diff --git a/channels/chan_brcm.c b/channels/chan_brcm.c
index 4f710d60f29395d22643fc11b4611daaccc3ebdf..ca1c19287bdad4d391812a76dbed3ae493cffc0a 100644
--- a/channels/chan_brcm.c
+++ b/channels/chan_brcm.c
@@ -76,6 +76,7 @@ static int brcm_signal_dialtone(struct brcm_pvt *p);
static int brcm_in_conference(const struct brcm_pvt *p);
static int cwtimeout_cb(const void *data);
static int r4hanguptimeout_cb(const void *data);
+static void brcm_generate_rtp_packet(struct brcm_subchannel *p, uint8_t *packet_buf, int type, int marker, int dtmf_timestamp);
static int brcm_mute_connection(struct brcm_subchannel *p);
static int brcm_unmute_connection(struct brcm_subchannel *p);
static int brcm_close_connection(struct brcm_subchannel *p);
@@ -1190,7 +1191,7 @@ static char* brcm_get_codec_string(int id) {
case G726: return "g726"; break;
case G729: return "g729"; break;
case G722: return "g722"; break;
- case CN: return "cn"; break;
+ case CN: return "cn"; break;
case -1: return "none set"; break;
default: return "unknown id"; break;
}
@@ -1201,15 +1202,14 @@ static int brcm_classify_rtp_packet(int id) {
//ast_debug(5, "RTP Packet ID = %d\n", id);
switch (id) {
- case PCMU: return BRCM_AUDIO;
- case G726: return BRCM_AUDIO;
- case G723: return BRCM_AUDIO;
- case PCMA: return BRCM_AUDIO;
- case G729: return BRCM_AUDIO;
- case G722: return BRCM_AUDIO;
- case CN: return BRCM_AUDIO;
- case RTCP_SR: return BRCM_RTCP_SR;
- case RTCP_RR: return BRCM_RTCP_RR;
+ case PCMU: return BRCM_AUDIO;
+ case G726: return BRCM_AUDIO;
+ case G723: return BRCM_AUDIO;
+ case PCMA: return BRCM_AUDIO;
+ case G729: return BRCM_AUDIO;
+ case G722: return BRCM_AUDIO;
+ case CN: return BRCM_AUDIO;
+ case RTCP: return BRCM_UNKNOWN; // Ignored packet.
default:
ast_verbose("Unknown rtp packet id %d\n", id);
return BRCM_UNKNOWN;
@@ -1269,9 +1269,10 @@ static int brcm_write(struct ast_channel *ast, struct ast_frame *frame)
if (sub->channel_state == ONHOLD) {
return 0;
}
-
- if(frame->frametype == AST_FRAME_VOICE || frame->frametype == AST_FRAME_CNG) {
- packet_size = sizeof(audio_packet_t) - 1 + frame->datalen;
+ if (frame->frametype == AST_FRAME_CNG) {
+ /*packet_size = sizeof(audio_packet_t) - 1 + RTP_HEADER_SIZE + frame->datalen;
+ CNG paket has only 1 data for the noise lvl, datalen=1*/
+ packet_size = sizeof(audio_packet_t) + RTP_HEADER_SIZE ;
ap = calloc(1, packet_size);
if (!ap) {
ast_log(LOG_ERROR, "Out of memory\n");
@@ -1280,26 +1281,29 @@ static int brcm_write(struct ast_channel *ast, struct ast_frame *frame)
ap->line = sub->parent->line_id;
ap->cnx_id = sub->parent->line_id;
- ap->rtp_size = frame->datalen;
+ ap->rtp_size = RTP_HEADER_SIZE + 1;
/* copy frame data to audio packet */
- memcpy(ap->rtp, frame->data.ptr, frame->datalen);
+ /* level = 127 - (level & 0x7f); */
+ ap->rtp[12]=127 - (frame->subclass.integer & 0x7f);
//ast_mutex_lock(&sub->parent->lock);
pvt_lock(sub->parent, "TELCHAN write frame");
+ /* generate the rtp header */
+ brcm_generate_rtp_packet(sub, ap->rtp, CN, 0, 0);
+
/* set rtp id sent to endpoint */
- sub->codec = frame->frametype == AST_FRAME_CNG ? CN : map_ast_codec_id_to_rtp(frame->subclass.format);
+ sub->codec = CN;
//ast_mutex_unlock(&sub->parent->lock);
pvt_unlock(sub->parent);
-
pe_bus_send(audio_tx_bus, (uint8_t *)ap, packet_size);
free(ap);
- }
+ }
- if(frame->frametype == AST_FRAME_RTCP) {
- packet_size = sizeof(audio_packet_t) - 1 + frame->datalen;
+ if(frame->frametype == AST_FRAME_VOICE) {
+ packet_size = sizeof(audio_packet_t) - 1 + RTP_HEADER_SIZE + frame->datalen;
ap = calloc(1, packet_size);
if (!ap) {
ast_log(LOG_ERROR, "Out of memory\n");
@@ -1308,15 +1312,22 @@ static int brcm_write(struct ast_channel *ast, struct ast_frame *frame)
ap->line = sub->parent->line_id;
ap->cnx_id = sub->parent->line_id;
- ap->rtp_size = frame->datalen;
+ ap->rtp_size = RTP_HEADER_SIZE + frame->datalen;
/* copy frame data to audio packet */
- memcpy(ap->rtp, frame->data.ptr, frame->datalen);
+ memcpy(ap->rtp + RTP_HEADER_SIZE, frame->data.ptr, frame->datalen);
- if (frame->subclass.integer == RTCP_SR && frame->datalen >= 44) {
- sub->jitter_count++;
- sub->farEndInterrivalJitter += ((ap->rtp[40] << 24) | (ap->rtp[41] << 16) | (ap->rtp[42] << 8) | (ap->rtp[43]));
- }
+ //ast_mutex_lock(&sub->parent->lock);
+ pvt_lock(sub->parent, "TELCHAN write frame");
+
+ /* generate the rtp header */
+ brcm_generate_rtp_packet(sub, ap->rtp, map_ast_codec_id_to_rtp(frame->subclass.format), 0, 0);
+
+ /* set rtp id sent to endpoint */
+ sub->codec = map_ast_codec_id_to_rtp(frame->subclass.format);
+
+ //ast_mutex_unlock(&sub->parent->lock);
+ pvt_unlock(sub->parent);
pe_bus_send(audio_tx_bus, (uint8_t *)ap, packet_size);
free(ap);
@@ -2308,8 +2319,8 @@ static void audio_packet_handler(pe_packet_t *p) {
(ap->rtp[0] & 0x80) && ap->rtp_size) {
fr.frametype = AST_FRAME_VOICE;
fr.offset = 0;
- fr.data.ptr = ap->rtp;
- fr.datalen = ap->rtp_size;
+ fr.data.ptr = ap->rtp + 12;
+ fr.datalen = ap->rtp_size - 12;
switch (ap->rtp[1]) {
case PCMU:
@@ -2336,21 +2347,16 @@ static void audio_packet_handler(pe_packet_t *p) {
fr.subclass.format = ast_format_g722;
fr.samples = 160;
break;
- case CN:
- fr.frametype = AST_FRAME_CNG;
- fr.subclass.integer = ap->rtp[12];
- break;
+ case CN:
+ fr.frametype = AST_FRAME_CNG;
+ fr.subclass.integer = ap->rtp[12];
+ break;
default:
ast_log(LOG_WARNING, "Unknown rtp codec id [%d], %d\n", ap->rtp[1], p->data[1]);
return;
}
- } else if (rtp_packet_type == BRCM_RTCP_SR || rtp_packet_type == BRCM_RTCP_RR) {
- fr.frametype = AST_FRAME_RTCP;
- fr.data.ptr = ap->rtp;
- fr.datalen = ap->rtp_size;
- fr.subclass.integer = (rtp_packet_type == BRCM_RTCP_SR ? RTCP_SR : RTCP_RR);
} else {
- ast_debug(5, "Dropping RTP frame of type %d.\n", rtp_packet_type);
+ //ast_debug(5, "Dropping RTP frame of type %d.\n", rtp_packet_type);
drop_frame=1;
//pvt_unlock(sub->parent);
}
@@ -2790,6 +2796,9 @@ static struct brcm_pvt *brcm_allocate_pvt(void)
sub->connection_id = -1;
sub->connection_init = 0;
sub->channel_state = ONHOOK;
+ sub->time_stamp = 0;
+ sub->sequence_number = 0;
+ sub->ssrc = 0;
sub->codec = -1;
sub->parent = tmp;
sub->cw_timer_id = -1;
@@ -2799,8 +2808,6 @@ static struct brcm_pvt *brcm_allocate_pvt(void)
sub->period = 20; // 20 ms
sub->conference_initiator = 0;
tmp->sub[i] = sub;
- sub->jitter_count = 0;
- sub->farEndInterrivalJitter = 0;
ast_debug(2, "subchannel created\n");
} else {
ast_log(LOG_ERROR, "no subchannel created\n");
@@ -3044,6 +3051,9 @@ static void brcm_show_subchannels(struct ast_cli_args *a, struct brcm_pvt *p)
ast_cli(a->fd, " Channel state : %s\n", state2str(sub->channel_state));
ast_cli(a->fd, " Connection init : %d\n", sub->connection_init);
ast_cli(a->fd, " Codec used : %s\n", brcm_get_codec_string(sub->codec));
+ ast_cli(a->fd, " RTP sequence number : %d\n", sub->sequence_number);
+ ast_cli(a->fd, " RTP SSRC : %d\n", sub->ssrc);
+ ast_cli(a->fd, " RTP timestamp : %d\n", sub->time_stamp);
ast_cli(a->fd, " CW Timer id : %d\n", sub->cw_timer_id);
ast_cli(a->fd, " CW Rejected : %d\n", sub->cw_rejected);
ast_cli(a->fd, " R4 Hangup Timer id : %d\n", sub->r4_hangup_timer_id);
@@ -3816,23 +3826,20 @@ static void ubus_call_answer_rtp_stats(struct ubus_request *req, int type, struc
if (tb[RTP_STATS_AVERAGE_ROUND_TRIP_DELAY])
sub->rtp_stats.averageRoundTripDelay = blobmsg_get_u32(tb[RTP_STATS_AVERAGE_ROUND_TRIP_DELAY]);
- sub->rtp_stats.averageFarEndInterarrivalJitter = sub->jitter_count ? (sub->farEndInterrivalJitter / sub->jitter_count) : 0;
-
ast_log(LOG_DEBUG, "RTP stats received:\nlocalBurstDensity: %d\nremoteBurstDensity: %d\n"
"localBurstDuration: %d\nremoteBurstDuration: %d\nlocalGapDensity: %d\n"
"remoteGapDensity: %d\nlocalGapDuration: %d\nremoteGapDuration: %d\n"
"localJbRate: %d\nremoteJbRate: %d\nlocalJbMax: %d\nremoteJbMax: %d\n"
"localJbNominal: %d\nremoteJbNominal: %d\nlocalJbAbsMax: %d\n"
"remoteJbAbsMax: %d\ndiscarded: %d\nlost: %d\nrxpkts: %d\ntxpkts: %d\n"
- "jbAvg: %d\njitter: %d\nuLossRate: %d\nmaxJitter: %d\naverageRoundTripDelay: %d\n"
- "averageFarEndInterarrivalJitter: %d\n",
+ "jbAvg: %d\njitter: %d\nuLossRate: %d\nmaxJitter: %d\naverageRoundTripDelay: %d\n",
sub->rtp_stats.localBurstDensity, sub->rtp_stats.remoteBurstDensity, sub->rtp_stats.localBurstDuration,
sub->rtp_stats.remoteBurstDuration, sub->rtp_stats.localGapDensity, sub->rtp_stats.remoteGapDensity,
sub->rtp_stats.localGapDuration, sub->rtp_stats.remoteGapDuration, sub->rtp_stats.localJbRate,
sub->rtp_stats.remoteJbRate, sub->rtp_stats.localJbMax, sub->rtp_stats.remoteJbMax, sub->rtp_stats.localJbNominal,
sub->rtp_stats.remoteJbNominal, sub->rtp_stats.localJbAbsMax, sub->rtp_stats.remoteJbAbsMax, sub->rtp_stats.discarded,
sub->rtp_stats.lost, sub->rtp_stats.rxpkts, sub->rtp_stats.txpkts, sub->rtp_stats.jbAvg, sub->rtp_stats.jitter,
- sub->rtp_stats.uLossRate, sub->rtp_stats.maxJitter, sub->rtp_stats.averageRoundTripDelay, sub->rtp_stats.averageFarEndInterarrivalJitter);
+ sub->rtp_stats.uLossRate, sub->rtp_stats.maxJitter, sub->rtp_stats.averageRoundTripDelay);
}
static int endpt_get_rtp_stats(int line) {
@@ -4312,10 +4319,11 @@ static int brcm_signal_callerid(struct ast_channel *chan, struct brcm_subchannel
static int brcm_create_connection(struct brcm_subchannel *sub) {
if (!sub->connection_init) {
+ /* generate random nr for rtp header */
+ sub->ssrc = rand();
+
ast_debug(1, "Creating virtual Asterisk connection for pvt line_id=%i connection_id=%d\n", sub->parent->line_id, sub->connection_id);
sub->connection_init = 1;
- sub->jitter_count = 0;
- sub->farEndInterrivalJitter = 0;
if(!brcm_in_onhold(sub->parent) && !brcm_in_call(sub->parent)) { // Is there another connection already?
ast_debug(1, "Creating real endpoint connection for pvt line_id=%i\n", sub->parent->line_id);
endpt_connection(sub->parent->line_id, sub->connection_id, "create");
@@ -4442,6 +4450,27 @@ static int brcm_close_connection(struct brcm_subchannel *sub)
return 0;
}
+
+/* Generate rtp payload, 12 bytes of header and 160 bytes of ulaw payload */
+static void brcm_generate_rtp_packet(struct brcm_subchannel *sub, uint8_t *packet_buf, int type, int marker, int dtmf_timestamp) {
+ unsigned short* packet_buf16 = (unsigned short*)packet_buf;
+ unsigned int* packet_buf32 = (unsigned int*)packet_buf;
+
+ //Generate the rtp header, packet is zero from the start, that fact is used
+ packet_buf[0] |= 0x80; //Set version 2 of header
+ //Padding 0
+ //Extension 0
+ //CSRC count 0
+ packet_buf[1] = type;
+ packet_buf[1] |= marker?0x80:0x00;
+ packet_buf16[1] = htons(sub->sequence_number++); //Add sequence number
+ if (sub->sequence_number > 0xFFFF) sub->sequence_number=0;
+ packet_buf32[1] = htonl(sub->time_stamp); //Add timestamp
+ sub->time_stamp += sub->period*8;
+ packet_buf32[2] = sub->ssrc; //Random SSRC
+}
+
+
static void brcm_dialtone_set(struct brcm_pvt *p, dialtone_state state)
{
ast_debug(3, "Old dialtone: %s, new dialtone: %s\n",
diff --git a/channels/chan_brcm.h b/channels/chan_brcm.h
index 58d2b95777519c47e2180b190706b8f221f8f65c..8bce65d5dfe69e0d688fd614c72a4645e10d52f4 100644
--- a/channels/chan_brcm.h
+++ b/channels/chan_brcm.h
@@ -22,8 +22,7 @@
#define G722 9
#define DTMF_PAYLOAD 101
#define DTMF 128
-#define RTCP_SR 200
-#define RTCP_RR 201
+#define RTCP 200
#define CN 13
#define NOT_INITIALIZED -1
@@ -102,7 +101,10 @@ struct brcm_subchannel {
enum brcm_channel_state channel_state; /* Channel states */
unsigned int connection_init; /* State for endpoint id connection initialization */
struct ast_frame fr; /* Frame */
+ unsigned int sequence_number; /* Endpoint RTP sequence number state */
+ unsigned int time_stamp; /* Endpoint RTP time stamp state */
unsigned int period; /* Endpoint RTP period */
+ unsigned int ssrc; /* Endpoint RTP synchronization source */
int codec; /* Used codec */
struct brcm_pvt *parent; /* brcm_line owning this subchannel */
int cw_timer_id; /* Current call waiting timer id, -1 if no active timer */
@@ -113,8 +115,6 @@ struct brcm_subchannel {
char *conference_id; /* uuid of the conference initiated by this subchannel */
int conf_timer_id; /* Current conference call timer id, -1 if no active timer */
rtp_statistics rtp_stats; /* RTP statistics for currently hanging up channel */
- unsigned int jitter_count;
- unsigned long int farEndInterrivalJitter;
};
@@ -163,8 +163,6 @@ struct brcm_pvt {
enum rtp_type {
BRCM_UNKNOWN,
BRCM_AUDIO,
- BRCM_RTCP_SR,
- BRCM_RTCP_RR
};