Commit graph

8 commits

Author SHA1 Message Date
Johan Sternerup
8dbdfad914 webrtcbin: Support closing of data channels
Support for closing WebRTC data channels as described in RFC
8831 (section 6.7) now fully supported. This means that we can now
reuse data channels that have been closed properly. Previously, an
application that created a lot of short-lived on-demand data channels
would quickly exhaust resources held by lingering non-closed data
channels.

We now use a one-to-one style socket interface to SCTP just like the
Google implementation (i.e. SOCK_STREAM instead of SOCK_SEQPACKET, see
RFC 6458). For some reason the socket interface to use was made
optional through a property "use-sock-stream" even though code wasn't
written to handle the SOCK_SEQPACKET style. Specifically the
SCTP_RESET_STREAMS command wouldn't work without passing the correct
assocation id. Changing the default interface to use from
SOCK_SEQPACKET to SOCK_STREAM now means we don't have to bother about
the association id as there is only one association per socket. For
the SCTP_RESET_STREAMS command we set it to SCTP_ALL_ASSOC just to
match the Google implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2186>
2021-05-12 03:02:27 +00:00
Doug Nazar
4e29ba9fce webrtc: Fix sctp task's return type.
GstWebRTCBinFunc expects a GstStructure* return type.

Fixes segfault on PowerPC.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2188>
2021-04-22 16:14:41 -04:00
Olivier Crête
80a56c25a6 webrtc: Set the DSCP markings based on the priority
This matches how the WebRTC javascript API works and the Chrome implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
2020-10-30 16:24:40 -04:00
Sebastian Dröge
b565a7ef66 Revert "webrtc: Set the DSCP markings based on the priority"
This reverts commit 8ba08598bb.

It breaks the CI until the C# bindings are fixed.
2020-10-08 18:53:00 +03:00
Olivier Crête
8ba08598bb webrtc: Set the DSCP markings based on the priority
This matches how the WebRTC javascript API works and the Chrome implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
2020-10-06 16:49:08 -04:00
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
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
Matthew Waters
07e9374eff webrtcbin: add support for data channels based on SCTP
Mostly follows the W3C specification
https://www.w3.org/TR/webrtc/#peer-to-peer-data-api

With contributions from:
Mathieu Duponchelle <mathieu@centricular.com>

https://bugzilla.gnome.org/show_bug.cgi?id=794351
2018-09-21 19:45:12 +10:00