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8dbdfad914
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>
73 lines
2.8 KiB
C
73 lines
2.8 KiB
C
/* GStreamer
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* Copyright (C) 2018 Matthew Waters <matthew@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __WEBRTC_DATA_CHANNEL_H__
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#define __WEBRTC_DATA_CHANNEL_H__
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#include <gst/gst.h>
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#include <gst/webrtc/webrtc_fwd.h>
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#include <gst/webrtc/dtlstransport.h>
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#include <gst/webrtc/datachannel.h>
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#include "sctptransport.h"
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G_BEGIN_DECLS
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GType webrtc_data_channel_get_type(void);
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#define WEBRTC_TYPE_DATA_CHANNEL (webrtc_data_channel_get_type())
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#define WEBRTC_DATA_CHANNEL(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),WEBRTC_TYPE_DATA_CHANNEL,WebRTCDataChannel))
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#define WEBRTC_IS_DATA_CHANNEL(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),WEBRTC_TYPE_DATA_CHANNEL))
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#define WEBRTC_DATA_CHANNEL_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,WEBRTC_TYPE_DATA_CHANNEL,WebRTCDataChannelClass))
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#define WEBRTC_IS_DATA_CHANNEL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,WEBRTC_TYPE_DATA_CHANNEL))
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#define WEBRTC_DATA_CHANNEL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,WEBRTC_TYPE_DATA_CHANNEL,WebRTCDataChannelClass))
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typedef struct _WebRTCDataChannel WebRTCDataChannel;
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typedef struct _WebRTCDataChannelClass WebRTCDataChannelClass;
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struct _WebRTCDataChannel
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{
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GstWebRTCDataChannel parent;
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GstWebRTCSCTPTransport *sctp_transport;
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GstElement *appsrc;
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GstElement *appsink;
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GstWebRTCBin *webrtcbin;
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gboolean opened;
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gulong src_probe;
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GError *stored_error;
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gboolean peer_closed;
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gpointer _padding[GST_PADDING];
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};
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struct _WebRTCDataChannelClass
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{
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GstWebRTCDataChannelClass parent_class;
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gpointer _padding[GST_PADDING];
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};
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void webrtc_data_channel_start_negotiation (WebRTCDataChannel *channel);
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G_GNUC_INTERNAL
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void webrtc_data_channel_link_to_sctp (WebRTCDataChannel *channel,
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GstWebRTCSCTPTransport *sctp_transport);
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G_END_DECLS
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#endif /* __WEBRTC_DATA_CHANNEL_H__ */
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