- Implement datagram transport adaptor, which wraps datagram transport in DtlsTransportInternal. Datagram adaptor owns both ICE and Datagram Transports.
- Implement setup of datagram transport based on RTCConfiguration flag use_datagram_transport. This is very similar to MediaTransport setup with the exception that we create DTLS datagram adaptor.
- Propagate maximum datagram size to video encoder via MediaTransportConfig.
TODO: Currently this CL can only be tested in downstream projects. Once we add fake datagram transport, we will be able to implement unit tests similar to loopback media transport.
Bug: webrtc:9719
Change-Id: I4fa4a5725598dfee5da4f0f374269a7e289d48ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138100
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28047}
Prior to this CL, only the mline index of an ice candidate was used to
look up contents. However, due to recent changes, it is possible that
no mline index is specified, but that only a mid is specified.
No mline index is indicated with a -1 value.
This CL makes sure the mid is used if no mline index is given.
Bug: chromium:965483
Change-Id: I8962e71acb386f7b50349802f27358ba24c11921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138075
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28045}
This is a reland of df5731e44d510e9f23a35b77e9e102eb41919bf4 with fixes
to avoid existing chromium tests to fail.
Instead of replacing the existing RtpSender::set_stream_ids() to
also fire OnRenegotiationNeeded(), this CL keeps the old
set_stream_ids() and adds the new RtpSender::SetStreams() which sets
the stream IDs and fires the callback.
This allows existing callsites to maintain behavior, and reserve
SetStreams() for the cases when we want OnRenegotiationNeeded() to fire.
Using the SetStreams() name instead of SetStreamIDs() to match the W3C
spec and to make it more different that the existing set_stream_ids().
Original change's description:
> Improve spec compliance of SetStreamIDs in RtpSenderInterface
>
> This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
> event if needed and exposes the method on RtpSenderInterface.
>
> This is a spec-compliance change.
>
> Bug: webrtc:10129
> Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27974}
Bug: webrtc:10129
Change-Id: Ic0b322bfa25c157e3a39465ef8b486f898eaf6bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137439
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27992}
This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
event if needed and exposes the method on RtpSenderInterface.
This is a spec-compliance change.
Bug: webrtc:10129
Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27974}
This also refactors some of the code in peerconnection for
handling SCTP transports to be internal to the webrtc::SctpTransport
class, rather than being in peerconnection.
Bug: webrtc:10358, webrtc:10629
Change-Id: I15ecf95c199f56b08909e5a9311d446a412ed162
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137041
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27960}
This also changes the default when no max-message-size is set
to the protocol defined value of 64K, and prevents messages
from being sent when they are too large to send.
Bug: webrtc:10358
Change-Id: Iacc1dd774d1554d9f27315378fbea6351300b5cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135948
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27945}
Adding additional usage bits to the UsagePattern to:
- Track whether a mDNS candidate was collected
- Track whether a mDNS candidate was received from the remote peer
- Track whether a private IP address was received from the remote peer
The definition of a private IP address is extended to include 100.64/10 addresses.
Bug: None
Change-Id: I77182685120413d5c13c5f67e480d33fdcaefc6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134000
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Justin Uberti <juberti@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27747}
This patch makes VideoBitrateAllocatorFactory injectable
by adding to PeerConnectionDependencies instead of allowing it to be
overridden using MediaEngine (on PeerConnectionFactory).
With this patch VideoBitrateAllocatorFactory is owned
by the PeerConnection.
WANT_LGTM (examples) : sakal@
WANT_LGTM (api/pc) : steveanton@
Bug: webrtc:10547
Change-Id: I768d400a621f2b7a98795eb7f410adb48651bfd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132706
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27654}
The new processing applies only in Unified Plan mode.
Plan B retains the old-style processing.
This is a reland of 1fa06041bcd8a0119e557d16e7b54a9110c5ad03
Original change's description:
> Make negotiationneeded processing in PeerConnection spec compliant.
>
> This CL fixes the problem of misfired negotiationneeded notifications due
> to the lack of a NegotiationNeeded slot and the proper procedure to
> update it.
>
>
> Change-Id: Ie273c691f11316c9846606446f6cf838226b5d5c
> Bug: chromium:740501
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131283
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27594}
Bug: chromium:740501
Change-Id: I048ae81b2b00086f6d669e94eecf426f0db0ec08
TBR: steveanton@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133162
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27640}
This reverts commit 1fa06041bcd8a0119e557d16e7b54a9110c5ad03.
Reason for revert: Likely cause for breaking downstream projects
Original change's description:
> Make negotiationneeded processing in PeerConnection spec compliant.
>
> This CL fixes the problem of misfired negotiationneeded notifications due
> to the lack of a NegotiationNeeded slot and the proper procedure to
> update it.
>
>
> Change-Id: Ie273c691f11316c9846606446f6cf838226b5d5c
> Bug: chromium:740501
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131283
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27594}
TBR=steveanton@webrtc.org,magjed@webrtc.org,sakal@webrtc.org,hbos@webrtc.org,guidou@webrtc.org
Change-Id: Iad7b7d4e37227fa6a76ff830160ca3da9dbe4719
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:740501
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132761
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Jeroen de Borst <jeroendb@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27599}
This CL fixes the problem of misfired negotiationneeded notifications due
to the lack of a NegotiationNeeded slot and the proper procedure to
update it.
Change-Id: Ie273c691f11316c9846606446f6cf838226b5d5c
Bug: chromium:740501
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131283
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27594}
Plus all the annotations that were necessary to make things compile
again. I also had to send copies of some values owned by the signal
thread to the network thread, instead of letting the latter read them
itself.
Bug: webrtc:9987
Change-Id: Ic4b38696245584bab44956e60ac63753146e3ff4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131020
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27437}
After reviewer feedback, this CL was reduced to just adding scary
comments on two variables.
Bug: webrtc:9987
Change-Id: Id1e251ffd02e4ca8050235bd9f3971b5363f0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130960
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27431}
Previously, legacy GetStats would look up the track ID by querying the
local/remote SDP by SSRC. This doesn't work with Unified Plan since the
RtpSender/RtpReceiver track IDs may not correspond to the track ID
stored in the SDP.
This CL changes legacy GetStats to pull the track ID directly from the
RtpSenders and RtpReceivers as it generates the stats. This has a few
additional benefits:
1) Unsignaled receive SSRC stats should now get correctly matched to
the unsigneled RtpReceiver track ID for both Plan B and Unified
Plan.
2) Removes a couple methods on PeerConnection that were only used by
the legacy StatsCollector.
3) Keeps the SSRC -> track ID mapping more localized which should make
the code easier to understand.
Bug: chromium:943493
Change-Id: I43ecde8c3a3d1c5f9c749ba6c8dfb11e8c4950fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129782
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27324}
Earlier CLs assumed that the object pointed to by call_ had to be
accessed on the worker thread. While this is generally the case,
Call::MediaTransportChange is explicitly thread safe, so
PeerConnection::OnTransportChanged doesn't have to run on the worker
thread for that reason.
Which is fortunate, because it actually runs on the network thread.
The RTC_RUN_ON(worker_thread()) annotation on the method declaration
was ineffective because this method is being called via a base class
pointer; replacing it with a call to
RTC_DCHECK_RUN_ON(worker_thread()) in the function body immediately
triggered assertions in the unit tests.
Bug: webrtc:9987
Change-Id: I08cf558a74f4ca2b2eff8ef4810ebbd1287a9726
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129442
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27287}
Plus all the annotations that were necessary to make things compile
again.
We needed a special twist for call_. The value it points to is owned
by the worker thread, but the signal thread needs to read the pointer.
We could have made the pointer const, except that we explicitly reset
it in the destructor (in an invoke to the worker thread).
Bug: webrtc:9987
Change-Id: I31f024547f4be0e50967133b0d452c80ae38d7ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128863
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27278}
Plus all the annotations that were necessary to make things compile
again.
port_allocator_flags_ was accessed on both the signaling and the
network thread, but I was able to replace it with a return value.
Bug: webrtc:9987
Change-Id: Iab977a49d6588ce2240487475ec3588ae579caa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128772
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27254}
This involves inserting an extra layer between jsep_transport_controller
and the cricket::SctpTransportInternal layer. The objects at this layer
are reference counted.
Bug: chromium:818643
Change-Id: Ibed57c4a538de981cee63e0f7f1f319f029cab39
Reviewed-on: https://webrtc-review.googlesource.com/c/123884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26889}
Plus all the annotations that are necessary to make things compile
again.
Bug: webrtc:9987
Change-Id: If7bbd5a468a8c50ac3cfe03cd2ed4f5b5f461195
Reviewed-on: https://webrtc-review.googlesource.com/c/123047
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26727}
Plus all the annotations that are necessary to make things compile
again.
Bug: webrtc:9987
Change-Id: I68a51bfa3342c6d66d67276d5979144af34692c9
Reviewed-on: https://webrtc-review.googlesource.com/c/123046
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26716}
Plus all the annotations that are necessary to make things compile
again.
(The latter could only be made const, since it is accessed from both
the signal thread and the worker thread.)
Bug: webrtc:9987
Change-Id: I49f1568d26a327f48cd94c2c70bc5939e9fbd59a
Reviewed-on: https://webrtc-review.googlesource.com/c/122842
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26715}
Plus all the annotations that are necessary to make things compile
again.
Bug: webrtc:9987
Change-Id: I53caa3c6644c5d0fcb50f7c1e89e853eddd769a5
Reviewed-on: https://webrtc-review.googlesource.com/c/122882
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26705}
CreateOffer and CreateAnswer will now examine the layers on the
transceiver to determine if multiple layers are requested (Simulcast).
In this scenario RIDs will be used in the layers (instead of SSRCs).
When the offer is created, only RIDs are signalled in the offer.
When the offer is set locally SetLocalDescription() SSRCs will be
generated for each layer by the Channel and sent downstream to the
MediaChannel.
The MediaChannel receives configuration that looks identical to that of
legacy simulcast, and should be able to integrate the streams correctly
regardless of how they were signalled.
Setting multiple layers on the transciever is still not supported
through the API.
Bug: webrtc:10075
Change-Id: Id4ad3637b87b68ef6ca7eec69166fee2d9dfa36f
Reviewed-on: https://webrtc-review.googlesource.com/c/119780
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26428}
UniqueIdGenerator classes are useful outside the pc directory.
This change moves them to the rtc_base directory to enable code
in all directories to reference them.
Bug: None
Change-Id: I1c77da87ea26d9611f37dc1d4d2c16006a6589c6
Reviewed-on: https://webrtc-review.googlesource.com/c/119460
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26378}
This is in preparation for letting Chrome extract DTLSTransport
information after SLD/SRD instead of doing it on-demand.
Bug: chromium:907849
Change-Id: Iac6b174c98d3d14136e1fd25bce4a9292f6c8b41
Reviewed-on: https://webrtc-review.googlesource.com/c/116984
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26289}
