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review.jami.net / pjproject / 903154fe62e950423f77c9116048363d679d5f2e / . / pjsip / src / pjsua-lib / pjsua_call.c
blob: 64aeac4dc3ec0d768b47eb9f8e025649cbf11246 [file] [log] [blame]
/* $Id$ */
/*
* Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com)
* Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 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 General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include <pjsua-lib/pjsua.h>
#include <pjsua-lib/pjsua_internal.h>
#define THIS_FILE "pjsua_call.c"
/* Retry interval of sending re-INVITE for locking a codec when remote
* SDP answer contains multiple codec, in milliseconds.
*/
#define LOCK_CODEC_RETRY_INTERVAL 200
/*
* Max UPDATE/re-INVITE retry to lock codec
*/
#define LOCK_CODEC_MAX_RETRY 5
/* This callback receives notification from invite session when the
* session state has changed.
*/
static void pjsua_call_on_state_changed(pjsip_inv_session *inv,
pjsip_event *e);
/* This callback is called by invite session framework when UAC session
* has forked.
*/
static void pjsua_call_on_forked( pjsip_inv_session *inv,
pjsip_event *e);
/*
* Callback to be called when SDP offer/answer negotiation has just completed
* in the session. This function will start/update media if negotiation
* has succeeded.
*/
static void pjsua_call_on_media_update(pjsip_inv_session *inv,
pj_status_t status);
/*
* Called when session received new offer.
*/
static void pjsua_call_on_rx_offer(pjsip_inv_session *inv,
const pjmedia_sdp_session *offer);
/*
* Called to generate new offer.
*/
static void pjsua_call_on_create_offer(pjsip_inv_session *inv,
pjmedia_sdp_session **offer);
/*
* This callback is called when transaction state has changed in INVITE
* session. We use this to trap:
* - incoming REFER request.
* - incoming MESSAGE request.
*/
static void pjsua_call_on_tsx_state_changed(pjsip_inv_session *inv,
pjsip_transaction *tsx,
pjsip_event *e);
/*
* Redirection handler.
*/
static pjsip_redirect_op pjsua_call_on_redirected(pjsip_inv_session *inv,
const pjsip_uri *target,
const pjsip_event *e);
/* Create SDP for call hold. */
static pj_status_t create_sdp_of_call_hold(pjsua_call *call,
pjmedia_sdp_session **p_sdp);
/*
* Callback called by event framework when the xfer subscription state
* has changed.
*/
static void xfer_client_on_evsub_state( pjsip_evsub *sub, pjsip_event *event);
static void xfer_server_on_evsub_state( pjsip_evsub *sub, pjsip_event *event);
/*
* Reset call descriptor.
*/
static void reset_call(pjsua_call_id id)
{
pjsua_call *call = &pjsua_var.calls[id];
unsigned i;
pj_bzero(call, sizeof(*call));
call->index = id;
call->last_text.ptr = call->last_text_buf_;
for (i=0; i<PJ_ARRAY_SIZE(call->media); ++i) {
pjsua_call_media *call_med = &call->media[i];
call_med->ssrc = pj_rand();
call_med->strm.a.conf_slot = PJSUA_INVALID_ID;
call_med->strm.v.cap_win_id = PJSUA_INVALID_ID;
call_med->strm.v.rdr_win_id = PJSUA_INVALID_ID;
call_med->call = call;
call_med->idx = i;
call_med->tp_auto_del = PJ_TRUE;
}
}
/*
* Init call subsystem.
*/
pj_status_t pjsua_call_subsys_init(const pjsua_config *cfg)
{
pjsip_inv_callback inv_cb;
unsigned i;
const pj_str_t str_norefersub = { "norefersub", 10 };
pj_status_t status;
/* Init calls array. */
for (i=0; i<PJ_ARRAY_SIZE(pjsua_var.calls); ++i)
reset_call(i);
/* Copy config */
pjsua_config_dup(pjsua_var.pool, &pjsua_var.ua_cfg, cfg);
/* Verify settings */
if (pjsua_var.ua_cfg.max_calls >= PJSUA_MAX_CALLS) {
pjsua_var.ua_cfg.max_calls = PJSUA_MAX_CALLS;
}
/* Check the route URI's and force loose route if required */
for (i=0; i<pjsua_var.ua_cfg.outbound_proxy_cnt; ++i) {
status = normalize_route_uri(pjsua_var.pool,
&pjsua_var.ua_cfg.outbound_proxy[i]);
if (status != PJ_SUCCESS)
return status;
}
/* Initialize invite session callback. */
pj_bzero(&inv_cb, sizeof(inv_cb));
inv_cb.on_state_changed = &pjsua_call_on_state_changed;
inv_cb.on_new_session = &pjsua_call_on_forked;
inv_cb.on_media_update = &pjsua_call_on_media_update;
inv_cb.on_rx_offer = &pjsua_call_on_rx_offer;
inv_cb.on_create_offer = &pjsua_call_on_create_offer;
inv_cb.on_tsx_state_changed = &pjsua_call_on_tsx_state_changed;
inv_cb.on_redirected = &pjsua_call_on_redirected;
/* Initialize invite session module: */
status = pjsip_inv_usage_init(pjsua_var.endpt, &inv_cb);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
/* Add "norefersub" in Supported header */
pjsip_endpt_add_capability(pjsua_var.endpt, NULL, PJSIP_H_SUPPORTED,
NULL, 1, &str_norefersub);
return status;
}
/*
* Start call subsystem.
*/
pj_status_t pjsua_call_subsys_start(void)
{
/* Nothing to do */
return PJ_SUCCESS;
}
/*
* Get maximum number of calls configured in pjsua.
*/
PJ_DEF(unsigned) pjsua_call_get_max_count(void)
{
return pjsua_var.ua_cfg.max_calls;
}
/*
* Get number of currently active calls.
*/
PJ_DEF(unsigned) pjsua_call_get_count(void)
{
return pjsua_var.call_cnt;
}
/*
* Enum calls.
*/
PJ_DEF(pj_status_t) pjsua_enum_calls( pjsua_call_id ids[],
unsigned *count)
{
unsigned i, c;
PJ_ASSERT_RETURN(ids && *count, PJ_EINVAL);
PJSUA_LOCK();
for (i=0, c=0; c<*count && i<pjsua_var.ua_cfg.max_calls; ++i) {
if (!pjsua_var.calls[i].inv)
continue;
ids[c] = i;
++c;
}
*count = c;
PJSUA_UNLOCK();
return PJ_SUCCESS;
}
/* Allocate one call id */
static pjsua_call_id alloc_call_id(void)
{
pjsua_call_id cid;
#if 1
/* New algorithm: round-robin */
if (pjsua_var.next_call_id >= (int)pjsua_var.ua_cfg.max_calls ||
pjsua_var.next_call_id < 0)
{
pjsua_var.next_call_id = 0;
}
for (cid=pjsua_var.next_call_id;
cid<(int)pjsua_var.ua_cfg.max_calls;
++cid)
{
if (pjsua_var.calls[cid].inv == NULL) {
++pjsua_var.next_call_id;
return cid;
}
}
for (cid=0; cid < pjsua_var.next_call_id; ++cid) {
if (pjsua_var.calls[cid].inv == NULL) {
++pjsua_var.next_call_id;
return cid;
}
}
#else
/* Old algorithm */
for (cid=0; cid<(int)pjsua_var.ua_cfg.max_calls; ++cid) {
if (pjsua_var.calls[cid].inv == NULL)
return cid;
}
#endif
return PJSUA_INVALID_ID;
}
/* Get signaling secure level.
* Return:
* 0: if signaling is not secure
* 1: if TLS transport is used for immediate hop
* 2: if end-to-end signaling is secure.
*/
static int get_secure_level(pjsua_acc_id acc_id, const pj_str_t *dst_uri)
{
const pj_str_t tls = pj_str(";transport=tls");
const pj_str_t sips = pj_str("sips:");
pjsua_acc *acc = &pjsua_var.acc[acc_id];
if (pj_stristr(dst_uri, &sips))
return 2;
if (!pj_list_empty(&acc->route_set)) {
pjsip_route_hdr *r = acc->route_set.next;
pjsip_uri *uri = r->name_addr.uri;
pjsip_sip_uri *sip_uri;
sip_uri = (pjsip_sip_uri*)pjsip_uri_get_uri(uri);
if (pj_stricmp2(&sip_uri->transport_param, "tls")==0)
return 1;
} else {
if (pj_stristr(dst_uri, &tls))
return 1;
}
return 0;
}
/*
static int call_get_secure_level(pjsua_call *call)
{
if (call->inv->dlg->secure)
return 2;
if (!pj_list_empty(&call->inv->dlg->route_set)) {
pjsip_route_hdr *r = call->inv->dlg->route_set.next;
pjsip_uri *uri = r->name_addr.uri;
pjsip_sip_uri *sip_uri;
sip_uri = (pjsip_sip_uri*)pjsip_uri_get_uri(uri);
if (pj_stricmp2(&sip_uri->transport_param, "tls")==0)
return 1;
} else {
pjsip_sip_uri *sip_uri;
if (PJSIP_URI_SCHEME_IS_SIPS(call->inv->dlg->target))
return 2;
if (!PJSIP_URI_SCHEME_IS_SIP(call->inv->dlg->target))
return 0;
sip_uri = (pjsip_sip_uri*) pjsip_uri_get_uri(call->inv->dlg->target);
if (pj_stricmp2(&sip_uri->transport_param, "tls")==0)
return 1;
}
return 0;
}
*/
/* Outgoing call callback when media transport creation is completed. */
static pj_status_t
on_make_call_med_tp_complete(pjsua_call_id call_id,
const pjsua_med_tp_state_info *info)
{
pjmedia_sdp_session *offer;
pjsip_inv_session *inv = NULL;
pjsua_call *call = &pjsua_var.calls[call_id];
pjsua_acc *acc = &pjsua_var.acc[call->acc_id];
pjsip_dialog *dlg = call->async_call.dlg;
unsigned options = call->async_call.call_var.out_call.options;
pjsip_tx_data *tdata;
pj_status_t status = (info? info->status: PJ_SUCCESS);
PJSUA_LOCK();
/* Increment the dialog's lock otherwise when invite session creation
* fails the dialog will be destroyed prematurely.
*/
pjsip_dlg_inc_lock(dlg);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Error initializing media channel", status);
goto on_error;
}
/* pjsua_media_channel_deinit() has been called. */
if (call->async_call.med_ch_deinit)
goto on_error;
/* Create offer */
status = pjsua_media_channel_create_sdp(call->index, dlg->pool, NULL,
&offer, NULL);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Error initializing media channel", status);
goto on_error;
}
/* Create the INVITE session: */
options |= PJSIP_INV_SUPPORT_100REL;
if (acc->cfg.require_100rel)
options |= PJSIP_INV_REQUIRE_100REL;
if (acc->cfg.use_timer != PJSUA_SIP_TIMER_INACTIVE) {
options |= PJSIP_INV_SUPPORT_TIMER;
if (acc->cfg.use_timer == PJSUA_SIP_TIMER_REQUIRED)
options |= PJSIP_INV_REQUIRE_TIMER;
else if (acc->cfg.use_timer == PJSUA_SIP_TIMER_ALWAYS)
options |= PJSIP_INV_ALWAYS_USE_TIMER;
}
status = pjsip_inv_create_uac( dlg, offer, options, &inv);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Invite session creation failed", status);
goto on_error;
}
/* Init Session Timers */
status = pjsip_timer_init_session(inv, &acc->cfg.timer_setting);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Session Timer init failed", status);
goto on_error;
}
/* Create and associate our data in the session. */
call->inv = inv;
dlg->mod_data[pjsua_var.mod.id] = call;
inv->mod_data[pjsua_var.mod.id] = call;
/* If account is locked to specific transport, then lock dialog
* to this transport too.
*/
if (acc->cfg.transport_id != PJSUA_INVALID_ID) {
pjsip_tpselector tp_sel;
pjsua_init_tpselector(acc->cfg.transport_id, &tp_sel);
pjsip_dlg_set_transport(dlg, &tp_sel);
}
/* Set dialog Route-Set: */
if (!pj_list_empty(&acc->route_set))
pjsip_dlg_set_route_set(dlg, &acc->route_set);
/* Set credentials: */
if (acc->cred_cnt) {
pjsip_auth_clt_set_credentials( &dlg->auth_sess,
acc->cred_cnt, acc->cred);
}
/* Set authentication preference */
pjsip_auth_clt_set_prefs(&dlg->auth_sess, &acc->cfg.auth_pref);
/* Create initial INVITE: */
status = pjsip_inv_invite(inv, &tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create initial INVITE request",
status);
goto on_error;
}
/* Add additional headers etc */
pjsua_process_msg_data( tdata,
call->async_call.call_var.out_call.msg_data);
/* Must increment call counter now */
++pjsua_var.call_cnt;
/* Send initial INVITE: */
status = pjsip_inv_send_msg(inv, tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to send initial INVITE request",
status);
/* Upon failure to send first request, the invite
* session would have been cleared.
*/
inv = NULL;
goto on_error;
}
/* Done. */
pjsip_dlg_dec_lock(dlg);
PJSUA_UNLOCK();
return PJ_SUCCESS;
on_error:
if (dlg) {
/* This may destroy the dialog */
pjsip_dlg_dec_lock(dlg);
}
if (inv != NULL) {
pjsip_inv_terminate(inv, PJSIP_SC_OK, PJ_FALSE);
}
if (call_id != -1) {
reset_call(call_id);
pjsua_media_channel_deinit(call_id);
}
PJSUA_UNLOCK();
return status;
}
/*
* Make outgoing call to the specified URI using the specified account.
*/
PJ_DEF(pj_status_t) pjsua_call_make_call( pjsua_acc_id acc_id,
const pj_str_t *dest_uri,
unsigned options,
void *user_data,
const pjsua_msg_data *msg_data,
pjsua_call_id *p_call_id)
{
pj_pool_t *tmp_pool = NULL;
pjsip_dialog *dlg = NULL;
pjsua_acc *acc;
pjsua_call *call;
int call_id = -1;
pj_str_t contact;
pj_status_t status;
/* Check that account is valid */
PJ_ASSERT_RETURN(acc_id>=0 || acc_id<(int)PJ_ARRAY_SIZE(pjsua_var.acc),
PJ_EINVAL);
/* Check arguments */
PJ_ASSERT_RETURN(dest_uri, PJ_EINVAL);
PJ_LOG(4,(THIS_FILE, "Making call with acc #%d to %.*s", acc_id,
(int)dest_uri->slen, dest_uri->ptr));
pj_log_push_indent();
PJSUA_LOCK();
/* Create sound port if none is instantiated, to check if sound device
* can be used. But only do this with the conference bridge, as with
* audio switchboard (i.e. APS-Direct), we can only open the sound
* device once the correct format has been known
*/
if (!pjsua_var.is_mswitch && pjsua_var.snd_port==NULL &&
pjsua_var.null_snd==NULL && !pjsua_var.no_snd)
{
status = pjsua_set_snd_dev(pjsua_var.cap_dev, pjsua_var.play_dev);
if (status != PJ_SUCCESS)
goto on_error;
}
acc = &pjsua_var.acc[acc_id];
if (!acc->valid) {
pjsua_perror(THIS_FILE, "Unable to make call because account "
"is not valid", PJ_EINVALIDOP);
status = PJ_EINVALIDOP;
goto on_error;
}
/* Find free call slot. */
call_id = alloc_call_id();
if (call_id == PJSUA_INVALID_ID) {
pjsua_perror(THIS_FILE, "Error making call", PJ_ETOOMANY);
status = PJ_ETOOMANY;
goto on_error;
}
call = &pjsua_var.calls[call_id];
/* Associate session with account */
call->acc_id = acc_id;
call->call_hold_type = acc->cfg.call_hold_type;
/* Create temporary pool */
tmp_pool = pjsua_pool_create("tmpcall10", 512, 256);
/* Verify that destination URI is valid before calling
* pjsua_acc_create_uac_contact, or otherwise there
* a misleading "Invalid Contact URI" error will be printed
* when pjsua_acc_create_uac_contact() fails.
*/
if (1) {
pjsip_uri *uri;
pj_str_t dup;
pj_strdup_with_null(tmp_pool, &dup, dest_uri);
uri = pjsip_parse_uri(tmp_pool, dup.ptr, dup.slen, 0);
if (uri == NULL) {
pjsua_perror(THIS_FILE, "Unable to make call",
PJSIP_EINVALIDREQURI);
status = PJSIP_EINVALIDREQURI;
goto on_error;
}
}
/* Mark call start time. */
pj_gettimeofday(&call->start_time);
/* Reset first response time */
call->res_time.sec = 0;
/* Create suitable Contact header unless a Contact header has been
* set in the account.
*/
if (acc->contact.slen) {
contact = acc->contact;
} else {
status = pjsua_acc_create_uac_contact(tmp_pool, &contact,
acc_id, dest_uri);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to generate Contact header",
status);
goto on_error;
}
}
/* Create outgoing dialog: */
status = pjsip_dlg_create_uac( pjsip_ua_instance(),
&acc->cfg.id, &contact,
dest_uri, dest_uri, &dlg);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Dialog creation failed", status);
goto on_error;
}
/* Increment the dialog's lock otherwise when invite session creation
* fails the dialog will be destroyed prematurely.
*/
// pjsip_dlg_inc_lock(dlg);
/* Calculate call's secure level */
call->secure_level = get_secure_level(acc_id, dest_uri);
/* Attach user data */
call->user_data = user_data;
/* Store variables required for the callback after the async
* media transport creation is completed.
*/
call->async_call.call_var.out_call.options = options;
if (msg_data) {
call->async_call.call_var.out_call.msg_data = pjsua_msg_data_clone(
dlg->pool, msg_data);
}
call->async_call.dlg = dlg;
/* Init media channel */
status = pjsua_media_channel_init(call->index, PJSIP_ROLE_UAC,
call->secure_level, dlg->pool,
NULL, NULL, PJ_TRUE,
(pjsua_med_tp_state_cb)
&on_make_call_med_tp_complete);
if (status == PJ_SUCCESS) {
status = on_make_call_med_tp_complete(call->index, NULL);
if (status != PJ_SUCCESS)
goto on_error;
} else if (status != PJ_EPENDING) {
pjsua_perror(THIS_FILE, "Error initializing media channel", status);
goto on_error;
}
/* Done. */
if (p_call_id)
*p_call_id = call_id;
pj_pool_release(tmp_pool);
PJSUA_UNLOCK();
pj_log_pop_indent();
return PJ_SUCCESS;
on_error:
if (dlg) {
pjsip_dlg_inc_lock(dlg);
/* This may destroy the dialog */
pjsip_dlg_dec_lock(dlg);
}
if (call_id != -1) {
reset_call(call_id);
pjsua_media_channel_deinit(call_id);
}
if (tmp_pool)
pj_pool_release(tmp_pool);
PJSUA_UNLOCK();
pj_log_pop_indent();
return status;
}
/* Get the NAT type information in remote's SDP */
static void update_remote_nat_type(pjsua_call *call,
const pjmedia_sdp_session *sdp)
{
const pjmedia_sdp_attr *xnat;
xnat = pjmedia_sdp_attr_find2(sdp->attr_count, sdp->attr, "X-nat", NULL);
if (xnat) {
call->rem_nat_type = (pj_stun_nat_type) (xnat->value.ptr[0] - '0');
} else {
call->rem_nat_type = PJ_STUN_NAT_TYPE_UNKNOWN;
}
PJ_LOG(5,(THIS_FILE, "Call %d: remote NAT type is %d (%s)", call->index,
call->rem_nat_type, pj_stun_get_nat_name(call->rem_nat_type)));
}
/* Incoming call callback when media transport creation is completed. */
static pj_status_t
on_incoming_call_med_tp_complete(pjsua_call_id call_id,
const pjsua_med_tp_state_info *info)
{
pjsua_call *call = &pjsua_var.calls[call_id];
const pjmedia_sdp_session *offer=NULL;
pjmedia_sdp_session *answer;
pjsip_tx_data *response = NULL;
unsigned options = 0;
int sip_err_code = (info? info->sip_err_code: 0);
pj_status_t status = (info? info->status: PJ_SUCCESS);
PJSUA_LOCK();
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Error initializing media channel", status);
goto on_return;
}
/* pjsua_media_channel_deinit() has been called. */
if (call->async_call.med_ch_deinit) {
pjsua_media_channel_deinit(call->index);
call->med_ch_cb = NULL;
PJSUA_UNLOCK();
return PJ_SUCCESS;
}
/* Get remote SDP offer (if any). */
if (call->inv->neg)
pjmedia_sdp_neg_get_neg_remote(call->inv->neg, &offer);
status = pjsua_media_channel_create_sdp(call_id,
call->async_call.dlg->pool,
offer, &answer, &sip_err_code);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Error creating SDP answer", status);
goto on_return;
}
status = pjsip_inv_set_local_sdp(call->inv, answer);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Error setting local SDP", status);
sip_err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE;
goto on_return;
}
/* Verify that we can handle the request. */
status = pjsip_inv_verify_request3(NULL,
call->inv->pool_prov, &options, offer,
answer, NULL, pjsua_var.endpt, &response);
if (status != PJ_SUCCESS) {
/*
* No we can't handle the incoming INVITE request.
*/
goto on_return;
}
on_return:
if (status != PJ_SUCCESS) {
pjsip_tx_data *tdata;
pj_status_t status_;
status_ = pjsip_inv_end_session(call->inv, sip_err_code, NULL, &tdata);
if (status_ == PJ_SUCCESS && tdata)
status_ = pjsip_inv_send_msg(call->inv, tdata);
pjsua_media_channel_deinit(call->index);
}
/* Set the callback to NULL to indicate that the async operation
* has completed.
*/
call->med_ch_cb = NULL;
if (status == PJ_SUCCESS &&
!pj_list_empty(&call->async_call.call_var.inc_call.answers))
{
struct call_answer *answer, *next;
answer = call->async_call.call_var.inc_call.answers.next;
while (answer != &call->async_call.call_var.inc_call.answers) {
next = answer->next;
pjsua_call_answer(call_id, answer->code, answer->reason,
answer->msg_data);
pj_list_erase(answer);
answer = next;
}
}
PJSUA_UNLOCK();
return status;
}
/**
* Handle incoming INVITE request.
* Called by pjsua_core.c
*/
pj_bool_t pjsua_call_on_incoming(pjsip_rx_data *rdata)
{
pj_str_t contact;
pjsip_dialog *dlg = pjsip_rdata_get_dlg(rdata);
pjsip_dialog *replaced_dlg = NULL;
pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata);
pjsip_msg *msg = rdata->msg_info.msg;
pjsip_tx_data *response = NULL;
unsigned options = 0;
pjsip_inv_session *inv = NULL;
int acc_id;
pjsua_call *call;
int call_id = -1;
int sip_err_code;
pjmedia_sdp_session *offer=NULL;
pj_status_t status;
/* Don't want to handle anything but INVITE */
if (msg->line.req.method.id != PJSIP_INVITE_METHOD)
return PJ_FALSE;
/* Don't want to handle anything that's already associated with
* existing dialog or transaction.
*/
if (dlg || tsx)
return PJ_FALSE;
/* Don't want to accept the call if shutdown is in progress */
if (pjsua_var.thread_quit_flag) {
pjsip_endpt_respond_stateless(pjsua_var.endpt, rdata,
PJSIP_SC_TEMPORARILY_UNAVAILABLE, NULL,
NULL, NULL);
return PJ_TRUE;
}
PJ_LOG(4,(THIS_FILE, "Incoming %s", rdata->msg_info.info));
pj_log_push_indent();
PJSUA_LOCK();
/* Find free call slot. */
call_id = alloc_call_id();
if (call_id == PJSUA_INVALID_ID) {
pjsip_endpt_respond_stateless(pjsua_var.endpt, rdata,
PJSIP_SC_BUSY_HERE, NULL,
NULL, NULL);
PJ_LOG(2,(THIS_FILE,
"Unable to accept incoming call (too many calls)"));
goto on_return;
}
/* Clear call descriptor */
reset_call(call_id);
call = &pjsua_var.calls[call_id];
/* Mark call start time. */
pj_gettimeofday(&call->start_time);
/* Check INVITE request for Replaces header. If Replaces header is
* present, the function will make sure that we can handle the request.
*/
status = pjsip_replaces_verify_request(rdata, &replaced_dlg, PJ_FALSE,
&response);
if (status != PJ_SUCCESS) {
/*
* Something wrong with the Replaces header.
*/
if (response) {
pjsip_response_addr res_addr;
pjsip_get_response_addr(response->pool, rdata, &res_addr);
pjsip_endpt_send_response(pjsua_var.endpt, &res_addr, response,
NULL, NULL);
} else {
/* Respond with 500 (Internal Server Error) */
pjsip_endpt_respond_stateless(pjsua_var.endpt, rdata, 500, NULL,
NULL, NULL);
}
goto on_return;
}
/* If this INVITE request contains Replaces header, notify application
* about the request so that application can do subsequent checking
* if it wants to.
*/
if (replaced_dlg != NULL && pjsua_var.ua_cfg.cb.on_call_replace_request) {
pjsua_call *replaced_call;
int st_code = 200;
pj_str_t st_text = { "OK", 2 };
/* Get the replaced call instance */
replaced_call = (pjsua_call*) replaced_dlg->mod_data[pjsua_var.mod.id];
/* Notify application */
pjsua_var.ua_cfg.cb.on_call_replace_request(replaced_call->index,
rdata, &st_code, &st_text);
/* Must specify final response */
PJ_ASSERT_ON_FAIL(st_code >= 200, st_code = 200);
/* Check if application rejects this request. */
if (st_code >= 300) {
if (st_text.slen == 2)
st_text = *pjsip_get_status_text(st_code);
pjsip_endpt_respond(pjsua_var.endpt, NULL, rdata,
st_code, &st_text, NULL, NULL, NULL);
goto on_return;
}
}
/*
* Get which account is most likely to be associated with this incoming
* call. We need the account to find which contact URI to put for
* the call.
*/
acc_id = call->acc_id = pjsua_acc_find_for_incoming(rdata);
call->call_hold_type = pjsua_var.acc[acc_id].cfg.call_hold_type;
/* Get call's secure level */
if (PJSIP_URI_SCHEME_IS_SIPS(rdata->msg_info.msg->line.req.uri))
call->secure_level = 2;
else if (PJSIP_TRANSPORT_IS_SECURE(rdata->tp_info.transport))
call->secure_level = 1;
else
call->secure_level = 0;
/* Parse SDP from incoming request */
if (rdata->msg_info.msg->body) {
pjsip_rdata_sdp_info *sdp_info;
sdp_info = pjsip_rdata_get_sdp_info(rdata);
offer = sdp_info->sdp;
status = sdp_info->sdp_err;
if (status==PJ_SUCCESS && sdp_info->sdp==NULL)
status = PJSIP_ERRNO_FROM_SIP_STATUS(PJSIP_SC_NOT_ACCEPTABLE);
if (status != PJ_SUCCESS) {
const pj_str_t reason = pj_str("Bad SDP");
pjsip_hdr hdr_list;
pjsip_warning_hdr *w;
pjsua_perror(THIS_FILE, "Bad SDP in incoming INVITE",
status);
w = pjsip_warning_hdr_create_from_status(rdata->tp_info.pool,
pjsip_endpt_name(pjsua_var.endpt),
status);
pj_list_init(&hdr_list);
pj_list_push_back(&hdr_list, w);
pjsip_endpt_respond(pjsua_var.endpt, NULL, rdata, 400,
&reason, &hdr_list, NULL, NULL);
goto on_return;
}
/* Do quick checks on SDP before passing it to transports. More elabore
* checks will be done in pjsip_inv_verify_request2() below.
*/
if (offer->media_count==0) {
const pj_str_t reason = pj_str("Missing media in SDP");
pjsip_endpt_respond(pjsua_var.endpt, NULL, rdata, 400, &reason,
NULL, NULL, NULL);
goto on_return;
}
} else {
offer = NULL;
}
/* Verify that we can handle the request. */
options |= PJSIP_INV_SUPPORT_100REL;
options |= PJSIP_INV_SUPPORT_TIMER;
if (pjsua_var.acc[acc_id].cfg.require_100rel)
options |= PJSIP_INV_REQUIRE_100REL;
if (pjsua_var.media_cfg.enable_ice)
options |= PJSIP_INV_SUPPORT_ICE;
if (pjsua_var.acc[acc_id].cfg.use_timer == PJSUA_SIP_TIMER_REQUIRED)
options |= PJSIP_INV_REQUIRE_TIMER;
else if (pjsua_var.acc[acc_id].cfg.use_timer == PJSUA_SIP_TIMER_ALWAYS)
options |= PJSIP_INV_ALWAYS_USE_TIMER;
status = pjsip_inv_verify_request2(rdata, &options, offer, NULL, NULL,
pjsua_var.endpt, &response);
if (status != PJ_SUCCESS) {
/*
* No we can't handle the incoming INVITE request.
*/
if (response) {
pjsip_response_addr res_addr;
pjsip_get_response_addr(response->pool, rdata, &res_addr);
pjsip_endpt_send_response(pjsua_var.endpt, &res_addr, response,
NULL, NULL);
} else {
/* Respond with 500 (Internal Server Error) */
pjsip_endpt_respond(pjsua_var.endpt, NULL, rdata, 500, NULL,
NULL, NULL, NULL);
}
goto on_return;
}
/* Get suitable Contact header */
if (pjsua_var.acc[acc_id].contact.slen) {
contact = pjsua_var.acc[acc_id].contact;
} else {
status = pjsua_acc_create_uas_contact(rdata->tp_info.pool, &contact,
acc_id, rdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to generate Contact header",
status);
pjsip_endpt_respond_stateless(pjsua_var.endpt, rdata, 500, NULL,
NULL, NULL);
goto on_return;
}
}
/* Create dialog: */
status = pjsip_dlg_create_uas( pjsip_ua_instance(), rdata,
&contact, &dlg);
if (status != PJ_SUCCESS) {
pjsip_endpt_respond_stateless(pjsua_var.endpt, rdata, 500, NULL,
NULL, NULL);
goto on_return;
}
/* Set credentials */
if (pjsua_var.acc[acc_id].cred_cnt) {
pjsip_auth_clt_set_credentials(&dlg->auth_sess,
pjsua_var.acc[acc_id].cred_cnt,
pjsua_var.acc[acc_id].cred);
}
/* Set preference */
pjsip_auth_clt_set_prefs(&dlg->auth_sess,
&pjsua_var.acc[acc_id].cfg.auth_pref);
/* Disable Session Timers if not prefered and the incoming INVITE request
* did not require it.
*/
if (pjsua_var.acc[acc_id].cfg.use_timer == PJSUA_SIP_TIMER_INACTIVE &&
(options & PJSIP_INV_REQUIRE_TIMER) == 0)
{
options &= ~(PJSIP_INV_SUPPORT_TIMER);
}
/* Create invite session: */
status = pjsip_inv_create_uas( dlg, rdata, NULL, options, &inv);
if (status != PJ_SUCCESS) {
pjsip_hdr hdr_list;
pjsip_warning_hdr *w;
w = pjsip_warning_hdr_create_from_status(dlg->pool,
pjsip_endpt_name(pjsua_var.endpt),
status);
pj_list_init(&hdr_list);
pj_list_push_back(&hdr_list, w);
pjsip_dlg_respond(dlg, rdata, 500, NULL, &hdr_list, NULL);
/* Can't terminate dialog because transaction is in progress.
pjsip_dlg_terminate(dlg);
*/
pjsua_media_channel_deinit(call->index);
goto on_return;
}
/* If account is locked to specific transport, then lock dialog
* to this transport too.
*/
if (pjsua_var.acc[acc_id].cfg.transport_id != PJSUA_INVALID_ID) {
pjsip_tpselector tp_sel;
pjsua_init_tpselector(pjsua_var.acc[acc_id].cfg.transport_id, &tp_sel);
pjsip_dlg_set_transport(dlg, &tp_sel);
}
/* Create and attach pjsua_var data to the dialog: */
call->inv = inv;
dlg->mod_data[pjsua_var.mod.id] = call;
inv->mod_data[pjsua_var.mod.id] = call;
/* Store variables required for the callback after the async
* media transport creation is completed.
*/
call->async_call.dlg = dlg;
pj_list_init(&call->async_call.call_var.inc_call.answers);
/* Init media channel */
status = pjsua_media_channel_init(call->index, PJSIP_ROLE_UAS,
call->secure_level,
rdata->tp_info.pool,
offer,
&sip_err_code, PJ_TRUE,
(pjsua_med_tp_state_cb)
&on_incoming_call_med_tp_complete);
if (status == PJ_SUCCESS) {
status = on_incoming_call_med_tp_complete(call_id, NULL);
if (status != PJ_SUCCESS) {
sip_err_code = PJSIP_SC_NOT_ACCEPTABLE;
pjsip_dlg_respond(dlg, rdata, sip_err_code, NULL, NULL, NULL);
goto on_return;
}
} else if (status != PJ_EPENDING) {
pjsua_perror(THIS_FILE, "Error initializing media channel", status);
pjsip_dlg_respond(dlg, rdata, sip_err_code, NULL, NULL, NULL);
goto on_return;
}
/* Create answer */
/*
status = pjsua_media_channel_create_sdp(call->index, rdata->tp_info.pool,
offer, &answer, &sip_err_code);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Error creating SDP answer", status);
pjsip_endpt_respond(pjsua_var.endpt, NULL, rdata,
sip_err_code, NULL, NULL, NULL, NULL);
goto on_return;
}
*/
/* Init Session Timers */
status = pjsip_timer_init_session(inv,
&pjsua_var.acc[acc_id].cfg.timer_setting);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Session Timer init failed", status);
status = pjsip_inv_end_session(inv, PJSIP_SC_INTERNAL_SERVER_ERROR,
NULL, &response);
if (status == PJ_SUCCESS && response)
status = pjsip_inv_send_msg(inv, response);
pjsua_media_channel_deinit(call->index);
goto on_return;
}
/* Update NAT type of remote endpoint, only when there is SDP in
* incoming INVITE!
*/
if (pjsua_var.ua_cfg.nat_type_in_sdp && inv->neg &&
pjmedia_sdp_neg_get_state(inv->neg) > PJMEDIA_SDP_NEG_STATE_LOCAL_OFFER)
{
const pjmedia_sdp_session *remote_sdp;
if (pjmedia_sdp_neg_get_neg_remote(inv->neg, &remote_sdp)==PJ_SUCCESS)
update_remote_nat_type(call, remote_sdp);
}
/* Must answer with some response to initial INVITE. We'll do this before
* attaching the call to the invite session/dialog, so that the application
* will not get notification about this event (on another scenario, it is
* also possible that inv_send_msg() fails and causes the invite session to
* be disconnected. If we have the call attached at this time, this will
* cause the disconnection callback to be called before on_incoming_call()
* callback is called, which is not right).
*/
status = pjsip_inv_initial_answer(inv, rdata,
100, NULL, NULL, &response);
if (status != PJ_SUCCESS) {
if (response == NULL) {
pjsua_perror(THIS_FILE, "Unable to send answer to incoming INVITE",
status);
pjsip_dlg_respond(dlg, rdata, 500, NULL, NULL, NULL);
pjsip_inv_terminate(inv, 500, PJ_FALSE);
} else {
pjsip_inv_send_msg(inv, response);
pjsip_inv_terminate(inv, response->msg->line.status.code,
PJ_FALSE);
}
pjsua_media_channel_deinit(call->index);
goto on_return;
} else {
status = pjsip_inv_send_msg(inv, response);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to send 100 response", status);
goto on_return;
}
}
++pjsua_var.call_cnt;
/* Check if this request should replace existing call */
if (replaced_dlg) {
pjsip_inv_session *replaced_inv;
struct pjsua_call *replaced_call;
pjsip_tx_data *tdata;
/* Get the invite session in the dialog */
replaced_inv = pjsip_dlg_get_inv_session(replaced_dlg);
/* Get the replaced call instance */
replaced_call = (pjsua_call*) replaced_dlg->mod_data[pjsua_var.mod.id];
/* Notify application */
if (pjsua_var.ua_cfg.cb.on_call_replaced)
pjsua_var.ua_cfg.cb.on_call_replaced(replaced_call->index,
call_id);
PJ_LOG(4,(THIS_FILE, "Answering replacement call %d with 200/OK",
call_id));
/* Answer the new call with 200 response */
status = pjsip_inv_answer(inv, 200, NULL, NULL, &tdata);
if (status == PJ_SUCCESS)
status = pjsip_inv_send_msg(inv, tdata);
if (status != PJ_SUCCESS)
pjsua_perror(THIS_FILE, "Error answering session", status);
/* Note that inv may be invalid if 200/OK has caused error in
* starting the media.
*/
PJ_LOG(4,(THIS_FILE, "Disconnecting replaced call %d",
replaced_call->index));
/* Disconnect replaced invite session */
status = pjsip_inv_end_session(replaced_inv, PJSIP_SC_GONE, NULL,
&tdata);
if (status == PJ_SUCCESS && tdata)
status = pjsip_inv_send_msg(replaced_inv, tdata);
if (status != PJ_SUCCESS)
pjsua_perror(THIS_FILE, "Error terminating session", status);
} else {
/* Notify application if on_incoming_call() is overriden,
* otherwise hangup the call with 480
*/
if (pjsua_var.ua_cfg.cb.on_incoming_call) {
pjsua_var.ua_cfg.cb.on_incoming_call(acc_id, call_id, rdata);
} else {
pjsua_call_hangup(call_id, PJSIP_SC_TEMPORARILY_UNAVAILABLE,
NULL, NULL);
}
}
/* This INVITE request has been handled. */
on_return:
pj_log_pop_indent();
PJSUA_UNLOCK();
return PJ_TRUE;
}
/*
* Check if the specified call has active INVITE session and the INVITE
* session has not been disconnected.
*/
PJ_DEF(pj_bool_t) pjsua_call_is_active(pjsua_call_id call_id)
{
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
return pjsua_var.calls[call_id].inv != NULL &&
pjsua_var.calls[call_id].inv->state != PJSIP_INV_STATE_DISCONNECTED;
}
/*
* Check if call has an active media session.
*/
PJ_DEF(pj_bool_t) pjsua_call_has_media(pjsua_call_id call_id)
{
pjsua_call *call = &pjsua_var.calls[call_id];
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
return call->audio_idx >= 0 && call->media[call->audio_idx].strm.a.stream;
}
/* Acquire lock to the specified call_id */
pj_status_t acquire_call(const char *title,
pjsua_call_id call_id,
pjsua_call **p_call,
pjsip_dialog **p_dlg)
{
unsigned retry;
pjsua_call *call = NULL;
pj_bool_t has_pjsua_lock = PJ_FALSE;
pj_status_t status = PJ_SUCCESS;
pj_time_val time_start, timeout;
pj_gettimeofday(&time_start);
timeout.msec = PJSUA_ACQUIRE_CALL_TIMEOUT;
pj_time_val_normalize(&timeout);
for (retry=0; ; ++retry) {
if (retry % 10 == 9) {
pj_time_val dtime;
pj_gettimeofday(&dtime);
PJ_TIME_VAL_SUB(dtime, time_start);
if (!PJ_TIME_VAL_LT(dtime, timeout))
break;
}
has_pjsua_lock = PJ_FALSE;
status = PJSUA_TRY_LOCK();
if (status != PJ_SUCCESS) {
pj_thread_sleep(retry/10);
continue;
}
has_pjsua_lock = PJ_TRUE;
call = &pjsua_var.calls[call_id];
if (call->inv == NULL) {
PJSUA_UNLOCK();
PJ_LOG(3,(THIS_FILE, "Invalid call_id %d in %s", call_id, title));
return PJSIP_ESESSIONTERMINATED;
}
status = pjsip_dlg_try_inc_lock(call->inv->dlg);
if (status != PJ_SUCCESS) {
PJSUA_UNLOCK();
pj_thread_sleep(retry/10);
continue;
}
PJSUA_UNLOCK();
break;
}
if (status != PJ_SUCCESS) {
if (has_pjsua_lock == PJ_FALSE)
PJ_LOG(1,(THIS_FILE, "Timed-out trying to acquire PJSUA mutex "
"(possibly system has deadlocked) in %s",
title));
else
PJ_LOG(1,(THIS_FILE, "Timed-out trying to acquire dialog mutex "
"(possibly system has deadlocked) in %s",
title));
return PJ_ETIMEDOUT;
}
*p_call = call;
*p_dlg = call->inv->dlg;
return PJ_SUCCESS;
}
/*
* Get the conference port identification associated with the call.
*/
PJ_DEF(pjsua_conf_port_id) pjsua_call_get_conf_port(pjsua_call_id call_id)
{
pjsua_call *call;
pjsua_conf_port_id port_id;
pjsip_dialog *dlg;
pj_status_t status;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
status = acquire_call("pjsua_call_get_conf_port()", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
return PJSUA_INVALID_ID;
port_id = call->media[call->audio_idx].strm.a.conf_slot;
pjsip_dlg_dec_lock(dlg);
return port_id;
}
/*
* Obtain detail information about the specified call.
*/
PJ_DEF(pj_status_t) pjsua_call_get_info( pjsua_call_id call_id,
pjsua_call_info *info)
{
pjsua_call *call;
pjsip_dialog *dlg;
unsigned mi;
pj_status_t status;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
pj_bzero(info, sizeof(*info));
status = acquire_call("pjsua_call_get_info()", call_id, &call, &dlg);
if (status != PJ_SUCCESS) {
return status;
}
/* id and role */
info->id = call_id;
info->role = call->inv->role;
info->acc_id = call->acc_id;
/* local info */
info->local_info.ptr = info->buf_.local_info;
pj_strncpy(&info->local_info, &call->inv->dlg->local.info_str,
sizeof(info->buf_.local_info));
/* local contact */
info->local_contact.ptr = info->buf_.local_contact;
info->local_contact.slen = pjsip_uri_print(PJSIP_URI_IN_CONTACT_HDR,
call->inv->dlg->local.contact->uri,
info->local_contact.ptr,
sizeof(info->buf_.local_contact));
/* remote info */
info->remote_info.ptr = info->buf_.remote_info;
pj_strncpy(&info->remote_info, &call->inv->dlg->remote.info_str,
sizeof(info->buf_.remote_info));
/* remote contact */
if (call->inv->dlg->remote.contact) {
int len;
info->remote_contact.ptr = info->buf_.remote_contact;
len = pjsip_uri_print(PJSIP_URI_IN_CONTACT_HDR,
call->inv->dlg->remote.contact->uri,
info->remote_contact.ptr,
sizeof(info->buf_.remote_contact));
if (len < 0) len = 0;
info->remote_contact.slen = len;
} else {
info->remote_contact.slen = 0;
}
/* call id */
info->call_id.ptr = info->buf_.call_id;
pj_strncpy(&info->call_id, &call->inv->dlg->call_id->id,
sizeof(info->buf_.call_id));
/* state, state_text */
info->state = call->inv->state;
info->state_text = pj_str((char*)pjsip_inv_state_name(info->state));
/* If call is disconnected, set the last_status from the cause code */
if (call->inv->state >= PJSIP_INV_STATE_DISCONNECTED) {
/* last_status, last_status_text */
info->last_status = call->inv->cause;
info->last_status_text.ptr = info->buf_.last_status_text;
pj_strncpy(&info->last_status_text, &call->inv->cause_text,
sizeof(info->buf_.last_status_text));
} else {
/* last_status, last_status_text */
info->last_status = call->last_code;
info->last_status_text.ptr = info->buf_.last_status_text;
pj_strncpy(&info->last_status_text, &call->last_text,
sizeof(info->buf_.last_status_text));
}
/* Build array of media status and dir */
info->media_cnt = 0;
for (mi=0; mi < call->med_cnt &&
info->media_cnt < PJ_ARRAY_SIZE(info->media); ++mi)
{
pjsua_call_media *call_med = &call->media[mi];
info->media[info->media_cnt].index = mi;
info->media[info->media_cnt].status = call_med->state;
info->media[info->media_cnt].dir = call_med->dir;
info->media[info->media_cnt].type = call_med->type;
if (call_med->type == PJMEDIA_TYPE_AUDIO) {
info->media[info->media_cnt].stream.aud.conf_slot =
call_med->strm.a.conf_slot;
} else if (call_med->type == PJMEDIA_TYPE_VIDEO) {
pjmedia_vid_dev_index cap_dev = PJMEDIA_VID_INVALID_DEV;
info->media[info->media_cnt].stream.vid.win_in =
call_med->strm.v.rdr_win_id;
if (call_med->strm.v.cap_win_id != PJSUA_INVALID_ID) {
cap_dev = call_med->strm.v.cap_dev;
}
info->media[info->media_cnt].stream.vid.cap_dev = cap_dev;
} else {
continue;
}
++info->media_cnt;
}
if (call->audio_idx != -1) {
info->media_status = call->media[call->audio_idx].state;
info->media_dir = call->media[call->audio_idx].dir;
info->conf_slot = call->media[call->audio_idx].strm.a.conf_slot;
}
/* calculate duration */
if (info->state >= PJSIP_INV_STATE_DISCONNECTED) {
info->total_duration = call->dis_time;
PJ_TIME_VAL_SUB(info->total_duration, call->start_time);
if (call->conn_time.sec) {
info->connect_duration = call->dis_time;
PJ_TIME_VAL_SUB(info->connect_duration, call->conn_time);
}
} else if (info->state == PJSIP_INV_STATE_CONFIRMED) {
pj_gettimeofday(&info->total_duration);
PJ_TIME_VAL_SUB(info->total_duration, call->start_time);
pj_gettimeofday(&info->connect_duration);
PJ_TIME_VAL_SUB(info->connect_duration, call->conn_time);
} else {
pj_gettimeofday(&info->total_duration);
PJ_TIME_VAL_SUB(info->total_duration, call->start_time);
}
pjsip_dlg_dec_lock(dlg);
return PJ_SUCCESS;
}
/*
* Check if call remote peer support the specified capability.
*/
PJ_DEF(pjsip_dialog_cap_status) pjsua_call_remote_has_cap(
pjsua_call_id call_id,
int htype,
const pj_str_t *hname,
const pj_str_t *token)
{
pjsua_call *call;
pjsip_dialog *dlg;
pj_status_t status;
pjsip_dialog_cap_status cap_status;
status = acquire_call("pjsua_call_peer_has_cap()", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
return PJSIP_DIALOG_CAP_UNKNOWN;
cap_status = pjsip_dlg_remote_has_cap(dlg, htype, hname, token);
pjsip_dlg_dec_lock(dlg);
return cap_status;
}
/*
* Attach application specific data to the call.
*/
PJ_DEF(pj_status_t) pjsua_call_set_user_data( pjsua_call_id call_id,
void *user_data)
{
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
pjsua_var.calls[call_id].user_data = user_data;
return PJ_SUCCESS;
}
/*
* Get user data attached to the call.
*/
PJ_DEF(void*) pjsua_call_get_user_data(pjsua_call_id call_id)
{
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
NULL);
return pjsua_var.calls[call_id].user_data;
}
/*
* Get remote's NAT type.
*/
PJ_DEF(pj_status_t) pjsua_call_get_rem_nat_type(pjsua_call_id call_id,
pj_stun_nat_type *p_type)
{
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
PJ_ASSERT_RETURN(p_type != NULL, PJ_EINVAL);
*p_type = pjsua_var.calls[call_id].rem_nat_type;
return PJ_SUCCESS;
}
/*
* Get media stream info for the specified media index.
*/
PJ_DEF(pj_status_t) pjsua_call_get_stream_info( pjsua_call_id call_id,
unsigned med_idx,
pjsua_stream_info *psi)
{
pjsua_call *call;
pjsua_call_media *call_med;
pj_status_t status;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
PJ_ASSERT_RETURN(psi, PJ_EINVAL);
PJSUA_LOCK();
call = &pjsua_var.calls[call_id];
if (med_idx >= call->med_cnt) {
PJSUA_UNLOCK();
return PJ_EINVAL;
}
call_med = &call->media[med_idx];
psi->type = call_med->type;
switch (call_med->type) {
case PJMEDIA_TYPE_AUDIO:
status = pjmedia_stream_get_info(call_med->strm.a.stream,
&psi->info.aud);
break;
#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
case PJMEDIA_TYPE_VIDEO:
status = pjmedia_vid_stream_get_info(call_med->strm.v.stream,
&psi->info.vid);
break;
#endif
default:
status = PJMEDIA_EINVALIMEDIATYPE;
break;
}
PJSUA_UNLOCK();
return status;
}
/*
* Get media stream statistic for the specified media index.
*/
PJ_DEF(pj_status_t) pjsua_call_get_stream_stat( pjsua_call_id call_id,
unsigned med_idx,
pjsua_stream_stat *stat)
{
pjsua_call *call;
pjsua_call_media *call_med;
pj_status_t status;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
PJ_ASSERT_RETURN(stat, PJ_EINVAL);
PJSUA_LOCK();
call = &pjsua_var.calls[call_id];
if (med_idx >= call->med_cnt) {
PJSUA_UNLOCK();
return PJ_EINVAL;
}
call_med = &call->media[med_idx];
switch (call_med->type) {
case PJMEDIA_TYPE_AUDIO:
status = pjmedia_stream_get_stat(call_med->strm.a.stream,
&stat->rtcp);
if (status == PJ_SUCCESS)
status = pjmedia_stream_get_stat_jbuf(call_med->strm.a.stream,
&stat->jbuf);
break;
#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
case PJMEDIA_TYPE_VIDEO:
status = pjmedia_vid_stream_get_stat(call_med->strm.v.stream,
&stat->rtcp);
if (status == PJ_SUCCESS)
status = pjmedia_vid_stream_get_stat_jbuf(call_med->strm.v.stream,
&stat->jbuf);
break;
#endif
default:
status = PJMEDIA_EINVALIMEDIATYPE;
break;
}
PJSUA_UNLOCK();
return status;
}
/*
* Get media transport info for the specified media index.
*/
PJ_DEF(pj_status_t)
pjsua_call_get_med_transport_info(pjsua_call_id call_id,
unsigned med_idx,
pjmedia_transport_info *t)
{
pjsua_call *call;
pjsua_call_media *call_med;
pj_status_t status;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
PJ_ASSERT_RETURN(t, PJ_EINVAL);
PJSUA_LOCK();
call = &pjsua_var.calls[call_id];
if (med_idx >= call->med_cnt) {
PJSUA_UNLOCK();
return PJ_EINVAL;
}
call_med = &call->media[med_idx];
pjmedia_transport_info_init(t);
status = pjmedia_transport_get_info(call_med->tp, t);
PJSUA_UNLOCK();
return status;
}
/*
* Send response to incoming INVITE request.
*/
PJ_DEF(pj_status_t) pjsua_call_answer( pjsua_call_id call_id,
unsigned code,
const pj_str_t *reason,
const pjsua_msg_data *msg_data)
{
pjsua_call *call;
pjsip_dialog *dlg = NULL;
pjsip_tx_data *tdata;
pj_status_t status;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
PJ_LOG(4,(THIS_FILE, "Answering call %d: code=%d", call_id, code));
pj_log_push_indent();
status = acquire_call("pjsua_call_answer()", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
goto on_return;
PJSUA_LOCK();
/* If media transport creation is not yet completed, we will answer
* the call in the media transport creation callback instead.
*/
if (call->med_ch_cb) {
struct call_answer *answer;
PJ_LOG(4,(THIS_FILE, "Pending answering call %d upon completion "
"of media transport", call_id));
answer = PJ_POOL_ZALLOC_T(call->inv->pool_prov, struct call_answer);
answer->code = code;
if (reason) {
pj_strdup(call->inv->pool_prov, answer->reason, reason);
}
if (msg_data) {
answer->msg_data = pjsua_msg_data_clone(call->inv->pool_prov,
msg_data);
}
pj_list_push_back(&call->async_call.call_var.inc_call.answers,
answer);
PJSUA_UNLOCK();
if (dlg) pjsip_dlg_dec_lock(dlg);
pj_log_pop_indent();
return status;
}
PJSUA_UNLOCK();
if (call->res_time.sec == 0)
pj_gettimeofday(&call->res_time);
if (reason && reason->slen == 0)
reason = NULL;
/* Create response message */
status = pjsip_inv_answer(call->inv, code, reason, NULL, &tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Error creating response",
status);
goto on_return;
}
/* Call might have been disconnected if application is answering with
* 200/OK and the media failed to start.
*/
if (call->inv == NULL)
goto on_return;
/* Add additional headers etc */
pjsua_process_msg_data( tdata, msg_data);
/* Send the message */
status = pjsip_inv_send_msg(call->inv, tdata);
if (status != PJ_SUCCESS)
pjsua_perror(THIS_FILE, "Error sending response",
status);
on_return:
if (dlg) pjsip_dlg_dec_lock(dlg);
pj_log_pop_indent();
return status;
}
/*
* Hangup call by using method that is appropriate according to the
* call state.
*/
PJ_DEF(pj_status_t) pjsua_call_hangup(pjsua_call_id call_id,
unsigned code,
const pj_str_t *reason,
const pjsua_msg_data *msg_data)
{
pjsua_call *call;
pjsip_dialog *dlg = NULL;
pj_status_t status;
pjsip_tx_data *tdata;
if (call_id<0 || call_id>=(int)pjsua_var.ua_cfg.max_calls) {
PJ_LOG(1,(THIS_FILE, "pjsua_call_hangup(): invalid call id %d",
call_id));
}
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
PJ_LOG(4,(THIS_FILE, "Call %d hanging up: code=%d..", call_id, code));
pj_log_push_indent();
status = acquire_call("pjsua_call_hangup()", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
goto on_return;
if (code==0) {
if (call->inv->state == PJSIP_INV_STATE_CONFIRMED)
code = PJSIP_SC_OK;
else if (call->inv->role == PJSIP_ROLE_UAS)
code = PJSIP_SC_DECLINE;
else
code = PJSIP_SC_REQUEST_TERMINATED;
}
status = pjsip_inv_end_session(call->inv, code, reason, &tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE,
"Failed to create end session message",
status);
goto on_return;
}
/* pjsip_inv_end_session may return PJ_SUCCESS with NULL
* as p_tdata when INVITE transaction has not been answered
* with any provisional responses.
*/
if (tdata == NULL)
goto on_return;
/* Add additional headers etc */
pjsua_process_msg_data( tdata, msg_data);
/* Send the message */
status = pjsip_inv_send_msg(call->inv, tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE,
"Failed to send end session message",
status);
goto on_return;
}
/* Stop lock codec timer, if it is active */
if (call->lock_codec.reinv_timer.id) {
pjsip_endpt_cancel_timer(pjsua_var.endpt,
&call->lock_codec.reinv_timer);
call->lock_codec.reinv_timer.id = PJ_FALSE;
}
on_return:
if (dlg) pjsip_dlg_dec_lock(dlg);
pj_log_pop_indent();
return status;
}
/*
* Accept or reject redirection.
*/
PJ_DEF(pj_status_t) pjsua_call_process_redirect( pjsua_call_id call_id,
pjsip_redirect_op cmd)
{
pjsua_call *call;
pjsip_dialog *dlg;
pj_status_t status;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
status = acquire_call("pjsua_call_process_redirect()", call_id,
&call, &dlg);
if (status != PJ_SUCCESS)
return status;
status = pjsip_inv_process_redirect(call->inv, cmd, NULL);
pjsip_dlg_dec_lock(dlg);
return status;
}
/*
* Put the specified call on hold.
*/
PJ_DEF(pj_status_t) pjsua_call_set_hold(pjsua_call_id call_id,
const pjsua_msg_data *msg_data)
{
pjmedia_sdp_session *sdp;
pjsua_call *call;
pjsip_dialog *dlg = NULL;
pjsip_tx_data *tdata;
pj_status_t status;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
PJ_LOG(4,(THIS_FILE, "Putting call %d on hold", call_id));
pj_log_push_indent();
status = acquire_call("pjsua_call_set_hold()", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
goto on_return;
if (call->inv->state != PJSIP_INV_STATE_CONFIRMED) {
PJ_LOG(3,(THIS_FILE, "Can not hold call that is not confirmed"));
status = PJSIP_ESESSIONSTATE;
goto on_return;
}
status = create_sdp_of_call_hold(call, &sdp);
if (status != PJ_SUCCESS)
goto on_return;
/* Create re-INVITE with new offer */
status = pjsip_inv_reinvite( call->inv, NULL, sdp, &tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create re-INVITE", status);
goto on_return;
}
/* Add additional headers etc */
pjsua_process_msg_data( tdata, msg_data);
/* Send the request */
status = pjsip_inv_send_msg( call->inv, tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to send re-INVITE", status);
goto on_return;
}
/* Set flag that local put the call on hold */
call->local_hold = PJ_TRUE;
on_return:
if (dlg) pjsip_dlg_dec_lock(dlg);
pj_log_pop_indent();
return status;
}
/*
* Send re-INVITE (to release hold).
*/
PJ_DEF(pj_status_t) pjsua_call_reinvite( pjsua_call_id call_id,
unsigned options,
const pjsua_msg_data *msg_data)
{
pjmedia_sdp_session *sdp;
pj_str_t *new_contact = NULL;
pjsip_tx_data *tdata;
pjsua_call *call;
pjsip_dialog *dlg = NULL;
pj_status_t status;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
PJ_LOG(4,(THIS_FILE, "Sending re-INVITE on call %d", call_id));
pj_log_push_indent();
status = acquire_call("pjsua_call_reinvite()", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
goto on_return;
if (call->inv->state != PJSIP_INV_STATE_CONFIRMED) {
PJ_LOG(3,(THIS_FILE, "Can not re-INVITE call that is not confirmed"));
status = PJSIP_ESESSIONSTATE;
goto on_return;
}
/* Create SDP */
if (call->local_hold && (options & PJSUA_CALL_UNHOLD)==0) {
status = create_sdp_of_call_hold(call, &sdp);
} else {
status = pjsua_media_channel_create_sdp(call->index,
call->inv->pool_prov,
NULL, &sdp, NULL);
call->local_hold = PJ_FALSE;
}
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to get SDP from media endpoint",
status);
goto on_return;
}
if ((options & PJSUA_CALL_UPDATE_CONTACT) &
pjsua_acc_is_valid(call->acc_id))
{
new_contact = &pjsua_var.acc[call->acc_id].contact;
}
/* Create re-INVITE with new offer */
status = pjsip_inv_reinvite( call->inv, new_contact, sdp, &tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create re-INVITE", status);
goto on_return;
}
/* Add additional headers etc */
pjsua_process_msg_data( tdata, msg_data);
/* Send the request */
status = pjsip_inv_send_msg( call->inv, tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to send re-INVITE", status);
goto on_return;
}
on_return:
if (dlg) pjsip_dlg_dec_lock(dlg);
pj_log_pop_indent();
return status;
}
/*
* Send UPDATE request.
*/
PJ_DEF(pj_status_t) pjsua_call_update( pjsua_call_id call_id,
unsigned options,
const pjsua_msg_data *msg_data)
{
pjmedia_sdp_session *sdp;
pj_str_t *new_contact = NULL;
pjsip_tx_data *tdata;
pjsua_call *call;
pjsip_dialog *dlg = NULL;
pj_status_t status;
PJ_UNUSED_ARG(options);
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
PJ_LOG(4,(THIS_FILE, "Sending UPDATE on call %d", call_id));
pj_log_push_indent();
status = acquire_call("pjsua_call_update()", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
goto on_return;
/* Create SDP */
status = pjsua_media_channel_create_sdp(call->index,
call->inv->pool_prov,
NULL, &sdp, NULL);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to get SDP from media endpoint",
status);
goto on_return;
}
if ((options & PJSUA_CALL_UPDATE_CONTACT) &
pjsua_acc_is_valid(call->acc_id))
{
new_contact = &pjsua_var.acc[call->acc_id].contact;
}
/* Create UPDATE with new offer */
status = pjsip_inv_update(call->inv, new_contact, sdp, &tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create UPDATE request", status);
goto on_return;
}
/* Add additional headers etc */
pjsua_process_msg_data( tdata, msg_data);
/* Send the request */
status = pjsip_inv_send_msg( call->inv, tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to send UPDATE request", status);
goto on_return;
}
call->local_hold = PJ_FALSE;
on_return:
if (dlg) pjsip_dlg_dec_lock(dlg);
pj_log_pop_indent();
return status;
}
/*
* Initiate call transfer to the specified address.
*/
PJ_DEF(pj_status_t) pjsua_call_xfer( pjsua_call_id call_id,
const pj_str_t *dest,
const pjsua_msg_data *msg_data)
{
pjsip_evsub *sub;
pjsip_tx_data *tdata;
pjsua_call *call;
pjsip_dialog *dlg = NULL;
pjsip_generic_string_hdr *gs_hdr;
const pj_str_t str_ref_by = { "Referred-By", 11 };
struct pjsip_evsub_user xfer_cb;
pj_status_t status;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls &&
dest, PJ_EINVAL);
PJ_LOG(4,(THIS_FILE, "Transfering call %d to %.*s", call_id,
(int)dest->slen, dest->ptr));
pj_log_push_indent();
status = acquire_call("pjsua_call_xfer()", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
goto on_return;
/* Create xfer client subscription. */
pj_bzero(&xfer_cb, sizeof(xfer_cb));
xfer_cb.on_evsub_state = &xfer_client_on_evsub_state;
status = pjsip_xfer_create_uac(call->inv->dlg, &xfer_cb, &sub);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create xfer", status);
goto on_return;
}
/* Associate this call with the client subscription */
pjsip_evsub_set_mod_data(sub, pjsua_var.mod.id, call);
/*
* Create REFER request.
*/
status = pjsip_xfer_initiate(sub, dest, &tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create REFER request", status);
goto on_return;
}
/* Add Referred-By header */
gs_hdr = pjsip_generic_string_hdr_create(tdata->pool, &str_ref_by,
&dlg->local.info_str);
pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)gs_hdr);
/* Add additional headers etc */
pjsua_process_msg_data( tdata, msg_data);
/* Send. */
status = pjsip_xfer_send_request(sub, tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to send REFER request", status);
goto on_return;
}
/* For simplicity (that's what this program is intended to be!),
* leave the original invite session as it is. More advanced application
* may want to hold the INVITE, or terminate the invite, or whatever.
*/
on_return:
if (dlg) pjsip_dlg_dec_lock(dlg);
pj_log_pop_indent();
return status;
}
/*
* Initiate attended call transfer to the specified address.
*/
PJ_DEF(pj_status_t) pjsua_call_xfer_replaces( pjsua_call_id call_id,
pjsua_call_id dest_call_id,
unsigned options,
const pjsua_msg_data *msg_data)
{
pjsua_call *dest_call;
pjsip_dialog *dest_dlg;
char str_dest_buf[PJSIP_MAX_URL_SIZE*2];
pj_str_t str_dest;
int len;
pjsip_uri *uri;
pj_status_t status;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
PJ_ASSERT_RETURN(dest_call_id>=0 &&
dest_call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
PJ_LOG(4,(THIS_FILE, "Transfering call %d replacing with call %d",
call_id, dest_call_id));
pj_log_push_indent();
status = acquire_call("pjsua_call_xfer_replaces()", dest_call_id,
&dest_call, &dest_dlg);
if (status != PJ_SUCCESS) {
pj_log_pop_indent();
return status;
}
/*
* Create REFER destination URI with Replaces field.
*/
/* Make sure we have sufficient buffer's length */
PJ_ASSERT_ON_FAIL(dest_dlg->remote.info_str.slen +
dest_dlg->call_id->id.slen +
dest_dlg->remote.info->tag.slen +
dest_dlg->local.info->tag.slen + 32
< (long)sizeof(str_dest_buf),
{ status=PJSIP_EURITOOLONG; goto on_error; });
/* Print URI */
str_dest_buf[0] = '<';
str_dest.slen = 1;
uri = (pjsip_uri*) pjsip_uri_get_uri(dest_dlg->remote.info->uri);
len = pjsip_uri_print(PJSIP_URI_IN_REQ_URI, uri,
str_dest_buf+1, sizeof(str_dest_buf)-1);
if (len < 0) {
status = PJSIP_EURITOOLONG;
goto on_error;
}
str_dest.slen += len;
/* Build the URI */
len = pj_ansi_snprintf(str_dest_buf + str_dest.slen,
sizeof(str_dest_buf) - str_dest.slen,
"?%s"
"Replaces=%.*s"
"%%3Bto-tag%%3D%.*s"
"%%3Bfrom-tag%%3D%.*s>",
((options&PJSUA_XFER_NO_REQUIRE_REPLACES) ?
"" : "Require=replaces&"),
(int)dest_dlg->call_id->id.slen,
dest_dlg->call_id->id.ptr,
(int)dest_dlg->remote.info->tag.slen,
dest_dlg->remote.info->tag.ptr,
(int)dest_dlg->local.info->tag.slen,
dest_dlg->local.info->tag.ptr);
PJ_ASSERT_ON_FAIL(len > 0 && len <= (int)sizeof(str_dest_buf)-str_dest.slen,
{ status=PJSIP_EURITOOLONG; goto on_error; });
str_dest.ptr = str_dest_buf;
str_dest.slen += len;
pjsip_dlg_dec_lock(dest_dlg);
status = pjsua_call_xfer(call_id, &str_dest, msg_data);
pj_log_pop_indent();
return status;
on_error:
if (dest_dlg) pjsip_dlg_dec_lock(dest_dlg);
pj_log_pop_indent();
return status;
}
/*
* Send DTMF digits to remote using RFC 2833 payload formats.
*/
PJ_DEF(pj_status_t) pjsua_call_dial_dtmf( pjsua_call_id call_id,
const pj_str_t *digits)
{
pjsua_call *call;
pjsip_dialog *dlg = NULL;
pj_status_t status;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
PJ_LOG(4,(THIS_FILE, "Call %d dialing DTMF %.*s",
call_id, (int)digits->slen, digits->ptr));
pj_log_push_indent();
status = acquire_call("pjsua_call_dial_dtmf()", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
goto on_return;
if (!pjsua_call_has_media(call_id)) {
PJ_LOG(3,(THIS_FILE, "Media is not established yet!"));
status = PJ_EINVALIDOP;
goto on_return;
}
status = pjmedia_stream_dial_dtmf(
call->media[call->audio_idx].strm.a.stream, digits);
on_return:
if (dlg) pjsip_dlg_dec_lock(dlg);
pj_log_pop_indent();
return status;
}
/**
* Send instant messaging inside INVITE session.
*/
PJ_DEF(pj_status_t) pjsua_call_send_im( pjsua_call_id call_id,
const pj_str_t *mime_type,
const pj_str_t *content,
const pjsua_msg_data *msg_data,
void *user_data)
{
pjsua_call *call;
pjsip_dialog *dlg = NULL;
const pj_str_t mime_text_plain = pj_str("text/plain");
pjsip_media_type ctype;
pjsua_im_data *im_data;
pjsip_tx_data *tdata;
pj_status_t status;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
PJ_LOG(4,(THIS_FILE, "Call %d sending %d bytes MESSAGE..",
call_id, (int)content->slen));
pj_log_push_indent();
status = acquire_call("pjsua_call_send_im()", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
goto on_return;
/* Set default media type if none is specified */
if (mime_type == NULL) {
mime_type = &mime_text_plain;
}
/* Create request message. */
status = pjsip_dlg_create_request( call->inv->dlg, &pjsip_message_method,
-1, &tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create MESSAGE request", status);
goto on_return;
}
/* Add accept header. */
pjsip_msg_add_hdr( tdata->msg,
(pjsip_hdr*)pjsua_im_create_accept(tdata->pool));
/* Parse MIME type */
pjsua_parse_media_type(tdata->pool, mime_type, &ctype);
/* Create "text/plain" message body. */
tdata->msg->body = pjsip_msg_body_create( tdata->pool, &ctype.type,
&ctype.subtype, content);
if (tdata->msg->body == NULL) {
pjsua_perror(THIS_FILE, "Unable to create msg body", PJ_ENOMEM);
pjsip_tx_data_dec_ref(tdata);
goto on_return;
}
/* Add additional headers etc */
pjsua_process_msg_data( tdata, msg_data);
/* Create IM data and attach to the request. */
im_data = PJ_POOL_ZALLOC_T(tdata->pool, pjsua_im_data);
im_data->acc_id = call->acc_id;
im_data->call_id = call_id;
im_data->to = call->inv->dlg->remote.info_str;
pj_strdup_with_null(tdata->pool, &im_data->body, content);
im_data->user_data = user_data;
/* Send the request. */
status = pjsip_dlg_send_request( call->inv->dlg, tdata,
pjsua_var.mod.id, im_data);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to send MESSAGE request", status);
goto on_return;
}
on_return:
if (dlg) pjsip_dlg_dec_lock(dlg);
pj_log_pop_indent();
return status;
}
/*
* Send IM typing indication inside INVITE session.
*/
PJ_DEF(pj_status_t) pjsua_call_send_typing_ind( pjsua_call_id call_id,
pj_bool_t is_typing,
const pjsua_msg_data*msg_data)
{
pjsua_call *call;
pjsip_dialog *dlg = NULL;
pjsip_tx_data *tdata;
pj_status_t status;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
PJ_LOG(4,(THIS_FILE, "Call %d sending typing indication..",
call_id));
pj_log_push_indent();
status = acquire_call("pjsua_call_send_typing_ind", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
goto on_return;
/* Create request message. */
status = pjsip_dlg_create_request( call->inv->dlg, &pjsip_message_method,
-1, &tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create MESSAGE request", status);
goto on_return;
}
/* Create "application/im-iscomposing+xml" msg body. */
tdata->msg->body = pjsip_iscomposing_create_body(tdata->pool, is_typing,
NULL, NULL, -1);
/* Add additional headers etc */
pjsua_process_msg_data( tdata, msg_data);
/* Send the request. */
status = pjsip_dlg_send_request( call->inv->dlg, tdata, -1, NULL);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to send MESSAGE request", status);
goto on_return;
}
on_return:
if (dlg) pjsip_dlg_dec_lock(dlg);
pj_log_pop_indent();
return status;
}
/*
* Send arbitrary request.
*/
PJ_DEF(pj_status_t) pjsua_call_send_request(pjsua_call_id call_id,
const pj_str_t *method_str,
const pjsua_msg_data *msg_data)
{
pjsua_call *call;
pjsip_dialog *dlg = NULL;
pjsip_method method;
pjsip_tx_data *tdata;
pj_status_t status;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
PJ_LOG(4,(THIS_FILE, "Call %d sending %.*s request..",
call_id, (int)method_str->slen, method_str->ptr));
pj_log_push_indent();
status = acquire_call("pjsua_call_send_request", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
goto on_return;
/* Init method */
pjsip_method_init_np(&method, (pj_str_t*)method_str);
/* Create request message. */
status = pjsip_dlg_create_request( call->inv->dlg, &method, -1, &tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create request", status);
goto on_return;
}
/* Add additional headers etc */
pjsua_process_msg_data( tdata, msg_data);
/* Send the request. */
status = pjsip_dlg_send_request( call->inv->dlg, tdata, -1, NULL);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to send request", status);
goto on_return;
}
on_return:
if (dlg) pjsip_dlg_dec_lock(dlg);
pj_log_pop_indent();
return status;
}
/*
* Terminate all calls.
*/
PJ_DEF(void) pjsua_call_hangup_all(void)
{
unsigned i;
PJ_LOG(4,(THIS_FILE, "Hangup all calls.."));
pj_log_push_indent();
PJSUA_LOCK();
for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) {
if (pjsua_var.calls[i].inv)
pjsua_call_hangup(i, 0, NULL, NULL);
}
PJSUA_UNLOCK();
pj_log_pop_indent();
}
/* Proto */
static pj_status_t perform_lock_codec(pjsua_call *call);
/* Timer callback to send re-INVITE or UPDATE to lock codec */
static void reinv_timer_cb(pj_timer_heap_t *th,
pj_timer_entry *entry)
{
pjsua_call_id call_id = (pjsua_call_id)(pj_size_t)entry->user_data;
pjsip_dialog *dlg;
pjsua_call *call;
pj_status_t status;
PJ_UNUSED_ARG(th);
pjsua_var.calls[call_id].lock_codec.reinv_timer.id = PJ_FALSE;
status = acquire_call("reinv_timer_cb()", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
return;
status = perform_lock_codec(call);
pjsip_dlg_dec_lock(dlg);
}
/* Check if the specified format can be skipped in counting codecs */
static pj_bool_t is_non_av_fmt(const pjmedia_sdp_media *m,
const pj_str_t *fmt)
{
const pj_str_t STR_TEL = {"telephone-event", 15};
unsigned pt;
pt = pj_strtoul(fmt);
/* Check for comfort noise */
if (pt == PJMEDIA_RTP_PT_CN)
return PJ_TRUE;
/* Dynamic PT, check the format name */
if (pt >= 96) {
pjmedia_sdp_attr *a;
pjmedia_sdp_rtpmap rtpmap;
/* Get the format name */
a = pjmedia_sdp_attr_find2(m->attr_count, m->attr, "rtpmap", fmt);
if (a && pjmedia_sdp_attr_get_rtpmap(a, &rtpmap)==PJ_SUCCESS) {
/* Check for telephone-event */
if (pj_stricmp(&rtpmap.enc_name, &STR_TEL)==0)
return PJ_TRUE;
} else {
/* Invalid SDP, should not reach here */
pj_assert(!"SDP should have been validated!");
return PJ_TRUE;
}
}
return PJ_FALSE;
}
/* Send re-INVITE or UPDATE with new SDP offer to select only one codec
* out of several codecs presented by callee in his answer.
*/
static pj_status_t perform_lock_codec(pjsua_call *call)
{
const pj_str_t STR_UPDATE = {"UPDATE", 6};
const pjmedia_sdp_session *local_sdp = NULL, *new_sdp;
unsigned i;
pj_bool_t rem_can_update;
pj_bool_t need_lock_codec = PJ_FALSE;
pjsip_tx_data *tdata;
pj_status_t status;
PJ_ASSERT_RETURN(call->lock_codec.reinv_timer.id==PJ_FALSE,
PJ_EINVALIDOP);
/* Verify if another SDP negotiation is in progress, e.g: session timer
* or another re-INVITE.
*/
if (call->inv==NULL || call->inv->neg==NULL ||
pjmedia_sdp_neg_get_state(call->inv->neg)!=PJMEDIA_SDP_NEG_STATE_DONE)
{
return PJMEDIA_SDPNEG_EINSTATE;
}
/* Don't do this if call is disconnecting! */
if (call->inv->state > PJSIP_INV_STATE_CONFIRMED ||
call->inv->cause >= 200)
{
return PJ_EINVALIDOP;
}
/* Verify if another SDP negotiation has been completed by comparing
* the SDP version.
*/
status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &local_sdp);
if (status != PJ_SUCCESS)
return status;
if (local_sdp->origin.version > call->lock_codec.sdp_ver)
return PJMEDIA_SDP_EINVER;
PJ_LOG(3, (THIS_FILE, "Updating media session to use only one codec.."));
/* Update the new offer so it contains only a codec. Note that formats
* order in the offer should have been matched to the answer, so we can
* just directly update the offer without looking-up the answer.
*/
new_sdp = pjmedia_sdp_session_clone(call->inv->pool_prov, local_sdp);
for (i = 0; i < call->med_cnt; ++i) {
unsigned j = 0, codec_cnt = 0;
const pjmedia_sdp_media *ref_m;
pjmedia_sdp_media *m;
pjsua_call_media *call_med = &call->media[i];
/* Verify if media is deactivated */
if (call_med->state == PJSUA_CALL_MEDIA_NONE ||
call_med->state == PJSUA_CALL_MEDIA_ERROR ||
call_med->dir == PJMEDIA_DIR_NONE)
{
continue;
}
ref_m = local_sdp->media[i];
m = new_sdp->media[i];
/* Verify that media must be active. */
pj_assert(ref_m->desc.port);
while (j < m->desc.fmt_count) {
pjmedia_sdp_attr *a;
pj_str_t *fmt = &m->desc.fmt[j];
if (is_non_av_fmt(m, fmt) || (++codec_cnt == 1)) {
++j;
continue;
}
/* Remove format */
a = pjmedia_sdp_attr_find2(m->attr_count, m->attr, "rtpmap", fmt);
if (a) pjmedia_sdp_attr_remove(&m->attr_count, m->attr, a);
a = pjmedia_sdp_attr_find2(m->attr_count, m->attr, "fmtp", fmt);
if (a) pjmedia_sdp_attr_remove(&m->attr_count, m->attr, a);
pj_array_erase(m->desc.fmt, sizeof(m->desc.fmt[0]),
m->desc.fmt_count, j);
--m->desc.fmt_count;
}
need_lock_codec |= (ref_m->desc.fmt_count > m->desc.fmt_count);
}
/* Last check if SDP trully needs to be updated. It is possible that OA
* negotiations have completed and SDP has changed but we didn't
* increase the SDP version (should not happen!).
*/
if (!need_lock_codec)
return PJ_SUCCESS;
/* Send UPDATE or re-INVITE */
rem_can_update = pjsip_dlg_remote_has_cap(call->inv->dlg,
PJSIP_H_ALLOW,
NULL, &STR_UPDATE) ==
PJSIP_DIALOG_CAP_SUPPORTED;
if (rem_can_update) {
status = pjsip_inv_update(call->inv, NULL, new_sdp, &tdata);
} else {
status = pjsip_inv_reinvite(call->inv, NULL, new_sdp, &tdata);
}
if (status==PJ_EINVALIDOP &&
++call->lock_codec.retry_cnt <= LOCK_CODEC_MAX_RETRY)
{
/* Ups, let's reschedule again */
pj_time_val delay = {0, LOCK_CODEC_RETRY_INTERVAL};
pj_time_val_normalize(&delay);
call->lock_codec.reinv_timer.id = PJ_TRUE;
pjsip_endpt_schedule_timer(pjsua_var.endpt,
&call->lock_codec.reinv_timer, &delay);
return status;
} else if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Error creating UPDATE/re-INVITE to lock codec",
status);
return status;
}
/* Send the UPDATE/re-INVITE request */
status = pjsip_inv_send_msg(call->inv, tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Error sending UPDATE/re-INVITE in lock codec",
status);
return status;
}
return status;
}
/* Check if remote answerer has given us more than one codecs. If so,
* create another offer with one codec only to lock down the codec.
*/
static pj_status_t lock_codec(pjsua_call *call)
{
pjsip_inv_session *inv = call->inv;
const pjmedia_sdp_session *local_sdp, *remote_sdp;
pj_time_val delay = {0, 0};
const pj_str_t st_update = {"UPDATE", 6};
unsigned i;
pj_bool_t has_mult_fmt = PJ_FALSE;
pj_status_t status;
/* Stop lock codec timer, if it is active */
if (call->lock_codec.reinv_timer.id) {
pjsip_endpt_cancel_timer(pjsua_var.endpt,
&call->lock_codec.reinv_timer);
call->lock_codec.reinv_timer.id = PJ_FALSE;
}
/* Skip this if we are the answerer */
if (!inv->neg || !pjmedia_sdp_neg_was_answer_remote(inv->neg)) {
return PJ_SUCCESS;
}
/* Delay this when the SDP negotiation done in call state EARLY and
* remote does not support UPDATE method.
*/
if (inv->state == PJSIP_INV_STATE_EARLY &&
pjsip_dlg_remote_has_cap(inv->dlg, PJSIP_H_ALLOW, NULL, &st_update)!=
PJSIP_DIALOG_CAP_SUPPORTED)
{
call->lock_codec.pending = PJ_TRUE;
return PJ_SUCCESS;
}
status = pjmedia_sdp_neg_get_active_local(inv->neg, &local_sdp);
if (status != PJ_SUCCESS)
return status;
status = pjmedia_sdp_neg_get_active_remote(inv->neg, &remote_sdp);
if (status != PJ_SUCCESS)
return status;
/* Find multiple codecs answer in all media */
for (i = 0; i < call->med_cnt; ++i) {
pjsua_call_media *call_med = &call->media[i];
const pjmedia_sdp_media *rem_m, *loc_m;
unsigned codec_cnt = 0;
/* Skip this if the media is inactive or error */
if (call_med->state == PJSUA_CALL_MEDIA_NONE ||
call_med->state == PJSUA_CALL_MEDIA_ERROR ||
call_med->dir == PJMEDIA_DIR_NONE)
{
continue;
}
/* Remote may answer with less media lines. */
if (i >= remote_sdp->media_count)
continue;
rem_m = remote_sdp->media[i];
loc_m = local_sdp->media[i];
/* Verify that media must be active. */
pj_assert(loc_m->desc.port && rem_m->desc.port);
/* Count the formats in the answer. */
if (rem_m->desc.fmt_count==1) {
codec_cnt = 1;
} else {
unsigned j;
for (j=0; j<rem_m->desc.fmt_count && codec_cnt <= 1; ++j) {
if (!is_non_av_fmt(rem_m, &rem_m->desc.fmt[j]))
++codec_cnt;
}
}
if (codec_cnt > 1) {
has_mult_fmt = PJ_TRUE;
break;
}
}
/* Each media in the answer already contains single codec. */
if (!has_mult_fmt) {
call->lock_codec.retry_cnt = 0;
return PJ_SUCCESS;
}
/* Remote keeps answering with multiple codecs, let's just give up
* locking codec to avoid infinite retry loop.
*/
if (++call->lock_codec.retry_cnt > LOCK_CODEC_MAX_RETRY)
return PJ_SUCCESS;
PJ_LOG(4, (THIS_FILE, "Got answer with multiple codecs, scheduling "
"updating media session to use only one codec.."));
call->lock_codec.sdp_ver = local_sdp->origin.version;
/* Can't send UPDATE or re-INVITE now, so just schedule it immediately.
* See: https://trac.pjsip.org/repos/ticket/1149
*/
pj_timer_entry_init(&call->lock_codec.reinv_timer, PJ_TRUE,
(void*)(pj_size_t)call->index,
&reinv_timer_cb);
pjsip_endpt_schedule_timer(pjsua_var.endpt,
&call->lock_codec.reinv_timer, &delay);
return PJ_SUCCESS;
}
/*
* This callback receives notification from invite session when the
* session state has changed.
*/
static void pjsua_call_on_state_changed(pjsip_inv_session *inv,
pjsip_event *e)
{
pjsua_call *call;
pj_log_push_indent();
PJSUA_LOCK();
call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
if (!call) {
PJSUA_UNLOCK();
pj_log_pop_indent();
return;
}
/* Get call times */
switch (inv->state) {
case PJSIP_INV_STATE_EARLY:
case PJSIP_INV_STATE_CONNECTING:
if (call->res_time.sec == 0)
pj_gettimeofday(&call->res_time);
call->last_code = (pjsip_status_code)
e->body.tsx_state.tsx->status_code;
pj_strncpy(&call->last_text,
&e->body.tsx_state.tsx->status_text,
sizeof(call->last_text_buf_));
break;
case PJSIP_INV_STATE_CONFIRMED:
pj_gettimeofday(&call->conn_time);
/* See if lock codec was pended as media update was done in the
* EARLY state and remote does not support UPDATE.
*/
if (call->lock_codec.pending) {
pj_status_t status;
status = lock_codec(call);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to lock codec", status);
}
call->lock_codec.pending = PJ_FALSE;
}
break;
case PJSIP_INV_STATE_DISCONNECTED:
pj_gettimeofday(&call->dis_time);
if (call->res_time.sec == 0)
pj_gettimeofday(&call->res_time);
if (e->type == PJSIP_EVENT_TSX_STATE &&
e->body.tsx_state.tsx->status_code > call->last_code)
{
call->last_code = (pjsip_status_code)
e->body.tsx_state.tsx->status_code;
pj_strncpy(&call->last_text,
&e->body.tsx_state.tsx->status_text,
sizeof(call->last_text_buf_));
} else {
call->last_code = PJSIP_SC_REQUEST_TERMINATED;
pj_strncpy(&call->last_text,
pjsip_get_status_text(call->last_code),
sizeof(call->last_text_buf_));
}
/* Stop lock codec timer, if it is active */
if (call->lock_codec.reinv_timer.id) {
pjsip_endpt_cancel_timer(pjsua_var.endpt,
&call->lock_codec.reinv_timer);
call->lock_codec.reinv_timer.id = PJ_FALSE;
}
break;
default:
call->last_code = (pjsip_status_code)
e->body.tsx_state.tsx->status_code;
pj_strncpy(&call->last_text,
&e->body.tsx_state.tsx->status_text,
sizeof(call->last_text_buf_));
break;
}
/* If this is an outgoing INVITE that was created because of
* REFER/transfer, send NOTIFY to transferer.
*/
if (call->xfer_sub && e->type==PJSIP_EVENT_TSX_STATE) {
int st_code = -1;
pjsip_evsub_state ev_state = PJSIP_EVSUB_STATE_ACTIVE;
switch (call->inv->state) {
case PJSIP_INV_STATE_NULL:
case PJSIP_INV_STATE_CALLING:
/* Do nothing */
break;
case PJSIP_INV_STATE_EARLY:
case PJSIP_INV_STATE_CONNECTING:
st_code = e->body.tsx_state.tsx->status_code;
if (call->inv->state == PJSIP_INV_STATE_CONNECTING)
ev_state = PJSIP_EVSUB_STATE_TERMINATED;
else
ev_state = PJSIP_EVSUB_STATE_ACTIVE;
break;
case PJSIP_INV_STATE_CONFIRMED:
#if 0
/* We don't need this, as we've terminated the subscription in
* CONNECTING state.
*/
/* When state is confirmed, send the final 200/OK and terminate
* subscription.
*/
st_code = e->body.tsx_state.tsx->status_code;
ev_state = PJSIP_EVSUB_STATE_TERMINATED;
#endif
break;
case PJSIP_INV_STATE_DISCONNECTED:
st_code = e->body.tsx_state.tsx->status_code;
ev_state = PJSIP_EVSUB_STATE_TERMINATED;
break;
case PJSIP_INV_STATE_INCOMING:
/* Nothing to do. Just to keep gcc from complaining about
* unused enums.
*/
break;
}
if (st_code != -1) {
pjsip_tx_data *tdata;
pj_status_t status;
status = pjsip_xfer_notify( call->xfer_sub,
ev_state, st_code,
NULL, &tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create NOTIFY", status);
} else {
status = pjsip_xfer_send_request(call->xfer_sub, tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to send NOTIFY", status);
}
}
}
}
if (pjsua_var.ua_cfg.cb.on_call_state)
(*pjsua_var.ua_cfg.cb.on_call_state)(call->index, e);
/* call->inv may be NULL now */
/* Destroy media session when invite session is disconnected. */
if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
pj_assert(call != NULL);
if (call)
pjsua_media_channel_deinit(call->index);
/* Free call */
call->inv = NULL;
--pjsua_var.call_cnt;
/* Reset call */
reset_call(call->index);
}
PJSUA_UNLOCK();
pj_log_pop_indent();
}
/*
* This callback is called by invite session framework when UAC session
* has forked.
*/
static void pjsua_call_on_forked( pjsip_inv_session *inv,
pjsip_event *e)
{
PJ_UNUSED_ARG(inv);
PJ_UNUSED_ARG(e);
PJ_TODO(HANDLE_FORKED_DIALOG);
}
/*
* Callback from UA layer when forked dialog response is received.
*/
pjsip_dialog* on_dlg_forked(pjsip_dialog *dlg, pjsip_rx_data *res)
{
if (dlg->uac_has_2xx &&
res->msg_info.cseq->method.id == PJSIP_INVITE_METHOD &&
pjsip_rdata_get_tsx(res) == NULL &&
res->msg_info.msg->line.status.code/100 == 2)
{
pjsip_dialog *forked_dlg;
pjsip_tx_data *bye;
pj_status_t status;
/* Create forked dialog */
status = pjsip_dlg_fork(dlg, res, &forked_dlg);
if (status != PJ_SUCCESS)
return NULL;
pjsip_dlg_inc_lock(forked_dlg);
/* Disconnect the call */
status = pjsip_dlg_create_request(forked_dlg, &pjsip_bye_method,
-1, &bye);
if (status == PJ_SUCCESS) {
status = pjsip_dlg_send_request(forked_dlg, bye, -1, NULL);
}
pjsip_dlg_dec_lock(forked_dlg);
if (status != PJ_SUCCESS) {
return NULL;
}
return forked_dlg;
} else {
return dlg;
}
}
/*
* Disconnect call upon error.
*/
static void call_disconnect( pjsip_inv_session *inv,
int code )
{
pjsua_call *call;
pjsip_tx_data *tdata;
pj_status_t status;
call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
status = pjsip_inv_end_session(inv, code, NULL, &tdata);
if (status != PJ_SUCCESS)
return;
/* Add SDP in 488 status */
#if DISABLED_FOR_TICKET_1185
if (call && call->tp && tdata->msg->type==PJSIP_RESPONSE_MSG &&
code==PJSIP_SC_NOT_ACCEPTABLE_HERE)
{
pjmedia_sdp_session *local_sdp;
pjmedia_transport_info ti;
pjmedia_transport_info_init(&ti);
pjmedia_transport_get_info(call->med_tp, &ti);
status = pjmedia_endpt_create_sdp(pjsua_var.med_endpt, tdata->pool,
1, &ti.sock_info, &local_sdp);
if (status == PJ_SUCCESS) {
pjsip_create_sdp_body(tdata->pool, local_sdp,
&tdata->msg->body);
}
}
#endif
pjsip_inv_send_msg(inv, tdata);
}
/*
* Callback to be called when SDP offer/answer negotiation has just completed
* in the session. This function will start/update media if negotiation
* has succeeded.
*/
static void pjsua_call_on_media_update(pjsip_inv_session *inv,
pj_status_t status)
{
pjsua_call *call;
const pjmedia_sdp_session *local_sdp;
const pjmedia_sdp_session *remote_sdp;
//const pj_str_t st_update = {"UPDATE", 6};
pj_log_push_indent();
PJSUA_LOCK();
call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "SDP negotiation has failed", status);
/* Do not deinitialize media since this may be a re-INVITE or
* UPDATE (which in this case the media should not get affected
* by the failed re-INVITE/UPDATE). The media will be shutdown
* when call is disconnected anyway.
*/
/* Stop/destroy media, if any */
/*pjsua_media_channel_deinit(call->index);*/
/* Disconnect call if we're not in the middle of initializing an
* UAS dialog and if this is not a re-INVITE
*/
if (inv->state != PJSIP_INV_STATE_NULL &&
inv->state != PJSIP_INV_STATE_CONFIRMED)
{
call_disconnect(inv, PJSIP_SC_UNSUPPORTED_MEDIA_TYPE);
}
goto on_return;
}
/* Get local and remote SDP */
status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &local_sdp);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE,
"Unable to retrieve currently active local SDP",
status);
//call_disconnect(inv, PJSIP_SC_UNSUPPORTED_MEDIA_TYPE);
goto on_return;
}
status = pjmedia_sdp_neg_get_active_remote(call->inv->neg, &remote_sdp);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE,
"Unable to retrieve currently active remote SDP",
status);
//call_disconnect(inv, PJSIP_SC_UNSUPPORTED_MEDIA_TYPE);
goto on_return;
}
/* Update remote's NAT type */
if (pjsua_var.ua_cfg.nat_type_in_sdp) {
update_remote_nat_type(call, remote_sdp);
}
/* Update media channel with the new SDP */
status = pjsua_media_channel_update(call->index, local_sdp, remote_sdp);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create media session",
status);
call_disconnect(inv, PJSIP_SC_NOT_ACCEPTABLE_HERE);
/* No need to deinitialize; media will be shutdown when call
* state is disconnected anyway.
*/
/*pjsua_media_channel_deinit(call->index);*/
goto on_return;
}
/* Ticket #476: make sure only one codec is specified in the answer. */
status = lock_codec(call);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to lock codec", status);
}
/* Call application callback, if any */
if (pjsua_var.ua_cfg.cb.on_call_media_state)
pjsua_var.ua_cfg.cb.on_call_media_state(call->index);
on_return:
PJSUA_UNLOCK();
pj_log_pop_indent();
}
/* Modify SDP for call hold. */
static pj_status_t modify_sdp_of_call_hold(pjsua_call *call,
pj_pool_t *pool,
pjmedia_sdp_session *sdp)
{
unsigned mi;
/* Call-hold is done by set the media direction to 'sendonly'
* (PJMEDIA_DIR_ENCODING), except when current media direction is
* 'inactive' (PJMEDIA_DIR_NONE).
* (See RFC 3264 Section 8.4 and RFC 4317 Section 3.1)
*/
/* http://trac.pjsip.org/repos/ticket/880
if (call->dir != PJMEDIA_DIR_ENCODING) {
*/
/* https://trac.pjsip.org/repos/ticket/1142:
* configuration to use c=0.0.0.0 for call hold.
*/
for (mi=0; mi<sdp->media_count; ++mi) {
pjmedia_sdp_media *m = sdp->media[mi];
if (call->call_hold_type == PJSUA_CALL_HOLD_TYPE_RFC2543) {
pjmedia_sdp_conn *conn;
pjmedia_sdp_attr *attr;
/* Get SDP media connection line */
conn = m->conn;
if (!conn)
conn = sdp->conn;
/* Modify address */
conn->addr = pj_str("0.0.0.0");
/* Remove existing directions attributes */
pjmedia_sdp_media_remove_all_attr(m, "sendrecv");
pjmedia_sdp_media_remove_all_attr(m, "sendonly");
pjmedia_sdp_media_remove_all_attr(m, "recvonly");
pjmedia_sdp_media_remove_all_attr(m, "inactive");
/* Add inactive attribute */
attr = pjmedia_sdp_attr_create(pool, "inactive", NULL);
pjmedia_sdp_media_add_attr(m, attr);
} else {
pjmedia_sdp_attr *attr;
/* Remove existing directions attributes */
pjmedia_sdp_media_remove_all_attr(m, "sendrecv");
pjmedia_sdp_media_remove_all_attr(m, "sendonly");
pjmedia_sdp_media_remove_all_attr(m, "recvonly");
pjmedia_sdp_media_remove_all_attr(m, "inactive");
if (call->media[mi].dir & PJMEDIA_DIR_ENCODING) {
/* Add sendonly attribute */
attr = pjmedia_sdp_attr_create(pool, "sendonly", NULL);
pjmedia_sdp_media_add_attr(m, attr);
} else {
/* Add inactive attribute */
attr = pjmedia_sdp_attr_create(pool, "inactive", NULL);
pjmedia_sdp_media_add_attr(m, attr);
}
}
}
return PJ_SUCCESS;
}
/* Create SDP for call hold. */
static pj_status_t create_sdp_of_call_hold(pjsua_call *call,
pjmedia_sdp_session **p_sdp)
{
pj_status_t status;
pj_pool_t *pool;
pjmedia_sdp_session *sdp;
/* Use call's provisional pool */
pool = call->inv->pool_prov;
/* Create new offer */
status = pjsua_media_channel_create_sdp(call->index, pool, NULL, &sdp,
NULL);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create local SDP", status);
return status;
}
status = modify_sdp_of_call_hold(call, pool, sdp);
if (status != PJ_SUCCESS)
return status;
*p_sdp = sdp;
return PJ_SUCCESS;
}
/*
* Called when session received new offer.
*/
static void pjsua_call_on_rx_offer(pjsip_inv_session *inv,
const pjmedia_sdp_session *offer)
{
pjsua_call *call;
pjmedia_sdp_session *answer;
unsigned i;
pj_status_t status;
PJSUA_LOCK();
call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
/* Supply candidate answer */
PJ_LOG(4,(THIS_FILE, "Call %d: received updated media offer",
call->index));
pj_log_push_indent();
status = pjsua_media_channel_create_sdp(call->index,
call->inv->pool_prov,
offer, &answer, NULL);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create local SDP", status);
goto on_return;
}
/* Validate media count in the generated answer */
pj_assert(answer->media_count == offer->media_count);
/* Check if offer's conn address is zero */
for (i = 0; i < answer->media_count; ++i) {
pjmedia_sdp_conn *conn;
conn = offer->media[i]->conn;
if (!conn)
conn = offer->conn;
if (pj_strcmp2(&conn->addr, "0.0.0.0")==0 ||
pj_strcmp2(&conn->addr, "0")==0)
{
pjmedia_sdp_conn *a_conn = answer->media[i]->conn;
/* Modify answer address */
if (a_conn) {
a_conn->addr = pj_str("0.0.0.0");
} else if (answer->conn == NULL ||
pj_strcmp2(&answer->conn->addr, "0.0.0.0") != 0)
{
a_conn = PJ_POOL_ZALLOC_T(call->inv->pool_prov,
pjmedia_sdp_conn);
a_conn->net_type = pj_str("IN");
a_conn->addr_type = pj_str("IP4");
a_conn->addr = pj_str("0.0.0.0");
answer->media[i]->conn = a_conn;
}
}
}
/* Check if call is on-hold */
if (call->local_hold) {
modify_sdp_of_call_hold(call, call->inv->pool_prov, answer);
}
status = pjsip_inv_set_sdp_answer(call->inv, answer);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to set answer", status);
goto on_return;
}
on_return:
PJSUA_UNLOCK();
pj_log_pop_indent();
}
/*
* Called to generate new offer.
*/
static void pjsua_call_on_create_offer(pjsip_inv_session *inv,
pjmedia_sdp_session **offer)
{
pjsua_call *call;
pj_status_t status;
pj_log_push_indent();
PJSUA_LOCK();
call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
/* See if we've put call on hold. */
if (call->local_hold) {
PJ_LOG(4,(THIS_FILE,
"Call %d: call is on-hold locally, creating call-hold SDP ",
call->index));
status = create_sdp_of_call_hold( call, offer );
} else {
PJ_LOG(4,(THIS_FILE, "Call %d: asked to send a new offer",
call->index));
status = pjsua_media_channel_create_sdp(call->index,
call->inv->pool_prov,
NULL, offer, NULL);
}
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create local SDP", status);
goto on_return;
}
on_return:
PJSUA_UNLOCK();
pj_log_pop_indent();
}
/*
* Callback called by event framework when the xfer subscription state
* has changed.
*/
static void xfer_client_on_evsub_state( pjsip_evsub *sub, pjsip_event *event)
{
PJ_UNUSED_ARG(event);
pj_log_push_indent();
/*
* When subscription is accepted (got 200/OK to REFER), check if
* subscription suppressed.
*/
if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACCEPTED) {
pjsip_rx_data *rdata;
pjsip_generic_string_hdr *refer_sub;
const pj_str_t REFER_SUB = { "Refer-Sub", 9 };
pjsua_call *call;
call = (pjsua_call*) pjsip_evsub_get_mod_data(sub, pjsua_var.mod.id);
/* Must be receipt of response message */
pj_assert(event->type == PJSIP_EVENT_TSX_STATE &&
event->body.tsx_state.type == PJSIP_EVENT_RX_MSG);
rdata = event->body.tsx_state.src.rdata;
/* Find Refer-Sub header */
refer_sub = (pjsip_generic_string_hdr*)
pjsip_msg_find_hdr_by_name(rdata->msg_info.msg,
&REFER_SUB, NULL);
/* Check if subscription is suppressed */
if (refer_sub && pj_stricmp2(&refer_sub->hvalue, "false")==0) {
/* Since no subscription is desired, assume that call has been
* transfered successfully.
*/
if (call && pjsua_var.ua_cfg.cb.on_call_transfer_status) {
const pj_str_t ACCEPTED = { "Accepted", 8 };
pj_bool_t cont = PJ_FALSE;
(*pjsua_var.ua_cfg.cb.on_call_transfer_status)(call->index,
200,
&ACCEPTED,
PJ_TRUE,
&cont);
}
/* Yes, subscription is suppressed.
* Terminate our subscription now.
*/
PJ_LOG(4,(THIS_FILE, "Xfer subscription suppressed, terminating "
"event subcription..."));
pjsip_evsub_terminate(sub, PJ_TRUE);
} else {
/* Notify application about call transfer progress.
* Initially notify with 100/Accepted status.
*/
if (call && pjsua_var.ua_cfg.cb.on_call_transfer_status) {
const pj_str_t ACCEPTED = { "Accepted", 8 };
pj_bool_t cont = PJ_FALSE;
(*pjsua_var.ua_cfg.cb.on_call_transfer_status)(call->index,
100,
&ACCEPTED,
PJ_FALSE,
&cont);
}
}
}
/*
* On incoming NOTIFY, notify application about call transfer progress.
*/
else if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACTIVE ||
pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED)
{
pjsua_call *call;
pjsip_msg *msg;
pjsip_msg_body *body;
pjsip_status_line status_line;
pj_bool_t is_last;
pj_bool_t cont;
pj_status_t status;
call = (pjsua_call*) pjsip_evsub_get_mod_data(sub, pjsua_var.mod.id);
/* When subscription is terminated, clear the xfer_sub member of
* the inv_data.
*/
if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED) {
pjsip_evsub_set_mod_data(sub, pjsua_var.mod.id, NULL);
PJ_LOG(4,(THIS_FILE, "Xfer client subscription terminated"));
}
if (!call || !event || !pjsua_var.ua_cfg.cb.on_call_transfer_status) {
/* Application is not interested with call progress status */
goto on_return;
}
/* This better be a NOTIFY request */
if (event->type == PJSIP_EVENT_TSX_STATE &&
event->body.tsx_state.type == PJSIP_EVENT_RX_MSG)
{
pjsip_rx_data *rdata;
rdata = event->body.tsx_state.src.rdata;
/* Check if there's body */
msg = rdata->msg_info.msg;
body = msg->body;
if (!body) {
PJ_LOG(2,(THIS_FILE,
"Warning: received NOTIFY without message body"));
goto on_return;
}
/* Check for appropriate content */
if (pj_stricmp2(&body->content_type.type, "message") != 0 ||
pj_stricmp2(&body->content_type.subtype, "sipfrag") != 0)
{
PJ_LOG(2,(THIS_FILE,
"Warning: received NOTIFY with non message/sipfrag "
"content"));
goto on_return;
}
/* Try to parse the content */
status = pjsip_parse_status_line((char*)body->data, body->len,
&status_line);
if (status != PJ_SUCCESS) {
PJ_LOG(2,(THIS_FILE,
"Warning: received NOTIFY with invalid "
"message/sipfrag content"));
goto on_return;
}
} else {
status_line.code = 500;
status_line.reason = *pjsip_get_status_text(500);
}
/* Notify application */
is_last = (pjsip_evsub_get_state(sub)==PJSIP_EVSUB_STATE_TERMINATED);
cont = !is_last;
(*pjsua_var.ua_cfg.cb.on_call_transfer_status)(call->index,
status_line.code,
&status_line.reason,
is_last, &cont);
if (!cont) {
pjsip_evsub_set_mod_data(sub, pjsua_var.mod.id, NULL);
}
/* If the call transfer has completed but the subscription is
* not terminated, terminate it now.
*/
if (status_line.code/100 == 2 && !is_last) {
pjsip_tx_data *tdata;
status = pjsip_evsub_initiate(sub, &pjsip_subscribe_method,
0, &tdata);
if (status == PJ_SUCCESS)
status = pjsip_evsub_send_request(sub, tdata);
}
}
on_return:
pj_log_pop_indent();
}
/*
* Callback called by event framework when the xfer subscription state
* has changed.
*/
static void xfer_server_on_evsub_state( pjsip_evsub *sub, pjsip_event *event)
{
PJ_UNUSED_ARG(event);
pj_log_push_indent();
/*
* When subscription is terminated, clear the xfer_sub member of
* the inv_data.
*/
if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED) {
pjsua_call *call;
call = (pjsua_call*) pjsip_evsub_get_mod_data(sub, pjsua_var.mod.id);
if (!call)
goto on_return;
pjsip_evsub_set_mod_data(sub, pjsua_var.mod.id, NULL);
call->xfer_sub = NULL;
PJ_LOG(4,(THIS_FILE, "Xfer server subscription terminated"));
}
on_return:
pj_log_pop_indent();
}
/*
* Follow transfer (REFER) request.
*/
static void on_call_transfered( pjsip_inv_session *inv,
pjsip_rx_data *rdata )
{
pj_status_t status;
pjsip_tx_data *tdata;
pjsua_call *existing_call;
int new_call;
const pj_str_t str_refer_to = { "Refer-To", 8};
const pj_str_t str_refer_sub = { "Refer-Sub", 9 };
const pj_str_t str_ref_by = { "Referred-By", 11 };
pjsip_generic_string_hdr *refer_to;
pjsip_generic_string_hdr *refer_sub;
pjsip_hdr *ref_by_hdr;
pj_bool_t no_refer_sub = PJ_FALSE;
char *uri;
pjsua_msg_data msg_data;
pj_str_t tmp;
pjsip_status_code code;
pjsip_evsub *sub;
pj_log_push_indent();
existing_call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
/* Find the Refer-To header */
refer_to = (pjsip_generic_string_hdr*)
pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_refer_to, NULL);
if (refer_to == NULL) {
/* Invalid Request.
* No Refer-To header!
*/
PJ_LOG(4,(THIS_FILE, "Received REFER without Refer-To header!"));
pjsip_dlg_respond( inv->dlg, rdata, 400, NULL, NULL, NULL);
goto on_return;
}
/* Find optional Refer-Sub header */
refer_sub = (pjsip_generic_string_hdr*)
pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_refer_sub, NULL);
if (refer_sub) {
if (!pj_strnicmp2(&refer_sub->hvalue, "true", 4)==0)
no_refer_sub = PJ_TRUE;
}
/* Find optional Referred-By header (to be copied onto outgoing INVITE
* request.
*/
ref_by_hdr = (pjsip_hdr*)
pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_ref_by,
NULL);
/* Notify callback */
code = PJSIP_SC_ACCEPTED;
if (pjsua_var.ua_cfg.cb.on_call_transfer_request)
(*pjsua_var.ua_cfg.cb.on_call_transfer_request)(existing_call->index,
&refer_to->hvalue,
&code);
if (code < 200)
code = PJSIP_SC_ACCEPTED;
if (code >= 300) {
/* Application rejects call transfer request */
pjsip_dlg_respond( inv->dlg, rdata, code, NULL, NULL, NULL);
goto on_return;
}
PJ_LOG(3,(THIS_FILE, "Call to %.*s is being transfered to %.*s",
(int)inv->dlg->remote.info_str.slen,
inv->dlg->remote.info_str.ptr,
(int)refer_to->hvalue.slen,
refer_to->hvalue.ptr));
if (no_refer_sub) {
/*
* Always answer with 2xx.
*/
pjsip_tx_data *tdata;
const pj_str_t str_false = { "false", 5};
pjsip_hdr *hdr;
status = pjsip_dlg_create_response(inv->dlg, rdata, code, NULL,
&tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create 2xx response to REFER",
status);
goto on_return;
}
/* Add Refer-Sub header */
hdr = (pjsip_hdr*)
pjsip_generic_string_hdr_create(tdata->pool, &str_refer_sub,
&str_false);
pjsip_msg_add_hdr(tdata->msg, hdr);
/* Send answer */
status = pjsip_dlg_send_response(inv->dlg, pjsip_rdata_get_tsx(rdata),
tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create 2xx response to REFER",
status);
goto on_return;
}
/* Don't have subscription */
sub = NULL;
} else {
struct pjsip_evsub_user xfer_cb;
pjsip_hdr hdr_list;
/* Init callback */
pj_bzero(&xfer_cb, sizeof(xfer_cb));
xfer_cb.on_evsub_state = &xfer_server_on_evsub_state;
/* Init additional header list to be sent with REFER response */
pj_list_init(&hdr_list);
/* Create transferee event subscription */
status = pjsip_xfer_create_uas( inv->dlg, &xfer_cb, rdata, &sub);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create xfer uas", status);
pjsip_dlg_respond( inv->dlg, rdata, 500, NULL, NULL, NULL);
goto on_return;
}
/* If there's Refer-Sub header and the value is "true", send back
* Refer-Sub in the response with value "true" too.
*/
if (refer_sub) {
const pj_str_t str_true = { "true", 4 };
pjsip_hdr *hdr;
hdr = (pjsip_hdr*)
pjsip_generic_string_hdr_create(inv->dlg->pool,
&str_refer_sub,
&str_true);
pj_list_push_back(&hdr_list, hdr);
}
/* Accept the REFER request, send 2xx. */
pjsip_xfer_accept(sub, rdata, code, &hdr_list);
/* Create initial NOTIFY request */
status = pjsip_xfer_notify( sub, PJSIP_EVSUB_STATE_ACTIVE,
100, NULL, &tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create NOTIFY to REFER",
status);
goto on_return;
}
/* Send initial NOTIFY request */
status = pjsip_xfer_send_request( sub, tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to send NOTIFY to REFER", status);
goto on_return;
}
}
/* We're cheating here.
* We need to get a null terminated string from a pj_str_t.
* So grab the pointer from the hvalue and NULL terminate it, knowing
* that the NULL position will be occupied by a newline.
*/
uri = refer_to->hvalue.ptr;
uri[refer_to->hvalue.slen] = '\0';
/* Init msg_data */
pjsua_msg_data_init(&msg_data);
/* If Referred-By header is present in the REFER request, copy this
* to the outgoing INVITE request.
*/
if (ref_by_hdr != NULL) {
pjsip_hdr *dup = (pjsip_hdr*)
pjsip_hdr_clone(rdata->tp_info.pool, ref_by_hdr);
pj_list_push_back(&msg_data.hdr_list, dup);
}
/* Now make the outgoing call. */
tmp = pj_str(uri);
status = pjsua_call_make_call(existing_call->acc_id, &tmp, 0,
existing_call->user_data, &msg_data,
&new_call);
if (status != PJ_SUCCESS) {
/* Notify xferer about the error (if we have subscription) */
if (sub) {
status = pjsip_xfer_notify(sub, PJSIP_EVSUB_STATE_TERMINATED,
500, NULL, &tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create NOTIFY to REFER",
status);
goto on_return;
}
status = pjsip_xfer_send_request(sub, tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to send NOTIFY to REFER",
status);
goto on_return;
}
}
goto on_return;
}
if (sub) {
/* Put the server subscription in inv_data.
* Subsequent state changed in pjsua_inv_on_state_changed() will be
* reported back to the server subscription.
*/
pjsua_var.calls[new_call].xfer_sub = sub;
/* Put the invite_data in the subscription. */
pjsip_evsub_set_mod_data(sub, pjsua_var.mod.id,
&pjsua_var.calls[new_call]);
}
on_return:
pj_log_pop_indent();
}
/*
* This callback is called when transaction state has changed in INVITE
* session. We use this to trap:
* - incoming REFER request.
* - incoming MESSAGE request.
*/
static void pjsua_call_on_tsx_state_changed(pjsip_inv_session *inv,
pjsip_transaction *tsx,
pjsip_event *e)
{
pjsua_call *call;
pj_log_push_indent();
PJSUA_LOCK();
call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
if (call == NULL)
goto on_return;
if (call->inv == NULL) {
/* Shouldn't happen. It happens only when we don't terminate the
* server subscription caused by REFER after the call has been
* transfered (and this call has been disconnected), and we
* receive another REFER for this call.
*/
goto on_return;
}
/* Notify application callback first */
if (pjsua_var.ua_cfg.cb.on_call_tsx_state) {
(*pjsua_var.ua_cfg.cb.on_call_tsx_state)(call->index, tsx, e);
}
if (tsx->role==PJSIP_ROLE_UAS &&
tsx->state==PJSIP_TSX_STATE_TRYING &&
pjsip_method_cmp(&tsx->method, pjsip_get_refer_method())==0)
{
/*
* Incoming REFER request.
*/
on_call_transfered(call->inv, e->body.tsx_state.src.rdata);
}
else if (tsx->role==PJSIP_ROLE_UAS &&
tsx->state==PJSIP_TSX_STATE_TRYING &&
pjsip_method_cmp(&tsx->method, &pjsip_message_method)==0)
{
/*
* Incoming MESSAGE request!
*/
pjsip_rx_data *rdata;
pjsip_msg *msg;
pjsip_accept_hdr *accept_hdr;
pj_status_t status;
rdata = e->body.tsx_state.src.rdata;
msg = rdata->msg_info.msg;
/* Request MUST have message body, with Content-Type equal to
* "text/plain".
*/
if (pjsua_im_accept_pager(rdata, &accept_hdr) == PJ_FALSE) {
pjsip_hdr hdr_list;
pj_list_init(&hdr_list);
pj_list_push_back(&hdr_list, accept_hdr);
pjsip_dlg_respond( inv->dlg, rdata, PJSIP_SC_NOT_ACCEPTABLE_HERE,
NULL, &hdr_list, NULL );
goto on_return;
}
/* Respond with 200 first, so that remote doesn't retransmit in case
* the UI takes too long to process the message.
*/
status = pjsip_dlg_respond( inv->dlg, rdata, 200, NULL, NULL, NULL);
/* Process MESSAGE request */
pjsua_im_process_pager(call->index, &inv->dlg->remote.info_str,
&inv->dlg->local.info_str, rdata);
}
else if (tsx->role == PJSIP_ROLE_UAC &&
pjsip_method_cmp(&tsx->method, &pjsip_message_method)==0)
{
/* Handle outgoing pager status */
if (tsx->status_code >= 200) {
pjsua_im_data *im_data;
im_data = (pjsua_im_data*) tsx->mod_data[pjsua_var.mod.id];
/* im_data can be NULL if this is typing indication */
if (im_data && pjsua_var.ua_cfg.cb.on_pager_status) {
pjsua_var.ua_cfg.cb.on_pager_status(im_data->call_id,
&im_data->to,
&im_data->body,
im_data->user_data,
(pjsip_status_code)
tsx->status_code,
&tsx->status_text);
}
}
}
on_return:
PJSUA_UNLOCK();
pj_log_pop_indent();
}
/* Redirection handler */
static pjsip_redirect_op pjsua_call_on_redirected(pjsip_inv_session *inv,
const pjsip_uri *target,
const pjsip_event *e)
{
pjsua_call *call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
pjsip_redirect_op op;
pj_log_push_indent();
PJSUA_LOCK();
if (pjsua_var.ua_cfg.cb.on_call_redirected) {
op = (*pjsua_var.ua_cfg.cb.on_call_redirected)(call->index,
target, e);
} else {
PJ_LOG(4,(THIS_FILE, "Unhandled redirection for call %d "
"(callback not implemented by application). Disconnecting "
"call.",
call->index));
op = PJSIP_REDIRECT_STOP;
}
PJSUA_UNLOCK();
pj_log_pop_indent();
return op;
}