- 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}
Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files.
TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change.
Bug: webrtc:9719
Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28016}
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}
This also introduces an option in CreateOfferOptions for
getting the non-spec behavior (2013 vintage) back.
Bug: chromium:962860
Change-Id: I72267408a61d6eb03e9895fe38b4cc803d8cbbaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136809
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27941}
This reverts commit 46afbf9481fbcc939c998c898ca1031ce41cc6b1.
Reason for revert: Tightened protocol name handling.
Original change's description:
> Revert "Reland "Version 2 "Refactoring DataContentDescription class"""
>
> This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e.
>
> Reason for revert: fuzzer failures
>
> Original change's description:
> > Reland "Version 2 "Refactoring DataContentDescription class""
> >
> > This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c
> >
> > Original change's description:
> > > Version 2 "Refactoring DataContentDescription class"
> > >
> > > (substantial changes since version 1)
> > >
> > > This CL splits the cricket::DataContentDescription class into
> > > two classes: cricket::RtpDataContentDescription (used for RTP data)
> > > and cricket::SctpDataContentDescription (used for SCTP only).
> > >
> > > SctpDataContentDescription no longer inherits from
> > > MediaContentDescriptionImpl, and no longer contains "codecs".
> > >
> > > Due to usage of internal interfaces by consumers, shimming the old
> > > DataContentDescription API is needed.
> > >
> > > A new cricket::DataContentDescription class is defined, which is
> > > a shim over RtpDataContentDescription and SctpDataContentDescription.
> > > It exposes as little functionality as possible, but supports the
> > > concerned consumer's usage
> > >
> > > Design document:
> > > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> > >
> > > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> > >
Bug: webrtc:10358
Change-Id: Ia9fb8f4679e082e3d18fbbb6b03fc13a08e06110
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136581
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@{#27933}
This reverts commit 37f2b43274a0d718de53a4cfcf02226356edcf6e.
Reason for revert: fuzzer failures
Original change's description:
> Reland "Version 2 "Refactoring DataContentDescription class""
>
> This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c
>
> Original change's description:
> > Version 2 "Refactoring DataContentDescription class"
> >
> > (substantial changes since version 1)
> >
> > This CL splits the cricket::DataContentDescription class into
> > two classes: cricket::RtpDataContentDescription (used for RTP data)
> > and cricket::SctpDataContentDescription (used for SCTP only).
> >
> > SctpDataContentDescription no longer inherits from
> > MediaContentDescriptionImpl, and no longer contains "codecs".
> >
> > Due to usage of internal interfaces by consumers, shimming the old
> > DataContentDescription API is needed.
> >
> > A new cricket::DataContentDescription class is defined, which is
> > a shim over RtpDataContentDescription and SctpDataContentDescription.
> > It exposes as little functionality as possible, but supports the
> > concerned consumer's usage
> >
> > Design document:
> > https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
> >
> > Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
> >
> > Bug: webrtc:10358
> > Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27853}
>
> Bug: webrtc:10358
> Change-Id: Iff45c4694167f0b31b34ff2167c1f4ffa650bcc4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135281
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27896}
TBR=steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,shampson@webrtc.org
Change-Id: Ied6d9fb96aafe9c957f2658b34b5331b1f359b26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10358
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135986
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27917}
This is a reland of 14b2758726879d21671a21291dfed8fb4fd5c21c
Original change's description:
> Version 2 "Refactoring DataContentDescription class"
>
> (substantial changes since version 1)
>
> This CL splits the cricket::DataContentDescription class into
> two classes: cricket::RtpDataContentDescription (used for RTP data)
> and cricket::SctpDataContentDescription (used for SCTP only).
>
> SctpDataContentDescription no longer inherits from
> MediaContentDescriptionImpl, and no longer contains "codecs".
>
> Due to usage of internal interfaces by consumers, shimming the old
> DataContentDescription API is needed.
>
> A new cricket::DataContentDescription class is defined, which is
> a shim over RtpDataContentDescription and SctpDataContentDescription.
> It exposes as little functionality as possible, but supports the
> concerned consumer's usage
>
> Design document:
> https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
>
> Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
>
> Bug: webrtc:10358
> Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27853}
Bug: webrtc:10358
Change-Id: Iff45c4694167f0b31b34ff2167c1f4ffa650bcc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135281
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27896}
This reverts commit 14b2758726879d21671a21291dfed8fb4fd5c21c.
Reason for revert: Internal import failed.
Original change's description:
> Version 2 "Refactoring DataContentDescription class"
>
> (substantial changes since version 1)
>
> This CL splits the cricket::DataContentDescription class into
> two classes: cricket::RtpDataContentDescription (used for RTP data)
> and cricket::SctpDataContentDescription (used for SCTP only).
>
> SctpDataContentDescription no longer inherits from
> MediaContentDescriptionImpl, and no longer contains "codecs".
>
> Due to usage of internal interfaces by consumers, shimming the old
> DataContentDescription API is needed.
>
> A new cricket::DataContentDescription class is defined, which is
> a shim over RtpDataContentDescription and SctpDataContentDescription.
> It exposes as little functionality as possible, but supports the
> concerned consumer's usage
>
> Design document:
> https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
>
> Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
>
> Bug: webrtc:10358
> Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27853}
TBR=danilchap@webrtc.org,steveanton@webrtc.org,kwiberg@webrtc.org,hbos@webrtc.org,hta@webrtc.org,shampson@webrtc.org
Change-Id: Ibc16ba14c1cbf50345a9b79151b79df140482539
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10358
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135280
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27855}
(substantial changes since version 1)
This CL splits the cricket::DataContentDescription class into
two classes: cricket::RtpDataContentDescription (used for RTP data)
and cricket::SctpDataContentDescription (used for SCTP only).
SctpDataContentDescription no longer inherits from
MediaContentDescriptionImpl, and no longer contains "codecs".
Due to usage of internal interfaces by consumers, shimming the old
DataContentDescription API is needed.
A new cricket::DataContentDescription class is defined, which is
a shim over RtpDataContentDescription and SctpDataContentDescription.
It exposes as little functionality as possible, but supports the
concerned consumer's usage
Design document:
https://docs.google.com/document/d/1H5LfQxJA2ikMWTQ8FZ3_GAmaXM7knfVQWiSz6ph8VQ0/edit#
Version 1 reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132700
Bug: webrtc:10358
Change-Id: Icf95fb7308244d6f2ebfdb403aaffc544e358580
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133900
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27853}
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 is a reland of cd8d1cf68e4eeed71fba51c97006a91bfd41813d
Original change's description:
> Surface ICE candidates that match an updated candidate filter.
>
> After this change an ICE agent can surface candidates that do not match
> the previous filter but are allowed by the updated one. The candidate
> filter, as part of the internal implementation in the ICE transport,
> manifests the RTCIceTransportPolicy field in RTCConfiguration.
>
> This new feature would allow an ICE agent to gather new candidates when
> the transport policy changes from e.g. 'relay' to 'all' without an ICE
> restart.
>
> A caveat in the current implementation remains, and a candidate can
> surface multiple times if the transport policy, or the candidate filter
> directly, performs multiple transitions from a value that disallows to
> one that allows the underlying candidate type. For example, if the
> transport policy is updated by 'all' -> 'relay' -> 'all', the same host
> candidate can surface after the second update.
>
>
> Bug: webrtc:8939
> Change-Id: I92c2e07dafab225c702c5de28f47958a0d3270cc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132282
> Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27674}
Bug: webrtc:8939
Change-Id: I9c32b1ea05028ecd937ab4912779dd958faf734f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133582
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27694}
This reverts commit cd8d1cf68e4eeed71fba51c97006a91bfd41813d.
Reason for revert: breaks an internal project
Original change's description:
> Surface ICE candidates that match an updated candidate filter.
>
> After this change an ICE agent can surface candidates that do not match
> the previous filter but are allowed by the updated one. The candidate
> filter, as part of the internal implementation in the ICE transport,
> manifests the RTCIceTransportPolicy field in RTCConfiguration.
>
> This new feature would allow an ICE agent to gather new candidates when
> the transport policy changes from e.g. 'relay' to 'all' without an ICE
> restart.
>
> A caveat in the current implementation remains, and a candidate can
> surface multiple times if the transport policy, or the candidate filter
> directly, performs multiple transitions from a value that disallows to
> one that allows the underlying candidate type. For example, if the
> transport policy is updated by 'all' -> 'relay' -> 'all', the same host
> candidate can surface after the second update.
>
>
> Bug: webrtc:8939
> Change-Id: I92c2e07dafab225c702c5de28f47958a0d3270cc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132282
> Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
> Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
> Reviewed-by: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27674}
TBR=shampson@webrtc.org,qingsi@webrtc.org,jeroendb@webrtc.org,sukhanov@webrtc.org
Change-Id: Idd51a640e55a612b42fe8b69e05dff57a22d021a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133581
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27677}
After this change an ICE agent can surface candidates that do not match
the previous filter but are allowed by the updated one. The candidate
filter, as part of the internal implementation in the ICE transport,
manifests the RTCIceTransportPolicy field in RTCConfiguration.
This new feature would allow an ICE agent to gather new candidates when
the transport policy changes from e.g. 'relay' to 'all' without an ICE
restart.
A caveat in the current implementation remains, and a candidate can
surface multiple times if the transport policy, or the candidate filter
directly, performs multiple transitions from a value that disallows to
one that allows the underlying candidate type. For example, if the
transport policy is updated by 'all' -> 'relay' -> 'all', the same host
candidate can surface after the second update.
Bug: webrtc:8939
Change-Id: I92c2e07dafab225c702c5de28f47958a0d3270cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132282
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27674}
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}
This is a reland of 07f3279a730980583403b78c3762c5d246d1d9be
Original change's description:
> Adding a restriction for legal RID values.
>
> According to the spec, RID values should be constrained to only
> alpha-numeric values. This was not enforced in our implementation to
> allow for more flexibility.
> It has been brought to our attention that some values that we currently
> consider legal (such as the '~', '=' ';' characters) might cause confusion
> with the simulcast syntax that uses these characters to indicate other
> meanings.
> What's worse, is that some characters, when used in RIDs (such as
> \u{1f937} \u{1f4a9} and \u{1f926}) cause uncontrollable laughter for some
> users which might also be a health hazard.
> This change resolves these issues by restricting RIDs to alpha-numeric.
>
> Bug: webrtc:10491
> Change-Id: I16e262c87525d0289764beacd098e1525a355463
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132061
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27499}
TBR=steveanton@webrtc.org
Bug: webrtc:10491
Change-Id: I856581306a9258480ee9184f12b55c2a23dd8636
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131983
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27530}
This reverts commit 07f3279a730980583403b78c3762c5d246d1d9be.
Reason for revert: Suspect of producing consistent failure in some Chrome trybots, blocking rolls.
Failed test:
external/wpt/webrtc/RTCPeerConnection-addTransceiver.https.html
First failure:
https://ci.chromium.org/p/chromium/builders/try/linux-rel/64597
Original change's description:
> Adding a restriction for legal RID values.
>
> According to the spec, RID values should be constrained to only
> alpha-numeric values. This was not enforced in our implementation to
> allow for more flexibility.
> It has been brought to our attention that some values that we currently
> consider legal (such as the '~', '=' ';' characters) might cause confusion
> with the simulcast syntax that uses these characters to indicate other
> meanings.
> What's worse, is that some characters, when used in RIDs (such as
> \u{1f937} \u{1f4a9} and \u{1f926}) cause uncontrollable laughter for some
> users which might also be a health hazard.
> This change resolves these issues by restricting RIDs to alpha-numeric.
>
> Bug: webrtc:10491
> Change-Id: I16e262c87525d0289764beacd098e1525a355463
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132061
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27499}
TBR=steveanton@webrtc.org,amithi@webrtc.org
Change-Id: I89f9d8a8d3fa82de8a7d429f11ad7cc30812ba7c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10491
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132244
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27525}
According to the spec, RID values should be constrained to only
alpha-numeric values. This was not enforced in our implementation to
allow for more flexibility.
It has been brought to our attention that some values that we currently
consider legal (such as the '~', '=' ';' characters) might cause confusion
with the simulcast syntax that uses these characters to indicate other
meanings.
What's worse, is that some characters, when used in RIDs (such as
\u{1f937} \u{1f4a9} and \u{1f926}) cause uncontrollable laughter for some
users which might also be a health hazard.
This change resolves these issues by restricting RIDs to alpha-numeric.
Bug: webrtc:10491
Change-Id: I16e262c87525d0289764beacd098e1525a355463
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132061
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27499}
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}
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}
Metrics are added to measure:
1. The number of send encodings in calls to AddTransceiver.
2. The number of times that simulcast is disabled because there is no
support from remote peer.
3. The number of times simulcast is indicated in ApplyLocal and
ApplyRemote and with which API surface (no simulcast, legacy munging,
spec-compliant).
Bug: webrtc:10372
Change-Id: I84717a1911efdf8aaf43cd6c04c7f09fcf2c58f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125482
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26979}
Simulcast is disabled if the RIDs are not negotiated.
This change addresses the scenario in which RIDs are negotiated but
support for the RID extension is not negotiated.
In such cases, the RID extension cannot be used, so support for
simulcast should be turned off, as if RIDs were not negotiated.
A similar case can be made for MIDs, however MIDs are not explicitly
specified in simulcast. RIDs are only guaranteed to be unique within
a media section so it would seem that MIDs should be required.
However, applications supply RID values and can guarantee their
uniqueness, so unlike RIDs, the use of MIDs is not enforced as mandatory.
Bug: webrtc:10075
Change-Id: Ic1b27878ea152eaee43a38bbfda11144307766fe
Reviewed-on: https://webrtc-review.googlesource.com/c/125176
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26934}
We already support decoding of the x-mt line. This change adds the
a=x-mt line to the SDP offer. This is not a backward compatible change
for media transport (because of the changes in pre-shared key handling)
1) if media transport is enabled, and SDES is enabled, generate the
media transport offer.
2) if media transport generated the offer, add that offer to the x-mt
line.
3) in order to create media transport, require an x-mt line (backward incompatible).
The way it works is that
1) PeerConnection, on the offerer, asks jsep transport for the
configuration of the media transport.
2) Tentative media transport is created in JsepTransportController when
that happens.
3) SessionDescription will include configuration from this tentative
media transport.
4) When the LocalDescription is set on the offerer, the tentative media
transport is promoted to the real media transport.
Caveats:
- now we really only support MaxBundle. In the previous implementations,
two media transports were briefly created in some tests, and the second
one was destroyed shortly after instantiation.
- we, for now, enforce SDES. In the future, whether SDES is used will be
refactored out of the peer connection.
In the future (on the callee) we should ignore 'is_media_transport' setting. If
Offer contains x-mt, media transport should be used (if the factory is
present). However, we need to decide how to negotiate media transport
for data channels vs data transport for media (x-mt line at this point
doesn't differentiate the two, so we still need to use app setting).
This change also removes the negotation of pre-shared key from the
a=crypto line. Instead, media transport will have its own, 256bit key.
Such key should be transported in the x-mt line. This makes the code
much simpler, and simplifies the dependency / a=crypto lines parsing.
Also, adds a proper test for the connection re-offer (on both sides: callee and caller).
Before, it was possible that media transport could get recreated, based on the offer.
The tests we had didn't test this scenario, and the loopback media factory didn't allow for such test.
This change adds counts to that loopback media factory, and asserts that only 1 media transport is created, even
when there is a re-offer.
Bug: webrtc:9719
Change-Id: Ibd8739af90e914da40ab412454bba8e1529f5a01
Reviewed-on: https://webrtc-review.googlesource.com/c/125040
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26933}
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}
