1 /*****************************************************************************\
3 ** Linux Call Router **
5 **---------------------------------------------------------------------------**
6 ** Copyright: Andreas Eversberg **
10 \*****************************************************************************/
13 #include <sofia-sip/sip_status.h>
14 #include <sofia-sip/su_log.h>
15 #include <sofia-sip/sdp.h>
16 #include <sofia-sip/sip_header.h>
20 unsigned char flip[256];
22 int any_sip_interface = 0;
24 //pthread_mutex_t mutex_msg;
25 su_home_t sip_home[1];
28 char interface_name[64];
35 static int delete_event(struct lcr_work *work, void *instance, int index);
36 static int load_timer(struct lcr_timer *timer, void *instance, int index);
41 Psip::Psip(int type, char *portname, struct port_settings *settings, struct interface *interface) : Port(type, portname, settings, interface)
44 if (interface->rtp_bridge)
46 p_s_sip_inst = interface->sip_inst;
47 memset(&p_s_delete, 0, sizeof(p_s_delete));
48 add_work(&p_s_delete, delete_event, this, 0);
51 memset(&p_s_rtp_fd, 0, sizeof(p_s_rtp_fd));
52 memset(&p_s_rtcp_fd, 0, sizeof(p_s_rtcp_fd));
53 memset(&p_s_rtp_sin_local, 0, sizeof(p_s_rtp_sin_local));
54 memset(&p_s_rtcp_sin_local, 0, sizeof(p_s_rtcp_sin_local));
55 memset(&p_s_rtp_sin_remote, 0, sizeof(p_s_rtp_sin_remote));
56 memset(&p_s_rtcp_sin_remote, 0, sizeof(p_s_rtcp_sin_remote));
58 p_s_rtp_ip_remote = 0;
59 p_s_rtp_port_local = 0;
60 p_s_rtp_port_remote = 0;
65 p_s_rtp_tx_action = 0;
68 memset(&p_s_loadtimer, 0, sizeof(p_s_loadtimer));
69 add_timer(&p_s_loadtimer, load_timer, this, 0);
72 PDEBUG(DEBUG_SIP, "Created new Psip(%s).\n", portname);
74 FATAL("No SIP instance for interface\n");
83 PDEBUG(DEBUG_SIP, "Destroyed SIP process(%s).\n", p_name);
85 del_timer(&p_s_loadtimer);
86 del_work(&p_s_delete);
91 static const char *media_type2name(uint8_t media_type) {
99 case MEDIA_TYPE_GSM_HR:
101 case MEDIA_TYPE_GSM_EFR:
110 static void sip_trace_header(class Psip *sip, const char *message, int direction)
112 /* init trace with given values */
115 sip?numberrize_callerinfo(sip->p_callerinfo.id, sip->p_callerinfo.ntype, options.national, options.international):NULL,
116 sip?sip->p_dialinginfo.id:NULL,
127 /* according to RFC 3550 */
129 #if __BYTE_ORDER == __LITTLE_ENDIAN
130 uint8_t csrc_count:4,
134 uint8_t payload_type:7,
136 #elif __BYTE_ORDER == __BIG_ENDIAN
147 } __attribute__((packed));
152 } __attribute__((packed));
154 #define RTP_VERSION 2
156 #define PAYLOAD_TYPE_ULAW 0
157 #define PAYLOAD_TYPE_ALAW 8
158 #define PAYLOAD_TYPE_GSM 3
160 /* decode an rtp frame */
161 static int rtp_decode(class Psip *psip, unsigned char *data, int len)
163 struct rtp_hdr *rtph = (struct rtp_hdr *)data;
164 struct rtp_x_hdr *rtpxh;
168 unsigned char *from, *to;
172 PDEBUG(DEBUG_SIP, "received RTP frame too short (len = %d)\n", len);
175 if (rtph->version != RTP_VERSION) {
176 PDEBUG(DEBUG_SIP, "received RTP version %d not supported.\n", rtph->version);
179 payload = data + sizeof(struct rtp_hdr) + (rtph->csrc_count << 2);
180 payload_len = len - sizeof(struct rtp_hdr) - (rtph->csrc_count << 2);
181 if (payload_len < 0) {
182 PDEBUG(DEBUG_SIP, "received RTP frame too short (len = %d, "
183 "csrc count = %d)\n", len, rtph->csrc_count);
186 if (rtph->extension) {
187 if (payload_len < (int)sizeof(struct rtp_x_hdr)) {
188 PDEBUG(DEBUG_SIP, "received RTP frame too short for "
189 "extension header\n");
192 rtpxh = (struct rtp_x_hdr *)payload;
193 x_len = ntohs(rtpxh->length) * 4 + sizeof(struct rtp_x_hdr);
195 payload_len -= x_len;
196 if (payload_len < 0) {
197 PDEBUG(DEBUG_SIP, "received RTP frame too short, "
198 "extension header exceeds frame length\n");
203 if (payload_len < 0) {
204 PDEBUG(DEBUG_SIP, "received RTP frame too short for "
208 payload_len -= payload[payload_len - 1];
209 if (payload_len < 0) {
210 PDEBUG(DEBUG_SIP, "received RTP frame with padding "
211 "greater than payload\n");
216 switch (rtph->payload_type) {
218 we only support alaw and ulaw!
219 case RTP_PT_GSM_FULL:
220 if (payload_len != 33) {
221 PDEBUG(DEBUG_SIP, "received RTP full rate frame with "
222 "payload length != 33 (len = %d)\n",
228 if (payload_len != 31) {
229 PDEBUG(DEBUG_SIP, "received RTP full rate frame with "
230 "payload length != 31 (len = %d)\n",
235 case RTP_PT_GSM_HALF:
236 if (payload_len != 14) {
237 PDEBUG(DEBUG_SIP, "received RTP half rate frame with "
238 "payload length != 14 (len = %d)\n",
244 case PAYLOAD_TYPE_ALAW:
245 if (options.law != 'a') {
246 PDEBUG(DEBUG_SIP, "received Alaw, but we don't do Alaw\n");
250 case PAYLOAD_TYPE_ULAW:
251 if (options.law == 'a') {
252 PDEBUG(DEBUG_SIP, "received Ulaw, but we don't do Ulaw\n");
257 PDEBUG(DEBUG_SIP, "received RTP frame with unknown payload "
258 "type %d\n", rtph->payload_type);
262 if (payload_len <= 0) {
263 PDEBUG(DEBUG_SIP, "received RTP payload is too small: %d\n", payload_len);
269 psip->record(payload, payload_len, 0); // from down
271 psip->tap(payload, payload_len, 0); // from down
276 if (psip->p_echotest) {
277 /* echo rtp data we just received */
278 psip->rtp_send_frame(from, n, (options.law=='a')?PAYLOAD_TYPE_ALAW:PAYLOAD_TYPE_ULAW);
282 *to++ = flip[*from++];
284 psip->dov_rx(payload, payload_len);
285 psip->bridge_tx(payload, payload_len);
290 static int rtp_sock_callback(struct lcr_fd *fd, unsigned int what, void *instance, int index)
292 class Psip *psip = (class Psip *) instance;
294 unsigned char buffer[256];
297 if ((what & LCR_FD_READ)) {
298 len = read(fd->fd, &buffer, sizeof(buffer));
300 PDEBUG(DEBUG_SIP, "read result=%d\n", len);
301 // psip->rtp_close();
302 // psip->rtp_shutdown();
305 if (psip->p_s_rtp_is_connected)
306 rc = rtp_decode(psip, buffer, len);
312 static int rtcp_sock_callback(struct lcr_fd *fd, unsigned int what, void *instance, int index)
314 // class Psip *psip = (class Psip *) instance;
316 unsigned char buffer[256];
318 if ((what & LCR_FD_READ)) {
319 len = read(fd->fd, &buffer, sizeof(buffer));
321 PDEBUG(DEBUG_SIP, "read result=%d\n", len);
322 // psip->rtp_close();
323 // psip->rtp_shutdown();
326 PDEBUG(DEBUG_SIP, "rtcp!\n");
332 #define RTP_PORT_BASE 30000
333 #define RTP_PORT_MAX 39998
334 static unsigned short next_udp_port = RTP_PORT_BASE;
336 static int rtp_sub_socket_bind(int fd, struct sockaddr_in *sin_local, uint32_t ip, uint16_t port)
339 socklen_t alen = sizeof(*sin_local);
341 sin_local->sin_family = AF_INET;
342 sin_local->sin_addr.s_addr = htonl(ip);
343 sin_local->sin_port = htons(port);
345 rc = bind(fd, (struct sockaddr *) sin_local, sizeof(*sin_local));
349 /* retrieve the address we actually bound to, in case we
350 * passed INADDR_ANY as IP address */
351 return getsockname(fd, (struct sockaddr *) sin_local, &alen);
354 static int rtp_sub_socket_connect(int fd, struct sockaddr_in *sin_local, struct sockaddr_in *sin_remote, uint32_t ip, uint16_t port)
357 socklen_t alen = sizeof(*sin_local);
359 sin_remote->sin_family = AF_INET;
360 sin_remote->sin_addr.s_addr = htonl(ip);
361 sin_remote->sin_port = htons(port);
363 rc = connect(fd, (struct sockaddr *) sin_remote, sizeof(*sin_remote));
365 PERROR("failed to connect to ip %08x port %d rc=%d\n", ip, port, rc);
369 return getsockname(fd, (struct sockaddr *) sin_local, &alen);
372 int Psip::rtp_open(void)
377 unsigned short start_port;
380 rc = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
386 register_fd(&p_s_rtp_fd, LCR_FD_READ, rtp_sock_callback, this, 0);
388 rc = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
394 register_fd(&p_s_rtcp_fd, LCR_FD_READ, rtcp_sock_callback, this, 0);
397 ip = htonl(INADDR_ANY);
399 start_port = next_udp_port;
401 rc = rtp_sub_socket_bind(p_s_rtp_fd.fd, &p_s_rtp_sin_local, ip, next_udp_port);
405 rc = rtp_sub_socket_bind(p_s_rtcp_fd.fd, &p_s_rtcp_sin_local, ip, next_udp_port + 1);
407 p_s_rtp_port_local = next_udp_port;
408 next_udp_port = (next_udp_port + 2 > RTP_PORT_MAX) ? RTP_PORT_BASE : next_udp_port + 2;
411 /* reopen rtp socket and try again with next udp port */
412 unregister_fd(&p_s_rtp_fd);
413 close(p_s_rtp_fd.fd);
415 rc2 = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
421 register_fd(&p_s_rtp_fd, LCR_FD_READ, rtp_sock_callback, this, 0);
424 next_udp_port = (next_udp_port + 2 > RTP_PORT_MAX) ? RTP_PORT_BASE : next_udp_port + 2;
425 if (next_udp_port == start_port)
427 /* we must use rc2, in order to preserve rc */
430 PDEBUG(DEBUG_SIP, "failed to find port\n");
434 p_s_rtp_ip_local = ntohl(p_s_rtp_sin_local.sin_addr.s_addr);
435 PDEBUG(DEBUG_SIP, "local ip %08x port %d\n", p_s_rtp_ip_local, p_s_rtp_port_local);
436 PDEBUG(DEBUG_SIP, "remote ip %08x port %d\n", p_s_rtp_ip_remote, p_s_rtp_port_remote);
438 return p_s_rtp_port_local;
441 int Psip::rtp_connect(void)
446 ia.s_addr = htonl(p_s_rtp_ip_remote);
447 PDEBUG(DEBUG_SIP, "rtp_connect(ip=%s, port=%u)\n", inet_ntoa(ia), p_s_rtp_port_remote);
449 rc = rtp_sub_socket_connect(p_s_rtp_fd.fd, &p_s_rtp_sin_local, &p_s_rtp_sin_remote, p_s_rtp_ip_remote, p_s_rtp_port_remote);
453 rc = rtp_sub_socket_connect(p_s_rtcp_fd.fd, &p_s_rtcp_sin_local, &p_s_rtcp_sin_remote, p_s_rtp_ip_remote, p_s_rtp_port_remote + 1);
457 p_s_rtp_ip_local = ntohl(p_s_rtp_sin_local.sin_addr.s_addr);
458 PDEBUG(DEBUG_SIP, "local ip %08x port %d\n", p_s_rtp_ip_local, p_s_rtp_port_local);
459 PDEBUG(DEBUG_SIP, "remote ip %08x port %d\n", p_s_rtp_ip_remote, p_s_rtp_port_remote);
460 p_s_rtp_is_connected = 1;
464 void Psip::rtp_close(void)
466 if (p_s_rtp_fd.fd > 0) {
467 unregister_fd(&p_s_rtp_fd);
468 close(p_s_rtp_fd.fd);
471 if (p_s_rtcp_fd.fd > 0) {
472 unregister_fd(&p_s_rtcp_fd);
473 close(p_s_rtcp_fd.fd);
476 if (p_s_rtp_is_connected) {
477 PDEBUG(DEBUG_SIP, "rtp closed\n");
478 p_s_rtp_is_connected = 0;
483 void tv_difference(struct timeval *diff, const struct timeval *from,
484 const struct timeval *__to)
486 struct timeval _to = *__to, *to = &_to;
488 if (to->tv_usec < from->tv_usec) {
490 to->tv_usec += 1000000;
493 diff->tv_usec = to->tv_usec - from->tv_usec;
494 diff->tv_sec = to->tv_sec - from->tv_sec;
497 /* encode and send a rtp frame */
498 int Psip::rtp_send_frame(unsigned char *data, unsigned int len, uint8_t payload_type)
500 struct rtp_hdr *rtph;
502 int duration; /* in samples */
503 unsigned char buffer[256];
507 record(data, len, 1); // from up
509 tap(data, len, 1); // from up
511 if (!p_s_rtp_is_connected) {
516 if (!p_s_rtp_tx_action) {
517 /* initialize sequences */
518 p_s_rtp_tx_action = 1;
519 p_s_rtp_tx_ssrc = rand();
520 p_s_rtp_tx_sequence = random();
521 p_s_rtp_tx_timestamp = random();
522 memset(&p_s_rtp_tx_last_tv, 0, sizeof(p_s_rtp_tx_last_tv));
525 switch (payload_type) {
527 we only support alaw and ulaw!
528 case RTP_PT_GSM_FULL:
536 case RTP_PT_GSM_HALF:
541 case PAYLOAD_TYPE_ALAW:
542 case PAYLOAD_TYPE_ULAW:
547 PERROR("unsupported message type %d\n", payload_type);
553 struct timeval tv, tv_diff;
554 long int usec_diff, frame_diff;
556 gettimeofday(&tv, NULL);
557 tv_difference(&tv_diff, &p_s_rtp_tx_last_tv, &tv);
558 p_s_rtp_tx_last_tv = tv;
560 usec_diff = tv_diff.tv_sec * 1000000 + tv_diff.tv_usec;
561 frame_diff = (usec_diff / 20000);
563 if (abs(frame_diff) > 1) {
564 long int frame_diff_excess = frame_diff - 1;
566 PDEBUG(DEBUG_SIP, "Correcting frame difference of %ld frames\n", frame_diff_excess);
567 p_s_rtp_tx_sequence += frame_diff_excess;
568 p_s_rtp_tx_timestamp += frame_diff_excess * duration;
573 rtph = (struct rtp_hdr *) buffer;
574 rtph->version = RTP_VERSION;
577 rtph->csrc_count = 0;
579 rtph->payload_type = payload_type;
580 rtph->sequence = htons(p_s_rtp_tx_sequence++);
581 rtph->timestamp = htonl(p_s_rtp_tx_timestamp);
582 p_s_rtp_tx_timestamp += duration;
583 rtph->ssrc = htonl(p_s_rtp_tx_ssrc);
584 memcpy(buffer + sizeof(struct rtp_hdr), data, payload_len);
586 if (p_s_rtp_fd.fd > 0) {
587 len = write(p_s_rtp_fd.fd, &buffer, sizeof(struct rtp_hdr) + payload_len);
588 if (len != sizeof(struct rtp_hdr) + payload_len) {
589 PDEBUG(DEBUG_SIP, "write result=%d\n", len);
599 /* receive from remote */
600 int Psip::bridge_rx(unsigned char *data, int len)
604 /* don't bridge, if tones are provided */
605 if (p_tone_name[0] || p_dov_tx)
611 if ((ret = Port::bridge_rx(data, len)))
614 /* write to rx buffer */
616 p_s_rxdata[p_s_rxpos++] = flip[*data++];
617 if (p_s_rxpos == 160) {
620 /* transmit data via rtp */
621 rtp_send_frame(p_s_rxdata, 160, (options.law=='a')?PAYLOAD_TYPE_ALAW:PAYLOAD_TYPE_ULAW);
628 /* taken from freeswitch */
629 /* map sip responses to QSIG cause codes ala RFC4497 section 8.4.4 */
630 static int status2cause(int status)
634 return 16; //SWITCH_CAUSE_NORMAL_CLEARING;
640 return 21; //SWITCH_CAUSE_CALL_REJECTED;
642 return 1; //SWITCH_CAUSE_UNALLOCATED_NUMBER;
645 return 3; //SWITCH_CAUSE_NO_ROUTE_DESTINATION;
648 return 102; //SWITCH_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
650 return 22; //SWITCH_CAUSE_NUMBER_CHANGED;
659 return 127; //SWITCH_CAUSE_INTERWORKING;
661 return 180; //SWITCH_CAUSE_NO_USER_RESPONSE;
666 return 41; //SWITCH_CAUSE_NORMAL_TEMPORARY_FAILURE;
669 return 17; //SWITCH_CAUSE_USER_BUSY;
671 return 28; //SWITCH_CAUSE_INVALID_NUMBER_FORMAT;
674 return 88; //SWITCH_CAUSE_INCOMPATIBLE_DESTINATION;
676 return 38; //SWITCH_CAUSE_NETWORK_OUT_OF_ORDER;
678 return 63; //SWITCH_CAUSE_SERVICE_UNAVAILABLE;
682 return 79; //SWITCH_CAUSE_SERVICE_NOT_IMPLEMENTED;
685 return 25; //SWITCH_CAUSE_EXCHANGE_ROUTING_ERROR;
687 return 31; //??? SWITCH_CAUSE_ORIGINATOR_CANCEL;
689 return 31; //SWITCH_CAUSE_NORMAL_UNSPECIFIED;
693 static int cause2status(int cause, int location, const char **st)
699 s = 404; *st = sip_404_Not_found;
702 s = 404; *st = sip_404_Not_found;
705 s = 404; *st = sip_404_Not_found;
708 s = 486; *st = sip_486_Busy_here;
711 s = 408; *st = sip_408_Request_timeout;
714 s = 480; *st = sip_480_Temporarily_unavailable;
717 s = 480; *st = sip_480_Temporarily_unavailable;
720 if (location == LOCATION_USER) {
721 s = 603; *st = sip_603_Decline;
723 s = 403; *st = sip_403_Forbidden;
727 //s = 301; *st = sip_301_Moved_permanently;
728 s = 410; *st = sip_410_Gone;
731 s = 410; *st = sip_410_Gone;
734 s = 502; *st = sip_502_Bad_gateway;
737 s = 484; *st = sip_484_Address_incomplete;
740 s = 501; *st = sip_501_Not_implemented;
743 s = 480; *st = sip_480_Temporarily_unavailable;
746 s = 503; *st = sip_503_Service_unavailable;
749 s = 503; *st = sip_503_Service_unavailable;
752 s = 503; *st = sip_503_Service_unavailable;
755 s = 503; *st = sip_503_Service_unavailable;
758 s = 503; *st = sip_503_Service_unavailable;
761 s = 403; *st = sip_403_Forbidden;
764 s = 403; *st = sip_403_Forbidden;
767 s = 503; *st = sip_503_Service_unavailable;
770 s = 488; *st = sip_488_Not_acceptable;
773 s = 501; *st = sip_501_Not_implemented;
776 s = 488; *st = sip_488_Not_acceptable;
779 s = 501; *st = sip_501_Not_implemented;
782 s = 403; *st = sip_403_Forbidden;
785 s = 503; *st = sip_503_Service_unavailable;
788 s = 504; *st = sip_504_Gateway_time_out;
791 s = 468; *st = sip_486_Busy_here;
798 * endpoint sends messages to the SIP port
801 int Psip::message_connect(unsigned int epoint_id, int message_id, union parameter *param)
805 struct lcr_msg *message;
807 unsigned char payload_type;
809 if (param->connectinfo.rtpinfo.port) {
810 PDEBUG(DEBUG_SIP, "RTP info given by remote, forward that\n");
812 media_type = param->connectinfo.rtpinfo.media_types[0];
813 payload_type = param->connectinfo.rtpinfo.payload_types[0];
814 p_s_rtp_ip_local = param->connectinfo.rtpinfo.ip;
815 p_s_rtp_port_local = param->connectinfo.rtpinfo.port;
816 PDEBUG(DEBUG_SIP, "payload type %d\n", payload_type);
817 PDEBUG(DEBUG_SIP, "local ip %08x port %d\n", p_s_rtp_ip_local, p_s_rtp_port_local);
818 PDEBUG(DEBUG_SIP, "remote ip %08x port %d\n", p_s_rtp_ip_remote, p_s_rtp_port_remote);
820 PDEBUG(DEBUG_SIP, "RTP info not given by remote, so we do our own RTP\n");
821 media_type = (options.law=='a') ? MEDIA_TYPE_ALAW : MEDIA_TYPE_ULAW;
822 payload_type = (options.law=='a') ? PAYLOAD_TYPE_ALAW : PAYLOAD_TYPE_ULAW;
823 /* open local RTP peer (if not bridging) */
824 if (!p_s_rtp_is_connected && rtp_connect() < 0) {
825 nua_cancel(p_s_handle, TAG_END());
826 nua_handle_destroy(p_s_handle);
828 sip_trace_header(this, "CANCEL", DIRECTION_OUT);
829 add_trace("reason", NULL, "failed to connect RTP/RTCP sockts");
831 message = message_create(p_serial, epoint_id, PORT_TO_EPOINT, MESSAGE_RELEASE);
832 message->param.disconnectinfo.cause = 41;
833 message->param.disconnectinfo.location = LOCATION_PRIVATE_LOCAL;
834 message_put(message);
835 new_state(PORT_STATE_RELEASE);
836 trigger_work(&p_s_delete);
841 ia.s_addr = htonl(p_s_rtp_ip_local);
845 "o=LCR-Sofia-SIP 0 0 IN IP4 %s\n"
849 "m=audio %d RTP/AVP %d\n"
850 "a=rtpmap:%d %s/8000\n"
851 , inet_ntoa(ia), inet_ntoa(ia), p_s_rtp_port_local, payload_type, payload_type, media_type2name(media_type));
852 PDEBUG(DEBUG_SIP, "Using SDP response: %s\n", sdp_str);
854 nua_respond(p_s_handle, SIP_200_OK,
855 NUTAG_MEDIA_ENABLE(0),
856 SIPTAG_CONTENT_TYPE_STR("application/sdp"),
857 SIPTAG_PAYLOAD_STR(sdp_str), TAG_END());
858 new_state(PORT_STATE_CONNECT);
859 sip_trace_header(this, "RESPOND", DIRECTION_OUT);
860 add_trace("respond", "value", "200 OK");
861 add_trace("reason", NULL, "call connected");
862 add_trace("rtp", "ip", "%s", inet_ntoa(ia));
863 add_trace("rtp", "port", "%d,%d", p_s_rtp_port_local, p_s_rtp_port_local + 1);
864 add_trace("rtp", "payload", "%s:%d", media_type2name(media_type), payload_type);
870 int Psip::message_release(unsigned int epoint_id, int message_id, union parameter *param)
872 struct lcr_msg *message;
873 char cause_str[128] = "";
874 int cause = param->disconnectinfo.cause;
875 int location = param->disconnectinfo.cause;
877 const char *status_text;
879 if (cause > 0 && cause <= 127) {
880 SPRINT(cause_str, "Q.850;cause=%d;text=\"%s\"", cause, isdn_cause[cause].english);
884 case PORT_STATE_OUT_SETUP:
885 case PORT_STATE_OUT_PROCEEDING:
886 case PORT_STATE_OUT_ALERTING:
887 PDEBUG(DEBUG_SIP, "RELEASE/DISCONNECT will cancel\n");
888 sip_trace_header(this, "CANCEL", DIRECTION_OUT);
890 add_trace("cause", "value", "%d", cause);
892 nua_cancel(p_s_handle, TAG_IF(cause_str[0], SIPTAG_REASON_STR(cause_str)), TAG_END());
894 case PORT_STATE_IN_SETUP:
895 case PORT_STATE_IN_PROCEEDING:
896 case PORT_STATE_IN_ALERTING:
897 PDEBUG(DEBUG_SIP, "RELEASE/DISCONNECT will respond\n");
898 status = cause2status(cause, location, &status_text);
899 sip_trace_header(this, "RESPOND", DIRECTION_OUT);
901 add_trace("cause", "value", "%d", cause);
902 add_trace("respond", "value", "%d %s", status, status_text);
904 nua_respond(p_s_handle, status, status_text, TAG_IF(cause_str[0], SIPTAG_REASON_STR(cause_str)), TAG_END());
905 nua_handle_destroy(p_s_handle);
907 trigger_work(&p_s_delete);
910 PDEBUG(DEBUG_SIP, "RELEASE/DISCONNECT will perform nua_bye\n");
911 sip_trace_header(this, "BYE", DIRECTION_OUT);
913 add_trace("cause", "value", "%d", cause);
915 nua_bye(p_s_handle, TAG_IF(cause_str[0], SIPTAG_REASON_STR(cause_str)), TAG_END());
918 if (message_id == MESSAGE_DISCONNECT) {
919 while(p_epointlist) {
920 message = message_create(p_serial, p_epointlist->epoint_id, PORT_TO_EPOINT, MESSAGE_RELEASE);
921 message->param.disconnectinfo.cause = CAUSE_NORMAL;
922 message->param.disconnectinfo.location = LOCATION_BEYOND;
923 message_put(message);
925 free_epointlist(p_epointlist);
929 new_state(PORT_STATE_RELEASE);
934 int Psip::message_setup(unsigned int epoint_id, int message_id, union parameter *param)
936 struct sip_inst *inst = (struct sip_inst *) p_s_sip_inst;
939 const char *local = inst->local_peer;
941 const char *remote = inst->remote_peer;
942 char sdp_str[512], pt_str[32];
944 struct epoint_list *epointlist;
945 sip_cseq_t *cseq = NULL;
946 struct lcr_msg *message;
947 int lcr_media = { (options.law=='a') ? MEDIA_TYPE_ALAW : MEDIA_TYPE_ULAW };
948 unsigned char lcr_payload = { (options.law=='a') ? PAYLOAD_TYPE_ALAW : PAYLOAD_TYPE_ULAW };
950 unsigned char *payload_types;
954 PDEBUG(DEBUG_SIP, "Doing Setup (inst %p)\n", inst);
956 memcpy(&p_dialinginfo, ¶m->setup.dialinginfo, sizeof(p_dialinginfo));
957 memcpy(&p_callerinfo, ¶m->setup.callerinfo, sizeof(p_callerinfo));
958 memcpy(&p_redirinfo, ¶m->setup.redirinfo, sizeof(p_redirinfo));
960 if (param->setup.rtpinfo.port) {
961 PDEBUG(DEBUG_SIP, "RTP info given by remote, forward that\n");
963 media_types = param->setup.rtpinfo.media_types;
964 payload_types = param->setup.rtpinfo.payload_types;
965 payloads = param->setup.rtpinfo.payloads;
966 p_s_rtp_ip_local = param->setup.rtpinfo.ip;
967 p_s_rtp_port_local = param->setup.rtpinfo.port;
968 PDEBUG(DEBUG_SIP, "local ip %08x port %d\n", p_s_rtp_ip_local, p_s_rtp_port_local);
969 PDEBUG(DEBUG_SIP, "remote ip %08x port %d\n", p_s_rtp_ip_remote, p_s_rtp_port_remote);
971 PDEBUG(DEBUG_SIP, "RTP info not given by remote, so we do our own RTP\n");
973 media_types = &lcr_media;
974 payload_types = &lcr_payload;
977 /* open local RTP peer (if not bridging) */
978 if (rtp_open() < 0) {
979 PERROR("Failed to open RTP sockets\n");
980 /* send release message to endpoit */
981 message = message_create(p_serial, epoint_id, PORT_TO_EPOINT, MESSAGE_RELEASE);
982 message->param.disconnectinfo.cause = 41;
983 message->param.disconnectinfo.location = LOCATION_PRIVATE_LOCAL;
984 message_put(message);
985 new_state(PORT_STATE_RELEASE);
986 trigger_work(&p_s_delete);
991 p_s_handle = nua_handle(inst->nua, NULL, TAG_END());
993 PERROR("Failed to create handle\n");
994 /* send release message to endpoit */
995 message = message_create(p_serial, epoint_id, PORT_TO_EPOINT, MESSAGE_RELEASE);
996 message->param.disconnectinfo.cause = 41;
997 message->param.disconnectinfo.location = LOCATION_PRIVATE_LOCAL;
998 message_put(message);
999 new_state(PORT_STATE_RELEASE);
1000 trigger_work(&p_s_delete);
1004 sip_trace_header(this, "NEW handle", DIRECTION_IN);
1005 add_trace("handle", "new", "0x%x", p_s_handle);
1008 if (!p_s_rtp_ip_local) {
1011 /* extract IP from local peer */
1012 SCPY(local_ip, local);
1013 p = strchr(local_ip, ':');
1016 PDEBUG(DEBUG_SIP, "RTP local IP not known, so we use our local SIP ip %s\n", local_ip);
1017 inet_pton(AF_INET, local_ip, &p_s_rtp_ip_local);
1018 p_s_rtp_ip_local = ntohl(p_s_rtp_ip_local);
1020 ia.s_addr = htonl(p_s_rtp_ip_local);
1023 "o=LCR-Sofia-SIP 0 0 IN IP4 %s\n"
1027 "m=audio %d RTP/AVP"
1028 , inet_ntoa(ia), inet_ntoa(ia), p_s_rtp_port_local);
1029 for (i = 0; i < payloads; i++) {
1030 SPRINT(pt_str, " %d", payload_types[i]);
1031 SCAT(sdp_str, pt_str);
1033 SCAT(sdp_str, "\n");
1034 for (i = 0; i < payloads; i++) {
1035 SPRINT(pt_str, "a=rtpmap:%d %s/8000\n", payload_types[i], media_type2name(media_types[i]));
1036 SCAT(sdp_str, pt_str);
1038 PDEBUG(DEBUG_SIP, "Using SDP for invite: %s\n", sdp_str);
1040 SPRINT(from, "sip:%s@%s", param->setup.callerinfo.id, local);
1041 SPRINT(to, "sip:%s@%s", param->setup.dialinginfo.id, remote);
1043 sip_trace_header(this, "INVITE", DIRECTION_OUT);
1044 add_trace("from", "uri", "%s", from);
1045 add_trace("to", "uri", "%s", to);
1046 add_trace("rtp", "ip", "%s", inet_ntoa(ia));
1047 add_trace("rtp", "port", "%d,%d", p_s_rtp_port_local, p_s_rtp_port_local + 1);
1048 for (i = 0; i < payloads; i++)
1049 add_trace("rtp", "payload", "%s:%d", media_type2name(media_types[i]), payload_types[i]);
1052 // cseq = sip_cseq_create(sip_home, 123, SIP_METHOD_INVITE);
1054 nua_invite(p_s_handle,
1055 TAG_IF(from[0], SIPTAG_FROM_STR(from)),
1056 TAG_IF(to[0], SIPTAG_TO_STR(to)),
1057 TAG_IF(cseq, SIPTAG_CSEQ(cseq)),
1058 NUTAG_MEDIA_ENABLE(0),
1059 SIPTAG_CONTENT_TYPE_STR("application/sdp"),
1060 SIPTAG_PAYLOAD_STR(sdp_str), TAG_END());
1061 new_state(PORT_STATE_OUT_SETUP);
1064 PDEBUG(DEBUG_SIP, "do overlap\n");
1065 new_state(PORT_STATE_OUT_OVERLAP);
1066 message = message_create(p_serial, epoint_id, PORT_TO_EPOINT, MESSAGE_OVERLAP);
1067 message_put(message);
1069 PDEBUG(DEBUG_SIP, "do proceeding\n");
1070 new_state(PORT_STATE_OUT_PROCEEDING);
1071 message = message_create(p_serial, epoint_id, PORT_TO_EPOINT, MESSAGE_PROCEEDING);
1072 message_put(message);
1075 /* attach only if not already */
1076 epointlist = p_epointlist;
1078 if (epointlist->epoint_id == epoint_id)
1080 epointlist = epointlist->next;
1083 epointlist_new(epoint_id);
1088 int Psip::message_notify(unsigned int epoint_id, int message_id, union parameter *param)
1090 // char sdp_str[256];
1091 // struct in_addr ia;
1093 switch (param->notifyinfo.notify) {
1094 case INFO_NOTIFY_REMOTE_HOLD:
1098 "o=LCR-Sofia-SIP 0 0 IN IP4 0.0.0.0\n"
1100 "c=IN IP4 0.0.0.0\n"
1103 PDEBUG(DEBUG_SIP, "Using SDP for hold: %s\n", sdp_str);
1104 nua_info(p_s_handle,
1105 // TAG_IF(from[0], SIPTAG_FROM_STR(from)),
1106 // TAG_IF(to[0], SIPTAG_TO_STR(to)),
1107 // TAG_IF(cseq, SIPTAG_CSEQ(cseq)),
1108 NUTAG_MEDIA_ENABLE(0),
1109 SIPTAG_CONTENT_TYPE_STR("application/sdp"),
1110 SIPTAG_PAYLOAD_STR(sdp_str), TAG_END());
1113 case INFO_NOTIFY_REMOTE_RETRIEVAL:
1115 ia.s_addr = htonl(p_s_rtp_ip_local);
1118 "o=LCR-Sofia-SIP 0 0 IN IP4 %s\n"
1122 "m=audio %d RTP/AVP %d\n"
1123 "a=rtpmap:%d %s/8000\n"
1124 , 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));
1125 PDEBUG(DEBUG_SIP, "Using SDP for rertieve: %s\n", sdp_str);
1126 nua_info(p_s_handle,
1127 // TAG_IF(from[0], SIPTAG_FROM_STR(from)),
1128 // TAG_IF(to[0], SIPTAG_TO_STR(to)),
1129 // TAG_IF(cseq, SIPTAG_CSEQ(cseq)),
1130 NUTAG_MEDIA_ENABLE(0),
1131 SIPTAG_CONTENT_TYPE_STR("application/sdp"),
1132 SIPTAG_PAYLOAD_STR(sdp_str), TAG_END());
1140 int Psip::message_dtmf(unsigned int epoint_id, int message_id, union parameter *param)
1144 /* prepare DTMF info payload */
1150 /* start invite to handle DTMF */
1151 nua_info(p_s_handle,
1152 NUTAG_MEDIA_ENABLE(0),
1153 SIPTAG_CONTENT_TYPE_STR("application/dtmf-relay"),
1154 SIPTAG_PAYLOAD_STR(dtmf_str), TAG_END());
1159 /* NOTE: incomplete and not working */
1160 int Psip::message_information(unsigned int epoint_id, int message_id, union parameter *param)
1164 /* prepare DTMF info payload */
1168 , param->information.id);
1170 /* start invite to handle DTMF */
1171 nua_info(p_s_handle,
1172 NUTAG_MEDIA_ENABLE(0),
1173 SIPTAG_CONTENT_TYPE_STR("application/dtmf-relay"),
1174 SIPTAG_PAYLOAD_STR(dtmf_str), TAG_END());
1179 int Psip::message_epoint(unsigned int epoint_id, int message_id, union parameter *param)
1181 if (Port::message_epoint(epoint_id, message_id, param))
1184 switch(message_id) {
1185 case MESSAGE_ALERTING: /* call is ringing on LCR side */
1186 if (p_state != PORT_STATE_IN_SETUP
1187 && p_state != PORT_STATE_IN_PROCEEDING)
1189 nua_respond(p_s_handle, SIP_180_RINGING, TAG_END());
1190 sip_trace_header(this, "RESPOND", DIRECTION_OUT);
1191 add_trace("respond", "value", "180 Ringing");
1193 new_state(PORT_STATE_IN_ALERTING);
1196 case MESSAGE_CONNECT: /* call is connected on LCR side */
1197 if (p_state != PORT_STATE_IN_SETUP
1198 && p_state != PORT_STATE_IN_PROCEEDING
1199 && p_state != PORT_STATE_IN_ALERTING)
1201 message_connect(epoint_id, message_id, param);
1204 case MESSAGE_DISCONNECT: /* call has been disconnected */
1205 case MESSAGE_RELEASE: /* call has been released */
1206 message_release(epoint_id, message_id, param);
1209 case MESSAGE_SETUP: /* dial-out command received from epoint */
1210 message_setup(epoint_id, message_id, param);
1213 case MESSAGE_INFORMATION: /* overlap dialing */
1214 if (p_state != PORT_STATE_OUT_OVERLAP)
1216 message_information(epoint_id, message_id, param);
1219 case MESSAGE_DTMF: /* DTMF info to be transmitted via INFO transaction */
1220 if (p_state == PORT_STATE_CONNECT)
1221 message_dtmf(epoint_id, message_id, param);
1222 case MESSAGE_NOTIFY: /* notification about remote hold/retrieve */
1223 if (p_state == PORT_STATE_CONNECT)
1224 message_notify(epoint_id, message_id, param);
1228 PDEBUG(DEBUG_SIP, "PORT(%s) SP port with (caller id %s) received an unsupported message: %d\n", p_name, p_callerinfo.id, message_id);
1234 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)
1238 if (!sip->sip_payload) {
1239 PDEBUG(DEBUG_SIP, "no payload given\n");
1243 sdp_parser_t *parser;
1246 sdp_attribute_t *attr;
1248 sdp_connection_t *conn;
1250 PDEBUG(DEBUG_SIP, "payload given: %s\n", sip->sip_payload->pl_data);
1252 parser = sdp_parse(NULL, sip->sip_payload->pl_data, (int) strlen(sip->sip_payload->pl_data), 0);
1256 if (!(sdp = sdp_session(parser))) {
1257 sdp_parser_free(parser);
1260 for (m = sdp->sdp_media; m; m = m->m_next) {
1261 if (m->m_proto != sdp_proto_rtp)
1263 if (m->m_type != sdp_media_audio)
1265 PDEBUG(DEBUG_SIP, "RTP port:'%u'\n", m->m_port);
1267 for (attr = m->m_attributes; attr; attr = attr->a_next) {
1268 PDEBUG(DEBUG_SIP, "ATTR: name:'%s' value='%s'\n", attr->a_name, attr->a_value);
1270 if (m->m_connections) {
1271 conn = m->m_connections;
1272 PDEBUG(DEBUG_SIP, "CONN: address:'%s'\n", conn->c_address);
1273 inet_pton(AF_INET, conn->c_address, ip);
1274 *ip = ntohl(p_s_rtp_ip_remote);
1276 char *p = sip->sip_payload->pl_data;
1279 PDEBUG(DEBUG_SIP, "sofia cannot find connection tag, so we try ourself\n");
1280 p = strstr(p, "c=IN IP4 ");
1282 PDEBUG(DEBUG_SIP, "missing c-tag with internet address\n");
1283 sdp_parser_free(parser);
1287 if ((p = strchr(addr, '\n'))) *p = '\0';
1288 if ((p = strchr(addr, '\r'))) *p = '\0';
1289 PDEBUG(DEBUG_SIP, "CONN: address:'%s'\n", addr);
1290 inet_pton(AF_INET, addr, ip);
1291 *ip = ntohl(p_s_rtp_ip_remote);
1293 for (map = m->m_rtpmaps; map; map = map->rm_next) {
1296 PDEBUG(DEBUG_SIP, "RTPMAP: coding:'%s' rate='%d' pt='%d'\n", map->rm_encoding, map->rm_rate, map->rm_pt);
1297 /* append to payload list, if there is space */
1298 add_trace("rtp", "payload", "%s:%d", map->rm_encoding, map->rm_pt);
1299 if (map->rm_pt == PAYLOAD_TYPE_ALAW)
1300 media_type = MEDIA_TYPE_ALAW;
1301 else if (map->rm_pt == PAYLOAD_TYPE_ULAW)
1302 media_type = MEDIA_TYPE_ULAW;
1303 else if (map->rm_pt == PAYLOAD_TYPE_GSM)
1304 media_type = MEDIA_TYPE_GSM;
1305 else if (!strcmp(map->rm_encoding, "GSM-EFR"))
1306 media_type = MEDIA_TYPE_GSM_EFR;
1307 else if (!strcmp(map->rm_encoding, "AMR"))
1308 media_type = MEDIA_TYPE_AMR;
1309 else if (!strcmp(map->rm_encoding, "GSM-HR"))
1310 media_type = MEDIA_TYPE_GSM_HR;
1311 if (media_type && *payloads <= max_payloads) {
1312 *payload_types++ = map->rm_pt;
1313 *media_types++ = media_type;
1319 sdp_parser_free(parser);
1324 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[])
1326 struct sip_inst *inst = (struct sip_inst *) p_s_sip_inst;
1327 const char *from = "", *to = "", *name = "";
1330 class Endpoint *epoint;
1331 struct lcr_msg *message;
1332 struct interface *interface;
1333 int media_types[32];
1334 uint8_t payload_types[32];
1338 interface = getinterfacebyname(inst->interface_name);
1340 PERROR("Cannot find interface %s.\n", inst->interface_name);
1344 if (sip->sip_from) {
1345 if (sip->sip_from->a_url)
1346 from = sip->sip_from->a_url->url_user;
1347 if (sip->sip_from->a_display) {
1348 name = sip->sip_from->a_display;
1349 if (!strncmp(name, "\"IMSI", 5)) {
1350 strncpy(imsi, name + 5, 15);
1357 if (sip->sip_to->a_url)
1358 to = sip->sip_to->a_url->url_user;
1360 PDEBUG(DEBUG_SIP, "invite received (%s->%s)\n", from, to);
1362 sip_trace_header(this, "Payload received", DIRECTION_NONE);
1363 ret = parse_sdp(sip, &p_s_rtp_ip_remote, &p_s_rtp_port_remote, payload_types, media_types, &payloads, sizeof(payload_types));
1365 /* if no RTP bridge, we must support LAW codec, otherwise we forward what we have */
1366 if (!p_s_rtp_bridge) {
1369 /* check if supported payload type exists */
1370 for (i = 0; i < payloads; i++) {
1371 if (media_types[i] == ((options.law=='a') ? MEDIA_TYPE_ALAW : MEDIA_TYPE_ULAW))
1374 if (i == payloads) {
1375 add_trace("error", NULL, "Expected LAW payload type (not bridged)");
1383 nua_respond(nh, SIP_400_BAD_REQUEST, TAG_END());
1385 nua_respond(nh, SIP_415_UNSUPPORTED_MEDIA, TAG_END());
1386 nua_handle_destroy(nh);
1388 sip_trace_header(this, "RESPOND", DIRECTION_OUT);
1390 add_trace("respond", "value", "415 Unsupported Media");
1392 add_trace("respond", "value", "400 Bad Request");
1393 add_trace("reason", NULL, "offered codec does not match");
1395 new_state(PORT_STATE_RELEASE);
1396 trigger_work(&p_s_delete);
1400 /* open local RTP peer (if not bridging) */
1401 if (!p_s_rtp_bridge && rtp_open() < 0) {
1402 nua_respond(nh, SIP_500_INTERNAL_SERVER_ERROR, TAG_END());
1403 nua_handle_destroy(nh);
1405 sip_trace_header(this, "RESPOND", DIRECTION_OUT);
1406 add_trace("respond", "value", "500 Internal Server Error");
1407 add_trace("reason", NULL, "failed to open RTP/RTCP sockts");
1409 new_state(PORT_STATE_RELEASE);
1410 trigger_work(&p_s_delete);
1415 sip_trace_header(this, "NEW handle", DIRECTION_IN);
1416 add_trace("handle", "new", "0x%x", nh);
1420 sip_trace_header(this, "INVITE", DIRECTION_IN);
1421 add_trace("rtp", "port", "%d", p_s_rtp_port_remote);
1422 /* caller information */
1424 p_callerinfo.present = INFO_PRESENT_NOTAVAIL;
1425 p_callerinfo.ntype = INFO_NTYPE_NOTPRESENT;
1426 add_trace("calling", "present", "unavailable");
1428 p_callerinfo.present = INFO_PRESENT_ALLOWED;
1429 add_trace("calling", "present", "allowed");
1430 p_callerinfo.screen = INFO_SCREEN_NETWORK;
1431 p_callerinfo.ntype = INFO_NTYPE_UNKNOWN;
1432 SCPY(p_callerinfo.id, from);
1433 add_trace("calling", "number", "%s", from);
1434 SCPY(p_callerinfo.name, name);
1436 add_trace("calling", "name", "%s", name);
1437 SCPY(p_callerinfo.imsi, imsi);
1439 add_trace("calling", "imsi", "%s", imsi);
1441 SCPY(p_callerinfo.interface, inst->interface_name);
1442 /* dialing information */
1444 p_dialinginfo.ntype = INFO_NTYPE_UNKNOWN;
1445 SCAT(p_dialinginfo.id, to);
1446 add_trace("dialing", "number", "%s", to);
1449 /* bearer capability */
1450 p_capainfo.bearer_capa = INFO_BC_SPEECH;
1451 p_capainfo.bearer_info1 = (options.law=='a')?3:2;
1452 p_capainfo.bearer_mode = INFO_BMODE_CIRCUIT;
1453 add_trace("bearer", "capa", "speech");
1454 add_trace("bearer", "mode", "circuit");
1455 /* if packet mode works some day, see dss1.cpp for conditions */
1456 p_capainfo.source_mode = B_MODE_TRANSPARENT;
1460 /* create endpoint */
1462 FATAL("Incoming call but already got an endpoint.\n");
1463 if (!(epoint = new Endpoint(p_serial, 0)))
1464 FATAL("No memory for Endpoint instance\n");
1465 epoint->ep_app = new_endpointapp(epoint, 0, interface->app); //incoming
1466 epointlist_new(epoint->ep_serial);
1468 #ifdef NUTAG_AUTO100
1469 /* send trying (proceeding) */
1470 nua_respond(nh, SIP_100_TRYING, TAG_END());
1471 sip_trace_header(this, "RESPOND", DIRECTION_OUT);
1472 add_trace("respond", "value", "100 Trying");
1476 new_state(PORT_STATE_IN_PROCEEDING);
1478 /* send setup message to endpoit */
1479 message = message_create(p_serial, ACTIVE_EPOINT(p_epointlist), PORT_TO_EPOINT, MESSAGE_SETUP);
1480 message->param.setup.port_type = p_type;
1481 // message->param.setup.dtmf = 0;
1482 memcpy(&message->param.setup.dialinginfo, &p_dialinginfo, sizeof(struct dialing_info));
1483 memcpy(&message->param.setup.callerinfo, &p_callerinfo, sizeof(struct caller_info));
1484 memcpy(&message->param.setup.capainfo, &p_capainfo, sizeof(struct capa_info));
1485 // SCPY((char *)message->param.setup.useruser.data, useruser.info);
1486 // message->param.setup.useruser.len = strlen(mncc->useruser.info);
1487 // message->param.setup.useruser.protocol = mncc->useruser.proto;
1488 if (p_s_rtp_bridge) {
1491 PDEBUG(DEBUG_SIP, "sending setup with RTP info\n");
1492 message->param.setup.rtpinfo.ip = p_s_rtp_ip_remote;
1493 message->param.setup.rtpinfo.port = p_s_rtp_port_remote;
1494 /* add codecs to setup message */
1495 for (i = 0; i < payloads; i++) {
1496 message->param.setup.rtpinfo.media_types[i] = media_types[i];
1497 message->param.setup.rtpinfo.payload_types[i] = payload_types[i];
1498 if (i == sizeof(message->param.setup.rtpinfo.payload_types))
1501 message->param.setup.rtpinfo.payloads = i;
1503 message_put(message);
1505 /* send progress, if tones are available and if we don't bridge */
1506 if (!p_s_rtp_bridge && interface->is_tones == IS_YES) {
1509 unsigned char payload_type;
1511 PDEBUG(DEBUG_SIP, "Connecting audio, since we have tones available\n");
1512 media_type = (options.law=='a') ? MEDIA_TYPE_ALAW : MEDIA_TYPE_ULAW;
1513 payload_type = (options.law=='a') ? PAYLOAD_TYPE_ALAW : PAYLOAD_TYPE_ULAW;
1514 /* open local RTP peer (if not bridging) */
1515 if (rtp_connect() < 0) {
1516 nua_respond(nh, SIP_500_INTERNAL_SERVER_ERROR, TAG_END());
1517 nua_handle_destroy(nh);
1519 sip_trace_header(this, "RESPOND", DIRECTION_OUT);
1520 add_trace("respond", "value", "500 Internal Server Error");
1521 add_trace("reason", NULL, "failed to connect RTP/RTCP sockts");
1523 new_state(PORT_STATE_RELEASE);
1524 trigger_work(&p_s_delete);
1525 message = message_create(p_serial, ACTIVE_EPOINT(p_epointlist), PORT_TO_EPOINT, MESSAGE_RELEASE);
1526 message->param.disconnectinfo.cause = 41;
1527 message->param.disconnectinfo.location = LOCATION_PRIVATE_LOCAL;
1528 message_put(message);
1529 new_state(PORT_STATE_RELEASE);
1530 trigger_work(&p_s_delete);
1534 ia.s_addr = htonl(p_s_rtp_ip_local);
1538 "o=LCR-Sofia-SIP 0 0 IN IP4 %s\n"
1542 "m=audio %d RTP/AVP %d\n"
1543 "a=rtpmap:%d %s/8000\n"
1544 , inet_ntoa(ia), inet_ntoa(ia), p_s_rtp_port_local, payload_type, payload_type, media_type2name(media_type));
1545 PDEBUG(DEBUG_SIP, "Using SDP response: %s\n", sdp_str);
1547 nua_respond(p_s_handle, SIP_183_SESSION_PROGRESS,
1548 NUTAG_MEDIA_ENABLE(0),
1549 SIPTAG_CONTENT_TYPE_STR("application/sdp"),
1550 SIPTAG_PAYLOAD_STR(sdp_str), TAG_END());
1551 sip_trace_header(this, "RESPOND", DIRECTION_OUT);
1552 add_trace("respond", "value", "183 SESSION PROGRESS");
1553 add_trace("reason", NULL, "audio available");
1554 add_trace("rtp", "ip", "%s", inet_ntoa(ia));
1555 add_trace("rtp", "port", "%d,%d", p_s_rtp_port_local, p_s_rtp_port_local + 1);
1556 add_trace("rtp", "payload", "%s:%d", media_type2name(media_type), payload_type);
1561 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[])
1563 struct lcr_msg *message;
1566 PDEBUG(DEBUG_SIP, "bye received\n");
1568 sip_trace_header(this, "BYE", DIRECTION_IN);
1569 if (sip->sip_reason && sip->sip_reason->re_protocol && !strcasecmp(sip->sip_reason->re_protocol, "Q.850") && sip->sip_reason->re_cause) {
1570 cause = atoi(sip->sip_reason->re_cause);
1571 add_trace("cause", "value", "%d", cause);
1575 // let stack do bye automaticall, since it will not accept our response for some reason
1576 // nua_respond(nh, SIP_200_OK, TAG_END());
1577 sip_trace_header(this, "RESPOND", DIRECTION_OUT);
1578 add_trace("respond", "value", "200 OK");
1580 // nua_handle_destroy(nh);
1585 while(p_epointlist) {
1586 /* send setup message to endpoit */
1587 message = message_create(p_serial, p_epointlist->epoint_id, PORT_TO_EPOINT, MESSAGE_RELEASE);
1588 message->param.disconnectinfo.cause = cause ? : 16;
1589 message->param.disconnectinfo.location = LOCATION_BEYOND;
1590 message_put(message);
1592 free_epointlist(p_epointlist);
1594 new_state(PORT_STATE_RELEASE);
1595 trigger_work(&p_s_delete);
1598 void Psip::i_cancel(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[])
1600 struct lcr_msg *message;
1602 PDEBUG(DEBUG_SIP, "cancel received\n");
1604 sip_trace_header(this, "CANCEL", DIRECTION_IN);
1607 nua_handle_destroy(nh);
1612 while(p_epointlist) {
1613 /* send setup message to endpoit */
1614 message = message_create(p_serial, p_epointlist->epoint_id, PORT_TO_EPOINT, MESSAGE_RELEASE);
1615 message->param.disconnectinfo.cause = 16;
1616 message->param.disconnectinfo.location = LOCATION_BEYOND;
1617 message_put(message);
1619 free_epointlist(p_epointlist);
1621 new_state(PORT_STATE_RELEASE);
1622 trigger_work(&p_s_delete);
1625 void Psip::r_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[])
1627 PDEBUG(DEBUG_SIP, "bye response received\n");
1629 nua_handle_destroy(nh);
1634 trigger_work(&p_s_delete);
1637 void Psip::r_cancel(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[])
1639 PDEBUG(DEBUG_SIP, "cancel response received\n");
1641 nua_handle_destroy(nh);
1646 trigger_work(&p_s_delete);
1649 void Psip::r_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[])
1651 struct lcr_msg *message;
1652 int cause = 0, location = 0;
1653 int media_types[32];
1654 uint8_t payload_types[32];
1657 PDEBUG(DEBUG_SIP, "response to invite received (status = %d)\n", status);
1659 sip_trace_header(this, "RESPOND", DIRECTION_OUT);
1660 add_trace("respond", "value", "%d", status);
1664 if (status == 183 || (status >= 200 && status <= 299)) {
1667 sip_trace_header(this, "Payload received", DIRECTION_NONE);
1668 ret = parse_sdp(sip, &p_s_rtp_ip_remote, &p_s_rtp_port_remote, payload_types, media_types, &payloads, sizeof(payload_types));
1672 else if (!p_s_rtp_bridge) {
1673 if (media_types[0] != ((options.law=='a') ? MEDIA_TYPE_ALAW : MEDIA_TYPE_ULAW)) {
1674 add_trace("error", NULL, "Expected LAW payload type (not bridged)");
1681 nua_cancel(nh, TAG_END());
1682 sip_trace_header(this, "CANCEL", DIRECTION_OUT);
1683 add_trace("reason", NULL, "accepted codec does not match");
1686 location = LOCATION_PRIVATE_LOCAL;
1687 goto release_with_cause;
1690 /* connect to remote RTP (if not bridging) */
1691 if (!p_s_rtp_bridge && rtp_connect() < 0) {
1692 nua_cancel(nh, TAG_END());
1693 sip_trace_header(this, "CANCEL", DIRECTION_OUT);
1694 add_trace("reason", NULL, "failed to open RTP/RTCP sockts");
1697 location = LOCATION_PRIVATE_LOCAL;
1698 goto release_with_cause;
1706 PDEBUG(DEBUG_SIP, "do proceeding\n");
1707 new_state(PORT_STATE_OUT_PROCEEDING);
1708 message = message_create(p_serial, ACTIVE_EPOINT(p_epointlist), PORT_TO_EPOINT, MESSAGE_PROCEEDING);
1709 message_put(message);
1713 PDEBUG(DEBUG_SIP, "do alerting\n");
1714 new_state(PORT_STATE_OUT_ALERTING);
1715 message = message_create(p_serial, ACTIVE_EPOINT(p_epointlist), PORT_TO_EPOINT, MESSAGE_ALERTING);
1716 message_put(message);
1719 PDEBUG(DEBUG_SIP, "do progress\n");
1720 message = message_create(p_serial, ACTIVE_EPOINT(p_epointlist), PORT_TO_EPOINT, MESSAGE_PROGRESS);
1721 message->param.progressinfo.progress = 8;
1722 message->param.progressinfo.location = 10;
1723 if (p_s_rtp_bridge) {
1724 message->param.progressinfo.rtpinfo.ip = p_s_rtp_ip_remote;
1725 message->param.progressinfo.rtpinfo.port = p_s_rtp_port_remote;
1726 message->param.progressinfo.rtpinfo.media_types[0] = media_types[0];
1727 message->param.progressinfo.rtpinfo.payload_types[0] = payload_types[0];
1728 message->param.progressinfo.rtpinfo.payloads = 1;
1730 message_put(message);
1733 if (status < 100 || status > 199)
1735 PDEBUG(DEBUG_SIP, "skipping 1xx message\n");
1741 if (status >= 200 && status <= 299) {
1742 PDEBUG(DEBUG_SIP, "do connect\n");
1743 nua_ack(nh, TAG_END());
1744 new_state(PORT_STATE_CONNECT);
1745 message = message_create(p_serial, ACTIVE_EPOINT(p_epointlist), PORT_TO_EPOINT, MESSAGE_CONNECT);
1746 if (p_s_rtp_bridge) {
1747 message->param.connectinfo.rtpinfo.ip = p_s_rtp_ip_remote;
1748 message->param.connectinfo.rtpinfo.port = p_s_rtp_port_remote;
1749 message->param.connectinfo.rtpinfo.media_types[0] = media_types[0];
1750 message->param.connectinfo.rtpinfo.payload_types[0] = payload_types[0];
1751 message->param.connectinfo.rtpinfo.payloads = 1;
1753 message_put(message);
1756 cause = status2cause(status);
1757 location = LOCATION_BEYOND;
1760 PDEBUG(DEBUG_SIP, "do release (cause %d)\n", cause);
1762 while(p_epointlist) {
1763 /* send setup message to endpoit */
1764 message = message_create(p_serial, p_epointlist->epoint_id, PORT_TO_EPOINT, MESSAGE_RELEASE);
1765 message->param.disconnectinfo.cause = cause;
1766 message->param.disconnectinfo.location = location;
1767 message_put(message);
1769 free_epointlist(p_epointlist);
1772 new_state(PORT_STATE_RELEASE);
1776 trigger_work(&p_s_delete);
1779 static void sip_callback(nua_event_t event, 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 tags[])
1781 struct sip_inst *inst = (struct sip_inst *) magic;
1783 class Psip *psip = NULL;
1785 PDEBUG(DEBUG_SIP, "Event %d from stack received (handle=%p)\n", event, nh);
1789 /* create or find port instance */
1790 if (event == nua_i_invite)
1793 struct interface *interface = interface_first;
1795 /* create call instance */
1796 SPRINT(name, "%s-%d-in", inst->interface_name, 0);
1798 if (!strcmp(interface->name, inst->interface_name))
1800 interface = interface->next;
1803 PERROR("Cannot find interface %s.\n", inst->interface_name);
1806 if (!(psip = new Psip(PORT_TYPE_SIP_IN, name, NULL, interface)))
1807 FATAL("Cannot create Port instance.\n");
1811 if ((port->p_type & PORT_CLASS_mISDN_MASK) == PORT_CLASS_SIP) {
1812 psip = (class Psip *)port;
1813 if (psip->p_s_handle == nh) {
1821 PERROR("no SIP Port found for handel %p\n", nh);
1822 nua_respond(nh, SIP_500_INTERNAL_SERVER_ERROR, TAG_END());
1823 nua_handle_destroy(nh);
1828 case nua_r_set_params:
1829 PDEBUG(DEBUG_SIP, "setparam response\n");
1832 PDEBUG(DEBUG_SIP, "error received\n");
1835 PDEBUG(DEBUG_SIP, "state change received\n");
1837 case nua_i_register:
1838 PDEBUG(DEBUG_SIP, "register received\n");
1841 psip->i_invite(status, phrase, nua, magic, nh, hmagic, sip, tags);
1844 PDEBUG(DEBUG_SIP, "ack received\n");
1847 PDEBUG(DEBUG_SIP, "active received\n");
1850 psip->i_bye(status, phrase, nua, magic, nh, hmagic, sip, tags);
1853 psip->i_cancel(status, phrase, nua, magic, nh, hmagic, sip, tags);
1856 psip->r_bye(status, phrase, nua, magic, nh, hmagic, sip, tags);
1859 psip->r_cancel(status, phrase, nua, magic, nh, hmagic, sip, tags);
1862 psip->r_invite(status, phrase, nua, magic, nh, hmagic, sip, tags);
1864 case nua_i_terminated:
1865 PDEBUG(DEBUG_SIP, "terminated received\n");
1868 PDEBUG(DEBUG_SIP, "Event %d not handled\n", event);
1872 /* received shutdown due to termination of RTP */
1873 void Psip::rtp_shutdown(void)
1875 struct lcr_msg *message;
1877 PDEBUG(DEBUG_SIP, "RTP stream terminated\n");
1879 sip_trace_header(this, "RTP terminated", DIRECTION_IN);
1882 nua_handle_destroy(p_s_handle);
1885 while(p_epointlist) {
1886 /* send setup message to endpoit */
1887 message = message_create(p_serial, p_epointlist->epoint_id, PORT_TO_EPOINT, MESSAGE_RELEASE);
1888 message->param.disconnectinfo.cause = 16;
1889 message->param.disconnectinfo.location = LOCATION_BEYOND;
1890 message_put(message);
1892 free_epointlist(p_epointlist);
1894 new_state(PORT_STATE_RELEASE);
1895 trigger_work(&p_s_delete);
1898 int sip_init_inst(struct interface *interface)
1900 struct sip_inst *inst = (struct sip_inst *) MALLOC(sizeof(*inst));
1903 interface->sip_inst = inst;
1904 SCPY(inst->interface_name, interface->name);
1905 SCPY(inst->local_peer, interface->sip_local_peer);
1906 SCPY(inst->remote_peer, interface->sip_remote_peer);
1908 /* init root object */
1909 inst->root = su_root_create(inst);
1911 PERROR("Failed to create SIP root\n");
1912 sip_exit_inst(interface);
1916 SPRINT(local, "sip:%s",inst->local_peer);
1917 if (!strchr(inst->local_peer, ':'))
1918 SCAT(local, ":5060");
1919 inst->nua = nua_create(inst->root, sip_callback, inst, NUTAG_URL(local), TAG_END());
1921 PERROR("Failed to create SIP stack object\n");
1922 sip_exit_inst(interface);
1925 nua_set_params(inst->nua,
1926 SIPTAG_ALLOW_STR("INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,INFO"),
1927 NUTAG_APPL_METHOD("INVITE"),
1928 NUTAG_APPL_METHOD("ACK"),
1929 // NUTAG_APPL_METHOD("BYE"), /* we must reply to BYE */
1930 NUTAG_APPL_METHOD("CANCEL"),
1931 NUTAG_APPL_METHOD("OPTIONS"),
1932 NUTAG_APPL_METHOD("NOTIFY"),
1933 NUTAG_APPL_METHOD("INFO"),
1935 #ifdef NUTAG_AUTO100
1939 NUTAG_AUTOANSWER(0),
1942 PDEBUG(DEBUG_SIP, "SIP interface created (inst=%p)\n", inst);
1944 any_sip_interface = 1;
1949 void sip_exit_inst(struct interface *interface)
1951 struct sip_inst *inst = (struct sip_inst *) interface->sip_inst;
1956 su_root_destroy(inst->root);
1958 nua_destroy(inst->nua);
1960 FREE(inst, sizeof(*inst));
1961 interface->sip_inst = NULL;
1963 PDEBUG(DEBUG_SIP, "SIP interface removed\n");
1965 /* check if there is any other SIP interface left */
1966 interface = interface_first;
1968 if (interface->sip_inst)
1970 interface = interface->next;
1973 any_sip_interface = 0;
1976 extern su_log_t su_log_default[];
1977 extern su_log_t nua_log[];
1978 //extern su_log_t soa_log[];
1984 /* init SOFIA lib */
1986 su_home_init(sip_home);
1988 if (options.deb & DEBUG_SIP) {
1989 su_log_set_level(su_log_default, 9);
1990 su_log_set_level(nua_log, 9);
1991 //su_log_set_level(soa_log, 9);
1994 for (i = 0; i < 256; i++)
1995 flip[i] = ((i & 1) << 7) + ((i & 2) << 5) + ((i & 4) << 3) + ((i & 8) << 1) + ((i & 16) >> 1) + ((i & 32) >> 3) + ((i & 64) >> 5) + ((i & 128) >> 7);
1997 PDEBUG(DEBUG_SIP, "SIP globals initialized\n");
2004 su_home_deinit(sip_home);
2007 PDEBUG(DEBUG_SIP, "SIP globals de-initialized\n");
2010 void sip_handle(void)
2012 struct interface *interface = interface_first;
2013 struct sip_inst *inst;
2016 if (interface->sip_inst) {
2017 inst = (struct sip_inst *) interface->sip_inst;
2018 su_root_step(inst->root, 0);
2020 interface = interface->next;
2024 /* deletes when back in event loop */
2025 static int delete_event(struct lcr_work *work, void *instance, int index)
2027 class Psip *psip = (class Psip *)instance;
2036 * generate audio, if no data is received from bridge
2039 void Psip::set_tone(const char *dir, const char *tone)
2041 Port::set_tone(dir, tone);
2046 void Psip::update_load(void)
2048 /* don't trigger load event if event already active */
2049 if (p_s_loadtimer.active)
2052 /* don't start timer if ... */
2053 if (!p_tone_name[0] && !p_dov_tx)
2056 p_s_next_tv_sec = 0;
2057 schedule_timer(&p_s_loadtimer, 0, 0); /* no delay the first time */
2060 static int load_timer(struct lcr_timer *timer, void *instance, int index)
2062 class Psip *psip = (class Psip *)instance;
2064 /* stop timer if ... */
2065 if (!psip->p_tone_name[0] && !psip->p_dov_tx)
2073 #define SEND_SIP_LEN 160
2075 void Psip::load_tx(void)
2078 struct timeval current_time;
2079 int tosend = SEND_SIP_LEN, i;
2080 unsigned char buf[SEND_SIP_LEN], *p = buf;
2083 gettimeofday(¤t_time, NULL);
2084 if (!p_s_next_tv_sec) {
2085 /* if timer expired the first time, set next expected timeout 160 samples in advance */
2086 p_s_next_tv_sec = current_time.tv_sec;
2087 p_s_next_tv_usec = current_time.tv_usec + SEND_SIP_LEN * 125;
2088 if (p_s_next_tv_usec >= 1000000) {
2089 p_s_next_tv_usec -= 1000000;
2092 schedule_timer(&p_s_loadtimer, 0, SEND_SIP_LEN * 125);
2094 diff = 1000000 * (current_time.tv_sec - p_s_next_tv_sec)
2095 + (current_time.tv_usec - p_s_next_tv_usec);
2096 if (diff < -SEND_SIP_LEN * 125 || diff > SEND_SIP_LEN * 125) {
2097 /* if clock drifts too much, set next timeout event to current timer + 160 */
2099 p_s_next_tv_sec = current_time.tv_sec;
2100 p_s_next_tv_usec = current_time.tv_usec + SEND_SIP_LEN * 125;
2101 if (p_s_next_tv_usec >= 1000000) {
2102 p_s_next_tv_usec -= 1000000;
2106 /* if diff is positive, it took too long, so next timeout will be earlier */
2107 p_s_next_tv_usec += SEND_SIP_LEN * 125;
2108 if (p_s_next_tv_usec >= 1000000) {
2109 p_s_next_tv_usec -= 1000000;
2113 schedule_timer(&p_s_loadtimer, 0, SEND_SIP_LEN * 125 - diff);
2117 if (p_tone_name[0]) {
2118 tosend -= read_audio(p, tosend);
2121 tosend -= dov_tx(p, tosend);
2124 PERROR("buffer is not completely filled\n");
2129 for (i = 0; i < SEND_SIP_LEN; i++) {
2133 /* transmit data via rtp */
2134 rtp_send_frame(buf, SEND_SIP_LEN, (options.law=='a')?PAYLOAD_TYPE_ALAW:PAYLOAD_TYPE_ULAW);