From 92626535c7136725dfab0857d6c3900e041084b7 Mon Sep 17 00:00:00 2001 From: Ilya Kreymer Date: Thu, 15 Aug 2019 08:25:26 -0700 Subject: webrtc ice: Add 'min/max-rtp-port' props for setting RTP port range default min port == 0, max port == 65535 -- if min port == 0, uses existing random port selection (range ignored) add 'gathering_started' flag to avoid changing ports after gathering has started validity checks: min port <= max port enforced, error thrown otherwise include tests to ensure port range is being utilized (by @hhardy) Part-of: --- docs/plugins/gst_plugins_cache.json | 28 ++++++++++++++ ext/webrtc/gstwebrtcice.c | 57 ++++++++++++++++++++++++++++ ext/webrtc/gstwebrtcice.h | 3 ++ ext/webrtc/icestream.c | 19 ++++++++++ tests/check/elements/webrtcbin.c | 75 ++++++++++++++++++++++++++++++++++++- 5 files changed, 181 insertions(+), 1 deletion(-) diff --git a/docs/plugins/gst_plugins_cache.json b/docs/plugins/gst_plugins_cache.json index fd4f6d031..ff2af7934 100644 --- a/docs/plugins/gst_plugins_cache.json +++ b/docs/plugins/gst_plugins_cache.json @@ -225773,6 +225773,34 @@ "readable": true, "type": "gboolean", "writable": true + }, + "max-rtp-port": { + "blurb": "Maximum port for local rtp port range. max-rtp-port must be >= min-rtp-port", + "conditionally-available": false, + "construct": true, + "construct-only": false, + "controllable": false, + "default": "65535", + "max": "65535", + "min": "0", + "mutable": "null", + "readable": true, + "type": "guint", + "writable": true + }, + "min-rtp-port": { + "blurb": "Minimum port for local rtp port range. min-rtp-port must be <= max-rtp-port", + "conditionally-available": false, + "construct": false, + "construct-only": false, + "controllable": false, + "default": "0", + "max": "65535", + "min": "0", + "mutable": "null", + "readable": true, + "type": "guint", + "writable": true } }, "signals": { diff --git a/ext/webrtc/gstwebrtcice.c b/ext/webrtc/gstwebrtcice.c index 8f3050835..886856cc9 100644 --- a/ext/webrtc/gstwebrtcice.c +++ b/ext/webrtc/gstwebrtcice.c @@ -56,6 +56,8 @@ enum PROP_AGENT, PROP_ICE_TCP, PROP_ICE_UDP, + PROP_MIN_RTP_PORT, + PROP_MAX_RTP_PORT, }; static guint gst_webrtc_ice_signals[LAST_SIGNAL] = { 0 }; @@ -991,6 +993,21 @@ gst_webrtc_ice_set_property (GObject * object, guint prop_id, g_object_set_property (G_OBJECT (ice->priv->nice_agent), "ice-udp", value); break; + + case PROP_MIN_RTP_PORT: + ice->min_rtp_port = g_value_get_uint (value); + if (ice->min_rtp_port > ice->max_rtp_port) + g_warning ("Set min-rtp-port to %u which is larger than" + " max-rtp-port %u", ice->min_rtp_port, ice->max_rtp_port); + break; + + case PROP_MAX_RTP_PORT: + ice->max_rtp_port = g_value_get_uint (value); + if (ice->min_rtp_port > ice->max_rtp_port) + g_warning ("Set max-rtp-port to %u which is smaller than" + " min-rtp-port %u", ice->max_rtp_port, ice->min_rtp_port); + break; + default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; @@ -1015,6 +1032,15 @@ gst_webrtc_ice_get_property (GObject * object, guint prop_id, g_object_get_property (G_OBJECT (ice->priv->nice_agent), "ice-udp", value); break; + + case PROP_MIN_RTP_PORT: + g_value_set_uint (value, ice->min_rtp_port); + break; + + case PROP_MAX_RTP_PORT: + g_value_set_uint (value, ice->max_rtp_port); + break; + default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; @@ -1101,6 +1127,37 @@ gst_webrtc_ice_class_init (GstWebRTCICEClass * klass) "Whether the agent should use ICE-UDP when gathering candidates", TRUE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + /** + * GstWebRTCICE:min-rtp-port: + * + * Minimum port for local rtp port range. + * min-rtp-port must be <= max-rtp-port + * + * Since: 1.20 + */ + g_object_class_install_property (gobject_class, + PROP_MIN_RTP_PORT, + g_param_spec_uint ("min-rtp-port", "ICE RTP candidate min port", + "Minimum port for local rtp port range. " + "min-rtp-port must be <= max-rtp-port", + 0, 65535, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + /** + * GstWebRTCICE:max-rtp-port: + * + * Maximum port for local rtp port range. + * min-rtp-port must be <= max-rtp-port + * + * Since: 1.20 + */ + g_object_class_install_property (gobject_class, + PROP_MAX_RTP_PORT, + g_param_spec_uint ("max-rtp-port", "ICE RTP candidate max port", + "Maximum port for local rtp port range. " + "max-rtp-port must be >= min-rtp-port", + 0, 65535, 65535, + G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); + /** * GstWebRTCICE::add-local-ip-address: * @object: the #GstWebRTCICE diff --git a/ext/webrtc/gstwebrtcice.h b/ext/webrtc/gstwebrtcice.h index 283672fe9..5e7543c80 100644 --- a/ext/webrtc/gstwebrtcice.h +++ b/ext/webrtc/gstwebrtcice.h @@ -51,6 +51,9 @@ struct _GstWebRTCICE GHashTable *turn_servers; GstWebRTCICEPrivate *priv; + + guint min_rtp_port; + guint max_rtp_port; }; struct _GstWebRTCICEClass diff --git a/ext/webrtc/icestream.c b/ext/webrtc/icestream.c index 36363f2c6..924c9cbd7 100644 --- a/ext/webrtc/icestream.c +++ b/ext/webrtc/icestream.c @@ -46,6 +46,7 @@ struct _GstWebRTCICEStreamPrivate { gboolean gathered; GList *transports; + gboolean gathering_started; }; #define gst_webrtc_ice_stream_parent_class parent_class @@ -187,6 +188,24 @@ gst_webrtc_ice_stream_gather_candidates (GstWebRTCICEStream * stream) } g_object_get (stream->ice, "agent", &agent, NULL); + + if (!stream->priv->gathering_started) { + if (stream->ice->min_rtp_port != 0 || stream->ice->max_rtp_port != 65535) { + if (stream->ice->min_rtp_port > stream->ice->max_rtp_port) { + GST_ERROR_OBJECT (stream->ice, + "invalid port range: min-rtp-port %d must be <= max-rtp-port %d", + stream->ice->min_rtp_port, stream->ice->max_rtp_port); + return FALSE; + } + + nice_agent_set_port_range (agent, stream->stream_id, + NICE_COMPONENT_TYPE_RTP, stream->ice->min_rtp_port, + stream->ice->max_rtp_port); + } + /* mark as gathering started to prevent changing ports again */ + stream->priv->gathering_started = TRUE; + } + if (!nice_agent_gather_candidates (agent, stream->stream_id)) { g_object_unref (agent); return FALSE; diff --git a/tests/check/elements/webrtcbin.c b/tests/check/elements/webrtcbin.c index 89a141cca..f6b46aa89 100644 --- a/tests/check/elements/webrtcbin.c +++ b/tests/check/elements/webrtcbin.c @@ -692,7 +692,6 @@ test_webrtc_wait_for_answer_error_eos (struct test_webrtc *t) test_webrtc_wait_for_state_mask (t, states); } -#if 0 static void test_webrtc_wait_for_ice_gathering_complete (struct test_webrtc *t) { @@ -709,6 +708,7 @@ test_webrtc_wait_for_ice_gathering_complete (struct test_webrtc *t) g_mutex_unlock (&t->lock); } +#if 0 static void test_webrtc_wait_for_ice_connection (struct test_webrtc *t, GstWebRTCICEConnectionState states) @@ -904,6 +904,78 @@ GST_START_TEST (test_audio) GST_END_TEST; +static void +_check_ice_port_restriction (struct test_webrtc *t, GstElement * element, + guint mlineindex, gchar * candidate, GstElement * other, gpointer user_data) +{ + GRegex *regex; + GMatchInfo *match_info; + + gchar *candidate_port; + gchar *candidate_protocol; + gchar *candidate_typ; + guint port_as_int; + guint peer_number; + + regex = + g_regex_new ("candidate:(\\d+) (1) (UDP|TCP) (\\d+) ([0-9.]+|[0-9a-f:]+)" + " (\\d+) typ ([a-z]+)", 0, 0, NULL); + + g_regex_match (regex, candidate, 0, &match_info); + fail_unless (g_match_info_get_match_count (match_info) == 8, candidate); + + candidate_protocol = g_match_info_fetch (match_info, 2); + candidate_port = g_match_info_fetch (match_info, 6); + candidate_typ = g_match_info_fetch (match_info, 7); + + peer_number = t->webrtc1 == element ? 1 : 2; + + port_as_int = atoi (candidate_port); + + if (!g_strcmp0 (candidate_typ, "host") && port_as_int != 9) { + guint expected_min = peer_number * 10000 + 1000; + guint expected_max = expected_min + 999; + + fail_unless (port_as_int >= expected_min); + fail_unless (port_as_int <= expected_max); + } + + g_free (candidate_port); + g_free (candidate_protocol); + g_free (candidate_typ); + g_match_info_free (match_info); + g_regex_unref (regex); +} + +GST_START_TEST (test_ice_port_restriction) +{ + struct test_webrtc *t = create_audio_test (); + GObject *webrtcice; + + VAL_SDP_INIT (offer, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL); + VAL_SDP_INIT (answer, _count_num_sdp_media, GUINT_TO_POINTER (1), NULL); + + /* + * Ports are defined as follows "{peer}{protocol}000" + * - peer number: "1" for t->webrtc1, "2" for t->webrtc2 + */ + g_object_get (t->webrtc1, "ice-agent", &webrtcice, NULL); + g_object_set (webrtcice, "min-rtp-port", 11000, "max-rtp-port", 11999, NULL); + g_object_unref (webrtcice); + + g_object_get (t->webrtc2, "ice-agent", &webrtcice, NULL); + g_object_set (webrtcice, "min-rtp-port", 21000, "max-rtp-port", 21999, NULL); + g_object_unref (webrtcice); + + t->on_ice_candidate = _check_ice_port_restriction; + test_validate_sdp (t, &offer, &answer); + + test_webrtc_wait_for_ice_gathering_complete (t); + test_webrtc_free (t); +} + +GST_END_TEST; + static struct test_webrtc * create_audio_video_test (void) { @@ -2896,6 +2968,7 @@ webrtcbin_suite (void) tcase_add_test (tc, test_sdp_no_media); tcase_add_test (tc, test_session_stats); tcase_add_test (tc, test_audio); + tcase_add_test (tc, test_ice_port_restriction); tcase_add_test (tc, test_audio_video); tcase_add_test (tc, test_media_direction); tcase_add_test (tc, test_media_setup); -- cgit v1.2.1