Downstreams have been updated, and this now updates all uses of label()
to id() within WebRTC code. This change also makes id() pure virtual and
removes label().
Bug: webrtc:8977
Change-Id: Ib045ea4fabba6f14447c64875c7aeba87dc2be24
Reviewed-on: https://webrtc-review.googlesource.com/60382
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22431}
Add configurable parameters in RTCConfiguration with the default value
given by the constants CONNECTION_WRITE_CONNECT_TIME and
CONNECTION_WRITE_CONNECT_FAILURES in the ICE implementation. These two
parameters define the time period for which a candidate pair must wait
for ping response and the minimum number of connectivity checks that
the pair must send without response before its state becomes unreliable
from writable as defined in the current ICE implementation.
Bug: webrtc:8988
Change-Id: I484599b7d776489a87741ffea8926df766095da9
Reviewed-on: https://webrtc-review.googlesource.com/60704
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22411}
Needed in order to return error codes to Chromium.
Bug: chromium:819629, chromium:589455
Change-Id: Iab22250db62a348eee21c6d8bfc44020a7380586
Reviewed-on: https://webrtc-review.googlesource.com/60522
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22367}
The connectivity check intervals for candidate pairs with strong and
weak connectivity are currently constants in the ICE implementation. A
set of suboptimal value of these constants for a given application may
result in undesirable behavior including excessive network switching
latency. This CL adds these intervals to RTCConfiguration that is
available to applications to configure, while maintaining the original
constants as their default value for compatibility with existing
applications.
Bug: webrtc:8988
Change-Id: I804b0f4cf7881be7d3c8aec2776bc9596de72482
Reviewed-on: https://webrtc-review.googlesource.com/60585
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22351}
This will give accurate stats for the number of calls
that use video that are using SDES.
Bug: chromium:804275
Change-Id: I35b045a2301fb5267b656b424b9b3482b1b72f9a
Reviewed-on: https://webrtc-review.googlesource.com/60481
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22324}
The naming convention according to the spec is stream id, not stream
labels.Changing things now to be spec compliant, before it is widely
used. This also includes changes to objective C wrapper code to be in
sync with the change.
Bug: webrtc:7932
Change-Id: I5705e6d8a647aaeed860316466a7320132f24b00
Reviewed-on: https://webrtc-review.googlesource.com/59301
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22316}
This change adds UMA stats that record the format of the remote offered
SDP. There are three classifications for the SDP format:
- Simple: No more than one audio and one video. Should be compatible
with both Plan B and Unified Plan endpoints.
- ComplexPlanB: More than one audio or more than one video in the Plan B
format (e.g., one audio mline with multiple streams).
- ComplexUnifiedPlan: More than one audio or more than one video in the
Unified Plan format (e.g., two audio mlines).
Bug: chromium:811683
Change-Id: If46394edfa6a812ef313d632e27ec27516c18867
Reviewed-on: https://webrtc-review.googlesource.com/57220
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22315}
These counters will register whether the media sections
used with SDES are for audio, video or data.
Bug: chromium:804275
Change-Id: I1da3bb6625af755c0897bf4cd349655cb283fbb6
Reviewed-on: https://webrtc-review.googlesource.com/59400
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22287}
This change replaces the use of sync_label from StreamParams with
the new stream_labels() and set_stream_labels() getter and setter.
Bug: webrtc:7932
Change-Id: Ibd6d38f7d4efed37ac07963e6fbe377c93a28fd6
Reviewed-on: https://webrtc-review.googlesource.com/58540
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22257}
There was an implementation, but it relied on SSLCertificate::GetChain,
which was never implemented. Except in the fake certificate classes
used by the stats collector tests, hence the tests were passing.
Instead of implementing GetChain, we decided (in
https://webrtc-review.googlesource.com/c/src/+/6500) to add
methods that return a SSLCertChain directly, since it results in a
somewhat cleaner object model.
So this CL switches everything to use the "chain" methods, and gets
rid of the obsolete methods and member variables.
Bug: webrtc:8920
Change-Id: Ie9d7d53654ba859535462521b54c788adec7badf
Reviewed-on: https://webrtc-review.googlesource.com/56961
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22177}
This is specified by the WebRTC specification:
https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
Bug: webrtc:7600
Change-Id: If45ba0383e5040d250cd3c1c2525ff3b03b1eb4f
Reviewed-on: https://webrtc-review.googlesource.com/55880
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22139}
Moving usage of bitrate configuration related interface from Call
interface to the corresponding methods in the RtpSendTransportController
interface.
SetBitrateConfig was replaced with SetSdpBitrateParameters
SetBitrateConfigMask was replaced with SetClientBitratePreferences
OnNetworkRouteChanged was replaced with OnNetworkRouteChanged
This makes it more clear that RtpSendTransportController owns bitrate
configuration and fits a longer term ambition to reduce the scope of
the Call class.
Bug: webrtc:8415
Change-Id: I6d04eaad22a54ecd5ed60096e01689b0c67e9c65
Reviewed-on: https://webrtc-review.googlesource.com/54365
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22131}
This is a reland of 5897fe27abcbe70f706cc23adc26147e0581f97e.
Adding back CallConfig::kDefaultStartBitrateBps as deprecated.
Also making BitrateContraints::kDefaultStartBitrateBps private to stop
it from being used in other places.
Original change's description:
> Moved BitrateConfig out of Call::Config.
>
> This prepares for a CL extracting the bitrate configuration logic from
> the Call class.
>
> Also renaming BitrateConfig to BitrateConstraints.
>
> Bug: webrtc:8415
> Change-Id: I7e472683034c57bdc8093cdf5e78e477d1732480
> Reviewed-on: https://webrtc-review.googlesource.com/54400
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22104}
Bug: webrtc:8415
Change-Id: Iacfe2d6daedff710832ab89210c7c66d4403c93b
Reviewed-on: https://webrtc-review.googlesource.com/55980
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22123}
The stats collectors now work with Unified Plan, so re-enable the
tests that were disabled.
Bug: webrtc:8764
Change-Id: I9ac97fd19d0024b3aaf26dd5ab09d3ffcb33210a
Reviewed-on: https://webrtc-review.googlesource.com/55800
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22114}
STUN candidates use STUN binding requests to keep NAT bindings open.
Related stats including packet loss and RTT can be now collected via the
legacy GetStats in PeerConnection.
Bug: None
Change-Id: I7b0eee1ccb07eb670a32ee303c9590047b25f31c
Reviewed-on: https://webrtc-review.googlesource.com/54100
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22113}
These methods no longer work with Unified Plan and have been
replaced by iterating over RtpTransceivers to get all the
VoiceChannels and VideoChannels.
Bug: webrtc:8587
Change-Id: I66ec282ee9f7eb987c32e30957733c13c6cf45b8
Reviewed-on: https://webrtc-review.googlesource.com/55760
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22110}
STUN candidates use STUN binding requests to keep NAT bindings open. The
interval at which the STUN keepalive pings are sent is configurable now
via RTCConfiguration.
TBR=sakal@webrtc.org
Bug: None
Change-Id: I5f99ea3fe1e9042fa2bf7dcab0aace78f57739e6
Reviewed-on: https://webrtc-review.googlesource.com/54180
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22109}
This reverts commit 5897fe27abcbe70f706cc23adc26147e0581f97e.
Reason for revert: Breaking internal builds
Original change's description:
> Moved BitrateConfig out of Call::Config.
>
> This prepares for a CL extracting the bitrate configuration logic from
> the Call class.
>
> Also renaming BitrateConfig to BitrateConstraints.
>
> Bug: webrtc:8415
> Change-Id: I7e472683034c57bdc8093cdf5e78e477d1732480
> Reviewed-on: https://webrtc-review.googlesource.com/54400
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22104}
TBR=nisse@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Change-Id: I598040edba7f1ff8b39d2d9c3c3ceca5627aaa0c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/55740
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22106}
This prepares for a CL extracting the bitrate configuration logic from
the Call class.
Also renaming BitrateConfig to BitrateConstraints.
Bug: webrtc:8415
Change-Id: I7e472683034c57bdc8093cdf5e78e477d1732480
Reviewed-on: https://webrtc-review.googlesource.com/54400
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22104}
Original change's description:
> Set session error if SetLocal/RemoteDescription ever fails
>
> This changes SetLocalDescription/SetRemoteDescription to set a
> session error which will cause any future calls to fail early if
> there is an error when applying a session description.
>
> This is needed since until better error recovery is implemented
> failing a call to SetLocalDescription or SetRemoteDescription
> could leave the PeerConnection in an inconsistent state.
>
> Bug: chromium:800775
> Change-Id: If06fd73d6e902af15d072dc562bbe830d3b11ad5
> Reviewed-on: https://webrtc-review.googlesource.com/54061
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22061}
Bug: chromium:800775
Change-Id: I0016108264e013452e9d34239c012baf23240e99
Reviewed-on: https://webrtc-review.googlesource.com/54720
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22067}
This adds a callback corresponding to the ontrack event as defined
in the WebRTC specification.
Bug: webrtc:7600
Change-Id: Ied8c55e11dcea864428fb194623c1595c21657c7
Reviewed-on: https://webrtc-review.googlesource.com/52660
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22066}
Original change's description:
> Update RTCStatsCollector to work with RtpTransceivers
>
> Bug: webrtc:8764
> Change-Id: I8b442345869eb6d8b65fd12241ed7cb6e7d7ce3d
> Reviewed-on: https://webrtc-review.googlesource.com/49580
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22026}
Bug: webrtc:8764
Change-Id: I6a682824febf3f4f41397fc1a8dd7396c4ffa8e3
Reviewed-on: https://webrtc-review.googlesource.com/54160
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22064}
This reverts commit 71439a60e7915179be96dd42dc732dc51c279884.
Reason for revert: https://ci.chromium.org/buildbot/chromium.webrtc.fyi/Mac%20Tester/47796
Original change's description:
> Set session error if SetLocal/RemoteDescription ever fails
>
> This changes SetLocalDescription/SetRemoteDescription to set a
> session error which will cause any future calls to fail early if
> there is an error when applying a session description.
>
> This is needed since until better error recovery is implemented
> failing a call to SetLocalDescription or SetRemoteDescription
> could leave the PeerConnection in an inconsistent state.
>
> Bug: chromium:800775
> Change-Id: If06fd73d6e902af15d072dc562bbe830d3b11ad5
> Reviewed-on: https://webrtc-review.googlesource.com/54061
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22061}
TBR=steveanton@webrtc.org,deadbeef@webrtc.org
Change-Id: I8af271f2b6dd6a896e390a6fe736e809329b4f4a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:800775
Reviewed-on: https://webrtc-review.googlesource.com/54700
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22063}
This changes SetLocalDescription/SetRemoteDescription to set a
session error which will cause any future calls to fail early if
there is an error when applying a session description.
This is needed since until better error recovery is implemented
failing a call to SetLocalDescription or SetRemoteDescription
could leave the PeerConnection in an inconsistent state.
Bug: chromium:800775
Change-Id: If06fd73d6e902af15d072dc562bbe830d3b11ad5
Reviewed-on: https://webrtc-review.googlesource.com/54061
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22061}
Bug: webrtc:7600
Change-Id: Ic4e5560fdeb9848c65c59e0f45ca3a2a4a22a2ad
Reviewed-on: https://webrtc-review.googlesource.com/53401
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22042}
RTCConfiguration.
This bug holds IceConfig unchanged in PeerConnection::SetConfiguration
when the update of IceConfig is necessary, unless ice_check_min_interval
is part of the update.
TBR=deadbeef@webrtc.org
Bug: webrtc:8898
Change-Id: I87774863bfedd7c05408fb22937d7322e53417c3
Reviewed-on: https://webrtc-review.googlesource.com/54201
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#22041}
Bug: webrtc:8764
Change-Id: I8b442345869eb6d8b65fd12241ed7cb6e7d7ce3d
Reviewed-on: https://webrtc-review.googlesource.com/49580
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22026}
This change includes updates to the sdp logic, and transceiver
dissociation and also tests these updates. The sdp validation for
unified plan is updated to consider both the stored remote and local
descriptions for an offer, because either could be the most up to date.
This is important when considering a recycled m section. This also
updates to only dissociate a transceiver when we are setting the remote
or local description from an offer. The final small update allows us to
properly create a media description for a transceiver that is not new
but is part of a recycled m section that has only been set locally for
an offer and we are re-offering.
Bug: webrtc:8765
Change-Id: Ia86e54fcd977478824cfa88ebaf992215ed68aae
Reviewed-on: https://webrtc-review.googlesource.com/52080
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22025}
This changes the StatsCollector to handle stats from multiple
MediaChannels of the same type (e.g., audio or video).
Bug: webrtc:8764
Change-Id: I91ba50d10cf469420189a311acdafbf6f78579b2
Reviewed-on: https://webrtc-review.googlesource.com/49560
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22009}
We count a) what semantics are asked for explicitly (if any),
and b) what semantics are reflected in the successfully
processed answer, as indicated by presence of msid lines
of type Unified Plan vs Plan B.
This gives an indication of usage in sessions initiated by
the browser. It does not indicate usage in sessions where the
browser is the answerer.
Bug: chromium:811683
Change-Id: I2e28a6a83df1664e1aa1e17cd4ff2921de1fba7e
Reviewed-on: https://webrtc-review.googlesource.com/52101
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22008}
This cl fixes various minor issues found during a quick scan of the current log
usage.
Bug: webrtc:8529
Change-Id: I1e1eb02ef220177dbb327203509736ad7f70cc1c
Reviewed-on: https://webrtc-review.googlesource.com/52262
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Grunell <henrikg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21996}
Media type is not part of the WebRTC spec for RtpTransceiver, but it is
handy and the RtpSender/RtpReceiver also have it.
Bug: webrtc:7600
Change-Id: I8350069502588bff478db4dc1318329626dcf9be
Reviewed-on: https://webrtc-review.googlesource.com/50560
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21988}
This changes CreateAnswer to become compliant with the WebRTC 1.0
specification which details that createAnswer should fail if the
PeerConnection is in a state other than 'have-remote-offer' or
'have-local-pranswer'.
Bug: webrtc:8813
Change-Id: I7ca41bdebda1ea163aec8815267c1bbfd7d6d11e
Reviewed-on: https://webrtc-review.googlesource.com/47581
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21923}
The bug 8432 is caused by trying to connect through a
"link-local" interface (IP address 169.254.0.x/16),
which is listed among the iPhone network interfaces.
The bug is not happening if the link-local network interfaces
are skipped in the ICE candidate gethering process.
To control this behaviour an option - disable_link_local_networks -
is added inside the RTCConfiguration.
It is used to set the new BasicPortAllocatorSession flag -
PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS.
The port allocator flag is added if the configuration option is set.
IPIsLinkLocal IPAddress function and its friends (IPIsLoopback, IPIsPrivate)
are refactored to work on both IPv4 and IPv6.
Unit test IPIsLinkLocal.
Bonus: fix a bug in IPIsLinkLocalV6:
take into account just 10 network mask bits instead of 16.
Bug: webrtc:8432
Change-Id: Ibe8f677a36098057b7fcad5c798380727b23359b
Reviewed-on: https://webrtc-review.googlesource.com/36380
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21922}
This removes the SessionStats object and replaces it with two
methods on PeerConnection: GetTransportNamesByMid and
GetTransportStatsByNames for use by the stats collectors. These
methods are more flexible and can cover cases where there are more
than one video/audio channel.
Bug: webrtc:8764
Change-Id: Id400cc548fc43675462ff6175a7fa9c9f4fd5948
Reviewed-on: https://webrtc-review.googlesource.com/47244
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21921}
This change list passes the instance of RtcEventLog from Peerconnection
down to P2PTransportChannel, and binds the structured ICE logging with
ICE layer objects. Logs of ICE connectivity checks are injected for
candidate pairs.
TBR=terelius@webrtc.org
Bug: None
Change-Id: Ia979dbbac6d31dcf0f8988da1065bdfc3e461821
Reviewed-on: https://webrtc-review.googlesource.com/34660
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21884}
Previously, the code which reported cipher stats to UMA for all
transports would classify the media type based on the transport name,
which is brittle and misleading with BUNDLE. This corrects the code to
track all media types (audio, video, data) which use the transport and
report once for each.
Bug: None
Change-Id: I8506f64f0011788b744b8386ac58518a21914b52
Reviewed-on: https://webrtc-review.googlesource.com/47247
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21863}
Original change's description:
> Parameterize PeerConnection signaling tests for Unified Plan
>
> This also changes the behavior of CreateAnswer to fail unless
> the signaling state is kHaveRemoteOffer or kHaveLocalPranswer,
> as per the WebRTC specification.
>
> Bug: webrtc:8765
> Change-Id: I60ac67cd92b17fcbff964afc14d049481e816a28
> Reviewed-on: https://webrtc-review.googlesource.com/41042
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21779}
Bug: webrtc:8813
Change-Id: I9f608fcd0b7aca00b4c1092e271dbd9cd710c38a
Reviewed-on: https://webrtc-review.googlesource.com/46861
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21860}
This is intended to ensure compatibility between Plan B and
Unified Plan endpoints for the single audio - single video case.
If Unified Plan is the offerer, it will add a=msid and a=ssrc MSID
entries to its offer.
If Unified Plan is the answerer, it will use whatever MSID
signaling mechanism was used in the offer (either a=msid or
a=ssrc).
Bug: webrtc:7600
Change-Id: I6192dec19123fbb56f5d04540d2175c7fb30b9b6
Reviewed-on: https://webrtc-review.googlesource.com/44162
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21859}
This changes the behavior of CreateOffer/CreateAnswer when Unified
Plan is enabled to be in line with that specified in JSEP.
In particular, MSID information is now only included if the
RtpTransceiver is not stopped and either is sending or has ever
sent.
Bug: webrtc:7600
Change-Id: I6400f0583525c7776331eeb0e1bb53973bc02dfb
Reviewed-on: https://webrtc-review.googlesource.com/46400
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21857}
The network preference is added to RTCConfiguration and passed to ICE.
ICE considers now the preference set by applications over network
interface types when making decisions in candidate pair switching.
Bug: webrtc:8816
Change-Id: I40d2612705b54c83dd45772ac855808e0a76b1e1
Reviewed-on: https://webrtc-review.googlesource.com/44020
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21855}
This method returns the DTLS SSL certificate chain associated with the
audio transport on the remote side. This will become populated once the
DTLS connection with the peer has been completed.
TBR=deadbeef@webrtc.org
Bug: webrtc:8800
Change-Id: Ib90ccb3463415e798c17c187c5bdbfc4da26f11f
Reviewed-on: https://webrtc-review.googlesource.com/44140
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21785}
This implements the WebRTC specification for handling
the legacy offer options offer_to_receive_audio and
offer_to_receive_video. They are not implemented for CreateAnswer.
With Unified Plan semantics, clients should switch to the
RtpTransceiver API for ensuring the correct media sections are
offered.
Bug: webrtc:7600
Change-Id: I6ced00b86b165a352bd0ca3d64b48fadcfd12235
Reviewed-on: https://webrtc-review.googlesource.com/41341
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21784}
These tests verify the behavior between Plan B and
Unified Plan PeerConnections.
Bug: webrtc:7600
Change-Id: Ic41a0e692d32cde6fe7719ada2dbffd4281c008c
Reviewed-on: https://webrtc-review.googlesource.com/43244
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21782}
