[Asterisk-code-review] rtp: Add support for transport-cc in receiver direction. (...asterisk[16])
Friendly Automation
asteriskteam at digium.com
Fri May 3 10:08:17 CDT 2019
Friendly Automation has submitted this change and it was merged. ( https://gerrit.asterisk.org/c/asterisk/+/11315 )
Change subject: rtp: Add support for transport-cc in receiver direction.
......................................................................
rtp: Add support for transport-cc in receiver direction.
The transport-cc draft is a mechanism by which additional information
about packet reception can be provided to the sender of packets so
they can do sender side bandwidth estimation. This is accomplished
by having a transport specific sequence number and an RTCP feedback
message. This change implements this in the receiver direction.
For each received RTP packet where transport-cc is negotiated we store
the time at which the RTP packet was received and its sequence number.
At a 1 second interval we go through all packets in that period of time
and use the stored time of each in comparison to its preceding packet to
calculate its delta. This delta information is placed in the RTCP
feedback message, along with indicators for any packets which were not
received.
The browser then uses this information to better estimate available
bandwidth and adjust accordingly. This may result in it lowering the
available send bandwidth or adjusting how "bursty" it can be.
ASTERISK-28400
Change-Id: I654a2cff5bd5554ab94457a14f70adb71f574afc
---
M include/asterisk/rtp_engine.h
M main/rtp_engine.c
M res/res_pjsip_sdp_rtp.c
M res/res_rtp_asterisk.c
4 files changed, 458 insertions(+), 7 deletions(-)
Approvals:
Benjamin Keith Ford: Looks good to me, but someone else must approve
Kevin Harwell: Looks good to me, approved
Friendly Automation: Approved for Submit
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index 57aaefe..206ed63 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -306,6 +306,8 @@
#define AST_RTP_RTCP_FMT_FIR 4
/*! REMB Information (From draft-alvestrand-rmcat-remb-03) */
#define AST_RTP_RTCP_FMT_REMB 15
+/*! Transport-wide congestion control feedback (From draft-holmer-rmcat-transport-wide-cc-extensions-01) */
+#define AST_RTP_RTCP_FMT_TRANSPORT_WIDE_CC 15
/*!
* \since 12
@@ -541,6 +543,8 @@
AST_RTP_EXTENSION_UNSUPPORTED = 0,
/*! abs-send-time from https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 */
AST_RTP_EXTENSION_ABS_SEND_TIME,
+ /*! transport-cc from https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01 */
+ AST_RTP_EXTENSION_TRANSPORT_WIDE_CC,
/*! The maximum number of known RTP extensions */
AST_RTP_EXTENSION_MAX,
};
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 90ade04..f409bc2 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -235,6 +235,7 @@
static const char * const rtp_extension_uris[AST_RTP_EXTENSION_MAX] = {
[AST_RTP_EXTENSION_UNSUPPORTED] = "",
[AST_RTP_EXTENSION_ABS_SEND_TIME] = "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time",
+ [AST_RTP_EXTENSION_TRANSPORT_WIDE_CC] = "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
};
/*! List of RTP engines that are currently registered */
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index cba494a..80ac253 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -274,6 +274,7 @@
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_REMB, session->endpoint->media.webrtc);
if (session->endpoint->media.webrtc) {
enable_rtp_extension(session, session_media, AST_RTP_EXTENSION_ABS_SEND_TIME, AST_RTP_EXTENSION_DIRECTION_SENDRECV, sdp);
+ enable_rtp_extension(session, session_media, AST_RTP_EXTENSION_TRANSPORT_WIDE_CC, AST_RTP_EXTENSION_DIRECTION_SENDRECV, sdp);
}
if (session->endpoint->media.tos_video || session->endpoint->media.cos_video) {
ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_video,
@@ -1184,7 +1185,18 @@
pj_str_t stmp;
pjmedia_sdp_attr *attr;
- if (!session->endpoint->media.webrtc || session_media->type != AST_MEDIA_TYPE_VIDEO) {
+ if (!session->endpoint->media.webrtc) {
+ return;
+ }
+
+ /* transport-cc is supposed to be for the entire transport, and any media sources so
+ * while the header does not appear in audio streams and isn't negotiated there, we still
+ * place this attribute in as Chrome does.
+ */
+ attr = pjmedia_sdp_attr_create(pool, "rtcp-fb", pj_cstr(&stmp, "* transport-cc"));
+ pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
+
+ if (session_media->type != AST_MEDIA_TYPE_VIDEO) {
return;
}
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 80f3d06..023273a 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -312,6 +312,32 @@
struct ast_rtp_instance *instance;
};
+/*! \brief Packet statistics (used for transport-cc) */
+struct rtp_transport_wide_cc_packet_statistics {
+ /*! The transport specific sequence number */
+ unsigned int seqno;
+ /*! The time at which the packet was received */
+ struct timeval received;
+ /*! The delta between this packet and the previous */
+ int delta;
+};
+
+/*! \brief Statistics information (used for transport-cc) */
+struct rtp_transport_wide_cc_statistics {
+ /*! A vector of packet statistics */
+ AST_VECTOR(, struct rtp_transport_wide_cc_packet_statistics) packet_statistics; /*!< Packet statistics, used for transport-cc */
+ /*! The last sequence number received */
+ unsigned int last_seqno;
+ /*! The last extended sequence number */
+ unsigned int last_extended_seqno;
+ /*! How many feedback packets have gone out */
+ unsigned int feedback_count;
+ /*! How many cycles have occurred for the sequence numbers */
+ unsigned int cycles;
+ /*! Scheduler id for periodic feedback transmission */
+ int schedid;
+};
+
/*! \brief RTP session description */
struct ast_rtp {
int s;
@@ -387,6 +413,8 @@
struct ast_data_buffer *send_buffer; /*!< Buffer for storing sent packets for retransmission */
struct ast_data_buffer *recv_buffer; /*!< Buffer for storing received packets for retransmission */
+ struct rtp_transport_wide_cc_statistics transport_wide_cc; /*!< Transport-cc statistics information */
+
#ifdef HAVE_PJPROJECT
ast_cond_t cond; /*!< ICE/TURN condition for signaling */
@@ -3676,6 +3704,11 @@
return -1;
}
+ if (AST_VECTOR_INIT(&rtp->transport_wide_cc.packet_statistics, 0)) {
+ return -1;
+ }
+ rtp->transport_wide_cc.schedid = -1;
+
rtp->f.subclass.format = ao2_bump(ast_format_none);
rtp->lastrxformat = ao2_bump(ast_format_none);
rtp->lasttxformat = ao2_bump(ast_format_none);
@@ -3752,6 +3785,8 @@
ast_data_buffer_free(rtp->recv_buffer);
}
+ AST_VECTOR_FREE(&rtp->transport_wide_cc.packet_statistics);
+
ao2_cleanup(rtp->lasttxformat);
ao2_cleanup(rtp->lastrxformat);
ao2_cleanup(rtp->f.subclass.format);
@@ -6308,6 +6343,377 @@
}
}
+static int rtp_transport_wide_cc_packet_statistics_cmp(struct rtp_transport_wide_cc_packet_statistics a,
+ struct rtp_transport_wide_cc_packet_statistics b)
+{
+ return a.seqno - b.seqno;
+}
+
+static void rtp_transport_wide_cc_feedback_status_vector_append(unsigned char *rtcpheader, int *packet_len, int *status_vector_chunk_bits,
+ uint16_t *status_vector_chunk, int status)
+{
+ /* Appending this status will use up 2 bits */
+ *status_vector_chunk_bits -= 2;
+
+ /* We calculate which bits we want to update the status of. Since a status vector
+ * is 16 bits we take away 2 (for the header), and then we take away any that have
+ * already been used.
+ */
+ *status_vector_chunk |= (status << (16 - 2 - (14 - *status_vector_chunk_bits)));
+
+ /* If there are still bits available we can return early */
+ if (*status_vector_chunk_bits) {
+ return;
+ }
+
+ /* Otherwise we have to place this chunk into the packet */
+ put_unaligned_uint16(rtcpheader + *packet_len, htons(*status_vector_chunk));
+ *status_vector_chunk_bits = 14;
+
+ /* The first bit being 1 indicates that this is a status vector chunk and the second
+ * bit being 1 indicates that we are using 2 bits to represent each status for a
+ * packet.
+ */
+ *status_vector_chunk = (1 << 15) | (1 << 14);
+ *packet_len += 2;
+}
+
+static void rtp_transport_wide_cc_feedback_status_append(unsigned char *rtcpheader, int *packet_len, int *status_vector_chunk_bits,
+ uint16_t *status_vector_chunk, int *run_length_chunk_count, int *run_length_chunk_status, int status)
+{
+ if (*run_length_chunk_status != status) {
+ while (*run_length_chunk_count > 0 && *run_length_chunk_count < 8) {
+ /* Realistically it only makes sense to use a run length chunk if there were 8 or more
+ * consecutive packets of the same type, otherwise we could end up making the packet larger
+ * if we have lots of small blocks of the same type. To help with this we backfill the status
+ * vector (since it always represents 7 packets). Best case we end up with only that single
+ * status vector and the rest are run length chunks.
+ */
+ rtp_transport_wide_cc_feedback_status_vector_append(rtcpheader, packet_len, status_vector_chunk_bits,
+ status_vector_chunk, *run_length_chunk_status);
+ *run_length_chunk_count -= 1;
+ }
+
+ if (*run_length_chunk_count) {
+ /* There is a run length chunk which needs to be written out */
+ put_unaligned_uint16(rtcpheader + *packet_len, htons((0 << 15) | (*run_length_chunk_status << 13) | *run_length_chunk_count));
+ *packet_len += 2;
+ }
+
+ /* In all cases the run length chunk has to be reset */
+ *run_length_chunk_count = 0;
+ *run_length_chunk_status = -1;
+
+ if (*status_vector_chunk_bits == 14) {
+ /* We aren't in the middle of a status vector so we can try for a run length chunk */
+ *run_length_chunk_status = status;
+ *run_length_chunk_count = 1;
+ } else {
+ /* We're doing a status vector so populate it accordingly */
+ rtp_transport_wide_cc_feedback_status_vector_append(rtcpheader, packet_len, status_vector_chunk_bits,
+ status_vector_chunk, status);
+ }
+ } else {
+ /* This is easy, the run length chunk count can just get bumped up */
+ *run_length_chunk_count += 1;
+ }
+}
+
+static int rtp_transport_wide_cc_feedback_produce(const void *data)
+{
+ struct ast_rtp_instance *instance = (struct ast_rtp_instance *) data;
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ unsigned char *rtcpheader;
+ char bdata[1024];
+ struct rtp_transport_wide_cc_packet_statistics *first_packet;
+ struct rtp_transport_wide_cc_packet_statistics *previous_packet;
+ int i;
+ int status_vector_chunk_bits = 14;
+ uint16_t status_vector_chunk = (1 << 15) | (1 << 14);
+ int run_length_chunk_count = 0;
+ int run_length_chunk_status = -1;
+ int packet_len = 20;
+ int delta_len = 0;
+ int packet_count = 0;
+ unsigned int received_msw;
+ unsigned int received_lsw;
+ struct ast_sockaddr remote_address = { { 0, } };
+ int res;
+ int ice;
+ unsigned int large_delta_count = 0;
+ unsigned int small_delta_count = 0;
+ unsigned int lost_count = 0;
+
+ if (!rtp || !rtp->rtcp || rtp->transport_wide_cc.schedid == -1) {
+ ao2_ref(instance, -1);
+ return 0;
+ }
+
+ ao2_lock(instance);
+
+ rtcpheader = (unsigned char *)bdata;
+
+ /* The first packet in the vector acts as our base sequence number and reference time */
+ first_packet = AST_VECTOR_GET_ADDR(&rtp->transport_wide_cc.packet_statistics, 0);
+ previous_packet = first_packet;
+
+ /* We go through each packet that we have statistics for, adding it either to a status
+ * vector chunk or a run length chunk. The code tries to be as efficient as possible to
+ * reduce packet size and will favor run length chunks when it makes sense.
+ */
+ for (i = 0; i < AST_VECTOR_SIZE(&rtp->transport_wide_cc.packet_statistics); ++i) {
+ struct rtp_transport_wide_cc_packet_statistics *statistics;
+ int lost = 0;
+ int res = 0;
+
+ statistics = AST_VECTOR_GET_ADDR(&rtp->transport_wide_cc.packet_statistics, i);
+
+ packet_count++;
+
+ if (first_packet != statistics) {
+ /* The vector stores statistics in a sorted fashion based on the sequence
+ * number. This ensures we can detect any packets that have been lost/not
+ * received by comparing the sequence numbers.
+ */
+ lost = statistics->seqno - (previous_packet->seqno + 1);
+ lost_count += lost;
+ }
+
+ while (lost) {
+ /* We append a not received status until all the lost packets have been accounted for */
+ rtp_transport_wide_cc_feedback_status_append(rtcpheader, &packet_len, &status_vector_chunk_bits,
+ &status_vector_chunk, &run_length_chunk_count, &run_length_chunk_status, 0);
+ packet_count++;
+
+ /* If there is no more room left for storing packets stop now, we leave 20
+ * extra bits at the end just in case.
+ */
+ if ((sizeof(bdata) - (packet_len + delta_len + 20)) < 0) {
+ res = -1;
+ break;
+ }
+
+ lost--;
+ }
+
+ /* If the lost packet appending bailed out because we have no more space, then exit here too */
+ if (res) {
+ break;
+ }
+
+ /* Per the spec the delta is in increments of 250 */
+ statistics->delta = ast_tvdiff_us(statistics->received, previous_packet->received) / 250;
+
+ /* Based on the delta determine the status of this packet */
+ if (statistics->delta < 0 || statistics->delta > 127) {
+ /* Large or negative delta */
+ rtp_transport_wide_cc_feedback_status_append(rtcpheader, &packet_len, &status_vector_chunk_bits,
+ &status_vector_chunk, &run_length_chunk_count, &run_length_chunk_status, 2);
+ delta_len += 2;
+ large_delta_count++;
+ } else {
+ /* Small delta */
+ rtp_transport_wide_cc_feedback_status_append(rtcpheader, &packet_len, &status_vector_chunk_bits,
+ &status_vector_chunk, &run_length_chunk_count, &run_length_chunk_status, 1);
+ delta_len += 1;
+ small_delta_count++;
+ }
+
+ previous_packet = statistics;
+
+ /* If there is no more room left in the packet stop handling of any subsequent packets */
+ if ((sizeof(bdata) - (packet_len + delta_len + 20)) < 0) {
+ break;
+ }
+ }
+
+ if (status_vector_chunk_bits != 14) {
+ /* If the status vector chunk has packets in it then place it in the RTCP packet */
+ put_unaligned_uint16(rtcpheader + packet_len, htons(status_vector_chunk));
+ packet_len += 2;
+ } else if (run_length_chunk_count) {
+ /* If there is a run length chunk in progress then place it in the RTCP packet */
+ put_unaligned_uint16(rtcpheader + packet_len, htons((0 << 15) | (run_length_chunk_status << 13) | run_length_chunk_count));
+ packet_len += 2;
+ }
+
+ /* We iterate again to build delta chunks */
+ for (i = 0; i < AST_VECTOR_SIZE(&rtp->transport_wide_cc.packet_statistics); ++i) {
+ struct rtp_transport_wide_cc_packet_statistics *statistics;
+
+ statistics = AST_VECTOR_GET_ADDR(&rtp->transport_wide_cc.packet_statistics, i);
+
+ if (statistics->delta < 0 || statistics->delta > 127) {
+ /* We need 2 bytes to store this delta */
+ put_unaligned_uint16(rtcpheader + packet_len, htons(statistics->delta));
+ packet_len += 2;
+ } else {
+ /* We can store this delta in 1 byte */
+ rtcpheader[packet_len] = statistics->delta;
+ packet_len += 1;
+ }
+
+ /* If this is the last packet handled by the run length chunk or status vector chunk code
+ * then we can go no further.
+ */
+ if (statistics == previous_packet) {
+ break;
+ }
+ }
+
+ /* Zero pad the end of the packet */
+ while (packet_len % 4) {
+ rtcpheader[packet_len++] = 0;
+ }
+
+ /* Add the general RTCP header information */
+ put_unaligned_uint32(rtcpheader, htonl((2 << 30) | (AST_RTP_RTCP_FMT_TRANSPORT_WIDE_CC << 24)
+ | (AST_RTP_RTCP_RTPFB << 16) | ((packet_len / 4) - 1)));
+ put_unaligned_uint32(rtcpheader + 4, htonl(rtp->ssrc));
+ put_unaligned_uint32(rtcpheader + 8, htonl(rtp->themssrc));
+
+ /* Add the transport-cc specific header information */
+ put_unaligned_uint32(rtcpheader + 12, htonl((first_packet->seqno << 16) | packet_count));
+
+ timeval2ntp(first_packet->received, &received_msw, &received_lsw);
+ put_unaligned_time24(rtcpheader + 16, received_msw, received_lsw);
+ rtcpheader[19] = rtp->transport_wide_cc.feedback_count;
+
+ /* The packet is now fully constructed so send it out */
+ ast_sockaddr_copy(&remote_address, &rtp->rtcp->them);
+
+ ast_debug(2, "Sending transport-cc feedback packet of size '%d' on '%s' with packet count of %d (small = %d, large = %d, lost = %d)\n",
+ packet_len, ast_rtp_instance_get_channel_id(instance), packet_count, small_delta_count, large_delta_count, lost_count);
+
+ res = rtcp_sendto(instance, (unsigned int *)rtcpheader, packet_len, 0, &remote_address, &ice);
+ if (res < 0) {
+ ast_log(LOG_ERROR, "RTCP transport-cc feedback error to %s due to %s\n",
+ ast_sockaddr_stringify(&remote_address), strerror(errno));
+ }
+
+ AST_VECTOR_RESET(&rtp->transport_wide_cc.packet_statistics, AST_VECTOR_ELEM_CLEANUP_NOOP);
+
+ rtp->transport_wide_cc.feedback_count++;
+
+ ao2_unlock(instance);
+
+ return 1000;
+}
+
+static void rtp_instance_parse_transport_wide_cc(struct ast_rtp_instance *instance, struct ast_rtp *rtp,
+ unsigned char *data, int len)
+{
+ uint16_t *seqno = (uint16_t *)data;
+ struct rtp_transport_wide_cc_packet_statistics statistics;
+ struct ast_rtp_instance *transport = rtp->bundled ? rtp->bundled : instance;
+ struct ast_rtp *transport_rtp = ast_rtp_instance_get_data(transport);
+
+ /* If the sequence number has cycled over then record it as such */
+ if (((int)transport_rtp->transport_wide_cc.last_seqno - (int)ntohs(*seqno)) > 100) {
+ transport_rtp->transport_wide_cc.cycles += RTP_SEQ_MOD;
+ }
+
+ /* Populate the statistics information for this packet */
+ statistics.seqno = transport_rtp->transport_wide_cc.cycles + ntohs(*seqno);
+ statistics.received = ast_tvnow();
+
+ /* We allow at a maximum 1000 packet statistics in play at a time, if we hit the
+ * limit we give up and start fresh.
+ */
+ if (AST_VECTOR_SIZE(&transport_rtp->transport_wide_cc.packet_statistics) > 1000) {
+ AST_VECTOR_RESET(&rtp->transport_wide_cc.packet_statistics, AST_VECTOR_ELEM_CLEANUP_NOOP);
+ }
+
+ if (!AST_VECTOR_SIZE(&transport_rtp->transport_wide_cc.packet_statistics) ||
+ statistics.seqno > transport_rtp->transport_wide_cc.last_extended_seqno) {
+ /* This is the expected path */
+ if (AST_VECTOR_APPEND(&transport_rtp->transport_wide_cc.packet_statistics, statistics)) {
+ return;
+ }
+
+ transport_rtp->transport_wide_cc.last_extended_seqno = statistics.seqno;
+ transport_rtp->transport_wide_cc.last_seqno = ntohs(*seqno);
+ } else {
+ /* This packet was out of order, so reorder it within the vector accordingly */
+ if (AST_VECTOR_ADD_SORTED(&transport_rtp->transport_wide_cc.packet_statistics, statistics,
+ rtp_transport_wide_cc_packet_statistics_cmp)) {
+ return;
+ }
+ }
+
+ /* If we have not yet scheduled the periodic sending of feedback for this transport then do so */
+ if (transport_rtp->transport_wide_cc.schedid < 0 && transport_rtp->rtcp) {
+ ast_debug(1, "Starting RTCP transport-cc feedback transmission on RTP instance '%p'\n", transport);
+ ao2_ref(transport, +1);
+ ast_log(LOG_NOTICE, "Starting feedback\n");
+ transport_rtp->transport_wide_cc.schedid = ast_sched_add(rtp->sched, 1000,
+ rtp_transport_wide_cc_feedback_produce, transport);
+ if (transport_rtp->transport_wide_cc.schedid < 0) {
+ ao2_ref(transport, -1);
+ ast_log(LOG_WARNING, "Scheduling RTCP transport-cc feedback transmission failed on RTP instance '%p'\n",
+ transport);
+ }
+ }
+}
+
+static void rtp_instance_parse_extmap_extensions(struct ast_rtp_instance *instance, struct ast_rtp *rtp,
+ unsigned char *extension, int len)
+{
+ int transport_wide_cc_id = ast_rtp_instance_extmap_get_id(instance, AST_RTP_EXTENSION_TRANSPORT_WIDE_CC);
+ int pos = 0;
+
+ /* We currently only care about the transport-cc extension, so if that's not negotiated then do nothing */
+ if (transport_wide_cc_id == -1) {
+ return;
+ }
+
+ /* Only while we do not exceed available extension data do we continue */
+ while (pos < len) {
+ int id = extension[pos] >> 4;
+ int extension_len = (extension[pos] & 0xF) + 1;
+
+ /* We've handled the first byte as it contains the extension id and length, so always
+ * skip ahead now
+ */
+ pos += 1;
+
+ if (id == 0) {
+ /* From the RFC:
+ * In both forms, padding bytes have the value of 0 (zero). They may be
+ * placed between extension elements, if desired for alignment, or after
+ * the last extension element, if needed for padding. A padding byte
+ * does not supply the ID of an element, nor the length field. When a
+ * padding byte is found, it is ignored and the parser moves on to
+ * interpreting the next byte.
+ */
+ continue;
+ } else if (id == 15) {
+ /* From the RFC:
+ * The local identifier value 15 is reserved for future extension and
+ * MUST NOT be used as an identifier. If the ID value 15 is
+ * encountered, its length field should be ignored, processing of the
+ * entire extension should terminate at that point, and only the
+ * extension elements present prior to the element with ID 15
+ * considered.
+ */
+ break;
+ } else if ((pos + extension_len) > len) {
+ /* The extension is corrupted and is stating that it contains more data than is
+ * available in the extensions data.
+ */
+ break;
+ }
+
+ /* If this is transport-cc then we need to parse it further */
+ if (id == transport_wide_cc_id) {
+ rtp_instance_parse_transport_wide_cc(instance, rtp, extension + pos, extension_len);
+ }
+
+ /* Skip ahead to the next extension */
+ pos += extension_len;
+ }
+}
+
static struct ast_frame *ast_rtp_interpret(struct ast_rtp_instance *instance, struct ast_srtp *srtp,
const struct ast_sockaddr *remote_address, unsigned char *read_area, int length, int prev_seqno)
{
@@ -6353,18 +6759,24 @@
/* Look for any RTP extensions, currently we do not support any */
if (ext) {
- hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2;
- hdrlen += 4;
- if (DEBUG_ATLEAST(1)) {
- unsigned int profile;
- profile = (ntohl(rtpheader[3]) & 0xffff0000) >> 16;
+ int extensions_size = (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2;
+ unsigned int profile;
+ profile = (ntohl(rtpheader[3]) & 0xffff0000) >> 16;
+
+ if (profile == 0xbede) {
+ /* We skip over the first 4 bytes as they are just for the one byte extension header */
+ rtp_instance_parse_extmap_extensions(instance, rtp, read_area + hdrlen + 4, extensions_size);
+ } else if (DEBUG_ATLEAST(1)) {
if (profile == 0x505a) {
ast_log(LOG_DEBUG, "Found Zfone extension in RTP stream - zrtp - not supported.\n");
- } else if (profile != 0xbede) {
+ } else {
/* SDP negotiated RTP extensions can not currently be output in logging */
ast_log(LOG_DEBUG, "Found unknown RTP Extensions %x\n", profile);
}
}
+
+ hdrlen += extensions_size;
+ hdrlen += 4;
}
/* Make sure after we potentially mucked with the header length that it is once again valid */
@@ -7316,6 +7728,18 @@
ao2_lock(instance);
rtp->rtcp->schedid = -1;
}
+ if (rtp->transport_wide_cc.schedid > -1) {
+ ao2_unlock(instance);
+ if (!ast_sched_del(rtp->sched, rtp->transport_wide_cc.schedid)) {
+ ao2_ref(instance, -1);
+ } else {
+ ast_debug(1, "Failed to tear down RTCP transport-cc feedback on RTP instance '%p'\n", instance);
+ ao2_lock(instance);
+ return;
+ }
+ ao2_lock(instance);
+ rtp->transport_wide_cc.schedid = -1;
+ }
if (rtp->rtcp->s > -1 && rtp->rtcp->s != rtp->s) {
close(rtp->rtcp->s);
}
@@ -7623,6 +8047,15 @@
rtp->rtcp->schedid = -1;
}
+ if (rtp->transport_wide_cc.schedid > -1) {
+ ao2_unlock(instance);
+ if (!ast_sched_del(rtp->sched, rtp->transport_wide_cc.schedid)) {
+ ao2_ref(instance, -1);
+ }
+ ao2_lock(instance);
+ rtp->transport_wide_cc.schedid = -1;
+ }
+
if (rtp->red) {
ao2_unlock(instance);
AST_SCHED_DEL(rtp->sched, rtp->red->schedid);
@@ -7763,6 +8196,7 @@
{
switch (extension) {
case AST_RTP_EXTENSION_ABS_SEND_TIME:
+ case AST_RTP_EXTENSION_TRANSPORT_WIDE_CC:
return 1;
default:
return 0;
--
To view, visit https://gerrit.asterisk.org/c/asterisk/+/11315
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Gerrit-Project: asterisk
Gerrit-Branch: 16
Gerrit-Change-Id: I654a2cff5bd5554ab94457a14f70adb71f574afc
Gerrit-Change-Number: 11315
Gerrit-PatchSet: 2
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Benjamin Keith Ford <bford at digium.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-MessageType: merged
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