From 7f70d818cdefbf726d1519a264cc552ef72049df Mon Sep 17 00:00:00 2001
From: Yalu Zhang <yalu.zhang@iopsys.eu>
Date: Tue, 25 Jun 2024 08:17:56 +0000
Subject: [PATCH] Forward RTP telephone event packets to voicemngr
The frame type of RTP telephone event packets is AST_FRAME_DTMF_BYPASS.
---
src/channels/chan_voicemngr.c | 144 +++++++++++++++++++---------------
1 file changed, 79 insertions(+), 65 deletions(-)
diff --git a/src/channels/chan_voicemngr.c b/src/channels/chan_voicemngr.c
index 00d89b0..188067f 100644
--- a/src/channels/chan_voicemngr.c
+++ b/src/channels/chan_voicemngr.c
@@ -1283,7 +1283,7 @@ static int chan_voicemngr_senddigit_begin(struct ast_channel *ast, char digit)
struct chan_voicemngr_subchannel *sub;
char signal[10];
- //ast_debug(5, "DTMF send begin: %c\n", digit);
+ ast_debug(5, "DTMF send begin: %c\n", digit);
sub = ast_channel_tech_pvt(ast);
pvt_lock(sub->parent, "DTMF senddigit_begin");
@@ -1301,7 +1301,7 @@ static int chan_voicemngr_senddigit_end(struct ast_channel *ast, char digit, uns
struct chan_voicemngr_subchannel *sub;
char signal[10];
- //ast_debug(5, "DTMF send end: %c, %d ms\n", digit, duration);
+ ast_debug(5, "DTMF send end: %c, %d ms\n", digit, duration);
sub = ast_channel_tech_pvt(ast);
pvt_lock(sub->parent, "DTMF senddigit_end");
@@ -1754,7 +1754,7 @@ static int chan_voicemngr_classify_rtp_packet(int payload_type) {
if (voicemngr_codecs[i].rtp_payload_type == payload_type)
return CHAN_VOICEMNGR_AUDIO;
- ast_verbose("Unknown rtp packet payload_type %d\n", payload_type);
+ ast_log(LOG_WARNING, "Unknown RTP payload_type %d\n", payload_type);
return CHAN_VOICEMNGR_UNKNOWN;
}
@@ -1764,147 +1764,161 @@ static int map_ast_codec_id_to_rtp(const struct ast_format *astcodec)
if (ast_format_cmp(astcodec, *voicemngr_codecs[i].ast_format) == AST_FORMAT_CMP_EQUAL)
return voicemngr_codecs[i].rtp_payload_type;
- ast_verbose("Unknown asterisk format/codec\n");
+ ast_log(LOG_WARNING, "Unknown asterisk format(%s), return PCMA\n", ast_format_get_name(astcodec));
return RTP_PT_PCMA;
}
-
static struct ast_frame *chan_voicemngr_read(struct ast_channel *ast)
{
return &ast_null_frame;
}
-/* Handle stream events on audio_bus. Parses raw stream and calls
- registered packet handlers. */
+/* Handle stream events on audio_bus. Parses raw stream and calls registered packet handlers. */
static void audio_rx_stream_handler(pe_stream_t *stream __attribute__((unused)), pe_event_t *event __attribute__((unused))) {
if (pe_bus_receive(audio_rx_bus, event) < 0) {
printf("audio_bus rx buffer full");
- return; // Drop packets if we can't cope the pace
+ return; // Drop packets if we can't cope the pace
}
pe_bus_dispatch(audio_rx_bus);
}
+/* Handle audio packets from Asterisk, i.e. from network */
static int chan_voicemngr_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct chan_voicemngr_subchannel *sub = ast_channel_tech_pvt(ast);
- int packet_size;
- audio_packet_t *ap;
- int sip_client_id = -1;
+ int packet_size, payload_type, sip_client_id = -1;
+ audio_packet_t *ap = NULL;
- if (ast_channel_state(ast) != AST_STATE_UP && ast_channel_state(ast) != AST_STATE_RING) {
- /* Silently ignore packets until channel is up */
- ast_debug(5, "error: channel not up\n");
+ if (sub->channel_state == ONHOLD || (ast_channel_state(ast) != AST_STATE_UP && ast_channel_state(ast) != AST_STATE_RING)) {
+ ast_debug(5, "Silently ignore the frame since the channel is not in the right state\n");
return 0;
}
- /* Ignore if on hold */
- if (sub->channel_state == ONHOLD) {
- return 0;
- }
- 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 ;
+ if (frame->frametype == AST_FRAME_VOICE) {
+ packet_size = sizeof(audio_packet_t) - 1 + RTP_HEADER_SIZE + frame->datalen;
ap = ast_calloc(1, packet_size);
if (!ap) {
ast_log(LOG_ERROR, "Out of memory\n");
return -1;
}
-
ap->line = sub->parent->line_id;
ap->connection_id = sub->parent->line_id;
- ap->rtp_size = RTP_HEADER_SIZE + 1;
+ ap->rtp_size = RTP_HEADER_SIZE + frame->datalen;
/* copy frame data to audio packet */
- /* level = 127 - (level & 0x7f); */
- ap->rtp[12]=127 - (frame->subclass.integer & 0x7f);
+ memcpy(ap->rtp + RTP_HEADER_SIZE, frame->data.ptr, frame->datalen);
- //ast_mutex_lock(&sub->parent->lock);
pvt_lock(sub->parent, "TELCHAN write frame");
/* generate the rtp header */
- chan_voicemngr_generate_rtp_packet(sub, ap->rtp, RTP_PT_CN, frame->seqno, frame->ts, frame->ssrc);
+ payload_type = map_ast_codec_id_to_rtp(frame->subclass.format);
+ chan_voicemngr_generate_rtp_packet(sub, ap->rtp, payload_type, frame->seqno, frame->ts, frame->ssrc);
sip_client_id = chan_voicemngr_get_sip_client_id(sub);
if (sip_client_id >= 0 && sip_client_id < MAX_SIP_CLIENTS) {
- line_stats[sip_client_id].rxpkts++;
- line_stats[sip_client_id].rxbytes += ap->rtp_size;
+ line_stats[sip_client_id].rxpkts++;
+ line_stats[sip_client_id].rxbytes += ap->rtp_size;
}
/* set rtp payload type sent to voicemngr */
- sub->codec = RTP_PT_CN;
+ sub->codec = payload_type;
+ /* in case of media played during calling on an outgoing call and there has no codec set
+ * and dsp do not have the cap to handle any payload type that received. */
+ if (sub->owner && ast_strlen_zero(ast_channel_codec_get(sub->owner))) {
+ ast_channel_codec_set(sub->owner, ast_format_get_name(frame->subclass.format));
+ ast_log(LOG_NOTICE, "set local codec to :%s \n", ast_channel_codec_get(sub->owner));
+ }
- //ast_mutex_unlock(&sub->parent->lock);
pvt_unlock(sub->parent);
- pe_bus_send(audio_tx_bus, (uint8_t *)ap, packet_size);
- ast_free(ap);
- }
-
- if(frame->frametype == AST_FRAME_VOICE) {
- packet_size = sizeof(audio_packet_t) - 1 + RTP_HEADER_SIZE + frame->datalen;
+ chan_voicemngr_modify_codec(sub); // in case of early media on outgoing
+ } else if(frame->frametype == AST_FRAME_RTCP) {
+ packet_size = sizeof(audio_packet_t) - 1 + frame->datalen;
ap = ast_calloc(1, packet_size);
if (!ap) {
ast_log(LOG_ERROR, "Out of memory\n");
return -1;
}
-
ap->line = sub->parent->line_id;
ap->connection_id = sub->parent->line_id;
- ap->rtp_size = RTP_HEADER_SIZE + frame->datalen;
+ ap->rtp_size = frame->datalen;
/* copy frame data to audio packet */
- memcpy(ap->rtp + RTP_HEADER_SIZE, frame->data.ptr, frame->datalen);
+ memcpy(ap->rtp, frame->data.ptr, frame->datalen);
+
+ chan_voicemngr_process_incoming_rtcp_packet(sub, ap->rtp, ap->rtp_size);
+ } else if (frame->frametype == AST_FRAME_CNG) {
+ /* CNG packet has only one byte for the noise level, i.e. frame->datalen=1*/
+ packet_size = sizeof(audio_packet_t) + RTP_HEADER_SIZE ;
+ ap = ast_calloc(1, packet_size);
+ if (!ap) {
+ ast_log(LOG_ERROR, "Out of memory\n");
+ return -1;
+ }
+ ap->line = sub->parent->line_id;
+ ap->connection_id = sub->parent->line_id;
+ ap->rtp_size = RTP_HEADER_SIZE + 1;
+
+ /* 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 */
- chan_voicemngr_generate_rtp_packet(sub, ap->rtp, map_ast_codec_id_to_rtp(frame->subclass.format), frame->seqno, frame->ts, frame->ssrc);
+ chan_voicemngr_generate_rtp_packet(sub, ap->rtp, RTP_PT_CN, frame->seqno, frame->ts, frame->ssrc);
sip_client_id = chan_voicemngr_get_sip_client_id(sub);
if (sip_client_id >= 0 && sip_client_id < MAX_SIP_CLIENTS) {
- line_stats[sip_client_id].rxpkts++;
- line_stats[sip_client_id].rxbytes += ap->rtp_size;
+ line_stats[sip_client_id].rxpkts++;
+ line_stats[sip_client_id].rxbytes += ap->rtp_size;
}
/* set rtp payload type sent to voicemngr */
- sub->codec = map_ast_codec_id_to_rtp(frame->subclass.format);
- //in case of media played during calling on an outgoing call and there has no codec set
- //and dsp do not have the cap to handle any payload type that received.
- if (sub->owner && ast_strlen_zero(ast_channel_codec_get(sub->owner))) {
- ast_channel_codec_set(sub->owner, ast_format_get_name(frame->subclass.format));
- ast_log(LOG_NOTICE, "set local codec to :%s \n", ast_channel_codec_get(sub->owner));
- }
+ sub->codec = RTP_PT_CN;
//ast_mutex_unlock(&sub->parent->lock);
pvt_unlock(sub->parent);
- chan_voicemngr_modify_codec(sub); //in case of early media on outgoing
-
- pe_bus_send(audio_tx_bus, (uint8_t *)ap, packet_size);
- ast_free(ap);
- }
-
- if(frame->frametype == AST_FRAME_RTCP) {
- packet_size = sizeof(audio_packet_t) - 1 + frame->datalen;
+ } else if (frame->frametype == AST_FRAME_DTMF_BYPASS) {
+ ast_debug(0, "RTP for Telephone Event: payload type = %d, len = %d, event = %hhu, timestamp = %ld, seqno = %d\n",
+ frame->subclass.integer, frame->datalen, *(unsigned char *)frame->data.ptr, frame->ts, frame->seqno);
+ packet_size = sizeof(audio_packet_t) - 1 + RTP_HEADER_SIZE + frame->datalen;
ap = ast_calloc(1, packet_size);
if (!ap) {
ast_log(LOG_ERROR, "Out of memory\n");
return -1;
}
-
ap->line = sub->parent->line_id;
ap->connection_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);
- chan_voicemngr_process_incoming_rtcp_packet(sub, ap->rtp, ap->rtp_size);
+ pvt_lock(sub->parent, "TELCHAN write frame");
+ /* generate the rtp header */
+ payload_type = frame->subclass.integer;
+ chan_voicemngr_generate_rtp_packet(sub, ap->rtp, payload_type, frame->seqno, frame->ts, frame->ssrc);
+
+ sip_client_id = chan_voicemngr_get_sip_client_id(sub);
+ if (sip_client_id >= 0 && sip_client_id < MAX_SIP_CLIENTS) {
+ line_stats[sip_client_id].rxpkts++;
+ line_stats[sip_client_id].rxbytes += ap->rtp_size;
+ }
+
+ pvt_unlock(sub->parent);
+ } else {
+ char ftype[40] = "Unknown frame type";
+
+ ast_frame_type2str(frame->frametype, ftype, sizeof(ftype));
+ ast_log(LOG_WARNING, "Unsupported frame type, %s(%d)\n", ftype, frame->frametype);
+ }
+
+ if (ap) {
pe_bus_send(audio_tx_bus, (uint8_t *)ap, packet_size);
ast_free(ap);
}
+
return 0;
}
@@ -3294,7 +3308,7 @@ static void audio_packet_handler(pe_packet_t *p) {
} else if(payload_type == sub->dtmf_pt) {
frame.frametype = AST_FRAME_DTMF_BYPASS;
frame.subclass.integer = ap->rtp[1]; //payload_type
- ast_debug(3, "DTMF rtp_event for digit: %d, with payload_type: %d \n", ap->rtp[12], ap->rtp[1]);
+ ast_debug(3, "DTMF rtp_event for digit: %d, with payload_type: %d \n", ap->rtp[12], payload_type);
if (sub->conference_initiator == 1) {
drop_frame=1;
}
--
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