Add DTMF detection via telephone-event payload

4 files changed, 164 insertions, 25 deletions

diff --git a/src/router/audio.c b/src/router/audio.c
index 0c4b6c3..16e83c0 100644
--- a/src/router/audio.c
+++ b/src/router/audio.c
@@ -127,6 +127,12 @@ void receive_originator(struct osmo_cc_session_codec *codec, uint16_t __attribut
 	len = len / 2;
 	sample_t samples[len];
 
+	if (codec->decoder == decode_te) {
+		struct telephone_event *te = (struct telephone_event *)data;
+		telephone_event(relation, te);
+		return;
+	}
+
 	/* convert int16 to samples */
 	int16_to_samples(samples, (int16_t *)data, len);
 
@@ -169,6 +175,11 @@ void receive_terminator(struct osmo_cc_session_codec *codec, uint16_t __attribut
 	len = len / 2;
 	sample_t samples[len];
 
+	if (codec->decoder == decode_te) {
+		PDEBUG(DROUTER, DEBUG_NOTICE, "Ignoring received telephony-events from terminator.\n");
+		return;
+	}
+
 	int16_to_samples(samples, (int16_t *)data, len);
 
 	/* forward to originator, if not a forking call */
@@ -284,3 +295,29 @@ void decode_l16(uint8_t *src_data, int src_len, uint8_t **dst_data, int *dst_len
 	*dst_len = len * 2;
 }
 
+void encode_te(uint8_t __attribute__((unused)) *src_data, int __attribute__((unused)) src_len, uint8_t **dst_data, int *dst_len)
+{
+	/* FIXME: TBD... */
+	*dst_data = NULL;
+	*dst_len = 0;
+}
+
+void decode_te(uint8_t *src_data, int src_len, uint8_t **dst_data, int *dst_len)
+{
+	uint8_t *src = src_data;
+	struct telephone_event *te;
+
+	if (src_len < 4)
+		return;
+
+	te = calloc(1, sizeof(*te));
+	if (!te)
+		return;
+	te->event = src[0];
+	te->e = src[1] >> 7;
+	te->r = (src[1] >> 6) & 0x1;
+	te->volume = src[1] & 0x3f;
+	te->duration = (src[2] << 16) | src[3];
+	*dst_data = (uint8_t *)te;
+	*dst_len = sizeof(*te);
+}
diff --git a/src/router/audio.h b/src/router/audio.h
index 390e54f..cf2d34c 100644
--- a/src/router/audio.h
+++ b/src/router/audio.h
@@ -1,6 +1,10 @@
+
 void receive_originator(struct osmo_cc_session_codec *codec, uint16_t sequence_number, uint32_t timestamp, uint8_t *data, int len);
 void receive_terminator(struct osmo_cc_session_codec *codec, uint16_t sequence_number, uint32_t timestamp, uint8_t *data, int len);
 void call_media_handle(void);
 void call_clock(int len);
 void encode_l16(uint8_t *src_data, int src_len, uint8_t **dst_data, int *dst_len);
 void decode_l16(uint8_t *src_data, int src_len, uint8_t **dst_data, int *dst_len);
+void encode_te(uint8_t *src_data, int src_len, uint8_t **dst_data, int *dst_len);
+void decode_te(uint8_t *src_data, int src_len, uint8_t **dst_data, int *dst_len);
+
diff --git a/src/router/call.c b/src/router/call.c
index 5b8c28b..13c2844 100644
--- a/src/router/call.c
+++ b/src/router/call.c
@@ -81,10 +81,11 @@ struct timer status_timer;
 static osmo_cc_endpoint_t *cc_ep;
 static const char *routing_script, *routing_shell;
 
-static struct osmo_cc_helper_audio_codecs codecs[] = {
-//	{ "L16", 8000, 1, encode_l16, decode_l16 }, FIXME: make codecs selectable, if more codecs are supported in the future
+static struct osmo_cc_helper_audio_codecs codecs_def[] = {
 	{ "PCMA", 8000, 1, g711_encode_alaw, g711_decode_alaw },
 	{ "PCMU", 8000, 1, g711_encode_ulaw, g711_decode_ulaw },
+	{ "telephone-event", 8000, 1, encode_te, decode_te },
+	{ "L16", 8000, 1, encode_l16, decode_l16 },
 	{ NULL, 0, 0, NULL, NULL},
 };
 
@@ -386,7 +387,7 @@ static void proxy_send_sdp_answer(call_relation_t *relation, osmo_cc_msg_t *msg)
 
 //	if (!relation->codec_negotiated || msg->type == OSMO_CC_MSG_SETUP_CNF) { gibt einen crash, da codec vor der antwort schon gesetzt ist. warum sollten wir nach einer antwort denn nochmal den codec schicken?
 	if (!relation->codec_negotiated) {
-		sdp = osmo_cc_helper_audio_accept(relation, codecs, receive_originator, relation->call->setup_msg, &relation->cc_session, &relation->codec, 0);
+		sdp = osmo_cc_helper_audio_accept(relation, relation->orig_codecs, receive_originator, relation->call->setup_msg, &relation->cc_session, &relation->codec, 0);
 		if (sdp) {
 			relation->codec_negotiated = 1;
 			PDEBUG(DROUTER, DEBUG_DEBUG, "Sending SDP answer to originator with supported codecs.\n");
@@ -1119,7 +1120,7 @@ void cc_message(osmo_cc_endpoint_t __attribute__((unused)) *ep, uint32_t callref
  * process message from routing
  */
 
-static const char *value_of_param(const char *arg, const char *param, const char **value_p)
+static const char *value_of_param(char *arg, char *param, char **value_p)
 {
 	if (!!strncmp(arg, param, strlen(param)))
 		return NULL;
@@ -1140,10 +1141,39 @@ static const char *value_of_param(const char *arg, const char *param, const char
 	return arg;
 }
 
+static void add_codecs_by_names(struct osmo_cc_helper_audio_codecs *codecs, char *names)
+{
+	int c = 0, i;
+	const char *name;
+
+	while ((name = strsep(&names, ","))) {
+	printf("name='%s'\n", name);
+		for (i = 0; codecs_def[i].payload_name; i++) {
+			if (!strcasecmp(name, codecs_def[i].payload_name)) {
+				if (c == MAX_CODECS) {
+					PDEBUG(DROUTER, DEBUG_ERROR, "Maximum of %d codecs are reached. Ignoring codec '%s'.\n", c, name);
+					break;
+				}
+				memcpy(&codecs[c], &codecs_def[i], sizeof(*codecs));
+				PDEBUG(DROUTER, DEBUG_INFO, "Adding given codec '%s'.\n", codecs[c].payload_name);
+				c++;
+				break;
+			}
+		}
+		if (!codecs_def[i].payload_name) {
+			PDEBUG(DROUTER, DEBUG_ERROR, "Given codec '%s' not supported!\n", name);
+		}
+	}
+	memset(&codecs[c], 0, sizeof(*codecs));
+}
+
 /* routing orders us to activate rtp proxy */
-static void routing_rtp_proxy(call_t *call)
+static void routing_rtp_proxy(call_t *call, int argc, char *argv[])
 {
 	call_relation_t *relation = call->relation_list;
+	int orig_given = 0, term_given = 0;
+	int i;
+	char *value;
 
 	/* ignore, if already enabled */
 	if (relation->rtp_proxy) {
@@ -1151,6 +1181,37 @@ static void routing_rtp_proxy(call_t *call)
 		return;
 	}
 
+	/* loop through all arguments and stop if there is a ':' */
+	for (i = 0; i < argc ; i++) {
+		if (value_of_param(argv[i], "orig-codecs", &value)) {
+			PDEBUG(DROUTER, DEBUG_INFO, "Originating codecs given: '%s'.\n", value);
+			add_codecs_by_names(relation->orig_codecs, value);
+			if (relation->orig_codecs[0].payload_name)
+				orig_given = 1;
+		} else if (value_of_param(argv[i], "term-codecs", &value)) {
+			PDEBUG(DROUTER, DEBUG_INFO, "Terminating codecs given: '%s'.\n", value);
+			add_codecs_by_names(relation->term_codecs, value);
+			if (relation->term_codecs[0].payload_name)
+				term_given = 1;
+		} else
+			PDEBUG(DROUTER, DEBUG_ERROR, "Unknown 'rtp-proxy' parameter '%s' from routing.\n", argv[i]);
+	}
+
+	if (!orig_given) {
+		memcpy(&relation->orig_codecs[0], &codecs_def[0], sizeof(relation->orig_codecs[0]));
+		memcpy(&relation->orig_codecs[1], &codecs_def[1], sizeof(relation->orig_codecs[1]));
+		memcpy(&relation->orig_codecs[2], &codecs_def[2], sizeof(relation->orig_codecs[2]));
+		memset(&relation->orig_codecs[3], 0, sizeof(relation->orig_codecs[3]));
+		PDEBUG(DROUTER, DEBUG_INFO, "No originating codeds given, using default '%s' and '%s' and '%s'.\n", relation->orig_codecs[0].payload_name, relation->orig_codecs[1].payload_name, relation->orig_codecs[2].payload_name);
+	}
+
+	if (!term_given) {
+		memcpy(&relation->term_codecs[0], &codecs_def[0], sizeof(relation->term_codecs[0]));
+		memcpy(&relation->term_codecs[1], &codecs_def[1], sizeof(relation->term_codecs[1]));
+		memset(&relation->term_codecs[2], 0, sizeof(relation->term_codecs[2]));
+		PDEBUG(DROUTER, DEBUG_INFO, "No terminating codeds given, using default '%s' and '%s'.\n", relation->term_codecs[0].payload_name, relation->term_codecs[1].payload_name);
+	}
+
 	/* error, if we already negotiated */
 	if (call->sdp_forwarded) {
 		PDEBUG(DROUTER, DEBUG_ERROR, "RTP-Proxy cannot be enabled now, because we already forwarded a call.\n");
@@ -1167,10 +1228,10 @@ static void routing_rtp_proxy(call_t *call)
 }
 
 /* routing orders us to play a wave file */
-static void routing_play(call_t *call, int argc, const char *argv[])
+static void routing_play(call_t *call, int argc, char *argv[])
 {
 	call_relation_t *relation = call->relation_list;
-	const char *filename = NULL, *volume = "1.0", *loop = NULL;
+	char *filename = NULL, *volume = "1.0", *loop = NULL;
 	int i;
 	int samplerate = 8000, channels = 0;
 	double deviation;
@@ -1228,10 +1289,10 @@ static void routing_play_stop(call_t *call)
 }
 
 /* routing orders us to record a wave file */
-static void routing_record(call_t *call, int argc, const char *argv[])
+static void routing_record(call_t *call, int argc, char *argv[])
 {
 	call_relation_t *relation = call->relation_list;
-	const char *filename = NULL, *volume = "1.0";
+	char *filename = NULL, *volume = "1.0";
 	int i;
 	int samplerate = 8000, channels = 2;
 	int rc;
@@ -1274,10 +1335,10 @@ static void routing_record_stop(call_t *call)
 }
 
 /* routing orders us to set local gain */
-static void routing_gain(call_t *call, int argc, const char *argv[], int tx)
+static void routing_gain(call_t *call, int argc, char *argv[], int tx)
 {
 	int i;
-	const char *gain = NULL;
+	char *gain = NULL;
 
 	if (!call->relation_list->rtp_proxy) {
 		PDEBUG(DROUTER, DEBUG_ERROR, "RTP-Proxy must be enabled to record a file!\n");
@@ -1306,15 +1367,15 @@ static void routing_gain(call_t *call, int argc, const char *argv[], int tx)
 }
 
 /* routing orders us to call remote end */
-static void routing_call(call_t *call, int argc, const char *argv[])
+static void routing_call(call_t *call, int argc, char *argv[])
 {
-	const char *interface;
-	const char *bearer_coding, *bearer_capability, *bearer_mode;
-	const char *calling, *calling_type, *calling_plan, *calling_present, *calling_screen, *no_calling;
-	const char *calling2, *calling2_type, *calling2_plan, *calling2_present, *calling2_screen, *no_calling2;
-	const char *redirecting, *redirecting_type, *redirecting_plan, *redirecting_present, *redirecting_screen, *redirecting_reason, *no_redirecting;
-	const char *dialing, *dialing_type, *dialing_plan;
-	const char *keypad;
+	char *interface;
+	char *bearer_coding, *bearer_capability, *bearer_mode;
+	char *calling, *calling_type, *calling_plan, *calling_present, *calling_screen, *no_calling;
+	char *calling2, *calling2_type, *calling2_plan, *calling2_present, *calling2_screen, *no_calling2;
+	char *redirecting, *redirecting_type, *redirecting_plan, *redirecting_present, *redirecting_screen, *redirecting_reason, *no_redirecting;
+	char *dialing, *dialing_type, *dialing_plan;
+	char *keypad;
 	uint8_t coding, capability, mode;
 	uint8_t type, plan, present, screen, reason;
 	char number[256];
@@ -1528,7 +1589,7 @@ next_call:
 	/* only if RTP-Proxy is used */
 	if (call->relation_list->rtp_proxy) {
 		PDEBUG(DROUTER, DEBUG_DEBUG, "Sending our codecs to the terminator.\n");
-		relation->cc_session = osmo_cc_helper_audio_offer(relation, codecs, receive_terminator, new_msg, 1);
+		relation->cc_session = osmo_cc_helper_audio_offer(relation, call->relation_list->term_codecs, receive_terminator, new_msg, 1);
 	} else
 	/* sdp from originator's setup message */
 	if (call->relation_list->sdp)
@@ -1666,7 +1727,7 @@ static void routing_answer(call_t *call)
 }
 
 /* routing orders us to dsiconnect/release the call */
-static void routing_disc_rel(call_t *call, int argc, const char *argv[], int disconnect)
+static void routing_disc_rel(call_t *call, int argc, char *argv[], int disconnect)
 {
 	call_relation_t *relation = call->relation_list;
 	uint8_t cause = 0;
@@ -1746,7 +1807,7 @@ static void routing_dtmf_stop(call_t *call)
 }
 
 /* routing failed, release the call */
-static void routing_error(call_t *call, const char *error)
+static void routing_error(call_t *call, char *error)
 {
 	osmo_cc_msg_t *new_msg;
 	call_relation_t *relation;
@@ -1768,7 +1829,7 @@ static void routing_error(call_t *call, const char *error)
 
 #if 0
 /* routing script says something */
-static void routing_say(call_t *call, const char *error)
+static void routing_say(call_t *call, char *error)
 {
 	char text[1024] = "";
 	fuer alle argv
@@ -1782,7 +1843,7 @@ void routing_receive_stdout(routing_t *routing, const char *string)
 {
 	call_t *call = routing->call;
 	int argc = 0;
-	const char *argv[256], *token;
+	char *argv[256], *token;
 
 	/* convert string into tokens */
 	while ((token = osmo_cc_strtok_quotes(&string)))
@@ -1791,7 +1852,7 @@ void routing_receive_stdout(routing_t *routing, const char *string)
 		return;
 
 	if (!strcasecmp(argv[0], "rtp-proxy"))
-		routing_rtp_proxy(call);
+		routing_rtp_proxy(call, argc - 1, argv + 1);
 	else
 	if (!strcasecmp(argv[0], "play"))
 		routing_play(call, argc - 1, argv + 1);
@@ -1899,6 +1960,30 @@ void routing_close(routing_t *routing)
 	call_destroy(call);
 }
 
+void telephone_event(call_relation_t *relation, struct telephone_event *te)
+{
+	char digit_string[7] = "dtmf x";
+
+	if (te->event < 16) {
+		if (!relation->te_started && !te->e && te->volume <= 36) {
+			PDEBUG(DROUTER, DEBUG_INFO, "Received start of Telephone-Event '%d'\n", te->event);
+			relation->te_started = 1;
+			digit_string[5] = "0123456789*#ABCD"[te->event];
+		}
+		if (relation->te_started && te->e) {
+			PDEBUG(DROUTER, DEBUG_INFO, "Received end of Telephone-Event '%d'\n", te->event);
+			relation->te_started = 0;
+		}
+	} else
+		PDEBUG(DROUTER, DEBUG_INFO, "Received unsupported Telephone-Event '%d'\n", te->event);
+
+	if (!relation->call->routing.routing)
+		return;
+
+	if (digit_string[5] != 'x')
+		routing_send(&relation->call->routing, digit_string);
+}
+
 #warning add progress, if terminating call sends sdp but call state already reached
 
 #warning beim disc muss progress geprueft werden und damit entschieden ob wir audio mitsenden sollen
diff --git a/src/router/call.h b/src/router/call.h
index 580f267..e782c26 100644
--- a/src/router/call.h
+++ b/src/router/call.h
@@ -21,6 +21,8 @@ enum call_state {
 
 #include "routing.h"
 
+#define MAX_CODECS	8
+
 /* relation to upper layer */
 typedef struct call_relation {
 	struct call_relation	*next;
@@ -39,6 +41,8 @@ typedef struct call_relation {
 	int			rtp_proxy;
 	osmo_cc_session_t	*cc_session;
 	int			codec_negotiated;
+	struct osmo_cc_helper_audio_codecs orig_codecs[MAX_CODECS + 1];	/* codecs for originator */
+	struct osmo_cc_helper_audio_codecs term_codecs[MAX_CODECS + 1]; /* codecs for terminator, stored at relation of originator */
 	osmo_cc_session_codec_t	*codec;
 
 	wave_play_t		play;		/* play a wave file */
@@ -48,6 +52,7 @@ typedef struct call_relation {
 	wave_rec_t		rec;		/* record a wave file */
 	dtmf_dec_t		dtmf_dec;	/* dtmf decoder */
 	int			dtmf_dec_enable;/* feed decoder with data */
+	int			te_started;	/* we got a digit via telephone-event */
 } call_relation_t;
 
 /* call instance */
@@ -77,8 +82,16 @@ typedef struct call {
 
 extern call_t *call_list;
 
+struct telephone_event {
+	uint8_t	event;
+	uint8_t e, r;
+	uint8_t volume;
+	uint16_t duration;
+};
+
 int call_init(osmo_cc_endpoint_t *ep, const char *_routing_script, const char *_routing_shell);
 void call_exit(void);
 int call_handle(void);
 void cc_message(osmo_cc_endpoint_t *ep, uint32_t callref, osmo_cc_msg_t *msg);
+void telephone_event(call_relation_t *relation, struct telephone_event *te);