Commit graph

96 commits

Author SHA1 Message Date
Matthew Waters
597c1b4ec6 webrtc: remove private properties/signals from the now public ice object
We don't want to expose all of the webrtcbin internals to the world.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1444>
2020-07-20 15:56:20 +10:00
Olivier Crête
cceca1ffe8 webrtcbin: Expose "latency" property
This property sets the latency both on the rtpbin/rtpjittbuffer, but
also on the RTPStorage elements currently used by the FEC decoder.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1367>
2020-06-29 22:45:31 -04:00
Sebastian Dröge
aa01e6ba22 webrtcbin: Don't call gst_ghost_pad_construct() anymore
It's deprecated, unneeded and doesn't do anything anymore.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1360>
2020-06-22 17:01:34 +00:00
Matthew Waters
0f41c0f000 webrtc: fix ice control mode when we offer initially
An initial offer means we have a local description not a remote
description.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1332

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1358>
2020-06-22 12:17:09 +00:00
Mathieu Duponchelle
a048ce81d4 plugins: uddate gst_type_mark_as_plugin_api() calls 2020-06-06 00:40:42 +02:00
Sebastian Dröge
74f2f733be plugins: Use gst_type_mark_as_plugin_api() for all non-element plugin types 2020-06-04 13:33:16 -04:00
Sebastian Dröge
b25d153c34 webrtc: Add GstWebRTCDataChannel to the library API
This makes it more discoverable for bindings and allows bindings to
generate static API for the signals and functions.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1168

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1313>
2020-06-02 21:04:37 +00:00
Matthew Waters
67ae885d4c webrtc: handle an ice-lite remote offer
When the remote peer offers an ice-lite SDP, we need to configure our
ICE negotiation to be in controlling mode as the peer will not be.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1304>
2020-05-28 19:57:45 +10:00
Chris Ayoup
9937101e51 webrtc: move filtering properties to webrtcice
We want webrtcbin to only expose properties that are defined in JSEP, so
these additional properties should be moved out.  In order to access
them, the webrtcice instance is exposed from webrtcbin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>
2020-05-11 05:30:59 +00:00
Chris Ayoup
ca754245e9 webrtc: allow setting local IP addresses
If a local IP address is specified, ICE gathering can be much faster
in environments where there are multiple IP addreses but only some are
usable (for example, if you are running docker on the machine).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>
2020-05-11 05:30:59 +00:00
Chris Ayoup
3fc8818824 webrtc: Allow toggling TCP and UDP candidates
Add some properties to allow TCP and UDP candidates to be toggled.  This
is useful in cases where someone is using this element in an environment
where it is known in advance whether a given transport will work or not
and will prevent wasting time generating and checking candidate pairs
that will not succeed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>
2020-05-11 05:30:59 +00:00
Matthew Waters
18de5f8f04 webrtc: remove debugging leftover
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1247>
2020-05-06 15:49:58 +10:00
Matthew Waters
50644f5718 webrtc: always reply to a promise
Otherwise, we defeat the purpose of a promise.

We were not replying when the state was closed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
2020-05-06 02:53:27 +00:00
Matthew Waters
1f395e3ddb webrtc: name threads based on the element name
Makes debugging a busy loop possibly easier

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
2020-05-06 02:53:27 +00:00
Matthew Waters
d552c6556c webrtc: correctly use the pad template
GstHarness uses this for releasing request pads correctly. Fixes
numerous leaks in the webrtc unit tests.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
2020-05-06 02:53:27 +00:00
Matthew Waters
46176fbcc7 webrtc: Fix a couple of renegotiation races
When negotiating the SDP we should only connect the streams that are
actually mentioned in the SDP.  All other streams are not relevant at
this time and would likely be part of a future SDP update.  Fixes a
couple of the renegotiation webrtc unit tests.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
2020-05-06 02:53:27 +00:00
Matthew Waters
b266652043 webrtcbin: also mark data channel transports as active
Fixes negotiation of a bundled sdp with only a data channel.

Without marking the transport as active, we would never unblock the
transportreceivebin and thus no data would ever reach us.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1231>
2020-05-01 03:13:46 +00:00
Matthew Waters
ce9b41f5d4 webrtcbin: fix bundle none case with remote offer bundling
If the remote is bundling, but we are not and remote is offering.
we cannot put the remote media sections into a bundled transport as that
is not how we are going to respond.

This specific failure case was that the remote ICE credentials were
never set on the ice stream and so ice connectivity would fail.

Technically, this whole bunde-policy=none handling should be removed
eventually when we implement bundle-policy=balanced.  Until such time,
we have this workaround.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1231>
2020-05-01 03:13:46 +00:00
Matthew Waters
80ede09193 webrtcbin: only start gathering on local descriptions
If we are in a state where we are answering, we would start gathering
when the offer is set which is incorrect for at least two reasons.

1. Sending ICE candidates before sending an answer is a hard error in
   all of the major browsers and will fail the negotiation.
2. If libnice ever adds the username fragment to the candidate for
   ice-restart hardening, the ice username and fragment would be
   incorrect.

JSEP also hints that the right call flow is to only start gathering when
a local description is set in 4.1.9 setLocalDescription

"This API indirectly controls the candidate gathering process."

as well as hints throughout other sections.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1226>
2020-04-30 14:47:55 +00:00
Matthew Waters
319a5e5779 webrtc: mark streams as active on renegotiation as well.
Otherwise when bundling, only the changed streams would be considered as
to whether the bundled transport needs to be blocked as all streams are
inactive.

Scenario is one transceiver changes direction to inactive and as that is
the only change in transciever direction, the entire bundled transport would
be blocked even if there are other active transceivers inside the same bundled
transport that are still active.

Fix by always checking the activeness of a stream regardless of if the
transceiverr has changed direction.
2020-03-25 14:46:15 +11:00
Sebastian Dröge
5a2053e0af webrtcbin: Use GPtrArrays or store items inline instead of using GArrays of pointers 2020-03-09 21:38:42 +02:00
Jan Schmidt
8274fcd311 webrtcbin: Prevent ICE gathering state reaching complete early
The ICE gathering state can transition to complete prematurely if the
underlying ICE components complete their gathering while the initial
ICE gathering state task is queued and still pending.

In that situation, the ice gathering state task will report complete
while there are still ICE candidates queued for emission.

Prevent that by storing ICE candidates in an array and checking if
there are any pending before reporting a completed ICE gathering
state.
2020-03-10 05:47:40 +11:00
Jan Schmidt
9410ef56b8 webrtc: Protect the pending ICE candidates array
ICE candidates can be added to the array directly from the application
or from the webrtc main loop. Rename it to make it clear that it's
holding remote ICE candidates from the peer, and protect it with a
new mutex
2020-03-10 05:25:40 +11:00
Jan Schmidt
ad53de1da1 webrtc: Don't crash in ICE gathering
Fix a crash collating ICE gathering states if there are
unassociated transceivers in the list with no TransportStream
2020-03-04 23:06:52 +00:00
Jan Schmidt
905988c63f webrtc: Unblock transportreceivebin for send-only bundled streams
If there is any active mline in a bundle, we need to unblock
the transportreceivebin for DTLS setup and RTCP reception,
otherwise no data can ever start flowing.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1206
2020-03-04 10:15:19 +00:00
Jan Schmidt
cb48733ff3 webrtc: Remove RECEIVE_STATE_DROP from transportreceivebin
As per discussion in the bug, remove the drop state from transportreceivebin.
Dropping data is necessary, but for bundled config, needs to happen
further downstream after mixed flows have been separated.

Also support switching back to BLOCK from PASS state.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1206
2020-03-04 10:15:19 +00:00
Jan Schmidt
96a407334d webrtc: Merge ICE candidates to local descriptions
When emitting ICE candidates, also merge them to the local and
pending description so they show up in the SDP if those are
retrieved from the current-local-description and
pending-local-description properties.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/676
2020-02-17 14:23:56 +00:00
Sebastian Dröge
4ffa6350e8 webrtc: In all blocking pad probes except for sink pads also handle serialized events
Otherwise it can happen that e.g. the stream-start event is tried to be
sent as part of pushing the first buffer. Downstream might not be in
PAUSED/PLAYING yet, so the event is rejected with GST_FLOW_FLUSHING and
because it's an event would not cause the blocking pad probe to trigger
first. This would then return GST_FLOW_FLUSHING for the buffer and shut
down all of upstream.

To solve this we return GST_PAD_PROBE_DROP for all events. In case of
sticky events they would be resent again later once we unblocked after
blocking on the buffer and everything works fine.

Don't handle events specifically in sink pad blocking pad probes as here
downstream is not linked yet and we are actually waiting for the
following CAPS event before unblocking can happen.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1172
2020-02-11 00:49:51 +00:00
Sebastian Dröge
c16d4d2c33 webrtcbin: Add a blocking pad probe for the receivebin -> sctpdec connection
Without this it might happen that received data from the DTLS transport
is already passed to sctpdec before its state was set to PLAYING. This
would cause the data to be dropped, GST_FLOW_FLUSHING to be returned and
the whole DTLS transport to shut down.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1172
among other things.
2020-02-11 00:49:51 +00:00
Mathieu Duponchelle
7cc185bd86 webrtcbin: connect rtp funnel after updating ptmaps
We need the streams' pt maps updated before requesting pads
on rtpbin, because this is what will trigger the requesting
of FEC encoders, and our handler for this request looks for
the payload types in the relevant stream's pt map.

Fixes #1187
2020-01-21 11:17:38 +00:00
Sebastian Dröge
0c39068c89 webrtcbin: Start datachannel SCTP elements only after the DTLS connection is established
Otherwise we would start sending data to the DTLS connection before, and
the DTLS elements consider this an error.

Also RFC 8261 mentions:
  o A DTLS connection MUST be established before an SCTP association can
    be set up.
2020-01-19 11:16:34 +00:00
Sebastian Dröge
2798a80ebe webrtcbin: Add handling of unspecified peer-connection-state situation
For us it can happen that the DTLS transports are still in the process
of connecting while the ICE transport is already completed. This
situation is not specified in the spec but conceptually that means it is
still in the process of connecting.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/758
2020-01-19 11:16:34 +00:00
Sebastian Dröge
4b73322333 webrtcbin: Return the old state if we ended up being in an unspecified situation
Previously we would've returned NEW, which is usually more wrong.
2020-01-19 11:16:34 +00:00
Sebastian Dröge
22869356db webrtcbin: Fix transitions for the peer connection state
They're now mapping exactly to what
  https://www.w3.org/TR/webrtc/#rtcpeerconnectionstate-enum
actually specifies.

Related to https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/758
2020-01-19 11:16:34 +00:00
Sebastian Dröge
41175f4ebe webrtcbin: Fix transitions for the connection state
They're now mapping exactly to what
  https://www.w3.org/TR/webrtc/#dom-rtciceconnectionstate
actually specifies.

Related to https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/758
2020-01-19 11:16:34 +00:00
Sebastian Dröge
47ce34d32c webrtcbin: Don't consider transceivers without mid as inactive during ICE gathering state updates
We don't have any mid before parsing the SDP, which happens after we
handled the SDP answer and that usually happens long after ICE candidate
gathering is finished.

Without this all transceivers are considered inactive and as such ICE
gathering is for active transceiver was considered complete from the
very beginning.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1126
2020-01-19 10:47:59 +00:00
Sebastian Dröge
de0f803d56 webrtcbin: Don't consider RTP receivers stopped
We don't support stopping RTP receivers currently so let's not consider
them all stopped all the time. This fixes some of the ICE/DTLS state
change handling and specifically fixes the ICE gathering state.

Previously the ICE gathering state was immediately going from NEW to
COMPLETE because it considered all transceivers stopped and as such all
activate transceivers were finished gathering ICE candidates.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1126
2020-01-19 10:47:59 +00:00
Sebastian Dröge
57c982a1dd webrtcbin: Improve logging related to ICE/DTLS state changes 2020-01-19 10:47:59 +00:00
Jan Schmidt
8e87fe42ad WebRTC: Support non-trickle ICE candidates in the SDP
Add any ICE candidates from the SDP before adding pending
trickle ICE candidates to support non-trickle peers

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/678
2020-01-13 02:30:44 +11:00
Edward Hervey
c026522084 webrtcbin: Fix memory leak
The structure is not used after this block

CID: 1455481
2019-11-21 16:25:21 +01:00
Niels De Graef
d8f61515d8 Don't pass default GLib marshallers for signals
By passing NULL to `g_signal_new` instead of a marshaller, GLib will
actually internally optimize the signal (if the marshaller is available
in GLib itself) by also setting the valist marshaller. This makes the
signal emission a bit more performant than the regular marshalling,
which still needs to box into `GValue` and call libffi in case of a
generic marshaller.

Note that for custom marshallers, one would use
`g_signal_set_va_marshaller()` with the valist marshaller instead.
2019-11-06 14:27:46 +00:00
Aaron Boxer
6d3429af34 documentation: fixed a heap o' typos 2019-11-05 09:11:25 -05:00
David Gunzinger
e2e86658f2 webrtc: fix type of max-retransmits, make it work 2019-08-13 12:17:13 +02:00
Jakub Adam
831b124976 webrtcbin: Support data channel SDP offers from Chrome
When negotiating a data channel, Chrome as recent as 75 still uses SDP
based on version 05 of the SCTP SDP draft, for example:

 m=application 9 DTLS/SCTP 5000
 a=sctpmap:5000 webrtc-datachannel 1024

Implement support for parsing SCTP port out of SDP message with sctpmap
attribute. Fixes data channel negotiation with Chrome browser.
2019-07-29 22:04:08 +00:00
Sebastian Dröge
329b2d3a6a webrtcbin: Don't assert if an SDP media can't be converted to caps
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1008
2019-07-08 07:18:41 +00:00
Matthew Waters
cef839533e webrtcbin: use the latest self-generated SDP as the basis for renegotiations
Fixes multiple errors when a webrtcbin renegotiation can switch between the
offerer and the answerer.
2019-07-03 23:44:15 +00:00
Matthew Waters
95488812b2 webrtc: fix the location of signalling-state change notification
1. The spec indicates that the notification should occur near the end of
   'setting the description' processing
2. The current location with the drop of the lock could cause the 'check
   if negotiation is needed' logic to execute and become confused about
   the state of the webrtcbin's current local descriptions.
   In the bad case, the following assertions could be hit:
   g_assert (trans->mline < gst_sdp_message_medias_len (webrtc->current_local_description->sdp));
   g_assert (trans->mline < gst_sdp_message_medias_len (webrtc->current_remote_description->sdp));

Moving the signalling state change later in the set description task
means that checking for a renegotiation will early abort as the
signalling state is not STABLE before the session description and
transceivers have been updated.
2019-06-04 05:43:43 +00:00
Matthew Waters
f8911deccf webrtc: only set sctp ports if they are different
SCTPassociation will complain if we do that while running and resetting
is not something we support at the moment
2019-05-30 21:33:09 +10:00
Matthew Waters
be011d2086 webrtc/dc: move some code from webrtcbin into the datachannel 2019-05-30 21:33:09 +10:00
Matthew Waters
a51db86ac4 webrtc: hold onto any unknown ICE candidates until the next SDP set
It is very possible for badly behaving signalling or peers to send
us ICE candidates before we receive an SDP.  While we had consideration
for that on the first set SDP, subsequent SDP's could result in
misconfigured ICE transports.  Expand the previous code to also take
into account reconfigurations.
2019-05-30 21:33:09 +10:00