In some cases, the DTLS ClientHello may arrive before the server's
transport is writable (before it receives a STUN ping response), or
even before it receives a remote fingerprint. If this packet is
discarded, it may take a second for a it to be sent again.
So, this CL caches it instead of dropping it, and feeds it into
the SSL library once the handshake has been started.
BUG=webrtc:5789
Review-Url: https://codereview.webrtc.org/1912323002
Cr-Commit-Position: refs/heads/master@{#12634}
The problem with gmock is worked around by commenting out any other override declarations in classes using gmock.
NOPRESUBMIT=True
BUG=webrtc:3970
Review-Url: https://codereview.webrtc.org/1921653002
Cr-Commit-Position: refs/heads/master@{#12563}
This propagated into various other places. Also had to #include headers that
were implicitly pulled by "scoped_ptr.h".
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1920043002
Cr-Commit-Position: refs/heads/master@{#12501}
Instead of using a raw pointer output parameter. This affects
SSLStreamAdapter::GetPeerCertificate
Transport::GetRemoteSSLCertificate
TransportChannel::GetRemoteSSLCertificate
TransportController::GetRemoteSSLCertificate
WebRtcSession::GetRemoteSSLCertificate
This is a good idea in general, but will also be very convenient when
scoped_ptr is gone, since unique_ptr doesn't have an .accept() method.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1802013002
Cr-Commit-Position: refs/heads/master@{#12262}
Also changed from unsigned to signed integer per the style guide.
By the way, I kept all delta-times to be 32-bit int.
The only things left in the p2p dir are
1. proberprober/main.cc where Time() is used as the input for a random number.
2. pseudotcp.cc: where 32-bit time info is sent over the wire.
BUG=webrtc:5636
Review URL: https://codereview.webrtc.org/1793553002
Cr-Commit-Position: refs/heads/master@{#12019}
and signaling the remote side to remove its remote candidate by setting the candidate priority to 0.
BUG=
Review URL: https://codereview.webrtc.org/1648813004
Cr-Commit-Position: refs/heads/master@{#11958}
Also change the type of "time interval" to int from uint32.
Fixed a few TODO therein. I think we should have the following convention:
1. All time delay/intervals should have type int although the time instant should have time uint32_t.
2. "interval" is preferred to "delay" if the delay will be repeated (like rescheduling).
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1762863002 .
Cr-Commit-Position: refs/heads/master@{#11888}
This appears to be dead code because GetTransport() is not used by WebRTC. It also adds dead code to DtlsTransportChannelWrapper and P2PTransportChannel.
BUG=
Review URL: https://codereview.webrtc.org/1691673002
Cr-Commit-Position: refs/heads/master@{#11662}
Reason for revert:
Broke chromium fyi build.
Original issue's description:
> Convert internal representation of Srtp cryptos from string to int.
>
> Note that the coversion from int to string happens in 3 places
> 1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
> 2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
> 3) stats collection also needs external names.
>
> External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
> Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.
>
> The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().
>
> BUG=webrtc:5043
>
> Committed: https://crrev.com/2764e1027a08a5543e04b854a27a520801faf6eb
> Cr-Commit-Position: refs/heads/master@{#10701}
TBR=juberti@webrtc.org,pthatcher@webrtc.org,juberti@google.com
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5043
Review URL: https://codereview.webrtc.org/1455233005
Cr-Commit-Position: refs/heads/master@{#10702}
Note that the coversion from int to string happens in 3 places
1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
3) stats collection also needs external names.
External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.
The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().
BUG=webrtc:5043
Review URL: https://codereview.webrtc.org/1416673006
Cr-Commit-Position: refs/heads/master@{#10701}
This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting.
For SRTP, currently it's still string internally but is reported as IANA number.
This is used by the ongoing CL https://codereview.chromium.org/1335023002.
BUG=523033
Review URL: https://codereview.webrtc.org/1337673002
Cr-Commit-Position: refs/heads/master@{#10124}
p2ptransportchannel. This CL does not use the new policy yet.
BUG=
Review URL: https://codereview.webrtc.org/1369773003
Cr-Commit-Position: refs/heads/master@{#10092}
Reason for revert:
This CL just landed: https://codereview.chromium.org/1323243006/
Which fixes the FYI bots for the original CL, and breaks them for this revert.
Original issue's description:
> Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
>
> Reason for revert:
> This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
>
> Original issue's description:
> > TransportController refactoring.
> >
> > Getting rid of TransportProxy, and in its place adding a
> > TransportController class which will facilitate access to and manage
> > the lifetimes of Transports. These Transports will now be accessed
> > solely from the worker thread, simplifying their implementation.
> >
> > This refactoring also pulls Transport-related code out of BaseSession.
> > Which means that BaseChannels will now rely on the TransportController
> > interface to create channels, rather than BaseSession.
> >
> > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> > Cr-Commit-Position: refs/heads/master@{#10022}
>
> TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c
> Cr-Commit-Position: refs/heads/master@{#10024}
TBR=pthatcher@webrtc.org,torbjorng@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1361773005
Cr-Commit-Position: refs/heads/master@{#10036}
Reason for revert:
This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
Original issue's description:
> TransportController refactoring.
>
> Getting rid of TransportProxy, and in its place adding a
> TransportController class which will facilitate access to and manage
> the lifetimes of Transports. These Transports will now be accessed
> solely from the worker thread, simplifying their implementation.
>
> This refactoring also pulls Transport-related code out of BaseSession.
> Which means that BaseChannels will now rely on the TransportController
> interface to create channels, rather than BaseSession.
>
> Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> Cr-Commit-Position: refs/heads/master@{#10022}
TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1358413003
Cr-Commit-Position: refs/heads/master@{#10024}
Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.
This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.
Review URL: https://codereview.webrtc.org/1350523003
Cr-Commit-Position: refs/heads/master@{#10022}
Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.
This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.
This CL also adds some unit tests, and does some renaming.
For example, from "CandidateReady" to "CandidateGathered".
Review URL: https://codereview.webrtc.org/1246913005
Cr-Commit-Position: refs/heads/master@{#9993}
