to avoid conflicts between
createOffer({voiceActivityDetection: false})
and the transceiver setCodecPreferences API
BUG=webrtc:12365
Change-Id: I369227103ab543f593b27145a37d3e5c19a59cd5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218343
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/master@{#33992}
In this CL, JsepTransportController and MediaSessionDescriptionFactory
are updated not to assume that there only exists at most a single BUNDLE
group but a list of N groups. This makes it possible to create multiple
BUNDLE groups by having multiple "a=group:BUNDLE" lines in the SDP.
This makes it possible to have some m= sections in one group and some
other m= sections in another group. For example, you could group all
audio m= sections in one group and all video m= sections in another
group. This enables "send all audio tracks on one transport and all
video tracks on another transport" in Unified Plan. This is something
that was possible in Plan B because all ssrcs in the same m= section
were implicitly bundled together forming a group of audio m= section and
video m= section (even without use of the BUNDLE tag).
PeerConnection will never create multiple BUNDLE groups by default, but
upon setting SDP with multiple BUNDLE groups the PeerConnection will
accept them if configured to accept BUNDLE. This makes it possible to
accept an SFU's BUNDLE offer without having to SDP munge the answer.
C++ unit tests are added. This fix has also been verified manually on:
https://jsfiddle.net/henbos/to89L6ce/43/
Without fix: 0+2 get bundled, 1+3 don't get bundled.
With fix: 0+2 get bundled in first group, 1+3 get bundled in second
group.
Bug: webrtc:10208
Change-Id: Iaf451fa5459c484730c8018274166ef154b19af8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214487
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33838}
This is a refactor to simplify a follow-up CL of adding
SdpVideoFormat::IsSameCodec.
The original files media/base/h264_profile_level_id.* and
media/base/vp9_profile.h must be kept until downstream projects
stop using them.
Bug: chroimium:1187565
Change-Id: Ib39eca095a3d61939a914d9bffaf4b891ddd222f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215236
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33782}
Since there is only a single type of DataChannel now, the enum was only used
when data channels were disabled at the PC API. That option has been
deprecated 4 years ago, it's now time to remove it.
Bug: webrtc:6625
Change-Id: I9e4ada1756da186e9639dd0fbf0249c55ea0b6c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215661
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33778}
This reverts commit a743303211b89bbcf4cea438ee797bbbc7b59e80.
Reason for revert: Breaks downstream tests that attempt to call FindHeaderExtensionByUri with 2 arguments. Could you keep the old 2-argument method declaration and just forward the call to the new 3-argument method with a suitable no-op filter?
Original change's description:
> Fix RTP header extension encryption
>
> Previously, RTP header extensions with encryption had been filtered
> if the encryption had been activated (not the other way around) which
> was likely an unintended logic inversion.
>
> In addition, it ensures that encrypted RTP header extensions are only
> negotiated if RTP header extension encryption is turned on. Formerly,
> which extensions had been negotiated depended on the order in which
> they were inserted, regardless of whether or not header encryption was
> actually enabled, leading to no extensions being sent on the wire.
>
> Further changes:
>
> - If RTP header encryption enabled, prefer encrypted extensions over
> non-encrypted extensions
> - Add most extensions to list of extensions supported for encryption
> - Discard encrypted extensions in a session description in case encryption
> is not supported for that extension
>
> Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get
> into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte
> header extensions will prevent any RTP packets being sent/received.
>
> Bug: webrtc:11713
> Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33723}
TBR=deadbeef@webrtc.org,terelius@webrtc.org,hta@webrtc.org,lennart.grahl@gmail.com
Change-Id: I7df6b0fa611c6496dccdfb09a65ff33ae4a52b26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215222
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33727}
Previously, RTP header extensions with encryption had been filtered
if the encryption had been activated (not the other way around) which
was likely an unintended logic inversion.
In addition, it ensures that encrypted RTP header extensions are only
negotiated if RTP header extension encryption is turned on. Formerly,
which extensions had been negotiated depended on the order in which
they were inserted, regardless of whether or not header encryption was
actually enabled, leading to no extensions being sent on the wire.
Further changes:
- If RTP header encryption enabled, prefer encrypted extensions over
non-encrypted extensions
- Add most extensions to list of extensions supported for encryption
- Discard encrypted extensions in a session description in case encryption
is not supported for that extension
Note that this depends on https://github.com/cisco/libsrtp/pull/491 to get
into libwebrtc (cherry-pick or bump libsrtp version). Otherwise, two-byte
header extensions will prevent any RTP packets being sent/received.
Bug: webrtc:11713
Change-Id: Ia0779453d342fa11e06996d9bc2d3c826f3466d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33723}
otherwise if the client receives a flexfec-enabled offer
and receiving flexfec is enabled by default, an answer
or subsequent offer will enable sending flexfec.
BUG=webrtc:8151
Change-Id: I632094f69ffa68518b6b8f31175eb093efaf51c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193862
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#32628}
And use it in a few places that were using RTC_CHECK(false) or FATAL()
to do the exact same job. There should be no change in behavior.
Bug: none
Change-Id: I36d5e6bcf35fd41534e08a8c879fa0811b4f1967
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191963
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32567}
"warning: control reaches end of non-void function [-Wreturn-type]"
Reported by gcc (8.3)
In all the reported cases, the end of function is never actually
reached. Add RTC_CHECK(false) to ensure the compiler is aware that
this path is a dead-end.
Bug: webrtc:12008
Change-Id: I7f816fde3d1897ed2774057c7e05da66e1895e60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189784
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fabien VALLÉE <fabien.vallee@netgem.com>
Cr-Commit-Position: refs/heads/master@{#32503}
This is a reland of 239f92ecf7fc8ca27e0376dd192b33ce33377b3c
Original change's description:
> introduce an unsupported content description type
>
> This carries around unsupported content descriptions
> (i.e. things where webrtc does not understand the media type
> or protocol) in a special data type so that a rejected content or
> mediasection is added to the answer SDP.
>
> BUG=webrtc:3513
>
> Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/master@{#32410}
Bug: webrtc:3513
Change-Id: I48e338100f829f1df5b8165217c89b5ef860fe79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32457}
This reverts commit 239f92ecf7fc8ca27e0376dd192b33ce33377b3c.
Reason for revert: Breaks downstream projects.
Original change's description:
> introduce an unsupported content description type
>
> This carries around unsupported content descriptions
> (i.e. things where webrtc does not understand the media type
> or protocol) in a special data type so that a rejected content or
> mediasection is added to the answer SDP.
>
> BUG=webrtc:3513
>
> Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/master@{#32410}
TBR=kthelgason@webrtc.org,hta@webrtc.org,philipp.hancke@googlemail.com
Change-Id: I055fe001fe2757d79be7c304eccc43a8e3104f69
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:3513
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188581
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32411}
This carries around unsupported content descriptions
(i.e. things where webrtc does not understand the media type
or protocol) in a special data type so that a rejected content or
mediasection is added to the answer SDP.
BUG=webrtc:3513
Change-Id: Ifc4168eae11e899f2504649de5e1eecb6801a9fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179082
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/master@{#32410}
This reverts commit 8e8b36a94a7a7a1fd0f8093979a406afa56e18c1.
Reason for revert: The CL has been improved with the following changes,
- Fixed negotiation of send/receive only clients.
- Handles the implicit assumption that any H264 decoder also can
decode H264 constraint baseline.
Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30284}
Change-Id: I834ed48ee78d04922c73e2836165e476925e1cc5
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168605
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30932}
This change adds exposure of a new transceiver method for getting
the total set of supported extensions stored as an attribute,
and their direction. If the direction is kStopped, the extension
is not signalled in Unified Plan SDP negotiation.
Note: SDP negotiation is not modified by this change.
Changes:
- RtpHeaderExtensionCapability gets a new RtpTransceiverDirection,
indicating either kStopped (extension available but not signalled),
or other (extension signalled).
- RtpTransceiver gets the new method as described above. The
default value of the attribute comes from the voice and video
engines as before.
https://chromestatus.com/feature/5680189201711104.
go/rtp-header-extension-ip
Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk
Bug: chromium:1051821
Change-Id: I440443b474db5b1cfe8c6b25b6c10a3ff9c21a8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170235
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30800}
Without exposing it in capabilities:
this extension is not stable enough to expose it by default,
but already in working state so with munge sdp can be experimented with.
Bug: webrtc:10342
Change-Id: I6bac123325a90431e4769e86da79638869e36cfc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168961
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30645}
The change ships GenericDescriptor00 and authentication by default,
but doesn't expose it by default, and makes WebRTC respond to
offers carrying it.
The change adds a unit test for the new semantics.
Tests well in munge-sdp. Frame marking replaced by
http://www.webrtc.org/experiments/rtp-hdrext/generic-frame-descriptor-00
in the offer results in an answer containing the
extension as first entry.
Bug: webrtc:11367
Change-Id: I0ef91b7d4096d949c3d547ece7d6c4d39aa241da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168661
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30542}
This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
Reason for revert: Breaks Chromium import due to flaky test in Chromium.
Original change's description:
> Reland "Distinguish between send and receive codecs"
>
> This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
>
> Reason for revert: Fixed negotiation of send-only clients.
>
> Original change's description:
> > Revert "Distinguish between send and receive codecs"
> >
> > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> >
> > Reason for revert: breaks negotiation with send-only clients
> >
> > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> >
> > Original change's description:
> > > Distinguish between send and receive codecs
> > >
> > > Even though send and receive codecs may be the same, they might have
> > > different support in HW. Distinguish between send and receive codecs
> > > to be able to keep track of which codecs have HW support.
> > >
> > > Bug: chromium:1029737
> > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30284}
> >
> > TBR=steveanton@webrtc.org,kron@webrtc.org
> >
> > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30292}
>
> TBR=steveanton@webrtc.org,kron@webrtc.org
>
>
> Bug: chromium:1029737
> Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30348}
TBR=steveanton@webrtc.org,kron@webrtc.org
Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30360}
This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
Reason for revert: Fixed negotiation of send-only clients.
Original change's description:
> Revert "Distinguish between send and receive codecs"
>
> This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
>
> Reason for revert: breaks negotiation with send-only clients
>
> (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
>
> Original change's description:
> > Distinguish between send and receive codecs
> >
> > Even though send and receive codecs may be the same, they might have
> > different support in HW. Distinguish between send and receive codecs
> > to be able to keep track of which codecs have HW support.
> >
> > Bug: chromium:1029737
> > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30284}
>
> TBR=steveanton@webrtc.org,kron@webrtc.org
>
> Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30292}
TBR=steveanton@webrtc.org,kron@webrtc.org
Bug: chromium:1029737
Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30348}
This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
Reason for revert: breaks negotiation with send-only clients
(webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
(peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
(peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30284}
TBR=steveanton@webrtc.org,kron@webrtc.org
Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30292}
Even though send and receive codecs may be the same, they might have
different support in HW. Distinguish between send and receive codecs
to be able to keep track of which codecs have HW support.
Bug: chromium:1029737
Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30284}
Video and audio senders are missing mid, rid and rrid extensions in
their GetCapabilities call.
Bug: chromium:1007894
Change-Id: Ie9edba28ae32fda5e501913cac694f43bfb185ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156560
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29493}
The plugin transport parameters (a=x-opaque: lines) relate to how to create and
set up a plugin transport. When SDP bundle is used, the x-opaque line needs to
be copied into the bundled m= section. This means x-opaque can appear on a
section even if the offerer does not intend to use the transport for the media
described by that section. Consequently, the answerer cannot currently tell
whether the caller is offering an alternate transport for media, data, or both.
This change adds an a=x-alt-protocol: line to SDP. The value following this
line matches the <protocol> part of the x-opaque:<protocol>:<params> line.
However, alt-protocol is not bundled--it only ever applies to the m= section
that contains the line. This allows the offerer to express which m= sections
should actually use an alternate transport, even in the case of bundle.
Note that this is still limited by the available configuration options:
datagram transport can be used for media (audio + video) and/or data. It is
still not possible to use it for audio but not video, or vice versa.
PeerConnection places an alt-protocol line in each media (audio/video) m=
section if it is configured to use a datagram transport for media. It places
an alt-protocol line in each data m= section if it is configured to use a
datagram transport for data channels. PeerConnection leaves alt-protocol in
media (audio/video) m= sections of the answer if it is configured to use a
datagram transport for media, and in data m= sections of the answer if it is
configured to use a datagram transport for data channels.
JsepTransport now negotiates use of the datagram transport independently for
media and data channels. It only uses it for media if the m= sections for
bundled audio/video have an alt-protocol line matching the x-opaque protocol,
and only uses it for data channels if a bundled m= section for data has an
alt-protocol line matching the x-opaque protocol.
Bug: webrtc:9719
Change-Id: I773e4fc10c57d815afcd76a2a74da38dd0c52b3b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154763
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29351}
PeerConnection now has a new setting in RTCConfiguration to enable use of
datagram transport for data channels. There is also a corresponding field
trial, which has both a kill-switch and a way to change the default value.
PeerConnection's interaction with MediaTransport for data channels has been
refactored to work with DataChannelTransportInterface instead.
Adds a DataChannelState and OnStateChanged() to the DataChannelSink
callbacks. This allows PeerConnection to listen to the data channel's
state directly, instead of indirectly by monitoring media transport
state. This is necessary to enable use of non-media-transport (eg.
datagram transport) data channel transports.
For now, PeerConnection watches the state through MediaTransport as well.
This will persist until MediaTransport implements the new callback.
Datagram transport use is negotiated. As such, an offer that requests to use
datagram transport for data channels may be rejected by the answerer. If the
offer includes DTLS, the data channels will be negotiated as SCTP/DTLS data
channels with an extra x-opaque parameter for datagram transport. If the
opaque parameter is rejected (by an answerer without datagram support), the
offerer may fall back to SCTP.
If DTLS is not enabled, there is no viable fallback. In this case, the data
channels are negotiated as media transport data channels. If the receiver does
not understand the x-opaque line, it will reject these data channels, and the
offerer's data channels will be closed.
Bug: webrtc:9719
Change-Id: Ic1bf3664c4bcf9d754482df59897f5f72fe68fcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147702
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28932}
This adds the RemoteEstimate rtcp packet and wires it up to GoogCC where
it's used to improve congestion controller behavior.
The functionality is negotiated using SDP.
It's added with a field trial that allow disabling the functionality in
case there's any issues.
Bug: webrtc:10742
Change-Id: I1ea8e4216a27cd2b00505c99b42d1e38726256c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146602
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28654}
