gstreamer/ext/webrtc
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
..
fwd.h webrtcbin: add support for data channels based on SCTP 2018-09-21 19:45:12 +10:00
gstwebrtc.c webrtcbin: an element that handles the transport aspects of webrtc connections 2018-02-02 15:02:21 +11:00
gstwebrtcbin.c webrtcbin: Support closing of data channels 2021-05-12 03:02:27 +00:00
gstwebrtcbin.h webrtcbin: Move GstPromise reply to operation framework 2021-04-12 18:37:27 -04:00
gstwebrtcice.c webrtc ice: Add 'min/max-rtp-port' props for setting RTP port range 2021-03-01 14:42:17 +00:00
gstwebrtcice.h webrtc ice: Add 'min/max-rtp-port' props for setting RTP port range 2021-03-01 14:42:17 +00:00
gstwebrtcstats.c webrtc/stats: provide the twcc stats when available 2021-04-29 22:01:54 +10:00
gstwebrtcstats.h webrtcbin: Implement getting stats for a specific pad 2020-11-24 04:27:52 +00:00
icestream.c webrtc ice: Add 'min/max-rtp-port' props for setting RTP port range 2021-03-01 14:42:17 +00:00
icestream.h webrtcbin: an element that handles the transport aspects of webrtc connections 2018-02-02 15:02:21 +11:00
meson.build webrtc: Update libnice version requirement to 0.1.17 2020-11-11 13:41:59 +02:00
nicetransport.c Fix missing unref of nice-agent causing sockets to never close. 2021-04-22 21:14:49 +00:00
nicetransport.h webrtc: Add properties to change the socket buffer sizes to ice object 2020-11-03 22:07:53 +00:00
sctptransport.c webrtcbin: Support closing of data channels 2021-05-12 03:02:27 +00:00
sctptransport.h webrtc: Set the DSCP markings based on the priority 2020-10-30 16:24:40 -04:00
transportreceivebin.c Use gst_element_request_pad_simple... 2021-05-05 06:17:14 +00:00
transportreceivebin.h webrtc: Remove RECEIVE_STATE_DROP from transportreceivebin 2020-03-04 10:15:19 +00:00
transportsendbin.c webrtcbin: Remove remnant of non-rtcp-mux mode 2021-01-06 23:02:37 +00:00
transportsendbin.h webrtcbin: Remove remnant of non-rtcp-mux mode 2021-01-06 23:02:37 +00:00
transportstream.c webrtcbin: Remove remnant of non-rtcp-mux mode 2021-01-06 23:02:37 +00:00
transportstream.h webrtcbin: Remove remnant of non-rtcp-mux mode 2021-01-06 23:02:37 +00:00
utils.c webrtc: only add nack pli by default if kind is video 2021-05-06 12:19:51 +00:00
utils.h webrtc: move webrtc_kind_from_caps() to utils 2021-05-06 12:19:51 +00:00
webrtcdatachannel.c webrtcbin: Support closing of data channels 2021-05-12 03:02:27 +00:00
webrtcdatachannel.h webrtcbin: Support closing of data channels 2021-05-12 03:02:27 +00:00
webrtcsdp.c webrtc: propagate more errors through the promise 2020-09-14 04:04:29 +00:00
webrtcsdp.h webrtc: propagate more errors through the promise 2020-09-14 04:04:29 +00:00
webrtctransceiver.c webrtc: expose transport property on sender and receiver 2021-01-13 19:22:42 +00:00
webrtctransceiver.h webrtcbin: Remember if a transceiver had a forced m-line 2021-04-12 17:55:07 -04:00