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-rw-r--r--apps/app_confbridge.c1
-rw-r--r--apps/confbridge/conf_config_parser.c13
-rw-r--r--apps/confbridge/include/confbridge.h1
-rw-r--r--bridges/bridge_softmix.c8
-rw-r--r--bridges/bridge_softmix/include/bridge_softmix_internal.h2
-rw-r--r--configs/samples/confbridge.conf.sample6
-rw-r--r--include/asterisk/bridge.h9
-rw-r--r--include/asterisk/frame.h2
-rw-r--r--main/bridge.c7
-rw-r--r--main/rtp_engine.c2
-rw-r--r--res/res_pjsip_session.c6
-rw-r--r--res/res_rtp_asterisk.c60
12 files changed, 108 insertions, 9 deletions
diff --git a/apps/app_confbridge.c b/apps/app_confbridge.c
index c6372fa2f..b2d612df3 100644
--- a/apps/app_confbridge.c
+++ b/apps/app_confbridge.c
@@ -1485,6 +1485,7 @@ static struct confbridge_conference *join_conference_bridge(const char *conferen
ast_bridge_set_talker_src_video_mode(conference->bridge);
} else if (ast_test_flag(&conference->b_profile, BRIDGE_OPT_VIDEO_SRC_SFU)) {
ast_bridge_set_sfu_video_mode(conference->bridge);
+ ast_bridge_set_video_update_discard(conference->bridge, conference->b_profile.video_update_discard);
}
/* Link it into the conference bridges container */
diff --git a/apps/confbridge/conf_config_parser.c b/apps/confbridge/conf_config_parser.c
index cc8fcfe5d..bfd9f4f56 100644
--- a/apps/confbridge/conf_config_parser.c
+++ b/apps/confbridge/conf_config_parser.c
@@ -450,6 +450,16 @@
</enumlist>
</description>
</configOption>
+ <configOption name="video_update_discard" default="2000">
+ <synopsis>Sets the amount of time in milliseconds after sending a video update to discard subsequent video updates</synopsis>
+ <description><para>
+ Sets the amount of time in milliseconds after sending a video update request
+ that subsequent video updates should be discarded. This means that if we
+ send a video update we will discard any other video update requests until
+ after the configured amount of time has elapsed. This prevents flooding of
+ video update requests from clients.
+ </para></description>
+ </configOption>
<configOption name="template">
<synopsis>When using the CONFBRIDGE dialplan function, use a bridge profile as a template for creating a new temporary profile</synopsis>
</configOption>
@@ -1652,6 +1662,8 @@ static char *handle_cli_confbridge_show_bridge_profile(struct ast_cli_entry *e,
break;
}
+ ast_cli(a->fd,"Video Update Discard: %u\n", b_profile.video_update_discard);
+
ast_cli(a->fd,"sound_only_person: %s\n", conf_get_sound(CONF_SOUND_ONLY_PERSON, b_profile.sounds));
ast_cli(a->fd,"sound_only_one: %s\n", conf_get_sound(CONF_SOUND_ONLY_ONE, b_profile.sounds));
ast_cli(a->fd,"sound_has_joined: %s\n", conf_get_sound(CONF_SOUND_HAS_JOINED, b_profile.sounds));
@@ -2220,6 +2232,7 @@ int conf_load_config(void)
aco_option_register(&cfg_info, "regcontext", ACO_EXACT, bridge_types, NULL, OPT_CHAR_ARRAY_T, 0, CHARFLDSET(struct bridge_profile, regcontext));
aco_option_register(&cfg_info, "language", ACO_EXACT, bridge_types, "en", OPT_CHAR_ARRAY_T, 0, CHARFLDSET(struct bridge_profile, language));
aco_option_register_custom(&cfg_info, "^sound_", ACO_REGEX, bridge_types, NULL, sound_option_handler, 0);
+ aco_option_register(&cfg_info, "video_update_discard", ACO_EXACT, bridge_types, "2000", OPT_UINT_T, 0, FLDSET(struct bridge_profile, video_update_discard));
/* This option should only be used with the CONFBRIDGE dialplan function */
aco_option_register_custom(&cfg_info, "template", ACO_EXACT, bridge_types, NULL, bridge_template_handler, 0);
diff --git a/apps/confbridge/include/confbridge.h b/apps/confbridge/include/confbridge.h
index cf30d5c62..adf9b867d 100644
--- a/apps/confbridge/include/confbridge.h
+++ b/apps/confbridge/include/confbridge.h
@@ -218,6 +218,7 @@ struct bridge_profile {
unsigned int mix_interval; /*!< The internal mixing interval used by the bridge. When set to 0 the bridgewill use a default interval. */
struct bridge_profile_sounds *sounds;
char regcontext[AST_MAX_CONTEXT];
+ unsigned int video_update_discard; /*!< Amount of time after sending a video update request that subsequent requests should be discarded */
};
/*! \brief The structure that represents a conference bridge */
diff --git a/bridges/bridge_softmix.c b/bridges/bridge_softmix.c
index f3b5b2aa9..c5428a854 100644
--- a/bridges/bridge_softmix.c
+++ b/bridges/bridge_softmix.c
@@ -985,6 +985,8 @@ static void softmix_bridge_write_voice(struct ast_bridge *bridge, struct ast_bri
*/
static int softmix_bridge_write_control(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame)
{
+ struct softmix_bridge_data *softmix_data = bridge->tech_pvt;
+
/*
* XXX Softmix needs to use channel roles to determine what to
* do with control frames.
@@ -992,7 +994,11 @@ static int softmix_bridge_write_control(struct ast_bridge *bridge, struct ast_br
switch (frame->subclass.integer) {
case AST_CONTROL_VIDUPDATE:
- ast_bridge_queue_everyone_else(bridge, NULL, frame);
+ if (!bridge->softmix.video_mode.video_update_discard ||
+ ast_tvdiff_ms(ast_tvnow(), softmix_data->last_video_update) > bridge->softmix.video_mode.video_update_discard) {
+ ast_bridge_queue_everyone_else(bridge, NULL, frame);
+ softmix_data->last_video_update = ast_tvnow();
+ }
break;
default:
break;
diff --git a/bridges/bridge_softmix/include/bridge_softmix_internal.h b/bridges/bridge_softmix/include/bridge_softmix_internal.h
index 9daae4ce8..f93e66391 100644
--- a/bridges/bridge_softmix/include/bridge_softmix_internal.h
+++ b/bridges/bridge_softmix/include/bridge_softmix_internal.h
@@ -198,6 +198,8 @@ struct softmix_bridge_data {
* (does not guarantee success)
*/
unsigned int binaural_init;
+ /*! The last time a video update was sent into the bridge */
+ struct timeval last_video_update;
};
struct softmix_mixing_array {
diff --git a/configs/samples/confbridge.conf.sample b/configs/samples/confbridge.conf.sample
index 0e07f6b16..265b95342 100644
--- a/configs/samples/confbridge.conf.sample
+++ b/configs/samples/confbridge.conf.sample
@@ -218,6 +218,12 @@ type=bridge
; Default is en (English).
;regcontext=conferences ; The name of the context into which to register conference names as extensions.
+;video_update_discard=2000 ; Amount of time (in milliseconds) to discard video update requests after sending a video
+ ; update request. Default is 2000. A video update request is a request for a full video
+ ; intra-frame. Clients can request this if they require a full frame in order to decode
+ ; the video stream. Since a full frame can be large limiting how often they occur can
+ ; reduce bandwidth usage at the cost of increasing how long it may take a newly joined
+ ; channel to receive the video stream.
; All sounds in the conference are customizable using the bridge profile options below.
; Simply state the option followed by the filename or full path of the filename after
diff --git a/include/asterisk/bridge.h b/include/asterisk/bridge.h
index bc0e9c81e..8d5c50211 100644
--- a/include/asterisk/bridge.h
+++ b/include/asterisk/bridge.h
@@ -134,6 +134,7 @@ struct ast_bridge_video_mode {
struct ast_bridge_video_single_src_data single_src_data;
struct ast_bridge_video_talker_src_data talker_src_data;
} mode_data;
+ unsigned int video_update_discard;
};
/*!
@@ -903,6 +904,14 @@ void ast_bridge_set_talker_src_video_mode(struct ast_bridge *bridge);
void ast_bridge_set_sfu_video_mode(struct ast_bridge *bridge);
/*!
+ * \brief Set the amount of time to discard subsequent video updates after a video update has been sent
+ *
+ * \param bridge Bridge to set the minimum video update wait time on
+ * \param video_update_discard Amount of time after sending a video update that others should be discarded
+ */
+void ast_bridge_set_video_update_discard(struct ast_bridge *bridge, unsigned int video_update_discard);
+
+/*!
* \brief Update information about talker energy for talker src video mode.
*/
void ast_bridge_update_talker_src_video_mode(struct ast_bridge *bridge, struct ast_channel *chan, int talker_energy, int is_keyfame);
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index 2f6c365ad..8f0daccb7 100644
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -137,6 +137,8 @@ enum {
AST_FRFLAG_HAS_TIMING_INFO = (1 << 0),
/*! This frame has been requeued */
AST_FRFLAG_REQUEUED = (1 << 1),
+ /*! This frame contains a valid sequence number */
+ AST_FRFLAG_HAS_SEQUENCE_NUMBER = (1 << 2),
};
struct ast_frame_subclass {
diff --git a/main/bridge.c b/main/bridge.c
index 3a358d9f8..a1a1a6f55 100644
--- a/main/bridge.c
+++ b/main/bridge.c
@@ -3816,6 +3816,13 @@ void ast_bridge_set_sfu_video_mode(struct ast_bridge *bridge)
ast_bridge_unlock(bridge);
}
+void ast_bridge_set_video_update_discard(struct ast_bridge *bridge, unsigned int video_update_discard)
+{
+ ast_bridge_lock(bridge);
+ bridge->softmix.video_mode.video_update_discard = video_update_discard;
+ ast_bridge_unlock(bridge);
+}
+
void ast_bridge_update_talker_src_video_mode(struct ast_bridge *bridge, struct ast_channel *chan, int talker_energy, int is_keyframe)
{
struct ast_bridge_video_talker_src_data *data;
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index fe60c4eae..e078b2400 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -3386,7 +3386,7 @@ int ast_rtp_instance_bundle(struct ast_rtp_instance *child, struct ast_rtp_insta
{
int res = -1;
- if (child->engine != parent->engine) {
+ if (parent && (child->engine != parent->engine)) {
return -1;
}
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index fe3680f3b..0ad2c8f30 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -785,6 +785,12 @@ static int handle_negotiated_sdp(struct ast_sip_session *session, const pjmedia_
* we remove it as a result of the stream limit being reached.
*/
if (ast_stream_get_state(stream) == AST_STREAM_STATE_REMOVED) {
+ /* This stream is no longer being used so release any resources the handler
+ * may have on it.
+ */
+ if (session_media->handler) {
+ session_media_set_handler(session_media, NULL);
+ }
continue;
}
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index a2e63ec0b..70561d0b6 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -266,7 +266,8 @@ struct ast_rtp {
unsigned int lastitexttimestamp;
unsigned int lastotexttimestamp;
unsigned int lasteventseqn;
- int lastrxseqno; /*!< Last received sequence number */
+ int lastrxseqno; /*!< Last received sequence number, from the network */
+ int expectedseqno; /*!< Next expected sequence number, from the core */
unsigned short seedrxseqno; /*!< What sequence number did they start with?*/
unsigned int seedrxts; /*!< What RTP timestamp did they start with? */
unsigned int rxcount; /*!< How many packets have we received? */
@@ -3245,6 +3246,7 @@ static int ast_rtp_new(struct ast_rtp_instance *instance,
rtp->ssrc = ast_random();
ast_uuid_generate_str(rtp->cname, sizeof(rtp->cname));
rtp->seqno = ast_random() & 0x7fff;
+ rtp->expectedseqno = -1;
rtp->sched = sched;
ast_sockaddr_copy(&rtp->bind_address, addr);
@@ -3274,7 +3276,7 @@ static int ast_rtp_new(struct ast_rtp_instance *instance,
* \return 0 if element does not match.
* \return Non-zero if element matches.
*/
-#define SSRC_MAPPING_ELEM_CMP(elem, value) ((elem).ssrc == (value))
+#define SSRC_MAPPING_ELEM_CMP(elem, value) (elem.instance == value)
/*! \pre instance is locked */
static int ast_rtp_destroy(struct ast_rtp_instance *instance)
@@ -3289,7 +3291,7 @@ static int ast_rtp_destroy(struct ast_rtp_instance *instance)
ao2_lock(rtp->bundled);
bundled_rtp = ast_rtp_instance_get_data(rtp->bundled);
- AST_VECTOR_REMOVE_CMP_UNORDERED(&bundled_rtp->ssrc_mapping, rtp->themssrc, SSRC_MAPPING_ELEM_CMP, AST_VECTOR_ELEM_CLEANUP_NOOP);
+ AST_VECTOR_REMOVE_CMP_UNORDERED(&bundled_rtp->ssrc_mapping, instance, SSRC_MAPPING_ELEM_CMP, AST_VECTOR_ELEM_CLEANUP_NOOP);
ao2_unlock(rtp->bundled);
ao2_lock(instance);
@@ -3897,6 +3899,7 @@ static int rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *fr
unsigned int ms = calc_txstamp(rtp, &frame->delivery);
struct ast_sockaddr remote_address = { {0,} };
int rate = rtp_get_rate(frame->subclass.format) / 1000;
+ unsigned int seqno;
if (ast_format_cmp(frame->subclass.format, ast_format_g722) == AST_FORMAT_CMP_EQUAL) {
frame->samples /= 2;
@@ -3963,6 +3966,40 @@ static int rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *fr
rtp->lastdigitts = rtp->lastts;
}
+ /* Assume that the sequence number we expect to use is what will be used until proven otherwise */
+ seqno = rtp->seqno;
+
+ /* If the frame contains sequence number information use it to influence our sequence number */
+ if (ast_test_flag(frame, AST_FRFLAG_HAS_SEQUENCE_NUMBER)) {
+ if (rtp->expectedseqno != -1) {
+ /* Determine where the frame from the core is in relation to where we expected */
+ int difference = frame->seqno - rtp->expectedseqno;
+
+ /* If there is a substantial difference then we've either got packets really out
+ * of order, or the source is RTP and it has cycled. If this happens we resync
+ * the sequence number adjustments to this frame. If we also have packet loss
+ * things won't be reflected correctly but it will sort itself out after a bit.
+ */
+ if (abs(difference) > 100) {
+ difference = 0;
+ }
+
+ /* Adjust the sequence number being used for this packet accordingly */
+ seqno += difference;
+
+ if (difference >= 0) {
+ /* This frame is on time or in the future */
+ rtp->expectedseqno = frame->seqno + 1;
+ rtp->seqno += difference;
+ }
+ } else {
+ /* This is the first frame with sequence number we've seen, so start keeping track */
+ rtp->expectedseqno = frame->seqno + 1;
+ }
+ } else {
+ rtp->expectedseqno = -1;
+ }
+
if (ast_test_flag(frame, AST_FRFLAG_HAS_TIMING_INFO)) {
rtp->lastts = frame->ts * rate;
}
@@ -3974,7 +4011,7 @@ static int rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *fr
int hdrlen = 12, res, ice;
unsigned char *rtpheader = (unsigned char *)(frame->data.ptr - hdrlen);
- put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (rtp->seqno) | (mark << 23)));
+ put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (seqno) | (mark << 23)));
put_unaligned_uint32(rtpheader + 4, htonl(rtp->lastts));
put_unaligned_uint32(rtpheader + 8, htonl(rtp->ssrc));
@@ -4011,7 +4048,13 @@ static int rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *fr
}
}
- rtp->seqno++;
+ /* If the sequence number that has been used doesn't match what we expected then this is an out of
+ * order late packet, so we don't need to increment as we haven't yet gotten the expected frame from
+ * the core.
+ */
+ if (seqno == rtp->seqno) {
+ rtp->seqno++;
+ }
return 0;
}
@@ -5474,6 +5517,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
rtp->f.datalen = res - hdrlen;
rtp->f.data.ptr = read_area + hdrlen;
rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
+ ast_set_flag(&rtp->f, AST_FRFLAG_HAS_SEQUENCE_NUMBER);
rtp->f.seqno = seqno;
rtp->f.stream_num = rtp->stream_num;
@@ -6082,7 +6126,7 @@ static void ast_rtp_set_stream_num(struct ast_rtp_instance *instance, int stream
static int ast_rtp_bundle(struct ast_rtp_instance *child, struct ast_rtp_instance *parent)
{
struct ast_rtp *child_rtp = ast_rtp_instance_get_data(child);
- struct ast_rtp *parent_rtp = ast_rtp_instance_get_data(parent);
+ struct ast_rtp *parent_rtp;
struct rtp_ssrc_mapping mapping;
struct ast_sockaddr them = { { 0, } };
@@ -6099,7 +6143,7 @@ static int ast_rtp_bundle(struct ast_rtp_instance *child, struct ast_rtp_instanc
/* The child lock can't be held while accessing the parent */
ao2_lock(child_rtp->bundled);
bundled_rtp = ast_rtp_instance_get_data(child_rtp->bundled);
- AST_VECTOR_REMOVE_CMP_UNORDERED(&bundled_rtp->ssrc_mapping, child_rtp->themssrc, SSRC_MAPPING_ELEM_CMP, AST_VECTOR_ELEM_CLEANUP_NOOP);
+ AST_VECTOR_REMOVE_CMP_UNORDERED(&bundled_rtp->ssrc_mapping, child, SSRC_MAPPING_ELEM_CMP, AST_VECTOR_ELEM_CLEANUP_NOOP);
ao2_unlock(child_rtp->bundled);
ao2_lock(child);
@@ -6113,6 +6157,8 @@ static int ast_rtp_bundle(struct ast_rtp_instance *child, struct ast_rtp_instanc
return 0;
}
+ parent_rtp = ast_rtp_instance_get_data(parent);
+
/* We no longer need any transport related resources as we will use our parent RTP instance instead */
rtp_deallocate_transport(child, child_rtp);
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-03 14:45:21 +0000