X-Git-Url: http://git.eversberg.eu/gitweb.cgi?p=lcr.git;a=blobdiff_plain;f=sip.cpp;h=075447cb347463dde39dffc74678b9fa8e089cb0;hp=c6f36ac0bb50cdfd9a61487152676266c4087547;hb=5566f74eb29be75da44e29ba72ee6f015249ce61;hpb=507d22099d8b7fce31bf2d575f609cc8d94131ea

diff --git a/sip.cpp b/sip.cpp
index c6f36ac..075447c 100644
--- a/sip.cpp
+++ b/sip.cpp
@@ -15,27 +15,38 @@
 #include <sofia-sip/sdp.h>
 #include <sofia-sip/sip_header.h>
 
+#ifndef SOFIA_SIP_GCC_4_8_PATCH_APLLIED
+#warning ********************************************************
+#warning Please apply the sofia-sip-gcc-4.8.patch !
+#warning If this issue is already fixed, just remove this check.
+#warning ********************************************************
+#error
+#endif
+
 #undef NUTAG_AUTO100
 
 unsigned char flip[256];
 
+int any_sip_interface = 0;
+
 //pthread_mutex_t mutex_msg;
 su_home_t	sip_home[1];
 
 struct sip_inst {
 	char			interface_name[64];
-	char			local_ip[16];
-	char			remote_ip[16];
+	char			local_peer[32];
+	char			remote_peer[32];
 	su_root_t		*root;
 	nua_t			*nua;
 };
 
 static int delete_event(struct lcr_work *work, void *instance, int index);
+static int load_timer(struct lcr_timer *timer, void *instance, int index);
 
 /*
  * initialize SIP port
  */
-Psip::Psip(int type, char *portname, struct port_settings *settings, struct interface *interface) : Port(type, portname, settings)
+Psip::Psip(int type, char *portname, struct port_settings *settings, struct interface *interface) : Port(type, portname, settings, interface)
 {
 	p_s_rtp_bridge = 0;
 	if (interface->rtp_bridge)
@@ -61,7 +72,14 @@ Psip::Psip(int type, char *portname, struct port_settings *settings, struct inte
 	p_s_rxpos = 0;
 	p_s_rtp_tx_action = 0;
 
+	/* audio */
+	memset(&p_s_loadtimer, 0, sizeof(p_s_loadtimer));
+	add_timer(&p_s_loadtimer, load_timer, this, 0);
+	p_s_next_tv_sec = 0;
+
 	PDEBUG(DEBUG_SIP, "Created new Psip(%s).\n", portname);
+	if (!p_s_sip_inst)
+		FATAL("No SIP instance for interface\n");
 }
 
 
@@ -72,22 +90,26 @@ Psip::~Psip()
 {
 	PDEBUG(DEBUG_SIP, "Destroyed SIP process(%s).\n", p_name);
 
+	del_timer(&p_s_loadtimer);
 	del_work(&p_s_delete);
 
 	rtp_close();
 }
 
-const char *payload_type2name(uint8_t payload_type) {
-	switch (payload_type) {
-	case 0:
+static const char *media_type2name(uint8_t media_type) {
+	switch (media_type) {
+	case MEDIA_TYPE_ULAW:
 		return "PCMU";
-	case 8:
+	case MEDIA_TYPE_ALAW:
 		return "PCMA";
-	case 3:
-	case 96:
-	case 97:
-	case 98:
+	case MEDIA_TYPE_GSM:
 		return "GSM";
+	case MEDIA_TYPE_GSM_HR:
+		return "GSM-HR";
+	case MEDIA_TYPE_GSM_EFR:
+		return "GSM-EFR";
+	case MEDIA_TYPE_AMR:
+		return "AMR";
 	}
 
 	return "UKN";
@@ -139,12 +161,9 @@ struct rtp_x_hdr {
 
 #define RTP_VERSION	2
 
-#define RTP_PT_ULAW 0
-#define RTP_PT_ALAW 8
-#define RTP_PT_GSM_FULL 3
-#define RTP_PT_GSM_HALF 96
-#define RTP_PT_GSM_EFR 97
-#define RTP_PT_GSM_AMR 98
+#define PAYLOAD_TYPE_ULAW 0
+#define PAYLOAD_TYPE_ALAW 8
+#define PAYLOAD_TYPE_GSM 3
 
 /* decode an rtp frame  */
 static int rtp_decode(class Psip *psip, unsigned char *data, int len)
@@ -203,6 +222,8 @@ static int rtp_decode(class Psip *psip, unsigned char *data, int len)
 	}
 
 	switch (rtph->payload_type) {
+#if 0
+we only support alaw and ulaw!
 	case RTP_PT_GSM_FULL:
 		if (payload_len != 33) {
 			PDEBUG(DEBUG_SIP, "received RTP full rate frame with "
@@ -227,13 +248,14 @@ static int rtp_decode(class Psip *psip, unsigned char *data, int len)
 			return -EINVAL;
 		}
 		break;
-	case RTP_PT_ALAW:
+#endif
+	case PAYLOAD_TYPE_ALAW:
 		if (options.law != 'a') {
 			PDEBUG(DEBUG_SIP, "received Alaw, but we don't do Alaw\n");
 			return -EINVAL;
 		}
 		break;
-	case RTP_PT_ULAW:
+	case PAYLOAD_TYPE_ULAW:
 		if (options.law == 'a') {
 			PDEBUG(DEBUG_SIP, "received Ulaw, but we don't do Ulaw\n");
 			return -EINVAL;
@@ -250,11 +272,24 @@ static int rtp_decode(class Psip *psip, unsigned char *data, int len)
 		return 0;
 	}
 
+	/* record audio */
+	if (psip->p_record)
+		psip->record(payload, payload_len, 0); // from down
+	if (psip->p_tap)
+		psip->tap(payload, payload_len, 0); // from down
+
 	n = payload_len;
 	from = payload;
 	to = payload;
+	if (psip->p_echotest) {
+		/* echo rtp data we just received */
+		psip->rtp_send_frame(from, n, (options.law=='a')?PAYLOAD_TYPE_ALAW:PAYLOAD_TYPE_ULAW);
+		return 0;
+	}
 	while(n--)
 		*to++ = flip[*from++];
+	if (psip->p_dov_rx)
+		psip->dov_rx(payload, payload_len);
 	psip->bridge_tx(payload, payload_len);
 
 	return 0;
@@ -275,7 +310,8 @@ static int rtp_sock_callback(struct lcr_fd *fd, unsigned int what, void *instanc
 //			psip->rtp_shutdown();
 			return len;
 		}
-		rc = rtp_decode(psip, buffer, len);
+		if (psip->p_s_rtp_is_connected)
+			rc = rtp_decode(psip, buffer, len);
 	}
 
 	return rc;
@@ -302,7 +338,8 @@ static int rtcp_sock_callback(struct lcr_fd *fd, unsigned int what, void *instan
 }
 
 #define RTP_PORT_BASE	30000
-static unsigned int next_udp_port = RTP_PORT_BASE;
+#define RTP_PORT_MAX	39998
+static unsigned short next_udp_port = RTP_PORT_BASE;
 
 static int rtp_sub_socket_bind(int fd, struct sockaddr_in *sin_local, uint32_t ip, uint16_t port)
 {
@@ -342,13 +379,14 @@ static int rtp_sub_socket_connect(int fd, struct sockaddr_in *sin_local, struct
 
 int Psip::rtp_open(void)
 {
-	int rc;
+	int rc, rc2;
 	struct in_addr ia;
 	unsigned int ip;
+	unsigned short start_port;
 
 	/* create socket */
 	rc = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
-	if (!rc) {
+	if (rc < 0) {
 		rtp_close();
 		return -EIO;
 	}
@@ -356,7 +394,7 @@ int Psip::rtp_open(void)
 	register_fd(&p_s_rtp_fd, LCR_FD_READ, rtp_sock_callback, this, 0);
 
 	rc = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
-	if (!rc) {
+	if (rc < 0) {
 		rtp_close();
 		return -EIO;
 	}
@@ -366,22 +404,41 @@ int Psip::rtp_open(void)
 	/* bind socket */
 	ip = htonl(INADDR_ANY);
 	ia.s_addr = ip;
-	for (next_udp_port = next_udp_port % 0xffff;
-	     next_udp_port < 0xffff; next_udp_port += 2) {
+	start_port = next_udp_port;
+	while (1) {
 		rc = rtp_sub_socket_bind(p_s_rtp_fd.fd, &p_s_rtp_sin_local, ip, next_udp_port);
 		if (rc != 0)
-			continue;
+			goto try_next_port;
 
-		rc = rtp_sub_socket_bind(p_s_rtcp_fd.fd, &p_s_rtcp_sin_local, ip, next_udp_port+1);
-		if (rc == 0)
+		rc = rtp_sub_socket_bind(p_s_rtcp_fd.fd, &p_s_rtcp_sin_local, ip, next_udp_port + 1);
+		if (rc == 0) {
+			p_s_rtp_port_local = next_udp_port;
+			next_udp_port = (next_udp_port + 2 > RTP_PORT_MAX) ? RTP_PORT_BASE : next_udp_port + 2;
 			break;
+		}
+		/* reopen rtp socket and try again with next udp port */
+		unregister_fd(&p_s_rtp_fd);
+		close(p_s_rtp_fd.fd);
+		p_s_rtp_fd.fd = 0;
+		rc2 = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
+		if (rc2 < 0) {
+			rtp_close();
+			return -EIO;
+		}
+		p_s_rtp_fd.fd = rc2;
+		register_fd(&p_s_rtp_fd, LCR_FD_READ, rtp_sock_callback, this, 0);
+
+try_next_port:
+		next_udp_port = (next_udp_port + 2 > RTP_PORT_MAX) ? RTP_PORT_BASE : next_udp_port + 2;
+		if (next_udp_port == start_port)
+			break;
+		/* we must use rc2, in order to preserve rc */
 	}
 	if (rc < 0) {
 		PDEBUG(DEBUG_SIP, "failed to find port\n");
 		rtp_close();
 		return rc;
 	}
-	p_s_rtp_port_local = next_udp_port;
 	p_s_rtp_ip_local = ntohl(p_s_rtp_sin_local.sin_addr.s_addr);
 	PDEBUG(DEBUG_SIP, "local ip %08x port %d\n", p_s_rtp_ip_local, p_s_rtp_port_local);
 	PDEBUG(DEBUG_SIP, "remote ip %08x port %d\n", p_s_rtp_ip_remote, p_s_rtp_port_remote);
@@ -453,6 +510,12 @@ int Psip::rtp_send_frame(unsigned char *data, unsigned int len, uint8_t payload_
 	int duration; /* in samples */
 	unsigned char buffer[256];
 
+	/* record audio */
+	if (p_record)
+		record(data, len, 1); // from up
+	if (p_tap)
+		tap(data, len, 1); // from up
+
 	if (!p_s_rtp_is_connected) {
 		/* drop silently */
 		return 0;
@@ -468,6 +531,8 @@ int Psip::rtp_send_frame(unsigned char *data, unsigned int len, uint8_t payload_
 	}
 
 	switch (payload_type) {
+#if 0
+we only support alaw and ulaw!
 	case RTP_PT_GSM_FULL:
 		payload_len = 33;
 		duration = 160;
@@ -480,8 +545,9 @@ int Psip::rtp_send_frame(unsigned char *data, unsigned int len, uint8_t payload_
 		payload_len = 14;
 		duration = 160;
 		break;
-	case RTP_PT_ALAW:
-	case RTP_PT_ULAW:
+#endif
+	case PAYLOAD_TYPE_ALAW:
+	case PAYLOAD_TYPE_ULAW:
 		payload_len = len;
 		duration = len;
 		break;
@@ -541,6 +607,18 @@ int Psip::rtp_send_frame(unsigned char *data, unsigned int len, uint8_t payload_
 /* receive from remote */
 int Psip::bridge_rx(unsigned char *data, int len)
 {
+	int ret;
+
+	/* don't bridge, if tones are provided */
+	if (p_tone_name[0] || p_dov_tx)
+		return -EBUSY;
+
+	if (p_dov_tx)
+		return -EBUSY;
+
+	if ((ret = Port::bridge_rx(data, len)))
+		return ret;
+
 	/* write to rx buffer */
 	while(len--) {
 		p_s_rxdata[p_s_rxpos++] = flip[*data++];
@@ -548,7 +626,7 @@ int Psip::bridge_rx(unsigned char *data, int len)
 			p_s_rxpos = 0;
 
 			/* transmit data via rtp */
-			rtp_send_frame(p_s_rxdata, 160, (options.law=='a')?RTP_PT_ALAW:RTP_PT_ULAW);
+			rtp_send_frame(p_s_rxdata, 160, (options.law=='a')?PAYLOAD_TYPE_ALAW:PAYLOAD_TYPE_ULAW);
 		}
 	}
 
@@ -733,11 +811,13 @@ int Psip::message_connect(unsigned int epoint_id, int message_id, union paramete
 	char sdp_str[256];
 	struct in_addr ia;
 	struct lcr_msg *message;
+	int media_type;
 	unsigned char payload_type;
 
 	if (param->connectinfo.rtpinfo.port) {
 		PDEBUG(DEBUG_SIP, "RTP info given by remote, forward that\n");
 		p_s_rtp_bridge = 1;
+		media_type = param->connectinfo.rtpinfo.media_types[0];
 		payload_type = param->connectinfo.rtpinfo.payload_types[0];
 		p_s_rtp_ip_local = param->connectinfo.rtpinfo.ip;
 		p_s_rtp_port_local = param->connectinfo.rtpinfo.port;
@@ -746,7 +826,8 @@ int Psip::message_connect(unsigned int epoint_id, int message_id, union paramete
 		PDEBUG(DEBUG_SIP, "remote ip %08x port %d\n", p_s_rtp_ip_remote, p_s_rtp_port_remote);
 	} else {
 		PDEBUG(DEBUG_SIP, "RTP info not given by remote, so we do our own RTP\n");
-		payload_type = (options.law=='a') ? RTP_PT_ALAW : RTP_PT_ULAW;
+		media_type = (options.law=='a') ? MEDIA_TYPE_ALAW : MEDIA_TYPE_ULAW;
+		payload_type = (options.law=='a') ? PAYLOAD_TYPE_ALAW : PAYLOAD_TYPE_ULAW;
 		/* open local RTP peer (if not bridging) */
 		if (!p_s_rtp_is_connected && rtp_connect() < 0) {
 			nua_cancel(p_s_handle, TAG_END());
@@ -775,9 +856,15 @@ int Psip::message_connect(unsigned int epoint_id, int message_id, union paramete
 		"t=0 0\n"
 		"m=audio %d RTP/AVP %d\n"
 		"a=rtpmap:%d %s/8000\n"
-		, inet_ntoa(ia), inet_ntoa(ia), p_s_rtp_port_local, payload_type, payload_type, payload_type2name(payload_type));
+		, inet_ntoa(ia), inet_ntoa(ia), p_s_rtp_port_local, payload_type, payload_type, media_type2name(media_type));
 	PDEBUG(DEBUG_SIP, "Using SDP response: %s\n", sdp_str);
 
+	/* NOTE:
+	 * If this response causes corrupt messages, like SDP body inside or
+	 * before header, check if the sofia-sip-gcc-4.8.patch was applied.
+	 * If it is still corrupted, try to disable optimization when compiling
+	 * sofia-sip.
+	 */
 	nua_respond(p_s_handle, SIP_200_OK,
 		NUTAG_MEDIA_ENABLE(0),
 		SIPTAG_CONTENT_TYPE_STR("application/sdp"),
@@ -788,7 +875,7 @@ int Psip::message_connect(unsigned int epoint_id, int message_id, union paramete
 	add_trace("reason", NULL, "call connected");
 	add_trace("rtp", "ip", "%s", inet_ntoa(ia));
 	add_trace("rtp", "port", "%d,%d", p_s_rtp_port_local, p_s_rtp_port_local + 1);
-	add_trace("rtp", "payload", "%d", payload_type);
+	add_trace("rtp", "payload", "%s:%d", media_type2name(media_type), payload_type);
 	end_trace();
 
 	return 0;
@@ -863,14 +950,17 @@ int Psip::message_setup(unsigned int epoint_id, int message_id, union parameter
 	struct sip_inst *inst = (struct sip_inst *) p_s_sip_inst;
 	char from[128];
 	char to[128];
-	const char *local = inst->local_ip;
-	const char *remote = inst->remote_ip;
+	const char *local = inst->local_peer;
+	char local_ip[16];
+	const char *remote = inst->remote_peer;
 	char sdp_str[512], pt_str[32];
 	struct in_addr ia;
 	struct epoint_list *epointlist;
 	sip_cseq_t *cseq = NULL;
 	struct lcr_msg *message;
-	unsigned char lcr_payload = { (options.law=='a') ? RTP_PT_ALAW : RTP_PT_ULAW };
+	int lcr_media = { (options.law=='a') ? MEDIA_TYPE_ALAW : MEDIA_TYPE_ULAW };
+	unsigned char lcr_payload = { (options.law=='a') ? PAYLOAD_TYPE_ALAW : PAYLOAD_TYPE_ULAW };
+	int *media_types;
 	unsigned char *payload_types;
 	int payloads = 0;
 	int i;
@@ -884,6 +974,7 @@ int Psip::message_setup(unsigned int epoint_id, int message_id, union parameter
 	if (param->setup.rtpinfo.port) {
 		PDEBUG(DEBUG_SIP, "RTP info given by remote, forward that\n");
 		p_s_rtp_bridge = 1;
+		media_types = param->setup.rtpinfo.media_types;
 		payload_types = param->setup.rtpinfo.payload_types;
 		payloads = param->setup.rtpinfo.payloads;
 		p_s_rtp_ip_local = param->setup.rtpinfo.ip;
@@ -893,6 +984,7 @@ int Psip::message_setup(unsigned int epoint_id, int message_id, union parameter
 	} else {
 		PDEBUG(DEBUG_SIP, "RTP info not given by remote, so we do our own RTP\n");
 		p_s_rtp_bridge = 0;
+		media_types = &lcr_media;
 		payload_types = &lcr_payload;
 		payloads = 1;
 
@@ -928,8 +1020,15 @@ int Psip::message_setup(unsigned int epoint_id, int message_id, union parameter
 	end_trace();
 
 	if (!p_s_rtp_ip_local) {
-		PDEBUG(DEBUG_SIP, "RTP local IP not known, so we use our local SIP ip %s\n", local);
-		inet_pton(AF_INET, local, &p_s_rtp_ip_local);
+		char *p;
+
+		/* extract IP from local peer */
+		SCPY(local_ip, local);
+		p = strchr(local_ip, ':');
+		if (p)
+			*p = '\0';
+		PDEBUG(DEBUG_SIP, "RTP local IP not known, so we use our local SIP ip %s\n", local_ip);
+		inet_pton(AF_INET, local_ip, &p_s_rtp_ip_local);
 		p_s_rtp_ip_local = ntohl(p_s_rtp_ip_local);
 	}
 	ia.s_addr = htonl(p_s_rtp_ip_local);
@@ -947,7 +1046,7 @@ int Psip::message_setup(unsigned int epoint_id, int message_id, union parameter
 	}
 	SCAT(sdp_str, "\n");
 	for (i = 0; i < payloads; i++) {
-		SPRINT(pt_str, "a=rtpmap:%d %s/8000\n", payload_types[i], payload_type2name(payload_types[i]));
+		SPRINT(pt_str, "a=rtpmap:%d %s/8000\n", payload_types[i], media_type2name(media_types[i]));
 		SCAT(sdp_str, pt_str);
 	}
 	PDEBUG(DEBUG_SIP, "Using SDP for invite: %s\n", sdp_str);
@@ -961,7 +1060,7 @@ int Psip::message_setup(unsigned int epoint_id, int message_id, union parameter
 	add_trace("rtp", "ip", "%s", inet_ntoa(ia));
 	add_trace("rtp", "port", "%d,%d", p_s_rtp_port_local, p_s_rtp_port_local + 1);
 	for (i = 0; i < payloads; i++)
-		add_trace("rtp", "payload", "%d", payload_types[i]);
+		add_trace("rtp", "payload", "%s:%d", media_type2name(media_types[i]), payload_types[i]);
 	end_trace();
 
 //	cseq = sip_cseq_create(sip_home, 123, SIP_METHOD_INVITE);
@@ -1036,7 +1135,7 @@ int Psip::message_notify(unsigned int epoint_id, int message_id, union parameter
 			"t=0 0\n"
 			"m=audio %d RTP/AVP %d\n"
 			"a=rtpmap:%d %s/8000\n"
-			, inet_ntoa(ia), inet_ntoa(ia), p_s_rtp_port_local, p_s_rtp_payload_type, p_s_rtp_payload_type, payload_type2name(p_s_rtp_payload_type));
+			, inet_ntoa(ia), inet_ntoa(ia), p_s_rtp_port_local, p_s_rtp_payload_type, p_s_rtp_payload_type, media_type2name(p_s_rtp_media_type));
 		PDEBUG(DEBUG_SIP, "Using SDP for rertieve: %s\n", sdp_str);
 		nua_info(p_s_handle,
 //			TAG_IF(from[0], SIPTAG_FROM_STR(from)),
@@ -1146,7 +1245,7 @@ int Psip::message_epoint(unsigned int epoint_id, int message_id, union parameter
 	return 0;
 }
 
-int Psip::parse_sdp(sip_t const *sip, unsigned int *ip, unsigned short *port, uint8_t *payload_types, int *payloads, int max_payloads)
+int Psip::parse_sdp(sip_t const *sip, unsigned int *ip, unsigned short *port, uint8_t *payload_types, int *media_types, int *payloads, int max_payloads)
 {
 	*payloads = 0;
 
@@ -1206,11 +1305,26 @@ int Psip::parse_sdp(sip_t const *sip, unsigned int *ip, unsigned short *port, ui
 			*ip = ntohl(p_s_rtp_ip_remote);
 		}
 		for (map = m->m_rtpmaps; map; map = map->rm_next) {
+			int media_type = 0;
+
 			PDEBUG(DEBUG_SIP, "RTPMAP: coding:'%s' rate='%d' pt='%d'\n", map->rm_encoding, map->rm_rate, map->rm_pt);
 			/* append to payload list, if there is space */
-			add_trace("rtp", "payload", "%d", map->rm_pt);
-			if (*payloads <= max_payloads) {
+			add_trace("rtp", "payload", "%s:%d", map->rm_encoding, map->rm_pt);
+			if (map->rm_pt == PAYLOAD_TYPE_ALAW)
+				media_type = MEDIA_TYPE_ALAW;
+			else if (map->rm_pt == PAYLOAD_TYPE_ULAW)
+				media_type = MEDIA_TYPE_ULAW;
+			else if (map->rm_pt == PAYLOAD_TYPE_GSM)
+				media_type = MEDIA_TYPE_GSM;
+			else if (!strcmp(map->rm_encoding, "GSM-EFR"))
+				media_type = MEDIA_TYPE_GSM_EFR;
+			else if (!strcmp(map->rm_encoding, "AMR"))
+				media_type = MEDIA_TYPE_AMR;
+			else if (!strcmp(map->rm_encoding, "GSM-HR"))
+				media_type = MEDIA_TYPE_GSM_HR;
+			if (media_type && *payloads <= max_payloads) {
 				*payload_types++ = map->rm_pt;
+				*media_types++ = media_type;
 				(*payloads)++;
 			}
 		}
@@ -1224,13 +1338,16 @@ int Psip::parse_sdp(sip_t const *sip, unsigned int *ip, unsigned short *port, ui
 void Psip::i_invite(int status, char const *phrase, nua_t *nua, nua_magic_t *magic, nua_handle_t *nh, nua_hmagic_t *hmagic, sip_t const *sip, tagi_t tagss[])
 {
 	struct sip_inst *inst = (struct sip_inst *) p_s_sip_inst;
-	const char *from = "", *to = "";
+	const char *from = "", *to = "", *name = "";
+	char imsi[16] = "";
 	int ret;
 	class Endpoint *epoint;
 	struct lcr_msg *message;
 	struct interface *interface;
+	int media_types[32];
 	uint8_t payload_types[32];
 	int payloads = 0;
+	int media_type;
 
 	interface = getinterfacebyname(inst->interface_name);
 	if (!interface) {
@@ -1238,14 +1355,26 @@ void Psip::i_invite(int status, char const *phrase, nua_t *nua, nua_magic_t *mag
 		return;
 	}
 
-	if (sip->sip_from && sip->sip_from->a_url)
-		from = sip->sip_from->a_url->url_user;
-	if (sip->sip_to && sip->sip_to->a_url)
-		to = sip->sip_to->a_url->url_user;
+	if (sip->sip_from) {
+		if (sip->sip_from->a_url)
+			from = sip->sip_from->a_url->url_user;
+		if (sip->sip_from->a_display) {
+			name = sip->sip_from->a_display;
+			if (!strncmp(name, "\"IMSI", 5)) {
+				strncpy(imsi, name + 5, 15);
+				imsi[15] = '\0';
+				name = "";
+			}
+		}
+	}
+	if (sip->sip_to) {
+		if (sip->sip_to->a_url)
+			to = sip->sip_to->a_url->url_user;
+	}
 	PDEBUG(DEBUG_SIP, "invite received (%s->%s)\n", from, to);
 
 	sip_trace_header(this, "Payload received", DIRECTION_NONE);
-	ret = parse_sdp(sip, &p_s_rtp_ip_remote, &p_s_rtp_port_remote, payload_types, &payloads, sizeof(payload_types));
+	ret = parse_sdp(sip, &p_s_rtp_ip_remote, &p_s_rtp_port_remote, payload_types, media_types, &payloads, sizeof(payload_types));
 	if (!ret) {
 		/* if no RTP bridge, we must support LAW codec, otherwise we forward what we have */
 		if (!p_s_rtp_bridge) {
@@ -1253,7 +1382,7 @@ void Psip::i_invite(int status, char const *phrase, nua_t *nua, nua_magic_t *mag
 
 			/* check if supported payload type exists */
 			for (i = 0; i < payloads; i++) {
-				if (payload_types[i] == ((options.law=='a') ? RTP_PT_ALAW : RTP_PT_ULAW))
+				if (media_types[i] == ((options.law=='a') ? MEDIA_TYPE_ALAW : MEDIA_TYPE_ULAW))
 					break;
 			}
 			if (i == payloads) {
@@ -1303,7 +1432,7 @@ void Psip::i_invite(int status, char const *phrase, nua_t *nua, nua_magic_t *mag
 	end_trace();
 
 	sip_trace_header(this, "INVITE", DIRECTION_IN);
-	add_trace("RTP", "port", "%d", p_s_rtp_port_remote);
+	add_trace("rtp", "port", "%d", p_s_rtp_port_remote);
 	/* caller information */
 	if (!from[0]) {
 		p_callerinfo.present = INFO_PRESENT_NOTAVAIL;
@@ -1316,6 +1445,12 @@ void Psip::i_invite(int status, char const *phrase, nua_t *nua, nua_magic_t *mag
 		p_callerinfo.ntype = INFO_NTYPE_UNKNOWN;
 		SCPY(p_callerinfo.id, from);
 		add_trace("calling", "number", "%s", from);
+		SCPY(p_callerinfo.name, name);
+		if (name[0])
+			add_trace("calling", "name", "%s", name);
+		SCPY(p_callerinfo.imsi, imsi);
+		if (imsi[0])
+			add_trace("calling", "imsi", "%s", imsi);
 	}
 	SCPY(p_callerinfo.interface, inst->interface_name);
 	/* dialing information */
@@ -1372,16 +1507,15 @@ void Psip::i_invite(int status, char const *phrase, nua_t *nua, nua_magic_t *mag
 		message->param.setup.rtpinfo.port = p_s_rtp_port_remote;
 		/* add codecs to setup message */
 		for (i = 0; i < payloads; i++) {
-			message->param.setup.rtpinfo.payload_types[message->param.setup.rtpinfo.payloads++] = payload_types[i];
-			if (message->param.setup.rtpinfo.payloads == sizeof(message->param.setup.rtpinfo.payload_types))
+			message->param.setup.rtpinfo.media_types[i] = media_types[i];
+			message->param.setup.rtpinfo.payload_types[i] = payload_types[i];
+			if (i == sizeof(message->param.setup.rtpinfo.payload_types))
 				break;
 		}
+		message->param.setup.rtpinfo.payloads = i;
 	}
 	message_put(message);
 
-#if 0
-issues:
-- send tones that are clocked with timer, unless data is received from bridge
 	/* send progress, if tones are available and if we don't bridge */
 	if (!p_s_rtp_bridge && interface->is_tones == IS_YES) {
 		char sdp_str[256];
@@ -1389,7 +1523,8 @@ issues:
 		unsigned char payload_type;
 
 		PDEBUG(DEBUG_SIP, "Connecting audio, since we have tones available\n");
-		payload_type = (options.law=='a') ? RTP_PT_ALAW : RTP_PT_ULAW;
+		media_type = (options.law=='a') ? MEDIA_TYPE_ALAW : MEDIA_TYPE_ULAW;
+		payload_type = (options.law=='a') ? PAYLOAD_TYPE_ALAW : PAYLOAD_TYPE_ULAW;
 		/* open local RTP peer (if not bridging) */
 		if (rtp_connect() < 0) {
 			nua_respond(nh, SIP_500_INTERNAL_SERVER_ERROR, TAG_END());
@@ -1420,23 +1555,21 @@ issues:
 			"t=0 0\n"
 			"m=audio %d RTP/AVP %d\n"
 			"a=rtpmap:%d %s/8000\n"
-			, inet_ntoa(ia), inet_ntoa(ia), p_s_rtp_port_local, payload_type, payload_type, payload_type2name(payload_type));
+			, inet_ntoa(ia), inet_ntoa(ia), p_s_rtp_port_local, payload_type, payload_type, media_type2name(media_type));
 		PDEBUG(DEBUG_SIP, "Using SDP response: %s\n", sdp_str);
 
 		nua_respond(p_s_handle, SIP_183_SESSION_PROGRESS,
 			NUTAG_MEDIA_ENABLE(0),
 			SIPTAG_CONTENT_TYPE_STR("application/sdp"),
 			SIPTAG_PAYLOAD_STR(sdp_str), TAG_END());
-		new_state(PORT_STATE_CONNECT);
 		sip_trace_header(this, "RESPOND", DIRECTION_OUT);
 		add_trace("respond", "value", "183 SESSION PROGRESS");
 		add_trace("reason", NULL, "audio available");
 		add_trace("rtp", "ip", "%s", inet_ntoa(ia));
 		add_trace("rtp", "port", "%d,%d", p_s_rtp_port_local, p_s_rtp_port_local + 1);
-		add_trace("rtp", "payload", "%d", payload_type);
+		add_trace("rtp", "payload", "%s:%d", media_type2name(media_type), payload_type);
 		end_trace();
 	}
-#endif
 }
 
 void Psip::i_bye(int status, char const *phrase, nua_t *nua, nua_magic_t *magic, nua_handle_t *nh, nua_hmagic_t *hmagic, sip_t const *sip, tagi_t tagss[])
@@ -1531,6 +1664,7 @@ void Psip::r_invite(int status, char const *phrase, nua_t *nua, nua_magic_t *mag
 {
 	struct lcr_msg *message;
 	int cause = 0, location = 0;
+	int media_types[32];
 	uint8_t payload_types[32];
 	int payloads = 0;
 
@@ -1545,12 +1679,12 @@ void Psip::r_invite(int status, char const *phrase, nua_t *nua, nua_magic_t *mag
 		int ret;
 
 		sip_trace_header(this, "Payload received", DIRECTION_NONE);
-		ret = parse_sdp(sip, &p_s_rtp_ip_remote, &p_s_rtp_port_remote, payload_types, &payloads, sizeof(payload_types));
+		ret = parse_sdp(sip, &p_s_rtp_ip_remote, &p_s_rtp_port_remote, payload_types, media_types, &payloads, sizeof(payload_types));
 		if (!ret) {
 			if (payloads != 1)
 				ret = 415;
 			else if (!p_s_rtp_bridge) {
-				if (payload_types[0] != ((options.law=='a') ? RTP_PT_ALAW : RTP_PT_ULAW)) {
+				if (media_types[0] != ((options.law=='a') ? MEDIA_TYPE_ALAW : MEDIA_TYPE_ULAW)) {
 					add_trace("error", NULL, "Expected LAW payload type (not bridged)");
 					ret = 415;
 				}
@@ -1603,6 +1737,7 @@ void Psip::r_invite(int status, char const *phrase, nua_t *nua, nua_magic_t *mag
 		if (p_s_rtp_bridge) {
 			message->param.progressinfo.rtpinfo.ip = p_s_rtp_ip_remote;
 			message->param.progressinfo.rtpinfo.port = p_s_rtp_port_remote;
+			message->param.progressinfo.rtpinfo.media_types[0] = media_types[0];
 			message->param.progressinfo.rtpinfo.payload_types[0] = payload_types[0];
 			message->param.progressinfo.rtpinfo.payloads = 1;
 		}
@@ -1625,6 +1760,7 @@ void Psip::r_invite(int status, char const *phrase, nua_t *nua, nua_magic_t *mag
 		if (p_s_rtp_bridge) {
 			message->param.connectinfo.rtpinfo.ip = p_s_rtp_ip_remote;
 			message->param.connectinfo.rtpinfo.port = p_s_rtp_port_remote;
+			message->param.connectinfo.rtpinfo.media_types[0] = media_types[0];
 			message->param.connectinfo.rtpinfo.payload_types[0] = payload_types[0];
 			message->param.connectinfo.rtpinfo.payloads = 1;
 		}
@@ -1776,11 +1912,12 @@ void Psip::rtp_shutdown(void)
 int sip_init_inst(struct interface *interface)
 {
 	struct sip_inst *inst = (struct sip_inst *) MALLOC(sizeof(*inst));
+	char local[64];
 
 	interface->sip_inst = inst;
 	SCPY(inst->interface_name, interface->name);
-	SCPY(inst->local_ip, interface->sip_local_ip);
-	SCPY(inst->remote_ip, interface->sip_remote_ip);
+	SCPY(inst->local_peer, interface->sip_local_peer);
+	SCPY(inst->remote_peer, interface->sip_remote_peer);
 
 	/* init root object */
 	inst->root = su_root_create(inst);
@@ -1790,7 +1927,10 @@ int sip_init_inst(struct interface *interface)
 		return -EINVAL;
 	}
 
-	inst->nua = nua_create(inst->root, sip_callback, inst, TAG_NULL());
+	SPRINT(local, "sip:%s",inst->local_peer);
+	if (!strchr(inst->local_peer, ':'))
+		SCAT(local, ":5060");
+	inst->nua = nua_create(inst->root, sip_callback, inst, NUTAG_URL(local), TAG_END());
 	if (!inst->nua) {
 		PERROR("Failed to create SIP stack object\n");
 		sip_exit_inst(interface);
@@ -1815,6 +1955,8 @@ int sip_init_inst(struct interface *interface)
 
 	PDEBUG(DEBUG_SIP, "SIP interface created (inst=%p)\n", inst);
 
+	any_sip_interface = 1;
+
 	return 0;
 }
 
@@ -1833,6 +1975,16 @@ void sip_exit_inst(struct interface *interface)
 	interface->sip_inst = NULL;
 
 	PDEBUG(DEBUG_SIP, "SIP interface removed\n");
+
+	/* check if there is any other SIP interface left */
+	interface = interface_first;
+	while (interface) {
+		if (interface->sip_inst)
+			break;
+		interface = interface->next;
+	}
+	if (!interface)
+		any_sip_interface = 0;
 }
 
 extern su_log_t su_log_default[];
@@ -1893,3 +2045,106 @@ static int delete_event(struct lcr_work *work, void *instance, int index)
 	return 0;
 }
 
+
+/*
+ * generate audio, if no data is received from bridge
+ */
+
+void Psip::set_tone(const char *dir, const char *tone)
+{
+	Port::set_tone(dir, tone);
+
+	update_load();
+}
+
+void Psip::update_load(void)
+{
+	/* don't trigger load event if event already active */
+	if (p_s_loadtimer.active)
+		return;
+
+	/* don't start timer if ... */
+	if (!p_tone_name[0] && !p_dov_tx)
+		return;
+
+	p_s_next_tv_sec = 0;
+	schedule_timer(&p_s_loadtimer, 0, 0); /* no delay the first time */
+}
+
+static int load_timer(struct lcr_timer *timer, void *instance, int index)
+{
+	class Psip *psip = (class Psip *)instance;
+
+	/* stop timer if ... */
+	if (!psip->p_tone_name[0] && !psip->p_dov_tx)
+		return 0;
+
+	psip->load_tx();
+
+	return 0;
+}
+
+#define SEND_SIP_LEN 160
+
+void Psip::load_tx(void)
+{
+	int diff;
+	struct timeval current_time;
+	int tosend = SEND_SIP_LEN, i;
+	unsigned char buf[SEND_SIP_LEN], *p = buf;
+
+	/* get elapsed */
+	gettimeofday(&current_time, NULL);
+	if (!p_s_next_tv_sec) {
+		/* if timer expired the first time, set next expected timeout 160 samples in advance */
+		p_s_next_tv_sec = current_time.tv_sec;
+		p_s_next_tv_usec = current_time.tv_usec + SEND_SIP_LEN * 125;
+		if (p_s_next_tv_usec >= 1000000) {
+			p_s_next_tv_usec -= 1000000;
+			p_s_next_tv_sec++;
+		}
+		schedule_timer(&p_s_loadtimer, 0, SEND_SIP_LEN * 125);
+	} else {
+		diff = 1000000 * (current_time.tv_sec - p_s_next_tv_sec)
+			+ (current_time.tv_usec - p_s_next_tv_usec);
+		if (diff < -SEND_SIP_LEN * 125 || diff > SEND_SIP_LEN * 125) {
+			/* if clock drifts too much, set next timeout event to current timer + 160 */
+			diff = 0;
+			p_s_next_tv_sec = current_time.tv_sec;
+			p_s_next_tv_usec = current_time.tv_usec + SEND_SIP_LEN * 125;
+			if (p_s_next_tv_usec >= 1000000) {
+				p_s_next_tv_usec -= 1000000;
+				p_s_next_tv_sec++;
+			}
+		} else {
+			/* if diff is positive, it took too long, so next timeout will be earlier */
+			p_s_next_tv_usec += SEND_SIP_LEN * 125;
+			if (p_s_next_tv_usec >= 1000000) {
+				p_s_next_tv_usec -= 1000000;
+				p_s_next_tv_sec++;
+			}
+		}
+		schedule_timer(&p_s_loadtimer, 0, SEND_SIP_LEN * 125 - diff);
+	}
+
+	/* copy tones */
+	if (p_tone_name[0]) {
+		tosend -= read_audio(p, tosend);
+	} else
+	if (p_dov_tx) {
+		tosend -= dov_tx(p, tosend);
+	}
+	if (tosend) {
+		PERROR("buffer is not completely filled\n");
+		return;
+	}
+
+	p = buf;
+	for (i = 0; i < SEND_SIP_LEN; i++) {
+		*p = flip[*p];
+		p++;
+	}
+	/* transmit data via rtp */
+	rtp_send_frame(buf, SEND_SIP_LEN, (options.law=='a')?PAYLOAD_TYPE_ALAW:PAYLOAD_TYPE_ULAW);
+}
+