#include <sofia-sip/sdp.h>
#include <sofia-sip/sip_header.h>
+#undef NUTAG_AUTO100
+
unsigned char flip[256];
//pthread_mutex_t mutex_msg;
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;
};
memset(&p_s_rtcp_sin_local, 0, sizeof(p_s_rtcp_sin_local));
memset(&p_s_rtp_sin_remote, 0, sizeof(p_s_rtp_sin_remote));
memset(&p_s_rtcp_sin_remote, 0, sizeof(p_s_rtcp_sin_remote));
- p_s_rtp_payload_type = 0;
p_s_rtp_ip_local = 0;
p_s_rtp_ip_remote = 0;
p_s_rtp_port_local = 0;
rtp_close();
}
-const char *payload_type2name(uint8_t payload_type) {
- switch (payload_type) {
- case 0:
+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";
#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_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)
}
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 "
return -EINVAL;
}
break;
- case RTP_PT_ALAW:
+ case RTP_PT_GSM_HALF:
+ if (payload_len != 14) {
+ PDEBUG(DEBUG_SIP, "received RTP half rate frame with "
+ "payload length != 14 (len = %d)\n",
+ payload_len);
+ return -EINVAL;
+ }
+ break;
+#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;
// psip->rtp_shutdown();
return len;
}
- PDEBUG(DEBUG_SIP, "rtcp!");
+ PDEBUG(DEBUG_SIP, "rtcp!\n");
}
return 0;
}
switch (payload_type) {
+#if 0
+we only support alaw and ulaw!
case RTP_PT_GSM_FULL:
payload_len = 33;
duration = 160;
payload_len = 31;
duration = 160;
break;
- case RTP_PT_ALAW:
- case RTP_PT_ULAW:
+ case RTP_PT_GSM_HALF:
+ payload_len = 14;
+ duration = 160;
+ break;
+#endif
+ case PAYLOAD_TYPE_ALAW:
+ case PAYLOAD_TYPE_ULAW:
payload_len = len;
duration = len;
break;
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);
}
}
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;
- p_s_rtp_payload_type = param->connectinfo.rtpinfo.payload_type;
+ 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;
+ PDEBUG(DEBUG_SIP, "payload type %d\n", payload_type);
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);
} else {
PDEBUG(DEBUG_SIP, "RTP info not given by remote, so we do our own RTP\n");
- p_s_rtp_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) {
+ if (!p_s_rtp_is_connected && rtp_connect() < 0) {
nua_cancel(p_s_handle, TAG_END());
nua_handle_destroy(p_s_handle);
p_s_handle = NULL;
"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, payload_type, payload_type, media_type2name(media_type));
PDEBUG(DEBUG_SIP, "Using SDP response: %s\n", sdp_str);
nua_respond(p_s_handle, SIP_200_OK,
sip_trace_header(this, "RESPOND", DIRECTION_OUT);
add_trace("respond", "value", "200 OK");
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", "%s:%d", media_type2name(media_type), payload_type);
end_trace();
return 0;
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;
- char sdp_str[256];
+ 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;
+ 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;
PDEBUG(DEBUG_SIP, "Doing Setup (inst %p)\n", inst);
if (param->setup.rtpinfo.port) {
PDEBUG(DEBUG_SIP, "RTP info given by remote, forward that\n");
p_s_rtp_bridge = 1;
- p_s_rtp_payload_type = param->setup.rtpinfo.payload_type;
+ 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;
p_s_rtp_port_local = param->setup.rtpinfo.port;
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);
} else {
PDEBUG(DEBUG_SIP, "RTP info not given by remote, so we do our own RTP\n");
- p_s_rtp_payload_type = (options.law=='a') ? RTP_PT_ALAW : RTP_PT_ULAW;
+ p_s_rtp_bridge = 0;
+ media_types = &lcr_media;
+ payload_types = &lcr_payload;
+ payloads = 1;
/* open local RTP peer (if not bridging) */
if (rtp_open() < 0) {
- struct lcr_msg *message;
-
PERROR("Failed to open RTP sockets\n");
/* send release message to endpoit */
message = message_create(p_serial, epoint_id, PORT_TO_EPOINT, MESSAGE_RELEASE);
return 0;
}
}
- SPRINT(from, "sip:%s@%s", param->setup.callerinfo.id, local);
- SPRINT(to, "sip:%s@%s", param->setup.dialinginfo.id, remote);
-
- sip_trace_header(this, "INVITE", DIRECTION_OUT);
- add_trace("from", "uri", "%s", from);
- add_trace("to", "uri", "%s", to);
- add_trace("rtp", "port", "%d,%d", p_s_rtp_port_local, p_s_rtp_port_local + 1);
- end_trace();
p_s_handle = nua_handle(inst->nua, NULL, TAG_END());
if (!p_s_handle) {
- struct lcr_msg *message;
-
PERROR("Failed to create handle\n");
/* send release message to endpoit */
message = message_create(p_serial, epoint_id, PORT_TO_EPOINT, MESSAGE_RELEASE);
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);
"s=SIP Call\n"
"c=IN IP4 %s\n"
"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));
+ "m=audio %d RTP/AVP"
+ , inet_ntoa(ia), inet_ntoa(ia), p_s_rtp_port_local);
+ for (i = 0; i < payloads; i++) {
+ SPRINT(pt_str, " %d", payload_types[i]);
+ SCAT(sdp_str, pt_str);
+ }
+ SCAT(sdp_str, "\n");
+ for (i = 0; i < payloads; 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);
+ SPRINT(from, "sip:%s@%s", param->setup.callerinfo.id, local);
+ SPRINT(to, "sip:%s@%s", param->setup.dialinginfo.id, remote);
+
+ sip_trace_header(this, "INVITE", DIRECTION_OUT);
+ add_trace("from", "uri", "%s", from);
+ add_trace("to", "uri", "%s", to);
+ 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", "%s:%d", media_type2name(media_types[i]), payload_types[i]);
+ end_trace();
+
// cseq = sip_cseq_create(sip_home, 123, SIP_METHOD_INVITE);
nua_invite(p_s_handle,
SIPTAG_PAYLOAD_STR(sdp_str), TAG_END());
new_state(PORT_STATE_OUT_SETUP);
+#if 0
+ PDEBUG(DEBUG_SIP, "do overlap\n");
+ new_state(PORT_STATE_OUT_OVERLAP);
+ message = message_create(p_serial, epoint_id, PORT_TO_EPOINT, MESSAGE_OVERLAP);
+ message_put(message);
+#else
+ PDEBUG(DEBUG_SIP, "do proceeding\n");
+ new_state(PORT_STATE_OUT_PROCEEDING);
+ message = message_create(p_serial, epoint_id, PORT_TO_EPOINT, MESSAGE_PROCEEDING);
+ message_put(message);
+#endif
+
/* attach only if not already */
epointlist = p_epointlist;
while(epointlist) {
"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)),
return 0;
}
-int Psip::message_epoint(unsigned int epoint_id, int message_id, union parameter *param)
+/* NOTE: incomplete and not working */
+int Psip::message_information(unsigned int epoint_id, int message_id, union parameter *param)
{
- class Endpoint *epoint;
+ char dtmf_str[64];
+
+ /* prepare DTMF info payload */
+ SPRINT(dtmf_str,
+ "Signal=%s\n"
+ "Duration=160\n"
+ , param->information.id);
+
+ /* start invite to handle DTMF */
+ nua_info(p_s_handle,
+ NUTAG_MEDIA_ENABLE(0),
+ SIPTAG_CONTENT_TYPE_STR("application/dtmf-relay"),
+ SIPTAG_PAYLOAD_STR(dtmf_str), TAG_END());
+
+ return 0;
+}
+int Psip::message_epoint(unsigned int epoint_id, int message_id, union parameter *param)
+{
if (Port::message_epoint(epoint_id, message_id, param))
return 1;
- epoint = find_epoint_id(epoint_id);
- if (!epoint) {
- PDEBUG(DEBUG_SIP, "PORT(%s) no endpoint object found where the message is from.\n", p_name);
- return 0;
- }
-
switch(message_id) {
case MESSAGE_ALERTING: /* call is ringing on LCR side */
if (p_state != PORT_STATE_IN_SETUP
message_setup(epoint_id, message_id, param);
return 1;
+ case MESSAGE_INFORMATION: /* overlap dialing */
+ if (p_state != PORT_STATE_OUT_OVERLAP)
+ return 0;
+ message_information(epoint_id, message_id, param);
+ return 1;
+
case MESSAGE_DTMF: /* DTMF info to be transmitted via INFO transaction */
if (p_state == PORT_STATE_CONNECT)
message_dtmf(epoint_id, message_id, param);
return 0;
}
-int Psip::parse_sdp(sip_t const *sip, unsigned int *ip, unsigned short *port, uint8_t payload_type)
+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)
{
- int codec_supported = 0;
+ *payloads = 0;
if (!sip->sip_payload) {
PDEBUG(DEBUG_SIP, "no payload given\n");
*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);
- if (map->rm_pt == payload_type) {
- PDEBUG(DEBUG_SIP, "supported codec found\n");
- codec_supported = 1;
- goto done_codec;
+ /* append to payload list, if there is space */
+ 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)++;
}
}
}
- done_codec:
sdp_parser_free(parser);
- if (!codec_supported)
- return 415;
-
return 0;
}
int ret;
class Endpoint *epoint;
struct lcr_msg *message;
- uint8_t payload_type;
+ struct interface *interface;
+ int media_types[32];
+ uint8_t payload_types[32];
+ int payloads = 0;
+
+ interface = getinterfacebyname(inst->interface_name);
+ if (!interface) {
+ PERROR("Cannot find interface %s.\n", inst->interface_name);
+ return;
+ }
if (sip->sip_from && sip->sip_from->a_url)
from = sip->sip_from->a_url->url_user;
to = sip->sip_to->a_url->url_user;
PDEBUG(DEBUG_SIP, "invite received (%s->%s)\n", from, to);
- if (p_s_rtp_bridge)
- payload_type = 3; // FIXME: use list and forward list to remote side
- else
- payload_type = (options.law=='a')?RTP_PT_ALAW:RTP_PT_ULAW;
- ret = parse_sdp(sip, &p_s_rtp_ip_remote, &p_s_rtp_port_remote, payload_type);
+ sip_trace_header(this, "Payload received", DIRECTION_NONE);
+ 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) {
+ int i;
+
+ /* check if supported payload type exists */
+ for (i = 0; i < payloads; i++) {
+ if (media_types[i] == ((options.law=='a') ? MEDIA_TYPE_ALAW : MEDIA_TYPE_ULAW))
+ break;
+ }
+ if (i == payloads) {
+ add_trace("error", NULL, "Expected LAW payload type (not bridged)");
+ ret = 415;
+ }
+ }
+ }
+ end_trace();
if (ret) {
if (ret == 400)
nua_respond(nh, SIP_400_BAD_REQUEST, TAG_END());
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;
FATAL("Incoming call but already got an endpoint.\n");
if (!(epoint = new Endpoint(p_serial, 0)))
FATAL("No memory for Endpoint instance\n");
- if (!(epoint->ep_app = new DEFAULT_ENDPOINT_APP(epoint, 0))) //incoming
- FATAL("No memory for Endpoint Application instance\n");
+ epoint->ep_app = new_endpointapp(epoint, 0, interface->app); //incoming
epointlist_new(epoint->ep_serial);
+#ifdef NUTAG_AUTO100
/* send trying (proceeding) */
nua_respond(nh, SIP_100_TRYING, TAG_END());
sip_trace_header(this, "RESPOND", DIRECTION_OUT);
add_trace("respond", "value", "100 Trying");
end_trace();
+#endif
new_state(PORT_STATE_IN_PROCEEDING);
// message->param.setup.useruser.len = strlen(mncc->useruser.info);
// message->param.setup.useruser.protocol = mncc->useruser.proto;
if (p_s_rtp_bridge) {
+ int i;
+
PDEBUG(DEBUG_SIP, "sending setup with RTP info\n");
- message->param.setup.rtpinfo.payload_type = payload_type;
message->param.setup.rtpinfo.ip = p_s_rtp_ip_remote;
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.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];
+ struct in_addr ia;
+ 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;
+ /* open local RTP peer (if not bridging) */
+ if (rtp_connect() < 0) {
+ nua_respond(nh, SIP_500_INTERNAL_SERVER_ERROR, TAG_END());
+ nua_handle_destroy(nh);
+ p_s_handle = NULL;
+ sip_trace_header(this, "RESPOND", DIRECTION_OUT);
+ add_trace("respond", "value", "500 Internal Server Error");
+ add_trace("reason", NULL, "failed to connect RTP/RTCP sockts");
+ end_trace();
+ new_state(PORT_STATE_RELEASE);
+ trigger_work(&p_s_delete);
+ message = message_create(p_serial, ACTIVE_EPOINT(p_epointlist), PORT_TO_EPOINT, MESSAGE_RELEASE);
+ message->param.disconnectinfo.cause = 41;
+ message->param.disconnectinfo.location = LOCATION_PRIVATE_LOCAL;
+ message_put(message);
+ new_state(PORT_STATE_RELEASE);
+ trigger_work(&p_s_delete);
+ return;
+ }
+
+ ia.s_addr = htonl(p_s_rtp_ip_local);
+
+ SPRINT(sdp_str,
+ "v=0\n"
+ "o=LCR-Sofia-SIP 0 0 IN IP4 %s\n"
+ "s=SIP Call\n"
+ "c=IN IP4 %s\n"
+ "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));
+ 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);
+ 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[])
{
struct lcr_msg *message;
int cause = 0, location = 0;
+ int media_types[32];
+ uint8_t payload_types[32];
+ int payloads = 0;
PDEBUG(DEBUG_SIP, "response to invite received (status = %d)\n", status);
/* connect audio */
if (status == 183 || (status >= 200 && status <= 299)) {
int ret;
- uint8_t payload_type;
- if (p_s_rtp_bridge)
- payload_type = p_s_rtp_payload_type;
- else
- payload_type = (options.law=='a')?RTP_PT_ALAW:RTP_PT_ULAW;
- ret = parse_sdp(sip, &p_s_rtp_ip_remote, &p_s_rtp_port_remote, payload_type);
+ sip_trace_header(this, "Payload received", DIRECTION_NONE);
+ 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 (media_types[0] != ((options.law=='a') ? MEDIA_TYPE_ALAW : MEDIA_TYPE_ULAW)) {
+ add_trace("error", NULL, "Expected LAW payload type (not bridged)");
+ ret = 415;
+ }
+ }
+ }
+ end_trace();
if (ret) {
- if (ret == 400)
- nua_cancel(nh, TAG_END());
- else
- nua_cancel(nh, TAG_END());
+ nua_cancel(nh, TAG_END());
sip_trace_header(this, "CANCEL", DIRECTION_OUT);
- add_trace("reason", NULL, "offered codec does not match");
+ add_trace("reason", NULL, "accepted codec does not match");
end_trace();
cause = 88;
location = LOCATION_PRIVATE_LOCAL;
/* process 1xx */
switch (status) {
case 100:
+#if 0
PDEBUG(DEBUG_SIP, "do proceeding\n");
new_state(PORT_STATE_OUT_PROCEEDING);
message = message_create(p_serial, ACTIVE_EPOINT(p_epointlist), PORT_TO_EPOINT, MESSAGE_PROCEEDING);
message_put(message);
+#endif
return;
case 180:
PDEBUG(DEBUG_SIP, "do alerting\n");
message->param.progressinfo.progress = 8;
message->param.progressinfo.location = 10;
if (p_s_rtp_bridge) {
- message->param.progressinfo.rtpinfo.payload_type = p_s_rtp_payload_type;
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;
}
message_put(message);
return;
new_state(PORT_STATE_CONNECT);
message = message_create(p_serial, ACTIVE_EPOINT(p_epointlist), PORT_TO_EPOINT, MESSAGE_CONNECT);
if (p_s_rtp_bridge) {
- message->param.connectinfo.rtpinfo.payload_type = p_s_rtp_payload_type;
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;
}
message_put(message);
return;
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);
NUTAG_APPL_METHOD("NOTIFY"),
NUTAG_APPL_METHOD("INFO"),
NUTAG_AUTOACK(0),
+#ifdef NUTAG_AUTO100
NUTAG_AUTO100(0),
+#endif
NUTAG_AUTOALERT(0),
NUTAG_AUTOANSWER(0),
TAG_NULL());
projects / lcr.git / blobdiff