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FS-10075: [freeswitch-core] WebRTC mods #resolve

This commit is contained in:
Anthony Minessale 2017-02-28 16:16:31 -06:00
parent 20bcb2edef
commit 50072f2ce2
3 changed files with 264 additions and 180 deletions
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4
src/include/switch_types.h
View File

@ -233,8 +233,8 @@ SWITCH_BEGIN_EXTERN_C
#define SWITCH_MAX_TRANS 2000
#define SWITCH_CORE_SESSION_MAX_PRIVATES 2
#define SWITCH_DEFAULT_VIDEO_SIZE 1200
#define SWITCH_RTCP_AUDIO_INTERVAL_MSEC "5000"
#define SWITCH_RTCP_VIDEO_INTERVAL_MSEC "2000"
#define SWITCH_RTCP_AUDIO_INTERVAL_MSEC "1000"
#define SWITCH_RTCP_VIDEO_INTERVAL_MSEC "1000"
#define TEXT_UNICODE_LINEFEED {0xe2, 0x80, 0xa8}
#define MAX_FMTP_LEN 256

66
src/switch_core_media.c
View File

@ -3931,6 +3931,8 @@ static switch_status_t check_ice(switch_media_handle_t *smh, switch_media_type_t
if (engine->ice_in.cands[engine->ice_in.chosen[0]][0].con_addr && engine->ice_in.cands[engine->ice_in.chosen[0]][0].con_port) {
char tmp[80] = "";
const char *media_varname = NULL, *port_varname = NULL;
engine->cur_payload_map->remote_sdp_ip = switch_core_session_strdup(smh->session, (char *) engine->ice_in.cands[engine->ice_in.chosen[0]][0].con_addr);
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(smh->session), SWITCH_LOG_DEBUG,
"setting remote %s ice addr to index %d %s:%d based on candidate\n", type2str(type), engine->ice_in.chosen[0],
@ -3947,12 +3949,26 @@ static switch_status_t check_ice(switch_media_handle_t *smh, switch_media_type_t
smh->mparams->remote_ip = engine->cur_payload_map->remote_sdp_ip;
}
if (engine->type == SWITCH_MEDIA_TYPE_VIDEO) {
media_varname = "remote_video_ip";
port_varname = "remote_video_port";
} else if (engine->type == SWITCH_MEDIA_TYPE_AUDIO) {
media_varname = "remote_audio_ip";
port_varname = "remote_audio_port";
} else if (engine->type == SWITCH_MEDIA_TYPE_TEXT) {
media_varname = "remote_text_ip";
port_varname = "remote_text_port";
}
switch_snprintf(tmp, sizeof(tmp), "%d", engine->ice_in.cands[engine->ice_in.chosen[0]][0].con_port);
switch_channel_set_variable(smh->session->channel, SWITCH_REMOTE_MEDIA_IP_VARIABLE, engine->ice_in.cands[engine->ice_in.chosen[0]][0].con_addr);
switch_channel_set_variable(smh->session->channel, SWITCH_REMOTE_MEDIA_PORT_VARIABLE, tmp);
switch_channel_set_variable(smh->session->channel, media_varname, engine->ice_in.cands[engine->ice_in.chosen[0]][0].con_addr);
switch_channel_set_variable(smh->session->channel, port_varname, tmp);
}
if (engine->ice_in.cands[engine->ice_in.chosen[1]][1].con_port) {
const char *media_varname = NULL, *port_varname = NULL;
char tmp[35] = "";
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(smh->session), SWITCH_LOG_DEBUG,
"Setting remote rtcp %s addr to %s:%d based on candidate\n", type2str(type),
engine->ice_in.cands[engine->ice_in.chosen[1]][1].con_addr, engine->ice_in.cands[engine->ice_in.chosen[1]][1].con_port);
@ -3960,6 +3976,23 @@ static switch_status_t check_ice(switch_media_handle_t *smh, switch_media_type_t
engine->remote_rtcp_ice_addr = switch_core_session_strdup(smh->session, engine->ice_in.cands[engine->ice_in.chosen[1]][1].con_addr);
engine->remote_rtcp_port = engine->ice_in.cands[engine->ice_in.chosen[1]][1].con_port;
if (engine->type == SWITCH_MEDIA_TYPE_VIDEO) {
media_varname = "remote_video_rtcp_ip";
port_varname = "remote_video_rtcp_port";
} else if (engine->type == SWITCH_MEDIA_TYPE_AUDIO) {
media_varname = "remote_audio_rtcp_ip";
port_varname = "remote_audio_rtcp_port";
} else if (engine->type == SWITCH_MEDIA_TYPE_TEXT) {
media_varname = "remote_text_rtcp_ip";
port_varname = "remote_text_rtcp_port";
}
switch_snprintf(tmp, sizeof(tmp), "%d", engine->ice_in.cands[engine->ice_in.chosen[1]][1].con_port);
switch_channel_set_variable(smh->session->channel, media_varname, engine->ice_in.cands[engine->ice_in.chosen[1]][1].con_addr);
switch_channel_set_variable(smh->session->channel, port_varname, tmp);
}
@ -5849,6 +5882,32 @@ SWITCH_DECLARE(uint8_t) switch_core_media_negotiate_sdp(switch_core_session_t *s
done:
if (v_engine->rtp_session) {
if (v_engine->fir) {
switch_rtp_set_flag(v_engine->rtp_session, SWITCH_RTP_FLAG_FIR);
} else {
switch_rtp_clear_flag(v_engine->rtp_session, SWITCH_RTP_FLAG_FIR);
}
if (v_engine->pli) {
switch_rtp_set_flag(v_engine->rtp_session, SWITCH_RTP_FLAG_PLI);
} else {
switch_rtp_clear_flag(v_engine->rtp_session, SWITCH_RTP_FLAG_PLI);
}
if (v_engine->nack) {
switch_rtp_set_flag(v_engine->rtp_session, SWITCH_RTP_FLAG_NACK);
} else {
switch_rtp_clear_flag(v_engine->rtp_session, SWITCH_RTP_FLAG_NACK);
}
if (v_engine->tmmbr) {
switch_rtp_set_flag(v_engine->rtp_session, SWITCH_RTP_FLAG_TMMBR);
} else {
switch_rtp_clear_flag(v_engine->rtp_session, SWITCH_RTP_FLAG_TMMBR);
}
}
if (match) {
switch_channel_set_flag(channel, CF_AUDIO);
} else {
@ -8757,6 +8816,7 @@ SWITCH_DECLARE(switch_status_t) switch_core_media_activate_rtp(switch_core_sessi
/******************************************************************************************/
if (v_engine->rtp_session) {
printf("XXXXXXXXXXXXXXXXXXXXXXXXXXXXx BLAH! XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXxxx\n");
goto video_up;
}
@ -8804,7 +8864,7 @@ SWITCH_DECLARE(switch_status_t) switch_core_media_activate_rtp(switch_core_sessi
if (v_engine->tmmbr) {
flags[SWITCH_RTP_FLAG_TMMBR]++;
}
v_engine->rtp_session = switch_rtp_new(a_engine->local_sdp_ip,
v_engine->local_sdp_port,
v_engine->cur_payload_map->remote_sdp_ip,

374
src/switch_rtp.c
View File

@ -322,6 +322,7 @@ struct switch_rtp {
rtcp_msg_t rtcp_send_msg;
switch_rtcp_frame_t rtcp_frame;
uint8_t send_rr;
uint8_t fir_seq;
uint16_t fir_count;
uint16_t pli_count;
@ -1030,7 +1031,6 @@ static void handle_ice(switch_rtp_t *rtp_session, switch_rtp_ice_t *ice, void *d
}
if ((ice->type & ICE_VANILLA)) {
char foo1[13] = "", foo2[13] = "";
if (!ok) ok = !memcmp(packet->header.id, ice->last_sent_id, 12);
if (packet->header.type == SWITCH_STUN_BINDING_RESPONSE) {
@ -1049,9 +1049,6 @@ static void handle_ice(switch_rtp_t *rtp_session, switch_rtp_ice_t *ice, void *d
}
}
memcpy(foo1, packet->header.id, 12);
memcpy(foo2, ice->last_sent_id, 12);
if (!ok && ice == &rtp_session->ice && rtp_session->rtcp_ice.ice_params && pri &&
*pri == rtp_session->rtcp_ice.ice_params->cands[rtp_session->rtcp_ice.ice_params->chosen[1]][1].priority) {
ice = &rtp_session->rtcp_ice;
@ -1766,7 +1763,7 @@ static void rtcp_generate_sender_info(switch_rtp_t *rtp_session, struct switch_r
switch_time_t now;
uint32_t sec, ntp_sec, ntp_usec;
switch_time_exp_t now_hr;
now = switch_time_now();
now = switch_micro_time_now();
sec = (uint32_t)(now/1000000); /* convert to seconds */
ntp_sec = sec+NTP_TIME_OFFSET; /* convert to NTP seconds */
sr->ntp_msw = htonl(ntp_sec); /* store result in "most significant word" */
@ -1795,7 +1792,7 @@ static void rtcp_generate_report_block(switch_rtp_t *rtp_session, struct switch_
uint32_t expected_pkt, dlsr;
int32_t pkt_lost;
uint32_t ntp_sec, ntp_usec, lsr_now, sec;
now = switch_time_now();
now = switch_micro_time_now();
sec = (uint32_t)(now/1000000); /* convert to seconds */
ntp_sec = sec+NTP_TIME_OFFSET; /* convert to NTP seconds */
ntp_usec = (uint32_t)(now - (sec*1000000)); /* remove seconds to keep only the microseconds */
@ -2007,7 +2004,7 @@ static int using_ice(switch_rtp_t *rtp_session)
static int check_rtcp_and_ice(switch_rtp_t *rtp_session)
{
int ret = 0;
int rtcp_ok = 0, rtcp_fb = 0;
int rtcp_ok = 0, rtcp_fb = 0, send_rr = 0;
switch_time_t now = switch_micro_time_now();
int rate = 0, nack_ttl = 0;
uint32_t cur_nack[MAX_NACK] = { 0 };
@ -2057,6 +2054,12 @@ static int check_rtcp_and_ice(switch_rtp_t *rtp_session)
}
}
if (rtp_session->send_rr) {
rtp_session->send_rr = 0;
rtcp_ok = 1;
send_rr = 1;
}
//switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_ERROR, "TIME CHECK %d > %d\n", (int)((now - rtp_session->rtcp_last_sent) / 1000), rate);
if (!rtcp_ok && (!rtp_session->rtcp_last_sent || (int)((now - rtp_session->rtcp_last_sent) / 1000) > rate)) {
@ -2100,7 +2103,7 @@ static int check_rtcp_and_ice(switch_rtp_t *rtp_session)
rtp_session->rtcp_send_msg.header.count = 1;
if (!rtp_session->stats.rtcp.sent_pkt_count) {
if (!rtp_session->stats.rtcp.sent_pkt_count || send_rr) {
rtp_session->rtcp_send_msg.header.type = _RTCP_PT_RR; /* Receiver report */
rr=(struct switch_rtcp_receiver_report*) rtp_session->rtcp_send_msg.body;
rr->ssrc = htonl(rtp_session->ssrc);
@ -3984,7 +3987,7 @@ SWITCH_DECLARE(switch_status_t) switch_rtp_create(switch_rtp_t **new_rtp_session
#endif
/* Jitter */
rtp_session->stats.inbound.last_proc_time = switch_time_now() / 1000;
rtp_session->stats.inbound.last_proc_time = switch_micro_time_now() / 1000;
rtp_session->stats.inbound.jitter_n = 0;
rtp_session->stats.inbound.jitter_add = 0;
rtp_session->stats.inbound.jitter_addsq = 0;
@ -6060,8 +6063,9 @@ static switch_status_t process_rtcp_report(switch_rtp_t *rtp_session, rtcp_msg_t
switch_core_media_gen_key_frame(rtp_session->session);
}
} else
} else {
struct switch_rtcp_report_block *report;
if (msg->header.type == _RTCP_PT_SR || msg->header.type == _RTCP_PT_RR) {
switch_time_t now;
@ -6072,7 +6076,11 @@ static switch_status_t process_rtcp_report(switch_rtp_t *rtp_session, rtcp_msg_t
double rtt_now = 0;
int rtt_increase = 0, packet_loss_increase=0;
now = switch_time_now(); /* number of microseconds since 00:00:00 january 1, 1970 UTC */
if (msg->header.type == _RTCP_PT_SR && rtp_session->ice.ice_user) {
rtp_session->send_rr = 1;
}
now = switch_micro_time_now(); /* number of microseconds since 00:00:00 january 1, 1970 UTC */
sec = (uint32_t)(now/1000000); /* converted to second (NTP most significant bits) */
ntp_sec = sec+NTP_TIME_OFFSET; /* 32bits most significant */
ntp_usec = (uint32_t)(now - (sec*1000000)); /* micro seconds */
@ -6088,7 +6096,7 @@ static switch_status_t process_rtcp_report(switch_rtp_t *rtp_session, rtcp_msg_t
lsr = (ntohl(sr->sender_info.ntp_lsw)&0xffff0000)>>16 | (ntohl(sr->sender_info.ntp_msw)&0x0000ffff)<<16; /* The middle 32 bits out of 64 in the NTP timestamp */
rtp_session->stats.rtcp.last_recv_lsr_peer = htonl(lsr); /* Save it include it in the next SR */
rtp_session->stats.rtcp.last_recv_lsr_local = lsr_now; /* Save it to calculate DLSR when generating next SR */
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG10,"Received a SR with %d report blocks, " \
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3,"Received a SR with %d report blocks, " \
"length in words = %d, " \
"SSRC = 0x%X, " \
"NTP MSW = %u, " \
@ -6114,172 +6122,188 @@ static switch_status_t process_rtcp_report(switch_rtp_t *rtp_session, rtcp_msg_t
rtp_session->rtcp_frame.packet_count = ntohl(sr->sender_info.pc);
rtp_session->rtcp_frame.octect_count = ntohl(sr->sender_info.oc);
for (i = 0; i < (int)msg->header.count && i < MAX_REPORT_BLOCKS ; i++) {
struct switch_rtcp_report_block *report = (struct switch_rtcp_report_block *) (msg->body + (sizeof(struct switch_rtcp_sr_head) + (i * sizeof(struct switch_rtcp_report_block))));
uint32_t old_avg = rtp_session->rtcp_frame.reports[i].loss_avg;
uint8_t percent_fraction = (uint8_t)report->fraction * 100 / 256 ;
if (!rtp_session->rtcp_frame.reports[i].loss_avg) {
rtp_session->rtcp_frame.reports[i].loss_avg = (uint8_t)percent_fraction;
} else {
rtp_session->rtcp_frame.reports[i].loss_avg = (uint32_t)(((float)rtp_session->rtcp_frame.reports[i].loss_avg * .7) +
((float)(uint8_t)percent_fraction * .3));
}
rtp_session->rtcp_frame.reports[i].ssrc = ntohl(report->ssrc);
rtp_session->rtcp_frame.reports[i].fraction = (uint8_t)report->fraction;
rtp_session->rtcp_frame.reports[i].lost = ntohl(report->lost);
rtp_session->rtcp_frame.reports[i].highest_sequence_number_received = ntohl(report->highest_sequence_number_received);
rtp_session->rtcp_frame.reports[i].jitter = ntohl(report->jitter);
rtp_session->rtcp_frame.reports[i].lsr = ntohl(report->lsr);
rtp_session->rtcp_frame.reports[i].dlsr = ntohl(report->dlsr);
if (rtp_session->rtcp_frame.reports[i].lsr && !rtp_session->flags[SWITCH_RTP_FLAG_RTCP_PASSTHRU]) {
switch_time_exp_gmt(&now_hr,now);
/* Calculating RTT = A - DLSR - LSR */
rtt_now = (double)(lsr_now - rtp_session->rtcp_frame.reports[i].dlsr - rtp_session->rtcp_frame.reports[i].lsr)/65536;
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3,
"Receiving an RTCP packet\n[%04d-%02d-%02d %02d:%02d:%02d.%d] SSRC[0x%x]\n"
"RTT[%f] = A[%u] - DLSR[%u] - LSR[%u]\n",
1900 + now_hr.tm_year, now_hr.tm_mday, now_hr.tm_mon, now_hr.tm_hour, now_hr.tm_min, now_hr.tm_sec, now_hr.tm_usec,
rtp_session->rtcp_frame.reports[i].ssrc, rtt_now,
lsr_now, rtp_session->rtcp_frame.reports[i].dlsr, rtp_session->rtcp_frame.reports[i].lsr);
if (!rtp_session->rtcp_frame.reports[i].rtt_avg) {
rtp_session->rtcp_frame.reports[i].rtt_avg = rtt_now;
} else {
rtp_session->rtcp_frame.reports[i].rtt_avg = (double)((rtp_session->rtcp_frame.reports[i].rtt_avg * .7) + (rtt_now * .3 ));
}
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3, "RTT average %f\n",
rtp_session->rtcp_frame.reports[i].rtt_avg);
}
if (rtp_session->flags[SWITCH_RTP_FLAG_ADJ_BITRATE_CAP] && rtp_session->flags[SWITCH_RTP_FLAG_ESTIMATORS] && !rtp_session->flags[SWITCH_RTP_FLAG_VIDEO]) {
/* SWITCH_RTP_FLAG_ADJ_BITRATE_CAP : Can the codec change its bitrate on the fly per API command ? */
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3, "Current packet loss: [%d %%] Current RTT: [%f ms]\n", percent_fraction, rtt_now);
#endif
switch_kalman_estimate(rtp_session->estimators[EST_RTT], rtt_now, EST_RTT);
if (switch_kalman_cusum_detect_change(rtp_session->detectors[EST_RTT], rtt_now, rtp_session->estimators[EST_RTT]->val_estimate_last)) {
/* sudden change in the mean value of RTT */
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3,"Sudden change in the mean value of RTT !\n");
#endif
rtt_increase = 1;
}
switch_kalman_estimate(rtp_session->estimators[EST_LOSS], percent_fraction, EST_LOSS);
if (switch_kalman_cusum_detect_change(rtp_session->detectors[EST_LOSS], percent_fraction, rtp_session->estimators[EST_LOSS]->val_estimate_last)){
/* sudden change in the mean value of packet loss */
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3,"Sudden change in the mean value of packet loss!\n");
#endif
packet_loss_increase = 1;
}
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3, "ESTIMATORS: Packet loss will be: [%f] RTT will be: [%f ms]\n",
rtp_session->estimators[EST_LOSS]->val_estimate_last, rtp_session->estimators[EST_RTT]->val_estimate_last);
#endif
if (rtp_session->rtcp_frame.reports[i].loss_avg != old_avg) {
/*getting bad*/
if (switch_kalman_is_slow_link(rtp_session->estimators[EST_LOSS],
rtp_session->estimators[EST_RTT])) {
/* going to minimum bitrate */
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3, "Slow link conditions: Loss average: [%d %%], Previous loss: [%d %%]. \
Going to minimum bitrate!",rtp_session->rtcp_frame.reports[i].loss_avg, old_avg);
#endif
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_ADJUST_BITRATE, SCCT_STRING, "minimum", SCCT_NONE, NULL, NULL, NULL);
/* if after going to minimum bitrate we still have packet loss then we increase ptime. TODO */
} else if (packet_loss_increase && (rtp_session->estimators[EST_LOSS]->val_estimate_last >= 5)) {
/* sudden change in the mean value of packet loss percentage */
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_ADJUST_BITRATE,
SCCT_STRING, "decrease",
SCCT_NONE, NULL, NULL, NULL);
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3,"Sudden change in the mean value of packet loss percentage !\n");
#endif
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_PACKET_LOSS, SCCT_INT,
(void *)&rtp_session->rtcp_frame.reports[i].loss_avg,
SCCT_NONE, NULL, NULL, NULL);
} else if (!rtt_increase && rtp_session->estimators[EST_LOSS]->val_estimate_last >= rtp_session->rtcp_frame.reports[i].loss_avg ) {
/* lossy because of congestion (queues full somewhere -> some packets are dropped , but RTT is good ), packet loss with many small gaps */
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3, "packet loss, but RTT is not bad\n");
#endif
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_PACKET_LOSS, SCCT_INT,
(void *)&rtp_session->rtcp_frame.reports[i].loss_avg,
SCCT_NONE, NULL, NULL, NULL);
} else if ((rtp_session->estimators[EST_LOSS]->val_estimate_last < 1) && packet_loss_increase) {
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3, "small packet loss average\n");
#endif
/*small loss_avg*/
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_ADJUST_BITRATE,
SCCT_STRING, "default",
SCCT_NONE, NULL, NULL, NULL);
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_PACKET_LOSS, SCCT_INT,
(void *)&rtp_session->rtcp_frame.reports[i].loss_avg,
SCCT_NONE, NULL, NULL, NULL);
} else if ((rtp_session->estimators[EST_LOSS]->val_estimate_last < 5) &&
(rtp_session->rtcp_frame.reports[i].rtt_avg < rtp_session->estimators[EST_RTT]->val_estimate_last)) {
/* estimate that packet loss will decrease, we can increase the bitrate */
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_ADJUST_BITRATE,
SCCT_STRING, "increase",
SCCT_NONE, NULL, NULL, NULL);
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_PACKET_LOSS, SCCT_INT,
(void *)&rtp_session->rtcp_frame.reports[i].loss_avg,
SCCT_NONE, NULL, NULL, NULL);
} else {
/* *do nothing about bitrate, just pass the packet loss to the codec */
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3,"do nothing about bitrate, just pass the packet loss to the codec\n");
#endif
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_PACKET_LOSS, SCCT_INT,
(void *)&rtp_session->rtcp_frame.reports[i].loss_avg,
SCCT_NONE, NULL, NULL, NULL);
}
}
} else {
if (!rtp_session->flags[SWITCH_RTP_FLAG_VIDEO] && rtp_session->rtcp_frame.reports[i].loss_avg != old_avg) {
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_PACKET_LOSS, SCCT_INT,
(void *)&rtp_session->rtcp_frame.reports[i].loss_avg,
SCCT_NONE, NULL, NULL, NULL);
}
}
}
rtp_session->rtcp_frame.report_count = (uint16_t)i;
report = &sr->report_block;
} else { /* Receiver report */
struct switch_rtcp_receiver_report* rr = (struct switch_rtcp_receiver_report*)msg->body;
packet_ssrc = rr->ssrc;
memset(&rtp_session->rtcp_frame, 0, sizeof(rtp_session->rtcp_frame));
//memset(&rtp_session->rtcp_frame, 0, sizeof(rtp_session->rtcp_frame));
report = &rr->report_block;
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3,"Received a RR with %d report blocks, " \
"length in words = %d, " \
"SSRC = 0x%X, ",
msg->header.count,
ntohs((uint16_t)msg->header.length),
ntohl(rr->ssrc));
}
for (i = 0; i < (int)msg->header.count && i < MAX_REPORT_BLOCKS ; i++) {
uint32_t old_avg = rtp_session->rtcp_frame.reports[i].loss_avg;
uint8_t percent_fraction = (uint8_t)report->fraction * 100 / 256 ;
if (!rtp_session->rtcp_frame.reports[i].loss_avg) {
rtp_session->rtcp_frame.reports[i].loss_avg = (uint8_t)percent_fraction;
} else {
rtp_session->rtcp_frame.reports[i].loss_avg = (uint32_t)(((float)rtp_session->rtcp_frame.reports[i].loss_avg * .7) +
((float)(uint8_t)percent_fraction * .3));
}
rtp_session->rtcp_frame.reports[i].ssrc = ntohl(report->ssrc);
rtp_session->rtcp_frame.reports[i].fraction = (uint8_t)report->fraction;
rtp_session->rtcp_frame.reports[i].lost = ntohl(report->lost);
rtp_session->rtcp_frame.reports[i].highest_sequence_number_received = ntohl(report->highest_sequence_number_received);
rtp_session->rtcp_frame.reports[i].jitter = ntohl(report->jitter);
rtp_session->rtcp_frame.reports[i].lsr = ntohl(report->lsr);
rtp_session->rtcp_frame.reports[i].dlsr = ntohl(report->dlsr);
if (rtp_session->rtcp_frame.reports[i].lsr && !rtp_session->flags[SWITCH_RTP_FLAG_RTCP_PASSTHRU]) {
switch_time_exp_gmt(&now_hr,now);
/* Calculating RTT = A - DLSR - LSR */
rtt_now = (double)(lsr_now - rtp_session->rtcp_frame.reports[i].dlsr - rtp_session->rtcp_frame.reports[i].lsr)/65536;
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3,
"Receiving an RTCP packet\n[%04d-%02d-%02d %02d:%02d:%02d.%d] SSRC[0x%x]\n"
"RTT[%f] = A[%u] - DLSR[%u] - LSR[%u]\n",
1900 + now_hr.tm_year, now_hr.tm_mday, now_hr.tm_mon, now_hr.tm_hour, now_hr.tm_min, now_hr.tm_sec, now_hr.tm_usec,
rtp_session->rtcp_frame.reports[i].ssrc, rtt_now,
lsr_now, rtp_session->rtcp_frame.reports[i].dlsr, rtp_session->rtcp_frame.reports[i].lsr);
if (!rtp_session->rtcp_frame.reports[i].rtt_avg) {
rtp_session->rtcp_frame.reports[i].rtt_avg = rtt_now;
} else {
rtp_session->rtcp_frame.reports[i].rtt_avg = (double)((rtp_session->rtcp_frame.reports[i].rtt_avg * .7) + (rtt_now * .3 ));
}
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3, "RTT average %f\n",
rtp_session->rtcp_frame.reports[i].rtt_avg);
}
if (rtp_session->flags[SWITCH_RTP_FLAG_ADJ_BITRATE_CAP] && rtp_session->flags[SWITCH_RTP_FLAG_ESTIMATORS] && !rtp_session->flags[SWITCH_RTP_FLAG_VIDEO]) {
/* SWITCH_RTP_FLAG_ADJ_BITRATE_CAP : Can the codec change its bitrate on the fly per API command ? */
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3, "Current packet loss: [%d %%] Current RTT: [%f ms]\n", percent_fraction, rtt_now);
#endif
switch_kalman_estimate(rtp_session->estimators[EST_RTT], rtt_now, EST_RTT);
if (switch_kalman_cusum_detect_change(rtp_session->detectors[EST_RTT], rtt_now, rtp_session->estimators[EST_RTT]->val_estimate_last)) {
/* sudden change in the mean value of RTT */
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3,"Sudden change in the mean value of RTT !\n");
#endif
rtt_increase = 1;
}
switch_kalman_estimate(rtp_session->estimators[EST_LOSS], percent_fraction, EST_LOSS);
if (switch_kalman_cusum_detect_change(rtp_session->detectors[EST_LOSS], percent_fraction, rtp_session->estimators[EST_LOSS]->val_estimate_last)){
/* sudden change in the mean value of packet loss */
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3,"Sudden change in the mean value of packet loss!\n");
#endif
packet_loss_increase = 1;
}
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3, "ESTIMATORS: Packet loss will be: [%f] RTT will be: [%f ms]\n",
rtp_session->estimators[EST_LOSS]->val_estimate_last, rtp_session->estimators[EST_RTT]->val_estimate_last);
#endif
if (rtp_session->rtcp_frame.reports[i].loss_avg != old_avg) {
/*getting bad*/
if (switch_kalman_is_slow_link(rtp_session->estimators[EST_LOSS],
rtp_session->estimators[EST_RTT])) {
/* going to minimum bitrate */
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3, "Slow link conditions: Loss average: [%d %%], Previous loss: [%d %%]. \
Going to minimum bitrate!",rtp_session->rtcp_frame.reports[i].loss_avg, old_avg);
#endif
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_ADJUST_BITRATE, SCCT_STRING, "minimum", SCCT_NONE, NULL, NULL, NULL);
/* if after going to minimum bitrate we still have packet loss then we increase ptime. TODO */
} else if (packet_loss_increase && (rtp_session->estimators[EST_LOSS]->val_estimate_last >= 5)) {
/* sudden change in the mean value of packet loss percentage */
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_ADJUST_BITRATE,
SCCT_STRING, "decrease",
SCCT_NONE, NULL, NULL, NULL);
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3,"Sudden change in the mean value of packet loss percentage !\n");
#endif
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_PACKET_LOSS, SCCT_INT,
(void *)&rtp_session->rtcp_frame.reports[i].loss_avg,
SCCT_NONE, NULL, NULL, NULL);
} else if (!rtt_increase && rtp_session->estimators[EST_LOSS]->val_estimate_last >= rtp_session->rtcp_frame.reports[i].loss_avg ) {
/* lossy because of congestion (queues full somewhere -> some packets are dropped , but RTT is good ), packet loss with many small gaps */
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3, "packet loss, but RTT is not bad\n");
#endif
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_PACKET_LOSS, SCCT_INT,
(void *)&rtp_session->rtcp_frame.reports[i].loss_avg,
SCCT_NONE, NULL, NULL, NULL);
} else if ((rtp_session->estimators[EST_LOSS]->val_estimate_last < 1) && packet_loss_increase) {
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3, "small packet loss average\n");
#endif
/*small loss_avg*/
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_ADJUST_BITRATE,
SCCT_STRING, "default",
SCCT_NONE, NULL, NULL, NULL);
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_PACKET_LOSS, SCCT_INT,
(void *)&rtp_session->rtcp_frame.reports[i].loss_avg,
SCCT_NONE, NULL, NULL, NULL);
} else if ((rtp_session->estimators[EST_LOSS]->val_estimate_last < 5) &&
(rtp_session->rtcp_frame.reports[i].rtt_avg < rtp_session->estimators[EST_RTT]->val_estimate_last)) {
/* estimate that packet loss will decrease, we can increase the bitrate */
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_ADJUST_BITRATE,
SCCT_STRING, "increase",
SCCT_NONE, NULL, NULL, NULL);
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_PACKET_LOSS, SCCT_INT,
(void *)&rtp_session->rtcp_frame.reports[i].loss_avg,
SCCT_NONE, NULL, NULL, NULL);
} else {
/* *do nothing about bitrate, just pass the packet loss to the codec */
#ifdef DEBUG_ESTIMATORS_
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG3,"do nothing about bitrate, just pass the packet loss to the codec\n");
#endif
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_PACKET_LOSS, SCCT_INT,
(void *)&rtp_session->rtcp_frame.reports[i].loss_avg,
SCCT_NONE, NULL, NULL, NULL);
}
}
} else {
if (!rtp_session->flags[SWITCH_RTP_FLAG_VIDEO] && rtp_session->rtcp_frame.reports[i].loss_avg != old_avg) {
switch_core_media_codec_control(rtp_session->session, SWITCH_MEDIA_TYPE_AUDIO,
SWITCH_IO_WRITE, SCC_AUDIO_PACKET_LOSS, SCCT_INT,
(void *)&rtp_session->rtcp_frame.reports[i].loss_avg,
SCCT_NONE, NULL, NULL, NULL);
}
}
report++;
}
rtp_session->rtcp_frame.report_count = (uint16_t)i;
rtp_session->rtcp_fresh_frame = 1;
rtp_session->stats.rtcp.peer_ssrc = ntohl(packet_ssrc);
}
}
if (msg->header.type > 194 && msg->header.type < 255) {
status = SWITCH_STATUS_SUCCESS;
@ -6820,7 +6844,7 @@ static int rtp_common_read(switch_rtp_t *rtp_session, switch_payload_t *payload_
if (rtcp_status == SWITCH_STATUS_SUCCESS) {
switch_rtp_reset_media_timer(rtp_session);
if (rtp_session->flags[SWITCH_RTP_FLAG_RTCP_PASSTHRU]) {
switch_channel_t *channel = switch_core_session_get_channel(rtp_session->session);
const char *uuid = switch_channel_get_partner_uuid(channel);
@ -7740,7 +7764,7 @@ static int rtp_common_write(switch_rtp_t *rtp_session,
switch_set_flag(&rtp_session->vad_data, SWITCH_VAD_FLAG_TALKING);
rtp_session->vad_data.start_talking = switch_time_now();
rtp_session->vad_data.start_talking = switch_micro_time_now();
if (!(rtp_session->rtp_bugs & RTP_BUG_NEVER_SEND_MARKER)) {
send_msg->header.m = 1;
@ -7763,7 +7787,7 @@ static int rtp_common_write(switch_rtp_t *rtp_session,
}
if (switch_test_flag(&rtp_session->vad_data, SWITCH_VAD_FLAG_TALKING)) {
if (++rtp_session->vad_data.hangover_hits >= rtp_session->vad_data.hangover) {
rtp_session->vad_data.stop_talking = switch_time_now();
rtp_session->vad_data.stop_talking = switch_micro_time_now();
rtp_session->vad_data.total_talk_time += (rtp_session->vad_data.stop_talking - rtp_session->vad_data.start_talking);
switch_clear_flag(&rtp_session->vad_data, SWITCH_VAD_FLAG_TALKING);
@ -8070,7 +8094,7 @@ SWITCH_DECLARE(switch_status_t) switch_rtp_enable_vad(switch_rtp_t *rtp_session,
rtp_session->vad_data.next_scan = switch_epoch_time_now(NULL);
rtp_session->vad_data.scan_freq = 0;
if (switch_test_flag(&rtp_session->vad_data, SWITCH_VAD_FLAG_TALKING)) {
rtp_session->vad_data.start_talking = switch_time_now();
rtp_session->vad_data.start_talking = switch_micro_time_now();
}
switch_rtp_set_flag(rtp_session, SWITCH_RTP_FLAG_VAD);
switch_set_flag(&rtp_session->vad_data, SWITCH_VAD_FLAG_CNG);