MNCC<->SIP bridge; attaches to OsmoMSC to interface with external SIP VoIP telephony https://osmocom.org/projects/osmo-sip-conector
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osmo-sip-connector/src/sip.c

755 lines
24 KiB

/*
* (C) 2016-2017 by Holger Hans Peter Freyther
*
* All Rights Reserved
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU Affero General Public License as published by
* the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU Affero General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*
*/
#include "sip.h"
#include "app.h"
#include "call.h"
#include "logging.h"
#include "sdp.h"
#include <osmocom/core/utils.h>
#include <osmocom/core/socket.h>
#include <osmocom/gsm/tlv.h>
#include <sofia-sip/sip_status.h>
#include <sofia-sip/su_log.h>
#include <sofia-sip/sdp.h>
#include <talloc.h>
#include <string.h>
#include <netinet/in.h>
extern void *tall_mncc_ctx;
static void sip_release_call(struct call_leg *_leg);
static void sip_ring_call(struct call_leg *_leg);
static void sip_connect_call(struct call_leg *_leg);
static void sip_dtmf_call(struct call_leg *_leg, int keypad);
static void sip_hold_call(struct call_leg *_leg);
static void sip_retrieve_call(struct call_leg *_leg);
/* Find a SIP Call leg by given nua_handle */
static struct sip_call_leg *sip_find_leg(nua_handle_t *nh)
{
struct call *call;
llist_for_each_entry(call, &g_call_list, entry) {
if (call->initial && call->initial->type == CALL_TYPE_SIP) {
struct sip_call_leg *leg = (struct sip_call_leg *) call->initial;
if (leg->nua_handle == nh)
return leg;
}
if (call->remote && call->remote->type == CALL_TYPE_SIP) {
struct sip_call_leg *leg = (struct sip_call_leg *) call->remote;
if (leg->nua_handle == nh)
return leg;
}
}
return NULL;
}
static void call_progress(struct sip_call_leg *leg, const sip_t *sip, int status)
{
struct call_leg *other = call_leg_other(&leg->base);
if (!other)
return;
/* Extract SDP for session in progress with matching codec */
if (status == 183)
sdp_extract_sdp(leg, sip, false);
LOGP(DSIP, LOGL_INFO, "leg(%p) is now progressing.\n", leg);
other->ring_call(other);
}
static void call_connect(struct sip_call_leg *leg, const sip_t *sip)
{
/* extract SDP file and if compatible continue */
struct call_leg *other = call_leg_other(&leg->base);
if (!other) {
LOGP(DSIP, LOGL_ERROR, "leg(%p) connected but leg gone\n", leg);
nua_cancel(leg->nua_handle, TAG_END());
return;
}
if (!sdp_extract_sdp(leg, sip, false)) {
LOGP(DSIP, LOGL_ERROR, "leg(%p) incompatible audio, releasing\n", leg);
nua_cancel(leg->nua_handle, TAG_END());
other->release_call(other);
return;
}
LOGP(DSIP, LOGL_INFO, "leg(%p) is now connected(%s).\n", leg, sip->sip_call_id->i_id);
leg->state = SIP_CC_CONNECTED;
other->connect_call(other);
nua_ack(leg->nua_handle, TAG_END());
}
static void new_call(struct sip_agent *agent, nua_handle_t *nh,
const sip_t *sip)
{
struct call *call;
struct sip_call_leg *leg;
const char *from = NULL, *to = NULL;
char ip_addr[INET6_ADDRSTRLEN];
bool xgcr_hdr_present = false;
uint8_t xgcr_hdr[28] = { 0 };
LOGP(DSIP, LOGL_INFO, "Incoming call(%s) handle(%p)\n", sip->sip_call_id->i_id, nh);
sip_unknown_t *unknown_header = sip->sip_unknown;
while (unknown_header != NULL) {
if (!strcmp("X-Global-Call-Ref", unknown_header->un_name)) {
osmo_hexparse(unknown_header->un_value, xgcr_hdr, sizeof(xgcr_hdr));
xgcr_hdr_present = true;
break;
}
unknown_header = unknown_header->un_next;
}
if (!sdp_screen_sdp(sip)) {
LOGP(DSIP, LOGL_ERROR, "No supported codec.\n");
nua_respond(nh, SIP_406_NOT_ACCEPTABLE, TAG_END());
nua_handle_destroy(nh);
return;
}
call = call_sip_create();
if (!call) {
LOGP(DSIP, LOGL_ERROR, "No supported codec.\n");
nua_respond(nh, SIP_500_INTERNAL_SERVER_ERROR, TAG_END());
nua_handle_destroy(nh);
return;
}
/* Decode Decode the Global Call Reference (if present) */
if (xgcr_hdr_present) {
if (osmo_dec_gcr(&call->gcr, xgcr_hdr, sizeof(xgcr_hdr)) < 0) {
LOGP(DSIP, LOGL_ERROR, "Failed to parse X-Global-Call-Ref.\n");
nua_respond(nh, SIP_406_NOT_ACCEPTABLE, TAG_END());
nua_handle_destroy(nh);
return;
}
call->gcr_present = true;
}
if (sip->sip_to)
to = sip->sip_to->a_url->url_user;
if (sip->sip_from)
from = sip->sip_from->a_url->url_user;
if (!to || !from) {
LOGP(DSIP, LOGL_ERROR, "Unknown from/to for invite.\n");
nua_respond(nh, SIP_406_NOT_ACCEPTABLE, TAG_END());
nua_handle_destroy(nh);
return;
}
leg = (struct sip_call_leg *) call->initial;
leg->state = SIP_CC_DLG_CNFD;
leg->dir = SIP_DIR_MO;
/*
* FIXME/TODO.. we need to select the codec at some point. But it is
* not this place. It starts with the TCH/F vs. TCH/H selection based
* on the offered codecs, and then RTP_CREATE should have it. So both
* are GSM related... and do not belong here. Just pick the first codec
* so the IP address, port and payload type is set.
*/
if (!sdp_extract_sdp(leg, sip, true)) {
LOGP(DSIP, LOGL_ERROR, "leg(%p) no audio, releasing\n", leg);
nua_respond(nh, SIP_406_NOT_ACCEPTABLE, TAG_END());
nua_handle_destroy(nh);
call_leg_release(&leg->base);
return;
}
LOGP(DSIP, LOGL_INFO, "SDP Extracted: IP=(%s) PORT=(%u) PAYLOAD=(%u).\n",
osmo_sockaddr_ntop((const struct sockaddr *)&leg->base.addr, ip_addr),
osmo_sockaddr_port((const struct sockaddr *)&leg->base.addr),
leg->base.payload_type);
leg->base.release_call = sip_release_call;
leg->base.ring_call = sip_ring_call;
leg->base.connect_call = sip_connect_call;
leg->base.dtmf = sip_dtmf_call;
leg->base.hold_call = sip_hold_call;
leg->base.retrieve_call = sip_retrieve_call;
leg->agent = agent;
leg->nua_handle = nh;
nua_handle_bind(nh, leg);
leg->sdp_payload = talloc_strdup(leg, sip->sip_payload->pl_data);
app_route_call(call,
talloc_strdup(leg, from),
talloc_strdup(leg, to));
}
static void sip_handle_reinvite(struct sip_call_leg *leg, nua_handle_t *nh, const sip_t *sip) {
char *sdp;
sdp_mode_t mode = sdp_sendrecv;
char ip_addr[INET6_ADDRSTRLEN];
struct sockaddr_storage prev_addr = leg->base.addr;
LOGP(DSIP, LOGL_INFO, "re-INVITE for call %s\n", sip->sip_call_id->i_id);
struct call_leg *other = call_leg_other(&leg->base);
if (!other) {
LOGP(DMNCC, LOGL_ERROR, "leg(%p) other leg gone!\n", leg);
sip_release_call(&leg->base);
return;
}
if (!sdp_get_sdp_mode(sip, &mode)) {
/* re-INVITE with no SDP.
* We should respond with SDP reflecting current session
*/
sdp = sdp_create_file(leg, other, sdp_sendrecv);
nua_respond(nh, SIP_200_OK,
NUTAG_MEDIA_ENABLE(0),
SIPTAG_CONTENT_TYPE_STR("application/sdp"),
SIPTAG_PAYLOAD_STR(sdp),
TAG_END());
talloc_free(sdp);
return;
}
LOGP(DSIP, LOGL_DEBUG, "pre re-INVITE have IP:port (%s:%u)\n",
osmo_sockaddr_ntop((struct sockaddr*)&prev_addr, ip_addr),
osmo_sockaddr_port((struct sockaddr*)&prev_addr));
if (mode == sdp_sendonly) {
/* SIP side places call on HOLD */
sdp = sdp_create_file(leg, other, sdp_recvonly);
/* TODO: Tell core network to stop sending RTP ? */
} else {
/* SIP re-INVITE may want to change media, IP, port */
if (!sdp_extract_sdp(leg, sip, true)) {
LOGP(DSIP, LOGL_ERROR, "leg(%p) no audio, releasing\n", leg);
nua_respond(nh, SIP_406_NOT_ACCEPTABLE, TAG_END());
nua_handle_destroy(nh);
call_leg_release(&leg->base);
return;
}
LOGP(DSIP, LOGL_DEBUG, "Media IP:port in re-INVITE: (%s:%u)\n",
osmo_sockaddr_ntop((struct sockaddr*)&leg->base.addr, ip_addr),
osmo_sockaddr_port((struct sockaddr*)&leg->base.addr));
if (osmo_sockaddr_cmp((struct osmo_sockaddr *)&prev_addr,
(struct osmo_sockaddr *)&leg->base.addr)) {
LOGP(DSIP, LOGL_INFO, "re-INVITE changes media connection to %s:%u\n",
osmo_sockaddr_ntop((struct sockaddr*)&leg->base.addr, ip_addr),
osmo_sockaddr_port((struct sockaddr*)&leg->base.addr));
if (other->update_rtp)
other->update_rtp(leg->base.call->remote);
} else {
LOGP(DSIP, LOGL_INFO, "re-INVITE does not change media connection (%s:%u)\n",
osmo_sockaddr_ntop((struct sockaddr*)&prev_addr, ip_addr),
osmo_sockaddr_port((struct sockaddr*)&prev_addr));
}
sdp = sdp_create_file(leg, other, sdp_sendrecv);
}
LOGP(DSIP, LOGL_DEBUG, "Sending 200 response to re-INVITE for mode(%u)\n", mode);
nua_respond(nh, SIP_200_OK,
NUTAG_MEDIA_ENABLE(0),
SIPTAG_CONTENT_TYPE_STR("application/sdp"),
SIPTAG_PAYLOAD_STR(sdp),
TAG_END());
talloc_free(sdp);
return;
}
/* Sofia SIP definitions come with error code numbers and strings, this
* map allows us to reuse the existing definitions.
* The map is in priority order. The first matching entry found
* will be used.
*/
static struct cause_map {
int sip_status;
const char *sip_phrase;
int gsm48_cause;
const char *q850_reason;
} cause_map[] = {
{ SIP_200_OK, GSM48_CC_CAUSE_NORM_CALL_CLEAR, "Normal Call Clearing" },
{ SIP_403_FORBIDDEN, GSM48_CC_CAUSE_CALL_REJECTED, "Call Rejected" },
{ SIP_401_UNAUTHORIZED, GSM48_CC_CAUSE_CALL_REJECTED, "Call Rejected" },
{ SIP_402_PAYMENT_REQUIRED, GSM48_CC_CAUSE_CALL_REJECTED, "Call Rejected" },
{ SIP_407_PROXY_AUTH_REQUIRED, GSM48_CC_CAUSE_CALL_REJECTED, "Call Rejected" },
{ SIP_603_DECLINE, GSM48_CC_CAUSE_CALL_REJECTED, "Call Rejected" },
{ SIP_406_NOT_ACCEPTABLE, GSM48_CC_CAUSE_CHAN_UNACCEPT, "Channel Unacceptable" },
{ SIP_404_NOT_FOUND, GSM48_CC_CAUSE_UNASSIGNED_NR, "Unallocated Number" },
{ SIP_485_AMBIGUOUS, GSM48_CC_CAUSE_NO_ROUTE, "No Route to Destination" },
{ SIP_604_DOES_NOT_EXIST_ANYWHERE, GSM48_CC_CAUSE_NO_ROUTE, "No Route to Destination" },
{ SIP_504_GATEWAY_TIME_OUT, GSM48_CC_CAUSE_RECOVERY_TIMER, "Recovery on Timer Expiry" },
{ SIP_408_REQUEST_TIMEOUT, GSM48_CC_CAUSE_RECOVERY_TIMER, "Recovery on Timer Expiry" },
{ SIP_410_GONE, GSM48_CC_CAUSE_NUMBER_CHANGED, "Number Changed" },
{ SIP_416_UNSUPPORTED_URI, GSM48_CC_CAUSE_INVAL_TRANS_ID, "Invalid Call Reference Value" },
{ SIP_420_BAD_EXTENSION, GSM48_CC_CAUSE_INTERWORKING, "Interworking, Unspecified" },
{ SIP_414_REQUEST_URI_TOO_LONG, GSM48_CC_CAUSE_INTERWORKING, "Interworking, Unspecified" },
{ SIP_413_REQUEST_TOO_LARGE, GSM48_CC_CAUSE_INTERWORKING, "Interworking, Unspecified" },
{ SIP_421_EXTENSION_REQUIRED, GSM48_CC_CAUSE_INTERWORKING, "Interworking, Unspecified" },
{ SIP_423_INTERVAL_TOO_BRIEF, GSM48_CC_CAUSE_INTERWORKING, "Interworking, Unspecified" },
{ SIP_505_VERSION_NOT_SUPPORTED, GSM48_CC_CAUSE_INTERWORKING, "Interworking, Unspecified" },
{ SIP_513_MESSAGE_TOO_LARGE, GSM48_CC_CAUSE_INTERWORKING, "Interworking, Unspecified" },
{ SIP_480_TEMPORARILY_UNAVAILABLE, GSM48_CC_CAUSE_USER_NOTRESPOND, "No User Responding" },
{ SIP_503_SERVICE_UNAVAILABLE, GSM48_CC_CAUSE_RESOURCE_UNAVAIL,"Resource Unavailable, Unspecified" },
{ SIP_503_SERVICE_UNAVAILABLE, GSM48_CC_CAUSE_TEMP_FAILURE, "Temporary Failure" },
{ SIP_503_SERVICE_UNAVAILABLE, GSM48_CC_CAUSE_SWITCH_CONG, "Switching Equipment Congestion" },
{ SIP_400_BAD_REQUEST, GSM48_CC_CAUSE_TEMP_FAILURE, "Temporary Failure" },
{ SIP_481_NO_CALL, GSM48_CC_CAUSE_TEMP_FAILURE, "Temporary Failure" },
{ SIP_500_INTERNAL_SERVER_ERROR, GSM48_CC_CAUSE_TEMP_FAILURE, "Temporary Failure" },
{ SIP_486_BUSY_HERE, GSM48_CC_CAUSE_USER_BUSY, "User Busy" },
{ SIP_600_BUSY_EVERYWHERE, GSM48_CC_CAUSE_USER_BUSY, "User Busy" },
{ SIP_484_ADDRESS_INCOMPLETE, GSM48_CC_CAUSE_INV_NR_FORMAT, "Invalid Number Format (addr incomplete)" },
{ SIP_488_NOT_ACCEPTABLE, GSM48_CC_CAUSE_INCOMPAT_DEST, "Incompatible Destination" },
{ SIP_606_NOT_ACCEPTABLE, GSM48_CC_CAUSE_INCOMPAT_DEST, "Incompatible Destination" },
{ SIP_502_BAD_GATEWAY, GSM48_CC_CAUSE_DEST_OOO, "Destination Out of Order" },
{ SIP_503_SERVICE_UNAVAILABLE, GSM48_CC_CAUSE_NETWORK_OOO, "Network Out of Order" },
{ SIP_405_METHOD_NOT_ALLOWED, GSM48_CC_CAUSE_SERV_OPT_UNAVAIL,"Service or Option Not Implemented" },
{ SIP_501_NOT_IMPLEMENTED, GSM48_CC_CAUSE_SERV_OPT_UNIMPL, "Service or Option Not Implemented" },
{ SIP_415_UNSUPPORTED_MEDIA, GSM48_CC_CAUSE_SERV_OPT_UNIMPL, "Service or Option Not Implemented" },
{ SIP_406_NOT_ACCEPTABLE, GSM48_CC_CAUSE_SERV_OPT_UNIMPL, "Service or Option Not Implemented" },
{ SIP_482_LOOP_DETECTED, GSM48_CC_CAUSE_PRE_EMPTION, "Exchange Routing Error" },
{ SIP_483_TOO_MANY_HOPS, GSM48_CC_CAUSE_PRE_EMPTION, "Exchange Routing Error" },
{ SIP_503_SERVICE_UNAVAILABLE, GSM48_CC_CAUSE_BEARER_CA_UNAVAIL,"Bearer Capability Not Available" },
{ SIP_480_TEMPORARILY_UNAVAILABLE, GSM48_CC_CAUSE_NORMAL_UNSPEC, "Normal, Unspecified" }
};
static int status2cause(int status)
{
uint8_t i;
for (i = 0; i < ARRAY_SIZE(cause_map) - 1; i++) {
if (cause_map[i].sip_status == status) {
return cause_map[i].gsm48_cause;
}
}
return GSM48_CC_CAUSE_NORMAL_UNSPEC;
}
void nua_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[])
{
LOGP(DSIP, LOGL_DEBUG, "SIP event[%s] status(%d) phrase(%s) %p\n",
nua_event_name(event), status, phrase, hmagic);
if (event == nua_r_invite) {
struct sip_call_leg *leg;
leg = (struct sip_call_leg *) hmagic;
/* MT call is moving forward */
/* The dialogue is now confirmed */
if (leg->state == SIP_CC_INITIAL)
leg->state = SIP_CC_DLG_CNFD;
if (status == 180 || status == 183)
call_progress(leg, sip, status);
else if (status == 200) {
if (leg->state == SIP_CC_CONNECTED || leg->state == SIP_CC_HOLD) {
/* This 200 is a response to our re-INVITE on
* a connected call. We just need to ACK it. */
nua_ack(leg->nua_handle, TAG_END());
} else {
call_connect(leg, sip);
}
}
else if (status >= 300) {
struct call_leg *other = call_leg_other(&leg->base);
LOGP(DSIP, LOGL_INFO, "INVITE got status(%d), releasing leg(%p).\n", status, leg);
nua_cancel(leg->nua_handle, TAG_END());
nua_handle_destroy(leg->nua_handle);
call_leg_release(&leg->base);
if (other) {
LOGP(DSIP, LOGL_INFO, "Releasing MNCC leg (%p) with status(%d)\n", other, status);
other->cause = status2cause(status);
other->release_call(other);
}
}
} else if (event == nua_i_ack) {
/* SDP comes back to us in 200 ACK after we
* respond to the re-INVITE query. */
if (sip->sip_payload && sip->sip_payload->pl_data) {
struct sip_call_leg *leg = sip_find_leg(nh);
if (leg)
sip_handle_reinvite(leg, nh, sip);
}
} else if (event == nua_r_bye || event == nua_r_cancel) {
/* our bye or hang up is answered */
struct sip_call_leg *leg = (struct sip_call_leg *) hmagic;
LOGP(DSIP, LOGL_INFO, "leg(%p) got resp to %s\n",
leg, event == nua_r_bye ? "bye" : "cancel");
nua_handle_destroy(leg->nua_handle);
call_leg_release(&leg->base);
} else if (event == nua_i_bye) {
/* our remote has hung up */
struct sip_call_leg *leg = (struct sip_call_leg *) hmagic;
struct call_leg *other = call_leg_other(&leg->base);
LOGP(DSIP, LOGL_INFO, "leg(%p) got bye, releasing.\n", leg);
nua_handle_destroy(leg->nua_handle);
call_leg_release(&leg->base);
if (other)
other->release_call(other);
} else if (event == nua_i_invite) {
/* new incoming leg or re-INVITE */
LOGP(DSIP, LOGL_INFO, "Processing INVITE Call-ID: %s\n", sip->sip_call_id->i_id);
if (status == 100) {
struct sip_call_leg *leg = sip_find_leg(nh);
if (leg)
sip_handle_reinvite(leg, nh, sip);
else
new_call((struct sip_agent *) magic, nh, sip);
}
} else if (event == nua_i_cancel) {
struct sip_call_leg *leg;
struct call_leg *other;
LOGP(DSIP, LOGL_INFO, "Cancelled on leg(%p)\n", hmagic);
leg = (struct sip_call_leg *) hmagic;
other = call_leg_other(&leg->base);
nua_handle_destroy(leg->nua_handle);
call_leg_release(&leg->base);
if (other)
other->release_call(other);
} else {
LOGP(DSIP, LOGL_DEBUG, "Did not handle event[%s] status(%d)\n", nua_event_name(event), status);
}
}
static void cause2status(int cause, int *sip_status, const char **sip_phrase, const char **reason_text)
{
uint8_t i;
for (i = 0; i < ARRAY_SIZE(cause_map) - 1; i++) {
if (cause_map[i].gsm48_cause == cause) {
*sip_status = cause_map[i].sip_status;
*sip_phrase = cause_map[i].sip_phrase;
*reason_text = cause_map[i].q850_reason;
LOGP(DSIP, LOGL_DEBUG, "%s(): Mapping cause(%s) to status(%d)\n",
__func__, gsm48_cc_cause_name(cause), *sip_status);
return;
}
}
LOGP(DSIP, LOGL_ERROR, "%s(): Cause(%s) not found in map.\n", __func__, gsm48_cc_cause_name(cause));
*sip_status = cause_map[i].sip_status;
*sip_phrase = cause_map[i].sip_phrase;
*reason_text = cause_map[i].q850_reason;
}
static void sip_release_call(struct call_leg *_leg)
{
struct sip_call_leg *leg;
char reason[64];
int sip_cause;
const char *sip_phrase;
const char *reason_text;
OSMO_ASSERT(_leg->type == CALL_TYPE_SIP);
leg = (struct sip_call_leg *) _leg;
/*
* If a dialogue is not confirmed yet, we can probably not do much
* but wait for the timeout. For a confirmed one we can send cancel
* and for a connected one bye. I don't see how sofia-sip is going
* to help us here.
*/
LOGP(DSIP, LOGL_INFO, "%s(): Release with MNCC cause(%s)\n", __func__, gsm48_cc_cause_name(_leg->cause));
cause2status(_leg->cause, &sip_cause, &sip_phrase, &reason_text);
snprintf(reason, sizeof reason, "Q.850;cause=%u;text=\"%s\"", _leg->cause, reason_text);
switch (leg->state) {
case SIP_CC_INITIAL:
LOGP(DSIP, LOGL_INFO, "Cancelling leg(%p) in initial state\n", leg);
nua_handle_destroy(leg->nua_handle);
call_leg_release(&leg->base);
break;
case SIP_CC_DLG_CNFD:
LOGP(DSIP, LOGL_INFO, "Cancelling leg(%p) in confirmed state\n", leg);
if (leg->dir == SIP_DIR_MT)
nua_cancel(leg->nua_handle, TAG_END());
else {
nua_respond(leg->nua_handle, sip_cause, sip_phrase,
SIPTAG_REASON_STR(reason),
TAG_END());
nua_handle_destroy(leg->nua_handle);
call_leg_release(&leg->base);
}
break;
case SIP_CC_CONNECTED:
case SIP_CC_HOLD:
LOGP(DSIP, LOGL_NOTICE, "Ending leg(%p) in connected state.\n", leg);
nua_bye(leg->nua_handle, TAG_END());
break;
}
}
static void sip_ring_call(struct call_leg *_leg)
{
struct sip_call_leg *leg;
OSMO_ASSERT(_leg->type == CALL_TYPE_SIP);
leg = (struct sip_call_leg *) _leg;
nua_respond(leg->nua_handle, SIP_180_RINGING, TAG_END());
}
static void sip_connect_call(struct call_leg *_leg)
{
struct call_leg *other;
struct sip_call_leg *leg;
char *sdp;
OSMO_ASSERT(_leg->type == CALL_TYPE_SIP);
leg = (struct sip_call_leg *) _leg;
/*
* TODO/FIXME: check if resulting codec is compatible..
*/
other = call_leg_other(&leg->base);
if (!other) {
sip_release_call(&leg->base);
return;
}
sdp = sdp_create_file(leg, other, sdp_sendrecv);
leg->state = SIP_CC_CONNECTED;
nua_respond(leg->nua_handle, SIP_200_OK,
NUTAG_MEDIA_ENABLE(0),
SIPTAG_CONTENT_TYPE_STR("application/sdp"),
SIPTAG_PAYLOAD_STR(sdp),
TAG_END());
talloc_free(sdp);
}
static void sip_dtmf_call(struct call_leg *_leg, int keypad)
{
struct sip_call_leg *leg;
char *buf;
OSMO_ASSERT(_leg->type == CALL_TYPE_SIP);
leg = (struct sip_call_leg *) _leg;
buf = talloc_asprintf(leg, "Signal=%c\nDuration=160\n", keypad);
nua_info(leg->nua_handle,
NUTAG_MEDIA_ENABLE(0),
SIPTAG_CONTENT_TYPE_STR("application/dtmf-relay"),
SIPTAG_PAYLOAD_STR(buf), TAG_END());
talloc_free(buf);
}
static void sip_hold_call(struct call_leg *_leg)
{
struct sip_call_leg *leg;
struct call_leg *other_leg;
OSMO_ASSERT(_leg->type == CALL_TYPE_SIP);
leg = (struct sip_call_leg *) _leg;
other_leg = call_leg_other(&leg->base);
if (!other_leg) {
LOGP(DMNCC, LOGL_ERROR, "leg(%p) other leg gone!\n", leg);
sip_release_call(&leg->base);
return;
}
char *sdp = sdp_create_file(leg, other_leg, sdp_sendonly);
nua_invite(leg->nua_handle,
NUTAG_MEDIA_ENABLE(0),
SIPTAG_CONTENT_TYPE_STR("application/sdp"),
SIPTAG_PAYLOAD_STR(sdp),
TAG_END());
talloc_free(sdp);
leg->state = SIP_CC_HOLD;
}
static void sip_retrieve_call(struct call_leg *_leg)
{
struct sip_call_leg *leg;
struct call_leg *other_leg;
OSMO_ASSERT(_leg->type == CALL_TYPE_SIP);
leg = (struct sip_call_leg *) _leg;
other_leg = call_leg_other(&leg->base);
if (!other_leg) {
LOGP(DMNCC, LOGL_ERROR, "leg(%p) other leg gone!\n", leg);
sip_release_call(&leg->base);
return;
}
char *sdp = sdp_create_file(leg, other_leg, sdp_sendrecv);
nua_invite(leg->nua_handle,
NUTAG_MEDIA_ENABLE(0),
SIPTAG_CONTENT_TYPE_STR("application/sdp"),
SIPTAG_PAYLOAD_STR(sdp),
TAG_END());
talloc_free(sdp);
leg->state = SIP_CC_CONNECTED;
}
static int send_invite(struct sip_agent *agent, struct sip_call_leg *leg,
const char *calling_num, const char *called_num)
{
struct call_leg *other = leg->base.call->initial;
char *from = talloc_asprintf(leg, "sip:%s@%s:%d",
calling_num,
agent->app->sip.local_addr,
agent->app->sip.local_port);
char *to = talloc_asprintf(leg, "sip:%s@%s:%d",
called_num,
agent->app->sip.remote_addr,
agent->app->sip.remote_port);
char *sdp = sdp_create_file(leg, other, sdp_sendrecv);
/* Encode the Global Call Reference (if present) */
char *x_gcr = NULL;
if (leg->base.call->gcr_present) {
struct msgb *msg = msgb_alloc(16, "SIP GCR");
if (msg != NULL && osmo_enc_gcr(msg, &leg->base.call->gcr) > 0)
x_gcr = talloc_asprintf(leg, "X-Global-Call-Ref: %s", msgb_hexdump(msg));
else
LOGP(DSIP, LOGL_ERROR, "Failed to encode GCR for leg(%p)\n", leg);
msgb_free(msg);
}
leg->state = SIP_CC_INITIAL;
leg->dir = SIP_DIR_MT;
nua_invite(leg->nua_handle,
SIPTAG_FROM_STR(from),
SIPTAG_TO_STR(to),
NUTAG_MEDIA_ENABLE(0),
SIPTAG_CONTENT_TYPE_STR("application/sdp"),
TAG_IF(x_gcr, SIPTAG_HEADER_STR(x_gcr)),
SIPTAG_PAYLOAD_STR(sdp),
TAG_END());
leg->base.call->remote = &leg->base;
talloc_free(from);
talloc_free(to);
talloc_free(sdp);
talloc_free(x_gcr);
return 0;
}
int sip_create_remote_leg(struct sip_agent *agent, struct call *call)
{
struct sip_call_leg *leg;
leg = talloc_zero(call, struct sip_call_leg);
if (!leg) {
LOGP(DSIP, LOGL_ERROR, "Failed to allocate leg for call(%u)\n",
call->id);
return -1;
}
leg->base.type = CALL_TYPE_SIP;
leg->base.call = call;
leg->base.release_call = sip_release_call;
leg->base.dtmf = sip_dtmf_call;
leg->base.hold_call = sip_hold_call;
leg->base.retrieve_call = sip_retrieve_call;
leg->agent = agent;
leg->nua_handle = nua_handle(agent->nua, leg, TAG_END());
if (!leg->nua_handle) {
LOGP(DSIP, LOGL_ERROR, "Failed to allocate nua for call(%u)\n",
call->id);
talloc_free(leg);
return -2;
}
return send_invite(agent, leg, call->source, call->dest);
}
char *make_sip_uri(struct sip_agent *agent)
{
const char *hostname = agent->app->sip.local_addr;
/* We need to map 0.0.0.0 to '*' to bind everywhere */
if (strcmp(hostname, "0.0.0.0") == 0)
hostname = "*";
return talloc_asprintf(tall_mncc_ctx, "sip:%s:%d",
agent->app->sip.local_addr,
agent->app->sip.local_port);
}
/* http://sofia-sip.sourceforge.net/refdocs/debug_logs.html */
static void sip_logger(void *stream, char const *fmt, va_list ap)
{
/* this is ugly, as unfortunately sofia-sip does not pass the log level to
* the log handler call-back function, so we have no clue what log level the
* currently logged message was sent for :( As a result, we can only use one
* hard-coded LOGL_NOTICE here */
if (!log_check_level(DSIP, LOGL_NOTICE))
return;
/* The sof