create TransportChannels with the TransportController.
When enabling bundling, WebRtcSession will get or create TransportChannels
with the TransportController.
When a TransportChannel of the BaseChannel needs to be destroyed, it will
fire a signal to notify the WebRtcSession.
BUG=none.
Review-Url: https://codereview.webrtc.org/2614263002
Cr-Commit-Position: refs/heads/master@{#16043}
Make P2PTransportChannel inherit from IceTransportInternal instead of
TransportChannelImpl and TransportChannel, so that the DTLS-related methods can
be separated from P2PTransportChannel.
BUG=webrtc:6951
Review-Url: https://codereview.webrtc.org/2608353003
Cr-Commit-Position: refs/heads/master@{#16041}
