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Code Issues 1 Projects Releases Wiki Activity

rtspclientsink: add RTSP address pool for unicast UDP

Browse source 
Adds an address pool for rtspclientsink in order to allow the
"port-range" property to restrict the ports available for the RTSP
streams rather than always using the ephemeral port-range.

If a value is not provided to the "port-range" property, rtspclientsink
will select random ports from the ephemeral port-range as before.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2606

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4828>
This commit is contained in:
James Oliver 2023-06-11 15:33:33 +08:00 committed by GStreamer Marge Bot
parent 33fb3bfd60
commit 87c177567d
3 changed files with 43 additions and 18 deletions
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2
subprojects/gst-rtsp-server/docs/plugins/gst_plugins_cache.json
View file

@ -103,7 +103,7 @@
"writable": true
},
"port-range": {
"blurb": "Client port range that can be used to receive RTCP data, eg. 3000-3005 (NULL = no restrictions)",
"blurb": "Client port range that can be used to send RTP data and receive RTCP data, eg. 3000-3005 (NULL = no restrictions)",
"conditionally-available": false,
"construct": false,
"construct-only": false,

38
subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.c
View file

@ -627,13 +627,13 @@ gst_rtsp_client_sink_class_init (GstRTSPClientSinkClass * klass)
/**
* GstRTSPClientSink:port-range:
*
* Configure the client port numbers that can be used to receive
* Configure the client port numbers that can be used to send RTP and receive
* RTCP.
*/
g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
g_param_spec_string ("port-range", "Port range",
"Client port range that can be used to receive RTCP data, "
"eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
"Client port range that can be used to send RTP data and receive RTCP "
"data, eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
@ -898,6 +898,8 @@ gst_rtsp_client_sink_init (GstRTSPClientSink * sink)
sink->user_agent = g_strdup (DEFAULT_USER_AGENT);
sink->publish_clock_mode = DEFAULT_PUBLISH_CLOCK_MODE;
sink->pool = NULL;
sink->profiles = DEFAULT_PROFILES;
/* protects the streaming thread in interleaved mode or the polling
@ -952,6 +954,10 @@ gst_rtsp_client_sink_finalize (GObject * object)
g_free (rtsp_client_sink->multi_iface);
g_free (rtsp_client_sink->user_agent);
if (rtsp_client_sink->pool) {
gst_object_unref (rtsp_client_sink->pool);
rtsp_client_sink->pool = NULL;
}
if (rtsp_client_sink->uri_sdp) {
gst_sdp_message_free (rtsp_client_sink->uri_sdp);
rtsp_client_sink->uri_sdp = NULL;
@ -1221,10 +1227,8 @@ gst_rtsp_client_sink_create_stream (GstRTSPClientSink * sink,
gst_rtsp_stream_set_ulpfec_percentage (stream, context->ulpfec_percentage);
gst_rtsp_stream_set_publish_clock_mode (stream, sink->publish_clock_mode);
#if 0
if (priv->pool)
gst_rtsp_stream_set_address_pool (stream, priv->pool);
#endif
if (sink->pool)
gst_rtsp_stream_set_address_pool (stream, sink->pool);
return stream;
no_free_pt:
@ -5065,6 +5069,21 @@ gst_rtsp_client_sink_change_state (GstElement * element,
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* init some state */
rtsp_client_sink->cur_protocols = rtsp_client_sink->protocols;
/* setup IPv4/IPv6 unicast port range. */
if (!rtsp_client_sink->pool)
rtsp_client_sink->pool = gst_rtsp_address_pool_new ();
if (rtsp_client_sink->client_port_range.max > 0) {
gst_rtsp_address_pool_add_range (rtsp_client_sink->pool,
GST_RTSP_ADDRESS_POOL_ANY_IPV4, GST_RTSP_ADDRESS_POOL_ANY_IPV4,
rtsp_client_sink->client_port_range.min,
rtsp_client_sink->client_port_range.max, 0);
gst_rtsp_address_pool_add_range (rtsp_client_sink->pool,
GST_RTSP_ADDRESS_POOL_ANY_IPV6, GST_RTSP_ADDRESS_POOL_ANY_IPV6,
rtsp_client_sink->client_port_range.min,
rtsp_client_sink->client_port_range.max, 0);
}
/* first attempt, don't ignore timeouts */
rtsp_client_sink->ignore_timeout = FALSE;
rtsp_client_sink->open_error = FALSE;
@ -5093,6 +5112,11 @@ gst_rtsp_client_sink_change_state (GstElement * element,
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtsp_client_sink_set_state (rtsp_client_sink, GST_STATE_READY);
if (rtsp_client_sink->pool) {
gst_object_unref (rtsp_client_sink->pool);
rtsp_client_sink->pool = NULL;
}
break;
default:
break;

21
subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.h
View file

@ -189,16 +189,17 @@ struct _GstRTSPClientSink {
GstRTSPPublishClockMode publish_clock_mode;
/* state */
GstRTSPState state;
gchar *content_base;
GstRTSPLowerTrans cur_protocols;
gboolean tried_url_auth;
gchar *addr;
gboolean need_redirect;
GstRTSPTimeRange *range;
gchar *control;
guint next_port_num;
GstClock *provided_clock;
GstRTSPState state;
gchar *content_base;
GstRTSPLowerTrans cur_protocols;
gboolean tried_url_auth;
gchar *addr;
GstRTSPAddressPool *pool;
gboolean need_redirect;
GstRTSPTimeRange *range;
gchar *control;
guint next_port_num;
GstClock *provided_clock;
/* supported methods */
gint methods;