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}
In short, the caller places a x-opaque line in SDP for each m= section that
uses datagram transport. If the answerer supports datagram transport, it will
parse this line and create a datagram transport. It will then echo the x-opaque
line into the answer (to indicate that it accepted use of datagram transport).
If the offer and answer contain exactly the same x-opaque line, both peers will
use datagram transport. If the x-opaque line is omitted from the answer (or is
different in the answer) they will fall back to RTP.
Note that a different x-opaque line in the answer means the answerer did not
understand something in the negotiation proto. Since WebRTC cannot know what
was misunderstood, or whether it's still possible to use the datagram transport,
it must fall back to RTP. This may change in the future, possibly by passing
the answer to the datagram transport, but it's good enough for now.
Negotiation consists of four parts:
1. DatagramTransport exposes transport parameters for both client and server
perspectives. The client just echoes what it received from the server (modulo
any fields it might not have understood).
2. SDP adds a x-opaque line for opaque transport parameters. Identical to
x-mt, but this is specific to datagram transport and goes in each m= section,
and appears in the answer as well as the offer.
- This is propagated to Jsep as part of the TransportDescription.
- SDP files: transport_description.h,cc, transport_description_factory.h,cc,
media_session.cc, webrtc_sdp.cc
3. JsepTransport/Controller:
- Exposes opaque parameters for each mid (m= section). On offerer, this means
pre-allocating a datagram transport and getting its parameters. On the
answerer, this means echoing the offerer's parameters.
- Uses a composite RTP transport to receive from either default RTP or
datagram transport until both offer and answer arrive.
- If a provisional answer arrives, sets the composite to send on the
provisionally selected transport.
- Once both offer and answer are set, deletes the unneeded transports and
keeps whichever transport is selected.
4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.
Bug: webrtc:9719
Change-Id: Ifcc428c8d76fb77dcc8abaa79507c620bcfb31b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140920
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28198}
This reverts commit 71c6482baf0ff17141c635e6a7639493db68a65c.
Reason for revert: Lands too much at once and breaks downstream tests that need to implement new interfaces first.
Original change's description:
> Implement true negotiation for DatagramTransport with fallback to RTP.
>
> In short, the caller places a x-opaque line in SDP for each m= section that
> uses datagram transport. If the answerer supports datagram transport, it will
> parse this line and create a datagram transport. It will then echo the x-opaque
> line into the answer (to indicate that it accepted use of datagram transport).
>
> If the offer and answer contain exactly the same x-opaque line, both peers will
> use datagram transport. If the x-opaque line is omitted from the answer (or is
> different in the answer) they will fall back to RTP.
>
> Note that a different x-opaque line in the answer means the answerer did not
> understand something in the negotiation proto. Since WebRTC cannot know what
> was misunderstood, or whether it's still possible to use the datagram transport,
> it must fall back to RTP. This may change in the future, possibly by passing
> the answer to the datagram transport, but it's good enough for now.
>
> Negotiation consists of four parts:
> 1. DatagramTransport exposes transport parameters for both client and server
> perspectives. The client just echoes what it received from the server (modulo
> any fields it might not have understood).
>
> 2. SDP adds a x-opaque line for opaque transport parameters. Identical to
> x-mt, but this is specific to datagram transport and goes in each m= section,
> and appears in the answer as well as the offer.
> - This is propagated to Jsep as part of the TransportDescription.
> - SDP files: transport_description.h,cc, transport_description_factory.h,cc,
> media_session.cc, webrtc_sdp.cc
>
> 3. JsepTransport/Controller:
> - Exposes opaque parameters for each mid (m= section). On offerer, this means
> pre-allocating a datagram transport and getting its parameters. On the
> answerer, this means echoing the offerer's parameters.
> - Uses a composite RTP transport to receive from either default RTP or
> datagram transport until both offer and answer arrive.
> - If a provisional answer arrives, sets the composite to send on the
> provisionally selected transport.
> - Once both offer and answer are set, deletes the unneeded transports and
> keeps whichever transport is selected.
>
> 4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.
>
> Bug: webrtc:9719
> Change-Id: Id8996eb1871e79d93b7923a5d7eb3431548c798d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140700
> Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28182}
TBR=steveanton@webrtc.org,mellem@webrtc.org,sukhanov@webrtc.org
Change-Id: I0d502c4a6d27516c35ed85154f3fa5869f88b3b7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140822
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28188}
In short, the caller places a x-opaque line in SDP for each m= section that
uses datagram transport. If the answerer supports datagram transport, it will
parse this line and create a datagram transport. It will then echo the x-opaque
line into the answer (to indicate that it accepted use of datagram transport).
If the offer and answer contain exactly the same x-opaque line, both peers will
use datagram transport. If the x-opaque line is omitted from the answer (or is
different in the answer) they will fall back to RTP.
Note that a different x-opaque line in the answer means the answerer did not
understand something in the negotiation proto. Since WebRTC cannot know what
was misunderstood, or whether it's still possible to use the datagram transport,
it must fall back to RTP. This may change in the future, possibly by passing
the answer to the datagram transport, but it's good enough for now.
Negotiation consists of four parts:
1. DatagramTransport exposes transport parameters for both client and server
perspectives. The client just echoes what it received from the server (modulo
any fields it might not have understood).
2. SDP adds a x-opaque line for opaque transport parameters. Identical to
x-mt, but this is specific to datagram transport and goes in each m= section,
and appears in the answer as well as the offer.
- This is propagated to Jsep as part of the TransportDescription.
- SDP files: transport_description.h,cc, transport_description_factory.h,cc,
media_session.cc, webrtc_sdp.cc
3. JsepTransport/Controller:
- Exposes opaque parameters for each mid (m= section). On offerer, this means
pre-allocating a datagram transport and getting its parameters. On the
answerer, this means echoing the offerer's parameters.
- Uses a composite RTP transport to receive from either default RTP or
datagram transport until both offer and answer arrive.
- If a provisional answer arrives, sets the composite to send on the
provisionally selected transport.
- Once both offer and answer are set, deletes the unneeded transports and
keeps whichever transport is selected.
4. PeerConnection pulls transport parameters out of Jsep and adds them to SDP.
Bug: webrtc:9719
Change-Id: Id8996eb1871e79d93b7923a5d7eb3431548c798d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140700
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28182}
Raw RTP packetization is done using the existing RtpPacketizerGeneric
without adding the generic payload header. It is intended to be used
together with generic frame descriptor RTP header extension.
Bug: webrtc:10625
Change-Id: I2e3d0a766e4933ddc4ad4abc1449b9b91ba6cd35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138061
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28154}
This documents in the API what is already true in the
implementation - that SessionDescription will eventually
delete MediaDescription objects passed to it.
The old API is preserved for backwards compatibility, but
marked as RTC_DEPRECATED.
Bug: webrtc:10701
Change-Id: I9a822b20cf3e58c5945fa51dbf6082960a332de8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28144}
This reverts commit 4436887ed2d3324279e0f2e091c9e9355392721a.
Reason for revert: The original revert was intended to be temporary.
Original change's description:
> Revert "Change default secure SCTP protocol to UDP/DTLS/SCTP"
>
> This reverts commit c3f4820e129d44471b366b8885a67b5392918d5a.
>
> Reason for revert: Will temporarily revert to fix an issue and reland afterwards.
>
> Original change's description:
> > Change default secure SCTP protocol to UDP/DTLS/SCTP
> >
> > The old value - DTLS/SCTP - is not standards conformant,
> > and the new value should be parsable since Chrome M61.
> >
> > Bug: webrtc:7706
> > Change-Id: I7468cc9597dec4ef4b102fccddc4e981fed7e8d8
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136804
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27940}
>
> TBR=steveanton@webrtc.org,mbonadei@webrtc.org,hbos@webrtc.org,hta@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:7706
> Change-Id: Ida8ae20767485c75edc44dff8a3fa1af2006f207
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139244
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28121}
TBR=steveanton@webrtc.org,mbonadei@webrtc.org,hbos@webrtc.org,hta@webrtc.org,guidou@webrtc.org
Change-Id: I381fa18b644874c20ddaa4cd13fec79a5fd9555a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7706
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139246
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28122}
This reverts commit c3f4820e129d44471b366b8885a67b5392918d5a.
Reason for revert: Will temporarily revert to fix an issue and reland afterwards.
Original change's description:
> Change default secure SCTP protocol to UDP/DTLS/SCTP
>
> The old value - DTLS/SCTP - is not standards conformant,
> and the new value should be parsable since Chrome M61.
>
> Bug: webrtc:7706
> Change-Id: I7468cc9597dec4ef4b102fccddc4e981fed7e8d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136804
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27940}
TBR=steveanton@webrtc.org,mbonadei@webrtc.org,hbos@webrtc.org,hta@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:7706
Change-Id: Ida8ae20767485c75edc44dff8a3fa1af2006f207
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139244
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28121}
