For ICE, we want sockets that are bound to specific network interfaces,
rather than to specific IP addresses. So, a while ago, we added a
"Network" class that gets passed into the Port constructor, in
addition to the IP address as before.
But we never finished the job of removing the IP address field, such that
a Port only guarantees something about the network interface it's
associated with, and not the specific IP address it ends up with.
This CL does that, and as a consequence, if a port ends up bound to
an IP address other than the "best" one (returned by Network::GetBestIP),
this *won't* be treated as an error.
This is relevant to Android, where even though we pass an IP address
into "Bind" as a way of identifying the network, the socket actually
gets bound using "android_setsocknetwork", which doesn't provide any
guarantees about the IP address. So, if a network interface has multiple
IPv6 addresses (for instance), we may not correctly predict the one
the OS will choose, and that's ok.
This CL also moves "SetAlternateLocalAddress" from VirtualSocket to
VirtualSocketServer, which makes for much more readable test code.
The next step, if there is one, is to pass along the Network class all
the way to SocketServer::Bind. Then the socket server could do smart
things with the network information. We could even stick a platform-
specific network handle in the Network object, such that the socket
server could use it for the binding, or for "sendmsg", for example.
See bug 7026 for more context about the sendmsg idea.
BUG=webrtc:7715
Review-Url: https://codereview.webrtc.org/2989303002
Cr-Commit-Position: refs/heads/master@{#19251}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
This CL adds the flag "PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS", which will
force the creation of ports not bound to any specific network interface.
These are normally only used when network enumeration fails or is disabled,
but in some circumstances (such as the one the test case adds), they're the
only thing that works.
This will result in extra ports being gathered, which is why it's only enabled
behind a flag for now. In the future, we could probably introduce more
sophisticated "pruning" logic that would lessen the impact of the extra ports
when they're redundant, and make the flag the default.
Some other minor changes that were required to make this use case work:
* Allow a TCPPort to be used for outgoing connections even if it tries and
fails to create a server socket.
* Allow Bind to fail if being called before Connect, and the IP is an "any"
address (0.0.0.0 or ::), since this bind would have been mostly pointless
anyway.
* Prevent P2PTransprotChannel from keeping a "backup" candidate pair using
an "any address" network; we only want this for actual networks.
BUG=webrtc:7798
Review-Url: https://codereview.webrtc.org/2936553003
Cr-Commit-Position: refs/heads/master@{#18578}
The signal was only being hooked up for incoming connections, not
outgoing connections.
As a result, the bandwidth estimator didn't know when packets were sent
and couldn't calculate delays.
BUG=webrtc:7509
Review-Url: https://codereview.webrtc.org/2834083002
Cr-Commit-Position: refs/heads/master@{#17817}
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}
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}
Bulk of the changes were produced using
git grep -l ' ASSERT(' | grep -v test | grep -v 'common\.h' |\
xargs -n1 sed -i 's/ ASSERT(/ RTC_DCHECK(/'
followed by additional includes of base/checks.h in affected files,
and git cl format.
Also had to do some tweaks to #if !defined(NDEBUG) logic in the
taskrunner code (webrtc/base/task.cc, webrtc/base/taskparent.cc,
webrtc/base/taskparent.h, webrtc/base/taskrunner.cc), replaced to
consistently use RTC_DCHECK_IS_ON, and some of the checks needed
additional #if protection.
Test code was excluded, because it should probably use RTC_CHECK
rather than RTC_DCHECK.
BUG=webrtc:6424
Review-Url: https://codereview.webrtc.org/2620303003
Cr-Commit-Position: refs/heads/master@{#16030}
Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
that it's not actually some kind of SSL over TCP. Also making it clear
that it's mutually exclusive with OPT_TLS. Maintaining deprecated
backwards compatible support for "OPT_SSLTCP".
Add "OPT_TLS_INSECURE" that implements the new certificate-check
disabled TLS mode, which is also mutually exclusive with the other
TLS options.
PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
the new insecure mode and added it as a RelayCredentials member.
TurnPort: Add new TLS policy member with appropriate getter and setter
to avoid constructor bloat. Initialize it from the RelayCredentials
after the TurnPort is created.
Expose the new feature in the PeerConnection API via
IceServer.tls_certificate_policy as well as via the Android JNI
PeerConnection API.
For security reasons we ensure that:
1) The policy is always explicitly initialized to secure.
2) API users have to explicitly integrate with the feature to
use it, and will otherwise get no change in behavior.
3) The feature is not immediately exposed in non-native
contexts. For example, disabling of certificate validation
is not implemented via URI parsing since this would
immediately allow it to be used from a web page.
This is a second attempt of https://codereview.webrtc.org/2557803002/
which was rolled back in https://codereview.webrtc.org/2590153002/
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2594623002
Cr-Commit-Position: refs/heads/master@{#15967}
Reason for revert:
This CL broke all Chromium WebRTC FYI bots. A roll+fix was attempted here: https://codereview.chromium.org/2590783003/, but failed to land. I'm reverting this CL now to make the tree green again. Make the API change gradual when you reland so that we can update Chromium between.
Original issue's description:
> Add disabled certificate check support to IceServer PeerConnection API.
>
> Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
> that it's not actually some kind of SSL over TCP. Also making it clear
> that it's mutually exclusive with OPT_TLS.
>
> Add "OPT_TLS_INSECURE" that implements the new certificate-check
> disabled TLS mode, which is also mutually exclusive with the other
> TLS options.
>
> PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
> the new insecure mode and added it as a RelayCredentials member.
>
> TurnPort: Add new TLS policy member with appropriate getter and setter
> to avoid constructor bloat. Initialize it from the RelayCredentials
> after the TurnPort is created.
>
> Expose the new feature in the PeerConnection API via
> IceServer.tls_certificate_policy as well as via the Android JNI
> PeerConnection API.
>
> For security reasons we ensure that:
>
> 1) The policy is always explicitly initialized to secure.
> 2) API users have to explicitly integrate with the feature to
> use it, and will otherwise get no change in behavior.
> 3) The feature is not immediately exposed in non-native
> contexts. For example, disabling of certificate validation
> is not implemented via URI parsing since this would
> immediately allow it to be used from a web page.
>
> BUG=webrtc:6840
>
> Review-Url: https://codereview.webrtc.org/2557803002
> Cr-Commit-Position: refs/heads/master@{#15670}
> Committed: b0f04fdb9eTBR=pthatcher@webrtc.org,deadbeef@webrtc.org,hnsl@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2590153002
Cr-Commit-Position: refs/heads/master@{#15703}
Refactor "OPT_SSLTCP" renaming it to "OPT_TLS_FAKE", making it clear
that it's not actually some kind of SSL over TCP. Also making it clear
that it's mutually exclusive with OPT_TLS.
Add "OPT_TLS_INSECURE" that implements the new certificate-check
disabled TLS mode, which is also mutually exclusive with the other
TLS options.
PortAllocator: Add a new TLS policy enum TlsCertPolicy which defines
the new insecure mode and added it as a RelayCredentials member.
TurnPort: Add new TLS policy member with appropriate getter and setter
to avoid constructor bloat. Initialize it from the RelayCredentials
after the TurnPort is created.
Expose the new feature in the PeerConnection API via
IceServer.tls_certificate_policy as well as via the Android JNI
PeerConnection API.
For security reasons we ensure that:
1) The policy is always explicitly initialized to secure.
2) API users have to explicitly integrate with the feature to
use it, and will otherwise get no change in behavior.
3) The feature is not immediately exposed in non-native
contexts. For example, disabling of certificate validation
is not implemented via URI parsing since this would
immediately allow it to be used from a web page.
BUG=webrtc:6840
Review-Url: https://codereview.webrtc.org/2557803002
Cr-Commit-Position: refs/heads/master@{#15670}
This change makes WebRTC no longer stop sending video when we receive an
EWOULDBLOCK error from the operating system. This was previously
causing calls on a slow link (where the first hop is slow) to rapidly
oscillate between starting and stopping video.
We still do need to stop sending packets if there is no known good
connection we can use for that. We used to generate a synthetic
EWOULDBLOCK error in that case. This CL replaces it with a different
code (ENOTCONN); EWOULDBLOCK no longer stops the stream but ENOTCONN
does.
I've updated all the places where we seemed to be generating EWOULDBLOCK
for reasons other than some buffer been full; please give it a thorough
look in case I missed something.
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/2192963002 .
Cr-Commit-Position: refs/heads/master@{#13566}
Every message will now be traced with the location from which it was
posted, including function name, file and line number.
This CL also writes a normal LOG message when the dispatch took more
than a certain amount of time (currently 50ms).
This logging should help us identify messages that are taking
longer than expected to be dispatched.
R=pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2019423006 .
Cr-Commit-Position: refs/heads/master@{#13104}
Add sent_ping_requests, recv_ping_responses to ConnectionInfo.
recv_ping_responses_ will be incremented when OnConnectionRequestResponse() is called.
ent_ping_requests_ will be incremented when OnConnectionRequestSent() is called.
BUG=webrtc:5695
Review-Url: https://codereview.webrtc.org/1940493002
Cr-Commit-Position: refs/heads/master@{#13001}
If the remote side sends a candidate with the same address and port with an existing candidate,
but with a new ufrag and pwd, the local client will create a new connection from it
and destroy the old connection with the same remote address.
BUG=webrtc:5915
Review-Url: https://codereview.webrtc.org/2018693002
Cr-Commit-Position: refs/heads/master@{#13000}
No operator== that accepts one unique_ptr<T> and one T*. No implicit
conversion to bool. No rtc_make_scoped_ptr function.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1803833002
Cr-Commit-Position: refs/heads/master@{#12048}
Chromium reported errors when building libjingle_nacl due to some methods used virtual instead of override when they were overriding the base class. My guess is that when one method starts using override, all other in the same class must too.
R=tommi@webrtc.orgTBR=pthatcher@webtrc.org
BUG=4173
Review URL: https://codereview.webrtc.org/1589563003 .
Cr-Commit-Position: refs/heads/master@{#11251}
To reduce the risk of future mistakes when connecting Ports, Port::OnSentPacket was made pure virtual to ensure that new implementations take care of it.
BUG=4173
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1577873003 .
Cr-Commit-Position: refs/heads/master@{#11247}
If the local port and the remote candidate's protocols do not match,
do not even try to pair them.
This avoids printing out confusing logs like
"Attempt to change a remote candidate..." in p2ptransportchannel
when two remote candidates have the same port number but different
protocols.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1516613002 .
Cr-Commit-Position: refs/heads/master@{#11034}
In tcp only call:
Tested with hangout.
Tested with firefox.
To test firefox, goto about:config, search for media.peerconnection.ice.tcp and turn it on.
Existing test case should be suffice to cover this.
R=juberti@google.comTBR=jubert@webrtc.org
BUG=webrtc:3849
Review URL: https://codereview.webrtc.org/1217463004 .
Cr-Commit-Position: refs/heads/master@{#10683}
Since the TCPConnection has never been connected, they are not scheduled for ping hence will never be detected.
Also fix the case when reconnect fails, as it has become READABLE before, it also will not be deleted.
BUG=webrtc:4936
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1307083002 .
Cr-Commit-Position: refs/heads/master@{#9782}
During the reconnection phase, EWOULDBLOCK has been returned to upper layer which stops the sending of video stream.
BUG=webrtc:4930
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1288553010 .
Cr-Commit-Position: refs/heads/master@{#9767}
The IceEndpointType has the format of <local_endpoint>_<remote_endpoint>. It is recorded on the BestConnection when we have the first OnTransportCompleted signaled.
BUG=webrtc:4918
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1277263002 .
Cr-Commit-Position: refs/heads/master@{#9737}
UDP case should not be changed.
Active TCPConnection will initiate Reconnect after OnClose and when Send or Ping fails.
Passive TCPConnection will prune itself as usual as the active side will create a new connection.
The Reconnect could make P2PCT choose a different best_connection in the case where connectivities exist b/w more than 1 Network.
Also, to avoid upper layer triggers ice restart, the WRITE_TIMEOUT caused by the socket disconnection is delayed to give the reconnect mechanism chance to kick in. The timeout event is only fired if the reconnect can't work in 5 sec. If the reconnect, there should be no ICE disconnected state trigger either in active or passive side.
BUG=1926
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/31359004
Cr-Commit-Position: refs/heads/master@{#8929}
