Gitiles
Code Review Sign In
review.jami.net / pjproject / 8d32e0b87ee3c747412be34fc9916a4e209f2e8f / . / pjsip / src / pjsua-lib / pjsua_call.c
blob: 96b24f7e6250b6e9f2e18be4352cf47f8d308945 [file] [log] [blame]
/*
* 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
/* Determine whether we should restart ICE upon receiving a re-INVITE
* with no SDP.
*/
#define RESTART_ICE_ON_REINVITE 1
/* Retry interval of trying to hangup a call. */
#define CALL_HANGUP_RETRY_INTERVAL 5000
/* Max number of hangup retries. */
#define CALL_HANGUP_MAX_RETRY 4
/*
* The INFO method.
*/
const pjsip_method pjsip_info_method =
{
PJSIP_OTHER_METHOD,
{ "INFO", 4 }
};
/* UPDATE method */
static const pjsip_method pjsip_update_method =
{
PJSIP_OTHER_METHOD,
{ "UPDATE", 6 }
};
/* 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,
struct pjsip_inv_on_rx_offer_cb_param *param);
/*
* Called when receiving re-INVITE.
*/
static pj_status_t pjsua_call_on_rx_reinvite(pjsip_inv_session *inv,
const pjmedia_sdp_session *offer,
pjsip_rx_data *rdata);
/*
* 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);
/* Timer callback to send re-INVITE/UPDATE to lock codec or ICE update */
static void reinv_timer_cb(pj_timer_heap_t *th, pj_timer_entry *entry);
/* Timer callback to hangup the call */
static void hangup_timer_cb(pj_timer_heap_t *th, pj_timer_entry *entry);
/* Check and send reinvite for lock codec and ICE update */
static pj_status_t process_pending_reinvite(pjsua_call *call);
/* Timer callbacks for trickle ICE */
static void trickle_ice_send_sip_info(pj_timer_heap_t *th,
struct pj_timer_entry *te);
static void trickle_ice_retrans_18x(pj_timer_heap_t *th,
struct pj_timer_entry *te);
/* End call session */
static pj_status_t call_inv_end_session(pjsua_call *call,
unsigned code,
const pj_str_t *reason,
const pjsua_msg_data *msg_data);
/*
* Reset call descriptor.
*/
static void reset_call(pjsua_call_id id)
{
pjsua_call *call = &pjsua_var.calls[id];
unsigned i;
if (call->incoming_data) {
pjsip_rx_data_free_cloned(call->incoming_data);
call->incoming_data = NULL;
}
pj_bzero(call, sizeof(*call));
call->index = id;
call->last_text.ptr = call->last_text_buf_;
call->cname.ptr = call->cname_buf;
call->cname.slen = sizeof(call->cname_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->strm.v.strm_dec_slot = PJSUA_INVALID_ID;
call_med->strm.v.strm_enc_slot = PJSUA_INVALID_ID;
call_med->call = call;
call_med->idx = i;
call_med->tp_auto_del = PJ_TRUE;
}
pjsua_call_setting_default(&call->opt);
pj_timer_entry_init(&call->reinv_timer, PJ_FALSE,
(void*)(pj_size_t)id, &reinv_timer_cb);
pj_bzero(&call->trickle_ice, sizeof(call->trickle_ice));
pj_timer_entry_init(&call->trickle_ice.timer, 0, call,
&trickle_ice_send_sip_info);
}
/* Get DTMF method type name */
static const char* get_dtmf_method_name(int type)
{
switch (type) {
case PJSUA_DTMF_METHOD_RFC2833:
return "RFC2833";
case PJSUA_DTMF_METHOD_SIP_INFO:
return "SIP INFO";
}
return "(Unknown)";
}
/*
* 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 };
const pj_str_t str_trickle_ice = { "trickle-ice", 11 };
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_offer2 = &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;
if (pjsua_var.ua_cfg.cb.on_call_rx_reinvite) {
inv_cb.on_rx_reinvite = &pjsua_call_on_rx_reinvite;
}
/* 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);
/* Add "INFO" in Allow header, for DTMF and video key frame request. */
pjsip_endpt_add_capability(pjsua_var.endpt, NULL, PJSIP_H_ALLOW,
NULL, 1, &pjsip_info_method.name);
/* Add "trickle-ice" in Supported header */
pjsip_endpt_add_capability(pjsua_var.endpt, NULL, PJSIP_H_SUPPORTED,
NULL, 1, &str_trickle_ice);
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.calls[cid].async_call.dlg == 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.calls[cid].async_call.dlg == 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 = NULL;
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 = 0;
pjsip_tx_data *tdata;
pj_bool_t cb_called = PJ_FALSE;
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);
/* Decrement dialog session. */
pjsip_dlg_dec_session(dlg, &pjsua_var.mod);
if (status != PJ_SUCCESS) {
pj_str_t err_str;
pj_ssize_t title_len;
call->last_code = PJSIP_SC_TEMPORARILY_UNAVAILABLE;
pj_strcpy2(&call->last_text, "Media init error: ");
title_len = call->last_text.slen;
err_str = pj_strerror(status, call->last_text_buf_ + title_len,
sizeof(call->last_text_buf_) - title_len);
call->last_text.slen += err_str.slen;
pjsua_perror(THIS_FILE, "Error initializing media channel", status);
goto on_error;
}
/* pjsua_media_channel_deinit() has been called or
* call has been hung up.
*/
if (call->async_call.med_ch_deinit ||
call->async_call.call_var.out_call.hangup)
{
PJ_LOG(4,(THIS_FILE, "Call has been hung up or media channel has "
"been deinitialized"));
goto on_error;
}
/* Create offer */
if ((call->opt.flag & PJSUA_CALL_NO_SDP_OFFER) == 0) {
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 == PJSUA_100REL_MANDATORY)
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;
}
if (acc->cfg.ice_cfg.enable_ice &&
acc->cfg.ice_cfg.ice_opt.trickle != PJ_ICE_SESS_TRICKLE_DISABLED)
{
options |= PJSIP_INV_SUPPORT_TRICKLE_ICE;
}
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) {
cb_called = PJ_TRUE;
/* Upon failure to send first request, the invite
* session would have been cleared.
*/
call->inv = inv = NULL;
goto on_error;
}
/* Done. */
call->med_ch_cb = NULL;
pjsip_dlg_dec_lock(dlg);
PJSUA_UNLOCK();
return PJ_SUCCESS;
on_error:
if (inv == NULL && call_id != -1 && !cb_called &&
!call->hanging_up &&
pjsua_var.ua_cfg.cb.on_call_state)
{
/* Use user event rather than NULL to avoid crash in
* unsuspecting app.
*/
pjsip_event user_event;
PJSIP_EVENT_INIT_USER(user_event, 0, 0, 0, 0);
(*pjsua_var.ua_cfg.cb.on_call_state)(call_id, &user_event);
}
if (dlg) {
/* This may destroy the dialog */
pjsip_dlg_dec_lock(dlg);
call->async_call.dlg = NULL;
}
if (inv != NULL) {
pjsip_inv_terminate(inv, PJSIP_SC_OK, PJ_FALSE);
call->inv = NULL;
}
if (call_id != -1) {
pjsua_media_channel_deinit(call_id);
reset_call(call_id);
}
call->med_ch_cb = NULL;
pjsua_check_snd_dev_idle();
PJSUA_UNLOCK();
return status;
}
/*
* Cleanup call setting flag to avoid one time flags, such as
* PJSUA_CALL_UNHOLD, PJSUA_CALL_UPDATE_CONTACT, or
* PJSUA_CALL_NO_SDP_OFFER, to be sticky (ticket #1793).
*/
void pjsua_call_cleanup_flag(pjsua_call_setting *opt)
{
opt->flag &= ~(PJSUA_CALL_UNHOLD | PJSUA_CALL_UPDATE_CONTACT |
PJSUA_CALL_NO_SDP_OFFER | PJSUA_CALL_REINIT_MEDIA |
PJSUA_CALL_UPDATE_VIA | PJSUA_CALL_SET_MEDIA_DIR);
}
/*
* Initialize call settings based on account ID.
*/
PJ_DEF(void) pjsua_call_setting_default(pjsua_call_setting *opt)
{
unsigned i;
pj_assert(opt);
pj_bzero(opt, sizeof(*opt));
opt->flag = PJSUA_CALL_INCLUDE_DISABLED_MEDIA;
opt->aud_cnt = 1;
#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
opt->vid_cnt = 1;
opt->req_keyframe_method = PJSUA_VID_REQ_KEYFRAME_SIP_INFO |
PJSUA_VID_REQ_KEYFRAME_RTCP_PLI;
#endif
for (i = 0; i < PJMEDIA_MAX_SDP_MEDIA; i++) {
opt->media_dir[i] = PJMEDIA_DIR_ENCODING_DECODING;
}
}
/*
* Initialize pjsua_call_send_dtmf_param default values.
*/
PJ_DEF(void) pjsua_call_send_dtmf_param_default(
pjsua_call_send_dtmf_param *param)
{
pj_bzero(param, sizeof(*param));
param->duration = PJSUA_CALL_SEND_DTMF_DURATION_DEFAULT;
}
static pj_status_t apply_call_setting(pjsua_call *call,
const pjsua_call_setting *opt,
const pjmedia_sdp_session *rem_sdp)
{
pj_assert(call);
if (!opt) {
pjsua_call_cleanup_flag(&call->opt);
} else {
call->opt = *opt;
}
#if !PJMEDIA_HAS_VIDEO
pj_assert(call->opt.vid_cnt == 0);
#endif
if (call->opt.flag & PJSUA_CALL_REINIT_MEDIA) {
PJ_LOG(4, (THIS_FILE, "PJSUA_CALL_REINIT_MEDIA"));
pjsua_media_channel_deinit(call->index);
}
/* If call is established or media channel hasn't been initialized,
* reinit media channel.
*/
if ((call->inv && call->inv->state == PJSIP_INV_STATE_CONNECTING &&
call->med_cnt == 0) ||
(call->inv && call->inv->state == PJSIP_INV_STATE_CONFIRMED) ||
(call->opt.flag & PJSUA_CALL_REINIT_MEDIA))
{
pjsip_role_e role = rem_sdp? PJSIP_ROLE_UAS : PJSIP_ROLE_UAC;
pj_status_t status;
status = pjsua_media_channel_init(call->index, role,
call->secure_level,
call->inv->pool_prov,
rem_sdp, NULL,
PJ_FALSE, NULL);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Error re-initializing media channel",
status);
return status;
}
}
return PJ_SUCCESS;
}
static void dlg_set_via(pjsip_dialog *dlg, pjsua_acc *acc)
{
if (acc->cfg.allow_via_rewrite && acc->via_addr.host.slen > 0) {
pjsip_dlg_set_via_sent_by(dlg, &acc->via_addr, acc->via_tp);
} else if (!pjsua_sip_acc_is_using_stun(acc->index) &&
!pjsua_sip_acc_is_using_upnp(acc->index))
{
/* Choose local interface to use in Via if acc is not using
* STUN nor UPnP. See https://github.com/pjsip/pjproject/issues/1804
*/
pjsip_host_port via_addr;
const void *via_tp;
if (pjsua_acc_get_uac_addr(acc->index, dlg->pool, &acc->cfg.id,
&via_addr, NULL, NULL,
&via_tp) == PJ_SUCCESS)
{
pjsip_dlg_set_via_sent_by(dlg, &via_addr,
(pjsip_transport*)via_tp);
}
}
}
static pj_status_t dlg_set_target(pjsip_dialog *dlg, const pj_str_t *target)
{
pjsip_uri *target_uri;
pj_str_t tmp;
pj_status_t status;
/* Parse target & verify */
pj_strdup_with_null(dlg->pool, &tmp, target);
target_uri = pjsip_parse_uri(dlg->pool, tmp.ptr, tmp.slen, 0);
if (!target_uri) {
return PJSIP_EINVALIDURI;
}
if (!PJSIP_URI_SCHEME_IS_SIP(target_uri) &&
!PJSIP_URI_SCHEME_IS_SIPS(target_uri))
{
return PJSIP_EINVALIDSCHEME;
}
/* Add the new target */
status = pjsip_target_set_add_uri(&dlg->target_set, dlg->pool,
target_uri, 0);
if (status != PJ_SUCCESS)
return status;
/* Set it as current target */
status = pjsip_target_set_set_current(&dlg->target_set,
pjsip_target_set_get_next(&dlg->target_set));
if (status != PJ_SUCCESS)
return status;
/* Update dialog target URI */
dlg->target = target_uri;
return PJ_SUCCESS;
}
/* Get account contact for call and update dialog transport */
void call_update_contact(pjsua_call *call, pj_str_t **new_contact)
{
pjsip_tpselector tp_sel;
pjsua_acc *acc = &pjsua_var.acc[call->acc_id];
if (acc->cfg.force_contact.slen)
*new_contact = &acc->cfg.force_contact;
else if (acc->contact.slen)
*new_contact = &acc->contact;
else {
/* Non-registering account */
pjsip_dialog *dlg = call->inv->dlg;
pj_str_t tmp_contact;
pj_status_t status;
status = pjsua_acc_create_uac_contact(dlg->pool,
&tmp_contact,
acc->index,
&dlg->remote.info_str);
if (status == PJ_SUCCESS) {
*new_contact = PJ_POOL_ZALLOC_T(dlg->pool, pj_str_t);
**new_contact = tmp_contact;
} else {
PJ_PERROR(3,(THIS_FILE, status,
"Call %d: failed creating contact "
"for contact update", call->index));
}
}
/* When contact is changed, the account transport may have been
* changed too, so let's update the dialog's transport too.
*/
pjsua_init_tpselector(acc->cfg.transport_id, &tp_sel);
pjsip_dlg_set_transport(call->inv->dlg, &tp_sel);
}
/*
* 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,
const pjsua_call_setting *opt,
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 = NULL;
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();
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;
}
/* Clear call descriptor */
reset_call(call_id);
call = &pjsua_var.calls[call_id];
/* Associate session with account */
call->acc_id = acc_id;
call->call_hold_type = acc->cfg.call_hold_type;
/* Generate per-session RTCP CNAME, according to RFC 7022. */
pj_create_random_string(call->cname_buf, call->cname.slen);
/* Apply call setting */
status = apply_call_setting(call, opt, NULL);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Failed to apply call setting", status);
goto on_error;
}
/* 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 && call->opt.aud_cnt > 0)
{
status = pjsua_set_snd_dev(pjsua_var.cap_dev, pjsua_var.play_dev);
if (status != PJ_SUCCESS)
goto on_error;
}
/* 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,
(msg_data && msg_data->target_uri.slen?
&msg_data->target_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);
dlg_set_via(dlg, acc);
/* 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.
*/
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;
/* Temporarily increment dialog session. Without this, dialog will be
* prematurely destroyed if dec_lock() is called on the dialog before
* the invite session is created.
*/
pjsip_dlg_inc_session(dlg, &pjsua_var.mod);
if ((call->opt.flag & PJSUA_CALL_NO_SDP_OFFER) == 0) {
/* Init media channel */
status = pjsua_media_channel_init(call->index, PJSIP_ROLE_UAC,
call->secure_level, dlg->pool,
NULL, NULL, PJ_TRUE,
&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);
pjsip_dlg_dec_session(dlg, &pjsua_var.mod);
goto on_error;
}
/* Done. */
if (p_call_id)
*p_call_id = call_id;
pjsip_dlg_dec_lock(dlg);
pj_pool_release(tmp_pool);
PJSUA_UNLOCK();
pj_log_pop_indent();
return PJ_SUCCESS;
on_error:
if (dlg && call) {
/* This may destroy the dialog */
pjsip_dlg_dec_lock(dlg);
call->async_call.dlg = NULL;
}
if (call_id != -1) {
pjsua_media_channel_deinit(call_id);
reset_call(call_id);
}
pjsua_check_snd_dev_idle();
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)));
}
static pj_status_t process_incoming_call_replace(pjsua_call *call,
pjsip_dialog *replaced_dlg)
{
pjsip_inv_session *replaced_inv;
struct pjsua_call *replaced_call;
pjsip_tx_data *tdata = NULL;
pj_status_t status = PJ_SUCCESS;
/* 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 (!replaced_call->hanging_up && pjsua_var.ua_cfg.cb.on_call_replaced)
pjsua_var.ua_cfg.cb.on_call_replaced(replaced_call->index,
call->index);
if (replaced_call->inv->state <= PJSIP_INV_STATE_EARLY &&
replaced_call->inv->role != PJSIP_ROLE_UAC)
{
if (replaced_call->last_code > 100 && replaced_call->last_code < 200)
{
pjsip_status_code code = replaced_call->last_code;
pj_str_t *text = &replaced_call->last_text;
PJ_LOG(4,(THIS_FILE, "Answering replacement call %d with %d/%.*s",
call->index, code, text->slen, text->ptr));
/* Answer the new call with last response in the replaced call */
status = pjsip_inv_answer(call->inv, code, text, NULL, &tdata);
}
} else {
PJ_LOG(4,(THIS_FILE, "Answering replacement call %d with 200/OK",
call->index));
/* Answer the new call with 200 response */
status = pjsip_inv_answer(call->inv, 200, NULL, NULL, &tdata);
}
if (status == PJ_SUCCESS && tdata)
status = pjsip_inv_send_msg(call->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);
return status;
}
static void process_pending_call_answer(pjsua_call *call)
{
struct call_answer *answer, *next;
/* No initial answer yet, this function should be called again later */
if (!call->inv->last_answer)
return;
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_answer2(call->index, answer->opt, answer->code,
answer->reason, answer->msg_data);
/* Call might have been disconnected if application is answering
* with 200/OK and the media failed to start.
* See pjsua_call_answer() below.
*/
if (!call->inv || !call->inv->pool_prov)
break;
pj_list_erase(answer);
answer = next;
}
}
static pj_status_t process_pending_call_hangup(pjsua_call *call)
{
pjsip_dialog *dlg = NULL;
pj_status_t status;
PJ_LOG(4,(THIS_FILE, "Call %d processing pending hangup: code=%d..",
call->index, call->last_code));
pj_log_push_indent();
status = acquire_call("pending_hangup()", call->index, &call, &dlg);
if (status != PJ_SUCCESS) {
PJ_LOG(3, (THIS_FILE, "Call %d failed to process pending hangup",
call->index));
goto on_return;
}
pjsua_media_channel_deinit(call->index);
pjsua_check_snd_dev_idle();
if (call->inv)
call_inv_end_session(call, call->last_code, &call->last_text, NULL);
on_return:
if (dlg) pjsip_dlg_dec_lock(dlg);
pj_log_pop_indent();
return status;
}
pj_status_t create_temp_sdp(pj_pool_t *pool,
const pjmedia_sdp_session *rem_sdp,
pjmedia_sdp_session **p_sdp)
{
const pj_str_t STR_AUDIO = { "audio", 5 };
const pj_str_t STR_VIDEO = { "video", 5 };
const pj_str_t STR_IP6 = { "IP6", 3};
pjmedia_sdp_session *sdp;
pj_sockaddr origin;
pj_uint16_t tmp_port = 50123;
pj_status_t status = PJ_SUCCESS;
pj_str_t tmp_st;
unsigned i = 0;
pj_bool_t sess_use_ipv4 = PJ_TRUE;
/* Get one address to use in the origin field */
pj_sockaddr_init(PJ_AF_INET, &origin, pj_strset2(&tmp_st, "127.0.0.1"), 0);
/* Create the base (blank) SDP */
status = pjmedia_endpt_create_base_sdp(pjsua_var.med_endpt, pool, NULL,
&origin, &sdp);
if (status != PJ_SUCCESS)
return status;
if (rem_sdp->conn && pj_stricmp(&rem_sdp->conn->addr_type, &STR_IP6)==0) {
sess_use_ipv4 = PJ_FALSE;
}
for (; i< rem_sdp->media_count ; ++i) {
pjmedia_sdp_media *m = NULL;
pjmedia_sock_info sock_info;
pj_bool_t med_use_ipv4 = sess_use_ipv4;
if (rem_sdp->media[i]->conn &&
pj_stricmp(&rem_sdp->media[i]->conn->addr_type, &STR_IP6) == 0)
{
med_use_ipv4 = PJ_FALSE;
}
pj_sockaddr_init(med_use_ipv4?PJ_AF_INET:PJ_AF_INET6,
&sock_info.rtp_addr_name,
med_use_ipv4?pj_strset2(&tmp_st, "127.0.0.1"):
pj_strset2(&tmp_st, "::1"),
rem_sdp->media[i]->desc.port? (tmp_port++):0);
pj_sockaddr_init(med_use_ipv4?PJ_AF_INET:PJ_AF_INET6,
&sock_info.rtcp_addr_name,
med_use_ipv4?pj_strset2(&tmp_st, "127.0.0.1"):
pj_strset2(&tmp_st, "::1"),
rem_sdp->media[i]->desc.port? (tmp_port++):0);
if (pj_stricmp(&rem_sdp->media[i]->desc.media, &STR_AUDIO)==0) {
m = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_media);
status = pjmedia_endpt_create_audio_sdp(pjsua_var.med_endpt,
pool, &sock_info, 0, &m);
if (status != PJ_SUCCESS)
return status;
} else if (pj_stricmp(&rem_sdp->media[i]->desc.media, &STR_VIDEO)==0) {
#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
m = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_media);
status = pjmedia_endpt_create_video_sdp(pjsua_var.med_endpt, pool,
&sock_info, 0, &m);
if (status != PJ_SUCCESS)
return status;
#else
m = pjmedia_sdp_media_clone_deactivate(pool, rem_sdp->media[i]);
#endif
} else {
m = pjmedia_sdp_media_clone_deactivate(pool, rem_sdp->media[i]);
}
if (status != PJ_SUCCESS)
return status;
/* Add connection line, if none */
if (m->conn == NULL && sdp->conn == NULL) {
m->conn = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_conn);
m->conn->net_type = pj_str("IN");
if (med_use_ipv4) {
m->conn->addr_type = pj_str("IP4");
m->conn->addr = pj_str("127.0.0.1");
} else {
m->conn->addr_type = pj_str("IP6");
m->conn->addr = pj_str("::1");
}
}
/* Disable media if it has zero format/codec */
if (m->desc.fmt_count == 0) {
m->desc.fmt[m->desc.fmt_count++] = pj_str("0");
pjmedia_sdp_media_deactivate(pool, m);
}
sdp->media[sdp->media_count++] = m;
}
*p_sdp = sdp;
return PJ_SUCCESS;
}
static pj_status_t verify_request(const pjsua_call *call,
pjsip_rx_data *rdata,
pj_bool_t use_tmp_sdp,
int *sip_err_code,
pjsip_tx_data **response)
{
const pjmedia_sdp_session *offer = NULL;
pjmedia_sdp_session *answer;
int err_code = 0;
pj_status_t status;
/* Get remote SDP offer (if any). */
if (call->inv->neg)
{
pjmedia_sdp_neg_get_neg_remote(call->inv->neg, &offer);
}
if (use_tmp_sdp) {
if (offer == NULL)
return PJ_SUCCESS;
/* Create temporary SDP to check for codec support and capability
* to handle the required SIP extensions.
*/
status = create_temp_sdp(call->inv->pool_prov, offer, &answer);
if (status != PJ_SUCCESS) {
err_code = PJSIP_SC_INTERNAL_SERVER_ERROR;
pjsua_perror(THIS_FILE, "Error creating SDP answer", status);
}
} else {
status = pjsua_media_channel_create_sdp(call->index,
call->async_call.dlg->pool,
offer, &answer, sip_err_code);
if (status != PJ_SUCCESS) {
err_code = *sip_err_code;
pjsua_perror(THIS_FILE, "Error creating SDP answer", status);
} else {
status = pjsip_inv_set_local_sdp(call->inv, answer);
if (status != PJ_SUCCESS) {
err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE;
pjsua_perror(THIS_FILE, "Error setting local SDP", status);
}
}
}
if (status == PJ_SUCCESS) {
unsigned options = 0;
/* Verify that we can handle the request. */
status = pjsip_inv_verify_request3(rdata,
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.
*/
pjsua_perror(THIS_FILE, "Request verification failed", status);
if (response)
err_code = (*response)->msg->line.status.code;
else
err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE;
}
}
if (sip_err_code && status != PJ_SUCCESS)
*sip_err_code = err_code? err_code:PJSIP_ERRNO_TO_SIP_STATUS(status);
return status;
}
/* Incoming call callback when media transport creation is completed. */
static pj_status_t
on_incoming_call_med_tp_complete2(pjsua_call_id call_id,
const pjsua_med_tp_state_info *info,
pjsip_rx_data *rdata,
int *sip_err_code,
pjsip_tx_data **tdata)
{
pjsua_call *call = &pjsua_var.calls[call_id];
pjsip_dialog *dlg = call->async_call.dlg;
pj_status_t status = (info? info->status: PJ_SUCCESS);
int err_code = (info? info->sip_err_code: 0);
pjsip_tx_data *response = NULL;
PJSUA_LOCK();
/* Increment the dialog's lock to prevent it to be destroyed prematurely,
* such as in case of transport error.
*/
pjsip_dlg_inc_lock(dlg);
/* Decrement dialog session. */
pjsip_dlg_dec_session(dlg, &pjsua_var.mod);
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;
pjsip_dlg_dec_lock(dlg);
PJSUA_UNLOCK();
return PJ_SUCCESS;
}
status = verify_request(call, rdata, PJ_FALSE, &err_code, &response);
on_return:
if (status != PJ_SUCCESS) {
if (err_code == 0)
err_code = PJSIP_ERRNO_TO_SIP_STATUS(status);
if (sip_err_code)
*sip_err_code = err_code;
/* If the callback is called from pjsua_call_on_incoming(), the
* invite's state is PJSIP_INV_STATE_NULL, so the invite session
* will be terminated later, otherwise we end the session here.
*/
if (call->inv->state > PJSIP_INV_STATE_NULL) {
pj_status_t status_ = PJ_SUCCESS;
if (response == NULL) {
status_ = pjsip_inv_end_session(call->inv, err_code, NULL,
&response);
}
if (status_ == PJ_SUCCESS && response)
status_ = pjsip_inv_send_msg(call->inv, response);
}
pjsua_media_channel_deinit(call->index);
}
/* Set the callback to NULL to indicate that the async operation
* has completed.
*/
call->med_ch_cb = NULL;
/* Finish any pending process */
if (status == PJ_SUCCESS) {
if (call->async_call.call_var.inc_call.replaced_dlg) {
/* Process pending call replace */
pjsip_dialog *replaced_dlg =
call->async_call.call_var.inc_call.replaced_dlg;
process_incoming_call_replace(call, replaced_dlg);
} else {
/* Process pending call answers */
process_pending_call_answer(call);
}
}
pjsip_dlg_dec_lock(dlg);
if (tdata)
*tdata = response;
PJSUA_UNLOCK();
return status;
}
static pj_status_t
on_incoming_call_med_tp_complete(pjsua_call_id call_id,
const pjsua_med_tp_state_info *info)
{
return on_incoming_call_med_tp_complete2(call_id, info, NULL, NULL, NULL);
}
/**
* 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 = NULL;
int call_id = -1;
int sip_err_code = PJSIP_SC_INTERNAL_SERVER_ERROR;
pjmedia_sdp_session *offer=NULL;
pj_bool_t should_dec_dlg = PJ_FALSE;
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];
/* Generate per-session RTCP CNAME, according to RFC 7022. */
pj_create_random_string(call->cname_buf, call->cname.slen);
/* 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);
status = pjsip_endpt_send_response(pjsua_var.endpt, &res_addr, response,
NULL, NULL);
if (status != PJ_SUCCESS) pjsip_tx_data_dec_ref(response);
} 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_var.ua_cfg.cb.on_call_replace_request2))
{
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];
/* Copy call setting from the replaced call */
call->opt = replaced_call->opt;
pjsua_call_cleanup_flag(&call->opt);
/* Notify application */
if (!replaced_call->hanging_up &&
pjsua_var.ua_cfg.cb.on_call_replace_request)
{
pjsua_var.ua_cfg.cb.on_call_replace_request(replaced_call->index,
rdata,
&st_code, &st_text);
}
if (!replaced_call->hanging_up &&
pjsua_var.ua_cfg.cb.on_call_replace_request2)
{
pjsua_var.ua_cfg.cb.on_call_replace_request2(replaced_call->index,
rdata,
&st_code, &st_text,
&call->opt);
}
/* 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;
}
/* Set the user_data of the new call to the existing/parent call,
* it is needed by PJSUA2 to update its states. While PJSUA app can
* always override it anytime.
*/
pjsua_call_set_user_data(call_id, replaced_call->user_data);
}
if (!replaced_dlg) {
/* Clone rdata. */
pjsip_rx_data_clone(rdata, 0, &call->incoming_data);
}
/*
* 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.
*/
if (replaced_dlg) {
/* For call replace, use the same account as the replaced call */
pjsua_call *replaced_call;
replaced_call = (pjsua_call*)replaced_dlg->mod_data[pjsua_var.mod.id];
acc_id = call->acc_id = replaced_call->acc_id;
} else {
acc_id = call->acc_id = pjsua_acc_find_for_incoming(rdata);
if (acc_id == PJSUA_INVALID_ID) {
pjsip_endpt_respond_stateless(pjsua_var.endpt, rdata,
PJSIP_SC_TEMPORARILY_UNAVAILABLE,
NULL, NULL, NULL);
PJ_LOG(2,(THIS_FILE,
"Unable to accept incoming call (no available account)"));
goto on_return;
}
}
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 &&
!PJSIP_INV_ACCEPT_UNKNOWN_BODY)
{
if (sdp_info->body.ptr == NULL) {
status = PJSIP_ERRNO_FROM_SIP_STATUS(
PJSIP_SC_UNSUPPORTED_MEDIA_TYPE);
} else {
status = PJSIP_ERRNO_FROM_SIP_STATUS(PJSIP_SC_NOT_ACCEPTABLE);
}
}
if (status != PJ_SUCCESS) {
pjsip_hdr hdr_list;
/* Check if body really contains SDP. */
if (sdp_info->body.ptr == NULL) {
/* Couldn't find "application/sdp" */
pjsip_accept_hdr *acc;
pjsua_perror(THIS_FILE, "Unknown Content-Type in incoming "\
"INVITE", status);
/* Add Accept header to response */
acc = pjsip_accept_hdr_create(rdata->tp_info.pool);
PJ_ASSERT_RETURN(acc, PJ_ENOMEM);
acc->values[acc->count++] = pj_str("application/sdp");
pj_list_init(&hdr_list);
pj_list_push_back(&hdr_list, acc);
pjsip_endpt_respond(pjsua_var.endpt, NULL, rdata,
PJSIP_SC_UNSUPPORTED_MEDIA_TYPE,
NULL, &hdr_list, NULL, NULL);
} else {
const pj_str_t reason = pj_str("Bad SDP");
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) && (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 == PJSUA_100REL_MANDATORY)
options |= PJSIP_INV_REQUIRE_100REL;
if (pjsua_var.acc[acc_id].cfg.ice_cfg.enable_ice) {
options |= PJSIP_INV_SUPPORT_ICE;
if (pjsua_var.acc[acc_id].cfg.ice_cfg.ice_opt.trickle !=
PJ_ICE_SESS_TRICKLE_DISABLED)
{
options |= PJSIP_INV_SUPPORT_TRICKLE_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);
status = pjsip_endpt_send_response(pjsua_var.endpt, &res_addr, response,
NULL, NULL);
if (status != PJ_SUCCESS) pjsip_tx_data_dec_ref(response);
} 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_and_inc_lock( 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;
}
if (pjsua_var.acc[acc_id].cfg.allow_via_rewrite &&
pjsua_var.acc[acc_id].via_addr.host.slen > 0)
{
pjsip_dlg_set_via_sent_by(dlg, &pjsua_var.acc[acc_id].via_addr,
pjsua_var.acc[acc_id].via_tp);
} else if (!pjsua_sip_acc_is_using_stun(acc_id) &&
!pjsua_sip_acc_is_using_upnp(acc_id))
{
/* Choose local interface to use in Via if acc is not using
* STUN nor UPnP. See https://github.com/pjsip/pjproject/issues/1804
*/
char target_buf[PJSIP_MAX_URL_SIZE];
pj_str_t target;
pjsip_host_port via_addr;
const void *via_tp;
target.ptr = target_buf;
target.slen = pjsip_uri_print(PJSIP_URI_IN_REQ_URI,
dlg->target,
target_buf, sizeof(target_buf));
if (target.slen < 0) target.slen = 0;
if (pjsua_acc_get_uac_addr(acc_id, dlg->pool, &target,
&via_addr, NULL, NULL,
&via_tp) == PJ_SUCCESS)
{
pjsip_dlg_set_via_sent_by(dlg, &via_addr,
(pjsip_transport*)via_tp);
}
}
/* 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);
}
/* If 100rel is optional and UAC supports it, use it. */
if ((options & PJSIP_INV_REQUIRE_100REL)==0 &&
pjsua_var.acc[acc_id].cfg.require_100rel == PJSUA_100REL_OPTIONAL)
{
const pj_str_t token = { "100rel", 6};
pjsip_dialog_cap_status cap_status;
cap_status = pjsip_dlg_remote_has_cap(dlg, PJSIP_H_SUPPORTED, NULL,
&token);
if (cap_status == PJSIP_DIALOG_CAP_SUPPORTED)
options |= PJSIP_INV_REQUIRE_100REL;
}
/* 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);
*/
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;
/* 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);
pjsip_dlg_inc_session(dlg, &pjsua_var.mod);
should_dec_dlg = PJ_TRUE;
/* Init media channel, only when there is offer or call replace request.
* For incoming call without SDP offer, media channel init will be done
* in pjsua_call_answer(), see ticket #1526.
*/
if (offer || replaced_dlg) {
/* This is only for initial verification, it will check the SDP for
* codec support and the capability to handle the required
* SIP extensions.
*/
status = verify_request(call, rdata, PJ_TRUE, &sip_err_code,
&response);
if (status != PJ_SUCCESS) {
pjsip_dlg_inc_lock(dlg);
if (response) {
pjsip_dlg_send_response(dlg, call->inv->invite_tsx, response);
} else {
pjsip_dlg_respond(dlg, rdata, sip_err_code, NULL, NULL, NULL);
}
if (call->inv && call->inv->dlg) {
pjsip_inv_terminate(call->inv, sip_err_code, PJ_FALSE);
}
pjsip_dlg_dec_lock(dlg);
call->inv = NULL;
call->async_call.dlg = NULL;
goto on_return;
}
status = pjsua_media_channel_init(call->index, PJSIP_ROLE_UAS,
call->secure_level,
rdata->tp_info.pool,
offer,
&sip_err_code, PJ_TRUE,
&on_incoming_call_med_tp_complete);
if (status == PJ_EPENDING) {
/* on_incoming_call_med_tp_complete() will call
* pjsip_dlg_dec_session().
*/
should_dec_dlg = PJ_FALSE;
} else if (status == PJ_SUCCESS) {
/* on_incoming_call_med_tp_complete2() will call
* pjsip_dlg_dec_session().
*/
should_dec_dlg = PJ_FALSE;
status = on_incoming_call_med_tp_complete2(call_id, NULL,
rdata, &sip_err_code,
&response);
if (status != PJ_SUCCESS) {
/* Since the call invite's state is still PJSIP_INV_STATE_NULL,
* the invite session was not ended in
* on_incoming_call_med_tp_complete(), so we need to send
* a response message and terminate the invite here.
*/
pjsip_dlg_inc_lock(dlg);
if (response) {
pjsip_dlg_send_response(dlg, call->inv->invite_tsx,
response);
} else {
pjsip_dlg_respond(dlg, rdata, sip_err_code, NULL, NULL,
NULL);
}
if (call->inv && call->inv->dlg) {
pjsip_inv_terminate(call->inv, sip_err_code, PJ_FALSE);
}
pjsip_dlg_dec_lock(dlg);
call->inv = NULL;
call->async_call.dlg = NULL;
goto on_return;
}
} else if (status != PJ_EPENDING) {
pjsua_perror(THIS_FILE, "Error initializing media channel", status);
pjsip_dlg_inc_lock(dlg);
pjsip_dlg_respond(dlg, rdata, sip_err_code, NULL, NULL, NULL);
if (call->inv && call->inv->dlg) {
pjsip_inv_terminate(call->inv, sip_err_code, PJ_FALSE);
}
pjsip_dlg_dec_lock(dlg);
call->inv = NULL;
call->async_call.dlg = 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);
pjsip_dlg_respond(dlg, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL, NULL);
pjsip_inv_terminate(inv, PJSIP_SC_INTERNAL_SERVER_ERROR, PJ_FALSE);
pjsua_media_channel_deinit(call->index);
call->inv = NULL;
call->async_call.dlg = NULL;
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);
call->inv = NULL;
call->async_call.dlg = NULL;
goto on_return;
} else {
#if !PJSUA_DISABLE_AUTO_SEND_100
status = pjsip_inv_send_msg(inv, response);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to send 100 response", status);
pjsua_media_channel_deinit(call->index);
call->inv = NULL;
call->async_call.dlg = NULL;
goto on_return;
}
#endif
}
/* Only do this after sending 100/Trying (really! see the long comment
* above)
*/
if (dlg->mod_data[pjsua_var.mod.id] == NULL) {
/* In PJSUA2, on_incoming_call() may be called from
* on_media_transport_created() hence this might already set
* to allow notification about fail events via on_call_state() and
* on_call_tsx_state().
*/
dlg->mod_data[pjsua_var.mod.id] = call;
inv->mod_data[pjsua_var.mod.id] = call;
++pjsua_var.call_cnt;
}
/* Check if this request should replace existing call */
if (replaced_dlg) {
/* Process call replace. If the media channel init has been completed,
* just process now, otherwise, just queue the replaced dialog so
* it will be processed once the media channel async init is finished
* successfully.
*/
if (call->med_ch_cb == NULL) {
process_incoming_call_replace(call, replaced_dlg);
} else {
call->async_call.call_var.inc_call.replaced_dlg = replaced_dlg;
}
} 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);
/* Notes:
* - the call might be reset when it's rejected or hangup
* by application from the callback.
* - onIncomingCall() may be simulated by onCreateMediaTransport()
* when media init is done synchrounously (see #1916). And if app
* happens to answer/hangup the call from the callback, the
* answer/hangup should have been delayed (see #1923),
* so let's process the answer/hangup now.
*/
if (call->async_call.call_var.inc_call.hangup) {
process_pending_call_hangup(call);
} else if (call->med_ch_cb == NULL && call->inv) {
process_pending_call_answer(call);
}
} else {
pjsua_call_hangup(call_id, PJSIP_SC_TEMPORARILY_UNAVAILABLE,
NULL, NULL);
}
}
/* This INVITE request has been handled. */
on_return:
if (dlg) {
if (should_dec_dlg)
pjsip_dlg_dec_session(dlg, &pjsua_var.mod);
pjsip_dlg_dec_lock(dlg);
}
if (call && call->incoming_data) {
pjsip_rx_data_free_cloned(call->incoming_data);
call->incoming_data = NULL;
}
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].hanging_up &&
pjsua_var.calls[call_id].inv != NULL &&
pjsua_var.calls[call_id].inv->state != PJSIP_INV_STATE_DISCONNECTED;
}
/* 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;
pjsip_dialog *dlg = NULL;
pj_gettimeofday(&time_start);
timeout.sec = 0;
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)
dlg = call->inv->dlg;
else
dlg = call->async_call.dlg;
if (dlg == 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(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 = dlg;
return PJ_SUCCESS;
}
/*
* 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_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
pj_bzero(info, sizeof(*info));
/* Use PJSUA_LOCK() instead of acquire_call():
* https://github.com/pjsip/pjproject/issues/1371
*/
PJSUA_LOCK();
call = &pjsua_var.calls[call_id];
dlg = (call->inv ? call->inv->dlg : call->async_call.dlg);
if (!dlg) {
PJSUA_UNLOCK();
return PJSIP_ESESSIONTERMINATED;
}
/* id and role */
info->id = call_id;
info->role = dlg->role;
info->acc_id = call->acc_id;
/* local info */
info->local_info.ptr = info->buf_.local_info;
pj_strncpy(&info->local_info, &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,
dlg->local.contact->uri,
info->local_contact.ptr,
sizeof(info->buf_.local_contact));
if (info->local_contact.slen < 0)
info->local_contact.slen = 0;
/* remote info */
info->remote_info.ptr = info->buf_.remote_info;
pj_strncpy(&info->remote_info, &dlg->remote.info_str,
sizeof(info->buf_.remote_info));
/* remote contact */
if (dlg->remote.contact) {
int len;
info->remote_contact.ptr = info->buf_.remote_contact;
len = pjsip_uri_print(PJSIP_URI_IN_CONTACT_HDR,
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, &dlg->call_id->id,
sizeof(info->buf_.call_id));
/* call setting */
pj_memcpy(&info->setting, &call->opt, sizeof(call->opt));
/* state, state_text */
if (call->hanging_up) {
info->state = PJSIP_INV_STATE_DISCONNECTED;
} else if (call->inv) {
info->state = call->inv->state;
if (call->inv->role == PJSIP_ROLE_UAS &&
info->state == PJSIP_INV_STATE_NULL)
{
info->state = PJSIP_INV_STATE_INCOMING;
}
} else if (call->async_call.dlg && call->last_code==0) {
info->state = PJSIP_INV_STATE_NULL;
} else {
info->state = PJSIP_INV_STATE_DISCONNECTED;
}
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 && 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));
}
/* Audio & video count offered by remote */
info->rem_offerer = call->rem_offerer;
if (call->rem_offerer) {
info->rem_aud_cnt = call->rem_aud_cnt;
info->rem_vid_cnt = call->rem_vid_cnt;
}
/* Build array of active media info */
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;
info->media[info->media_cnt].stream.vid.dec_slot =
call_med->strm.v.strm_dec_slot;
info->media[info->media_cnt].stream.vid.enc_slot =
call_med->strm.v.strm_enc_slot;
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;
}
/* Build array of provisional media info */
info->prov_media_cnt = 0;
for (mi=0; mi < call->med_prov_cnt &&
info->prov_media_cnt < PJ_ARRAY_SIZE(info->prov_media); ++mi)
{
pjsua_call_media *call_med = &call->media_prov[mi];
info->prov_media[info->prov_media_cnt].index = mi;
info->prov_media[info->prov_media_cnt].status = call_med->state;
info->prov_media[info->prov_media_cnt].dir = call_med->dir;
info->prov_media[info->prov_media_cnt].type = call_med->type;
if (call_med->type == PJMEDIA_TYPE_AUDIO) {
info->prov_media[info->prov_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->prov_media[info->prov_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->prov_media[info->prov_media_cnt].stream.vid.cap_dev=cap_dev;
} else {
continue;
}
++info->prov_media_cnt;
}
/* 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);
}
PJSUA_UNLOCK();
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 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;
}
/* Media channel init callback for pjsua_call_answer(). */
static pj_status_t
on_answer_call_med_tp_complete(pjsua_call_id call_id,
const pjsua_med_tp_state_info *info)
{
pjsua_call *call = &pjsua_var.calls[call_id];
pjmedia_sdp_session *sdp;
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;
}
status = pjsua_media_channel_create_sdp(call_id,
call->async_call.dlg->pool,
NULL, &sdp, &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, sdp);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Error setting local SDP", status);
sip_err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE;
goto on_return;
}
on_return:
if (status != PJ_SUCCESS) {
/* If the callback is called from pjsua_call_on_incoming(), the
* invite's state is PJSIP_INV_STATE_NULL, so the invite session
* will be terminated later, otherwise we end the session here.
*/
if (call->inv->state > PJSIP_INV_STATE_NULL) {
pjsip_tx_data *tdata;
pj_status_t status_;
if (sip_err_code == 0)
sip_err_code = PJSIP_ERRNO_TO_SIP_STATUS(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;
/* Finish any pending process */
if (status == PJ_SUCCESS) {
/* Process pending call answers */
process_pending_call_answer(call);
}
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)
{
return pjsua_call_answer2(call_id, NULL, code, reason, msg_data);
}
/*
* Send response to incoming INVITE request.
*/
PJ_DEF(pj_status_t) pjsua_call_answer2(pjsua_call_id call_id,
const pjsua_call_setting *opt,
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;
if (!call->inv->invite_tsx ||
call->inv->invite_tsx->role != PJSIP_ROLE_UAC ||
call->inv->invite_tsx->state >= PJSIP_TSX_STATE_COMPLETED)
{
PJ_LOG(3,(THIS_FILE, "Unable to answer call (no incoming INVITE or "
"already answered)"));
status = PJ_EINVALIDOP;
goto on_return;
}
/* Apply call setting, only if status code is 1xx or 2xx. */
if (opt && code < 300) {
/* Check if it has not been set previously or it is different to
* the previous one.
*/
if (!call->opt_inited) {
call->opt_inited = PJ_TRUE;
apply_call_setting(call, opt, NULL);
} else if (pj_memcmp(opt, &call->opt, sizeof(*opt)) != 0) {
/* Warn application about call setting inconsistency */
PJ_LOG(2,(THIS_FILE, "The call setting changes is ignored."));
}
}
PJSUA_LOCK();
/* Ticket #1526: When the incoming call contains no SDP offer, the media
* channel may have not been initialized at this stage. The media channel
* will be initialized here (along with SDP local offer generation) when
* the following conditions are met:
* - no pending media channel init
* - local SDP has not been generated
* - call setting has just been set, or SDP offer needs to be sent, i.e:
* answer code 183 or 2xx is issued
*/
if (!call->med_ch_cb &&
(call->opt_inited || (code==183 || code/100==2)) &&
(!call->inv->neg ||
pjmedia_sdp_neg_get_state(call->inv->neg) ==
PJMEDIA_SDP_NEG_STATE_NULL))
{
/* Mark call setting as initialized as it is just about to be used
* for initializing the media channel.
*/
call->opt_inited = PJ_TRUE;
status = pjsua_media_channel_init(call->index, PJSIP_ROLE_UAC,
call->secure_level,
dlg->pool,
NULL, NULL, PJ_TRUE,
&on_answer_call_med_tp_complete);
if (status == PJ_SUCCESS) {
status = on_answer_call_med_tp_complete(call->index, NULL);
if (status != PJ_SUCCESS) {
PJSUA_UNLOCK();
goto on_return;
}
} else if (status != PJ_EPENDING) {
PJSUA_UNLOCK();
pjsua_perror(THIS_FILE, "Error initializing media channel", status);
goto on_return;
}
}
/* If media transport creation is not yet completed, we will answer
* the call in the media transport creation callback instead.
* Or if initial answer is not sent yet, we will answer the call after
* initial answer is sent (see #1923).
*/
if (call->med_ch_cb || !call->inv->last_answer) {
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 (opt) {
answer->opt = PJ_POOL_ZALLOC_T(call->inv->pool_prov,
pjsua_call_setting);
*answer->opt = *opt;
}
if (reason) {
answer->reason = PJ_POOL_ZALLOC_T(call->inv->pool_prov, pj_str_t);
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;
}
/*
* Send response to incoming INVITE request.
*/
PJ_DEF(pj_status_t)
pjsua_call_answer_with_sdp(pjsua_call_id call_id,
const pjmedia_sdp_session *sdp,
const pjsua_call_setting *opt,
unsigned code,
const pj_str_t *reason,
const pjsua_msg_data *msg_data)
{
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);
status = acquire_call("pjsua_call_answer_with_sdp()",
call_id, &call, &dlg);
if (status != PJ_SUCCESS)
return status;
status = pjsip_inv_set_sdp_answer(call->inv, sdp);
pjsip_dlg_dec_lock(dlg);
if (status != PJ_SUCCESS)
return status;
return pjsua_call_answer2(call_id, opt, code, reason, msg_data);
}
static pj_status_t call_inv_end_session(pjsua_call *call,
unsigned code,
const pj_str_t *reason,
const pjsua_msg_data *msg_data)
{
pjsip_tx_data *tdata;
pj_status_t status;
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;
}
/* Stop hangup timer, if it is active. */
if (call->hangup_timer.id) {
pjsua_cancel_timer(&call->hangup_timer);
call->hangup_timer.id = PJ_FALSE;
}
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;
}
on_return:
/* Failure in pjsip_inv_send_msg() can cause
* pjsua_call_on_state_changed() to be called and call to be reset,
* so we need to check for call->inv as well.
*/
if (status != PJ_SUCCESS && call->inv) {
pj_time_val delay;
/* Schedule a retry */
if (call->hangup_retry >= CALL_HANGUP_MAX_RETRY) {
/* Forcefully terminate the invite session. */
PJ_LOG(1,(THIS_FILE,"Call %d: failed to hangup after %d retries, "
"terminating the session forcefully now!",
call->index, call->hangup_retry));
pjsip_inv_terminate(call->inv, call->hangup_code, PJ_TRUE);
return PJ_SUCCESS;
}
if (call->hangup_retry == 0) {
pj_timer_entry_init(&call->hangup_timer, PJ_FALSE,
(void*)call, &hangup_timer_cb);
call->hangup_code = code;
if (reason) {
pj_strdup(call->inv->pool_prov, &call->hangup_reason,
reason);
}
if (msg_data) {
call->hangup_msg_data = pjsua_msg_data_clone(
call->inv->pool_prov,
msg_data);
}
}
delay.sec = 0;
delay.msec = CALL_HANGUP_RETRY_INTERVAL;
pj_time_val_normalize(&delay);
call->hangup_timer.id = PJ_TRUE;
pjsua_schedule_timer(&call->hangup_timer, &delay);
call->hangup_retry++;
PJ_LOG(4, (THIS_FILE, "Will retry call %d hangup in %d msec",
call->index, CALL_HANGUP_RETRY_INTERVAL));
}
return PJ_SUCCESS;
}
/* Timer callback to hangup call */
static void hangup_timer_cb(pj_timer_heap_t *th, pj_timer_entry *entry)
{
pjsua_call* call = (pjsua_call *)entry->user_data;
pjsip_dialog *dlg;
pj_status_t status;
PJ_UNUSED_ARG(th);
pj_log_push_indent();
status = acquire_call("hangup_timer_cb()", call->index, &call, &dlg);
if (status != PJ_SUCCESS) {
pj_log_pop_indent();
return;
}
call->hangup_timer.id = PJ_FALSE;
call_inv_end_session(call, call->hangup_code, &call->hangup_reason,
call->hangup_msg_data);
pjsip_dlg_dec_lock(dlg);
pj_log_pop_indent();
}
/*
* 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;
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 (!call->hanging_up) {
pj_bool_t delay_hangup = PJ_FALSE;
pjsip_event user_event;
pj_gettimeofday(&call->dis_time);
if (call->res_time.sec == 0)
pj_gettimeofday(&call->res_time);
if (code==0) {
if (call->inv && call->inv->state == PJSIP_INV_STATE_CONFIRMED)
code = PJSIP_SC_OK;
else if (call->inv && call->inv->role == PJSIP_ROLE_UAS)
code = PJSIP_SC_DECLINE;
else
code = PJSIP_SC_REQUEST_TERMINATED;
}
call->last_code = code;
pj_strncpy(&call->last_text,
pjsip_get_status_text(call->last_code),
sizeof(call->last_text_buf_));
/* Stop reinvite timer, if it is active. */
if (call->reinv_timer.id) {
pjsua_cancel_timer(&call->reinv_timer);
call->reinv_timer.id = PJ_FALSE;
}
/* If media transport creation is not yet completed, we will continue
* from the media transport creation callback instead.
*/
if ((call->med_ch_cb && !call->inv) ||
((call->inv != NULL) &&
(call->inv->state == PJSIP_INV_STATE_NULL)))
{
delay_hangup = PJ_TRUE;
PJ_LOG(4,(THIS_FILE, "Will continue call %d hangup upon "
"completion of media transport", call_id));
if (call->inv && call->inv->role == PJSIP_ROLE_UAS)
call->async_call.call_var.inc_call.hangup = PJ_TRUE;
else
call->async_call.call_var.out_call.hangup = PJ_TRUE;
if (reason) {
pj_strncpy(&call->last_text, reason,
sizeof(call->last_text_buf_));
}
call->hanging_up = PJ_TRUE;
} else {
/* Destroy media session. */
pjsua_media_channel_deinit(call_id);
call->hanging_up = PJ_TRUE;
pjsua_check_snd_dev_idle();
}
/* Call callback which will report DISCONNECTED state.
* Use user event rather than NULL to avoid crash in
* unsuspecting app.
*/
PJSIP_EVENT_INIT_USER(user_event, 0, 0, 0, 0);
if (pjsua_var.ua_cfg.cb.on_call_state) {
(*pjsua_var.ua_cfg.cb.on_call_state)(call->index,
&user_event);
}
if (call->inv && !delay_hangup) {
call_inv_end_session(call, code, reason, msg_data);
}
} else {
/* Already requested and on progress */
PJ_LOG(4,(THIS_FILE, "Call %d hangup request ignored as "
"it is on progress", call_id));
}
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)
{
return pjsua_call_set_hold2(call_id, 0, msg_data);
}
PJ_DEF(pj_status_t) pjsua_call_set_hold2(pjsua_call_id call_id,
unsigned options,
const pjsua_msg_data *msg_data)
{
pjmedia_sdp_session *sdp;
pjsua_call *call;
pjsip_dialog *dlg = NULL;
pjsip_tx_data *tdata;
pj_str_t *new_contact = 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, "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;
}
/* We may need to re-initialize media before creating SDP */
if (call->med_prov_cnt == 0) {
status = apply_call_setting(call, &call->opt, NULL);
if (status != PJ_SUCCESS)
goto on_return;
}
status = create_sdp_of_call_hold(call, &sdp);
if (status != PJ_SUCCESS)
goto on_return;
if ((options & PJSUA_CALL_UPDATE_CONTACT) &&
pjsua_acc_is_valid(call->acc_id))
{
call_update_contact(call, &new_contact);
}
if ((options & PJSUA_CALL_UPDATE_VIA) &&
pjsua_acc_is_valid(call->acc_id))
{
dlg_set_via(call->inv->dlg, &pjsua_var.acc[call->acc_id]);
}
if ((call->opt.flag & PJSUA_CALL_UPDATE_TARGET) &&
msg_data && msg_data->target_uri.slen)
{
status = dlg_set_target(dlg, &msg_data->target_uri);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to set new target", status);
goto on_return;
}
}
/* 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);
/* Record the tx_data to keep track the operation */
call->hold_msg = (void*) tdata;
/* 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);
call->hold_msg = NULL;
goto on_return;
}
/* Set flag that local put the call on hold */
call->local_hold = PJ_TRUE;
/* Clear unhold flag */
call->opt.flag &= ~PJSUA_CALL_UNHOLD;
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)
{
pjsua_call *call;
pjsip_dialog *dlg = NULL;
pj_status_t status;
status = acquire_call("pjsua_call_reinvite()", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
goto on_return;
if (options != call->opt.flag)
call->opt.flag = options;
status = pjsua_call_reinvite2(call_id, &call->opt, msg_data);
on_return:
if (dlg) pjsip_dlg_dec_lock(dlg);
return status;
}
/*
* Send re-INVITE (to release hold).
*/
PJ_DEF(pj_status_t) pjsua_call_reinvite2(pjsua_call_id call_id,
const pjsua_call_setting *opt,
const pjsua_msg_data *msg_data)
{
pjmedia_sdp_session *sdp = NULL;
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_reinvite2()", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
goto on_return;
if (pjsua_call_media_is_changing(call)) {
PJ_LOG(1,(THIS_FILE, "Unable to reinvite" ERR_MEDIA_CHANGING));
status = PJ_EINVALIDOP;
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;
}
status = apply_call_setting(call, opt, NULL);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Failed to apply call setting", status);
goto on_return;
}
/* Create SDP */
if (call->local_hold && (call->opt.flag & PJSUA_CALL_UNHOLD)==0) {
status = create_sdp_of_call_hold(call, &sdp);
} else if ((call->opt.flag & PJSUA_CALL_NO_SDP_OFFER) == 0) {
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 ((call->opt.flag & PJSUA_CALL_UPDATE_CONTACT) &&
pjsua_acc_is_valid(call->acc_id))
{
call_update_contact(call, &new_contact);
}
if ((call->opt.flag & PJSUA_CALL_UPDATE_VIA) &&
pjsua_acc_is_valid(call->acc_id))
{
dlg_set_via(call->inv->dlg, &pjsua_var.acc[call->acc_id]);
}
if ((call->opt.flag & PJSUA_CALL_UPDATE_TARGET) &&
msg_data && msg_data->target_uri.slen)
{
status = dlg_set_target(dlg, &msg_data->target_uri);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to set new target", status);
goto on_return;
}
}
/* 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 */
call->med_update_success = PJ_FALSE;
status = pjsip_inv_send_msg( call->inv, tdata);
if (status == PJ_SUCCESS &&
((call->opt.flag & PJSUA_CALL_UNHOLD) &&
(call->opt.flag & PJSUA_CALL_NO_SDP_OFFER) == 0))
{
call->local_hold = PJ_FALSE;
} else 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)
{
pjsua_call *call;
pjsip_dialog *dlg = NULL;
pj_status_t status;
status = acquire_call("pjsua_call_update()", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
goto on_return;
if (options != call->opt.flag)
call->opt.flag = options;
status = pjsua_call_update2(call_id, &call->opt, msg_data);
on_return:
if (dlg) pjsip_dlg_dec_lock(dlg);
return status;
}
/*
* Send UPDATE request.
*/
PJ_DEF(pj_status_t) pjsua_call_update2(pjsua_call_id call_id,
const pjsua_call_setting *opt,
const pjsua_msg_data *msg_data)
{
pjmedia_sdp_session *sdp = NULL;
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 UPDATE on call %d", call_id));
pj_log_push_indent();
status = acquire_call("pjsua_call_update2()", call_id, &call, &dlg);
if (status != PJ_SUCCESS)
goto on_return;
/* Don't check media changing if UPDATE is sent without SDP */
if (pjsua_call_media_is_changing(call) &&
(opt && opt->flag & PJSUA_CALL_NO_SDP_OFFER) == 0)
{
PJ_LOG(1,(THIS_FILE, "Unable to send UPDATE" ERR_MEDIA_CHANGING));
status = PJ_EINVALIDOP;
goto on_return;
}
status = apply_call_setting(call, opt, NULL);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Failed to apply call setting", status);
goto on_return;
}
/* Create SDP */
if (call->local_hold && (call->opt.flag & PJSUA_CALL_UNHOLD)==0) {
status = create_sdp_of_call_hold(call, &sdp);
} else if ((call->opt.flag & PJSUA_CALL_NO_SDP_OFFER) == 0) {
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 ((call->opt.flag & PJSUA_CALL_UPDATE_CONTACT) &&
pjsua_acc_is_valid(call->acc_id))
{
call_update_contact(call, &new_contact);
}
if ((call->opt.flag & PJSUA_CALL_UPDATE_VIA) &&
pjsua_acc_is_valid(call->acc_id))
{
dlg_set_via(call->inv->dlg, &pjsua_var.acc[call->acc_id]);
}
if ((call->opt.flag & PJSUA_CALL_UPDATE_TARGET) &&
msg_data && msg_data->target_uri.slen)
{
status = dlg_set_target(dlg, &msg_data->target_uri);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to set new target", status);
goto on_return;
}
}
/* 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 */
call->med_update_success = PJ_FALSE;
status = pjsip_inv_send_msg( call->inv, tdata);
if (status == PJ_SUCCESS &&
((call->opt.flag & PJSUA_CALL_UNHOLD) &&
(call->opt.flag & PJSUA_CALL_NO_SDP_OFFER) == 0))
{
call->local_hold = PJ_FALSE;
} else if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to send UPDATE request", status);
goto on_return;
}
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, "Transferring 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;
char call_id_dest_buf[PJSIP_MAX_URL_SIZE * 2];
int call_id_len;
pjsip_uri *uri;
pj_status_t status;
const pjsip_parser_const_t *pconst;
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, "Transferring 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;
/* This uses the the same scanner definition used for SIP parsing
* to escape the call-id in the refer.
*
* A common pattern for call-ids is: name@domain. The '@' character,
* when used in a URL parameter, throws off many SIP parsers.
* URL escape it based off of the allowed characters for header values.
*/
pconst = pjsip_parser_const();
call_id_len = (int)pj_strncpy2_escape(call_id_dest_buf, &dest_dlg->call_id->id,
PJ_ARRAY_SIZE(call_id_dest_buf),
&pconst->pjsip_HDR_CHAR_SPEC);
if (call_id_len < 0) {
status = PJSIP_EURITOOLONG;
goto on_error;
}
/* 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&"),
call_id_len,
call_id_dest_buf,
(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.
*/
PJ_DEF(pj_status_t) pjsua_call_send_dtmf(pjsua_call_id call_id,
const pjsua_call_send_dtmf_param *param)
{
pj_status_t status = PJ_EINVAL;
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls &&
param, PJ_EINVAL);
PJ_LOG(4,(THIS_FILE, "Call %d sending DTMF %.*s using %s method",
call_id, (int)param->digits.slen, param->digits.ptr,
get_dtmf_method_name(param->method)));
if (param->method == PJSUA_DTMF_METHOD_RFC2833) {
status = pjsua_call_dial_dtmf(call_id, &param->digits);
} else if (param->method == PJSUA_DTMF_METHOD_SIP_INFO) {
const pj_str_t SIP_INFO = pj_str("INFO");
int i;
for (i = 0; i < param->digits.slen; ++i) {
char body[80];
pjsua_msg_data msg_data_;
pjsua_msg_data_init(&msg_data_);
msg_data_.content_type = pj_str("application/dtmf-relay");
pj_ansi_snprintf(body, sizeof(body),
"Signal=%c\r\n"
"Duration=%d",
param->digits.ptr[i], param->duration);
msg_data_.msg_body = pj_str(body);
status = pjsua_call_send_request(call_id, &SIP_INFO, &msg_data_);
}
}
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_bool_t content_in_msg_data;
pj_status_t status;
content_in_msg_data = msg_data && (msg_data->msg_body.slen ||
msg_data->multipart_ctype.type.slen);
PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
PJ_EINVAL);
/* Message body must be specified. */
PJ_ASSERT_RETURN(content || content_in_msg_data, PJ_EINVAL);
if (content) {
PJ_LOG(4,(THIS_FILE, "Call %d sending %d bytes MESSAGE..",
call_id, (int)content->slen));
} else {
PJ_LOG(4,(THIS_FILE, "Call %d sending MESSAGE..",
call_id));
}
pj_log_push_indent();
status = acquire_call("pjsua_call_send_im()", 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;
}
/* Add accept header. */
pjsip_msg_add_hdr( tdata->msg,
(pjsip_hdr*)pjsua_im_create_accept(tdata->pool));
/* Add message body, if content is set */
if (content) {
/* Set default media type if none is specified */
if (mime_type == NULL) {
mime_type = &mime_text_plain;
}
/* 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;
if (content)
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();
// This may deadlock, see https://github.com/pjsip/pjproject/issues/1305
//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();
}
/* Timer callback to send re-INVITE/UPDATE to lock codec or ICE update */
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].reinv_timer.id = PJ_FALSE;
pj_log_push_indent();
status = acquire_call("reinv_timer_cb()", call_id, &call, &dlg);
if (status != PJ_SUCCESS) {
pj_log_pop_indent();
return;
}
process_pending_reinvite(call);
pjsip_dlg_dec_lock(dlg);
pj_log_pop_indent();
}
/* 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;
}
/* Schedule check for the need of re-INVITE/UPDATE after media update, cases:
* - lock codec if remote answerer has given us more than one codecs
* - update ICE default transport address if it has changed after ICE
* connectivity check.
*/
void pjsua_call_schedule_reinvite_check(pjsua_call *call, unsigned delay_ms)
{
pj_time_val delay;
/* Stop reinvite timer, if it is active */
if (call->reinv_timer.id)
pjsua_cancel_timer(&call->reinv_timer);
delay.sec = 0;
delay.msec = delay_ms;
pj_time_val_normalize(&delay);
call->reinv_timer.id = PJ_TRUE;
pjsua_schedule_timer(&call->reinv_timer, &delay);
}
/* Check if lock codec is needed */
static pj_bool_t check_lock_codec(pjsua_call *call)
{
const pjmedia_sdp_session *local_sdp, *remote_sdp;
pj_bool_t has_mult_fmt = PJ_FALSE;
unsigned i;
pj_status_t status;
/* Check if lock codec is disabled */
if (!pjsua_var.acc[call->acc_id].cfg.lock_codec)
return PJ_FALSE;
/* Check lock codec retry count */
if (call->lock_codec.retry_cnt >= LOCK_CODEC_MAX_RETRY)
return PJ_FALSE;
/* Check if we are the answerer, we shouldn't need to lock codec */
if (!call->inv->neg || !pjmedia_sdp_neg_was_answer_remote(call->inv->neg))
return PJ_FALSE;
/* Check if remote answerer has given us more than one codecs. */
status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &local_sdp);
if (status != PJ_SUCCESS)
return PJ_FALSE;
status = pjmedia_sdp_neg_get_active_remote(call->inv->neg, &remote_sdp);
if (status != PJ_SUCCESS)
return PJ_FALSE;
for (i = 0; i < call->med_cnt && !has_mult_fmt; ++i) {
pjsua_call_media *call_med = &call->media[i];
const pjmedia_sdp_media *rem_m, *loc_m;
unsigned codec_cnt = 0;
unsigned j;
/* 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);
PJ_UNUSED_ARG(loc_m);
/* Count the formats in the answer. */
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 > 1)
has_mult_fmt = PJ_TRUE;
}
}
/* Reset retry count when remote answer has one codec */
if (!has_mult_fmt)
call->lock_codec.retry_cnt = 0;
return has_mult_fmt;
}
/* Check if ICE setup is complete and if it needs to send reinvite */
static pj_bool_t check_ice_complete(pjsua_call *call, pj_bool_t *need_reinv)
{
pj_bool_t ice_need_reinv = PJ_FALSE;
pj_bool_t ice_complete = PJ_TRUE;
unsigned i;
/* Check if ICE setup is complete and if it needs reinvite */
for (i = 0; i < call->med_cnt; ++i) {
pjsua_call_media *call_med = &call->media[i];
pjmedia_transport_info tpinfo;
pjmedia_ice_transport_info *ice_info;
if (call_med->tp_st == PJSUA_MED_TP_NULL ||
call_med->tp_st == PJSUA_MED_TP_DISABLED ||
call_med->state == PJSUA_CALL_MEDIA_ERROR)
{
continue;
}
pjmedia_transport_info_init(&tpinfo);
pjmedia_transport_get_info(call_med->tp, &tpinfo);
ice_info = (pjmedia_ice_transport_info*)
pjmedia_transport_info_get_spc_info(
&tpinfo, PJMEDIA_TRANSPORT_TYPE_ICE);
/* Check if ICE is active */
if (!ice_info || !ice_info->active)
continue;
/* Check if ICE setup not completed yet */
if (ice_info->sess_state < PJ_ICE_STRANS_STATE_RUNNING) {
ice_complete = PJ_FALSE;
break;
}
/* Check if ICE needs to send reinvite */
if (!ice_need_reinv &&
ice_info->sess_state == PJ_ICE_STRANS_STATE_RUNNING &&
ice_info->role == PJ_ICE_SESS_ROLE_CONTROLLING)
{
pjsua_ice_config *cfg=&pjsua_var.acc[call->acc_id].cfg.ice_cfg;
if ((cfg->ice_always_update && !call->reinv_ice_sent) ||
pj_sockaddr_cmp(&tpinfo.sock_info.rtp_addr_name,
&call_med->rtp_addr))
{
ice_need_reinv = PJ_TRUE;
}
}
}
if (ice_complete && need_reinv)
*need_reinv = ice_need_reinv;
return ice_complete;
}
/* Check and send reinvite for lock codec and ICE update */
static pj_status_t process_pending_reinvite(pjsua_call *call)
{
const pj_str_t ST_UPDATE = {"UPDATE", 6};
pj_pool_t *pool = call->inv->pool_prov;
pjsip_inv_session *inv = call->inv;
pj_bool_t ice_need_reinv;
pj_bool_t ice_completed;
pj_bool_t need_lock_codec;
pj_bool_t rem_can_update;
pjmedia_sdp_session *new_offer;
pjsip_tx_data *tdata;
unsigned i;
pj_status_t status;
/* Verify if another SDP negotiation is in progress, e.g: session timer
* or another re-INVITE.
*/
if (inv==NULL || inv->neg==NULL ||
pjmedia_sdp_neg_get_state(inv->neg)!=PJMEDIA_SDP_NEG_STATE_DONE)
{
return PJMEDIA_SDPNEG_EINSTATE;
}
/* Don't do this if call is disconnecting! */
if (inv->state > PJSIP_INV_STATE_CONFIRMED || inv->cause >= 200)
{
return PJ_EINVALIDOP;
}
if (inv->state == PJSIP_INV_STATE_EARLY) {
if (pjsip_dlg_remote_has_cap(inv->dlg, PJSIP_H_ALLOW, NULL,
&ST_UPDATE) == PJSIP_DIALOG_CAP_SUPPORTED &&
inv->sdp_done_early_rel &&
!PJSUA_LOCK_CODEC_DONT_USE_UPDATE)
{
/* Yes, remote supports UPDATE and SDP negotiation was done
* using reliable provisional responses. We can proceed.
*/
} else {
call->reinv_pending = PJ_TRUE;
return PJ_EPENDING;
}
}
/* Check if ICE setup is complete and if it needs reinvite */
ice_completed = check_ice_complete(call, &ice_need_reinv);
if (!ice_completed)
return PJ_EPENDING;
/* Check if we need to lock codec */
need_lock_codec = check_lock_codec(call);
/* Check if reinvite is really needed */
if (!need_lock_codec && !ice_need_reinv)
return PJ_SUCCESS;
/* Okay! So we need to send re-INVITE/UPDATE */
/* Check if remote support UPDATE */
rem_can_update = !PJSUA_LOCK_CODEC_DONT_USE_UPDATE &&
pjsip_dlg_remote_has_cap(inv->dlg, PJSIP_H_ALLOW, NULL,
&ST_UPDATE) ==
PJSIP_DIALOG_CAP_SUPPORTED;
/* Logging stuff */
{
const char *ST_ICE_UPDATE = "ICE transport address after "
"ICE negotiation";
const char *ST_LOCK_CODEC = "media session to use only one codec";
PJ_LOG(4,(THIS_FILE, "Call %d sending %s for updating %s%s%s",
call->index,
(rem_can_update? "UPDATE" : "re-INVITE"),
(ice_need_reinv? ST_ICE_UPDATE : ST_LOCK_CODEC),
(ice_need_reinv && need_lock_codec? " and " : ""),
(ice_need_reinv && need_lock_codec? ST_LOCK_CODEC : "")
));
}
/* Clear reinit media flag. Should we also cleanup other flags here? */
call->opt.flag &= ~PJSUA_CALL_REINIT_MEDIA;
/* Generate SDP re-offer */
status = pjsua_media_channel_create_sdp(call->index, pool, NULL,
&new_offer, NULL);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create local SDP", status);
return status;
}
/* Update the new offer so it contains only a codec. Note that
* SDP nego has removed unmatched codecs from the offer and the codec
* order in the offer has been matched to the answer, so we'll override
* the codecs in the just generated SDP with the ones from the active
* local SDP and leave just one codec for the next SDP re-offer.
*/
if (need_lock_codec) {
const pjmedia_sdp_session *ref_sdp;
/* Get local active SDP as reference */
status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &ref_sdp);
if (status != PJ_SUCCESS)
return status;
/* Verify media count. Note that remote may add/remove media line
* in the answer. When answer has less media, it must have been
* handled by pjsua_media_channel_update() as disabled media.
* When answer has more media, it must have been ignored (treated
* as non-exist) anywhere. Local media count should not be updated
* at this point, as modifying media count operation (i.e: reinvite,
* update, vid_set_strm) is currently blocking, protected with
* dialog mutex, and eventually reset SDP nego state to LOCAL OFFER.
*/
if (call->med_cnt != ref_sdp->media_count ||
ref_sdp->media_count != new_offer->media_count)
{
/* Anyway, just in case, let's just return error */
return PJMEDIA_SDPNEG_EINSTATE;
}
for (i = 0; i < call->med_cnt; ++i) {
unsigned j, codec_cnt = 0;
const pjmedia_sdp_media *ref_m = ref_sdp->media[i];
pjmedia_sdp_media *m = new_offer->media[i];
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;
}
/* Reset formats */
m->desc.fmt_count = 0;
pjmedia_sdp_attr_remove_all(&m->attr_count, m->attr, "rtpmap");
pjmedia_sdp_attr_remove_all(&m->attr_count, m->attr, "fmtp");
/* Copy only the first format + any non-AV formats from
* the active local SDP.
*/
for (j = 0; j < ref_m->desc.fmt_count; ++j) {
const pj_str_t *fmt = &ref_m->desc.fmt[j];
if (is_non_av_fmt(ref_m, fmt) || (++codec_cnt == 1)) {
pjmedia_sdp_attr *a;
m->desc.fmt[m->desc.fmt_count++] = *fmt;
a = pjmedia_sdp_attr_find2(ref_m->attr_count, ref_m->attr,
"rtpmap", fmt);
if (a) {
pjmedia_sdp_attr_add(&m->attr_count, m->attr,
pjmedia_sdp_attr_clone(pool, a));
}
a = pjmedia_sdp_attr_find2(ref_m->attr_count, ref_m->attr,
"fmtp", fmt);
if (a) {
pjmedia_sdp_attr_add(&m->attr_count, m->attr,
pjmedia_sdp_attr_clone(pool, a));
}
}
}
}
}
/* Put back original direction and "c=0.0.0.0" line */
{
const pjmedia_sdp_session *cur_sdp;
/* Get local active SDP */
status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &cur_sdp);
if (status != PJ_SUCCESS)
return status;
/* Make sure media count has not been changed */
if (call->med_cnt != cur_sdp->media_count)
return PJMEDIA_SDPNEG_EINSTATE;
for (i = 0; i < call->med_cnt; ++i) {
const pjmedia_sdp_media *m = cur_sdp->media[i];
pjmedia_sdp_media *new_m = new_offer->media[i];
pjsua_call_media *call_med = &call->media[i];
pjmedia_sdp_attr *a = NULL;
/* Update direction to the current dir */
pjmedia_sdp_media_remove_all_attr(new_m, "sendrecv");
pjmedia_sdp_media_remove_all_attr(new_m, "sendonly");
pjmedia_sdp_media_remove_all_attr(new_m, "recvonly");
pjmedia_sdp_media_remove_all_attr(new_m, "inactive");
if (call_med->dir == PJMEDIA_DIR_ENCODING_DECODING) {
a = pjmedia_sdp_attr_create(pool, "sendrecv", NULL);
} else if (call_med->dir == PJMEDIA_DIR_ENCODING) {
a = pjmedia_sdp_attr_create(pool, "sendonly", NULL);
} else if (call_med->dir == PJMEDIA_DIR_DECODING) {
a = pjmedia_sdp_attr_create(pool, "recvonly", NULL);
} else {
const pjmedia_sdp_conn *conn;
a = pjmedia_sdp_attr_create(pool, "inactive", NULL);
/* Also check if the original c= line address is zero */
conn = m->conn;
if (!conn)
conn = cur_sdp->conn;
if (pj_strcmp2(&conn->addr, "0.0.0.0")==0 ||
pj_strcmp2(&conn->addr, "0")==0)
{
if (!new_m->conn) {
new_m->conn = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_conn);
}
if (pj_strcmp2(&new_m->conn->addr, "0.0.0.0")) {
new_m->conn->net_type = pj_str("IN");
new_m->conn->addr_type = pj_str("IP4");
new_m->conn->addr = pj_str("0.0.0.0");
}
}
}
pj_assert(a);
pjmedia_sdp_media_add_attr(new_m, a);
}
}
if (rem_can_update) {
status = pjsip_inv_update(inv, NULL, new_offer, &tdata);
} else {
status = pjsip_inv_reinvite(inv, NULL, new_offer, &tdata);
}
if (status==PJ_EINVALIDOP &&
++call->lock_codec.retry_cnt < LOCK_CODEC_MAX_RETRY)
{
/* Ups, let's reschedule again */
pjsua_call_schedule_reinvite_check(call, LOCK_CODEC_RETRY_INTERVAL);
return PJ_SUCCESS;
} else if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Error creating UPDATE/re-INVITE",
status);
return status;
}
/* Send the UPDATE/re-INVITE request */
status = pjsip_inv_send_msg(inv, tdata);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Error sending UPDATE/re-INVITE",
status);
return status;
}
/* Update flags */
if (ice_need_reinv)
call->reinv_ice_sent = PJ_TRUE;
if (need_lock_codec)
++call->lock_codec.retry_cnt;
return PJ_SUCCESS;
}
static void trickle_ice_retrans_18x(pj_timer_heap_t *th,
struct pj_timer_entry *te)
{
pjsua_call *call = (pjsua_call*)te->user_data;
pjsip_tx_data *tdata = NULL;
pj_time_val delay;
PJ_UNUSED_ARG(th);
/* If trickling has been started or dialog has been established on
* both sides, stop 18x retransmission.
*/
if (call->trickle_ice.trickling >= PJSUA_OP_STATE_RUNNING ||
call->trickle_ice.remote_dlg_est)
{
return;
}
/* Make sure last tdata is 18x response */
if (call->inv->invite_tsx)
tdata = call->inv->invite_tsx->last_tx;
if (!tdata || tdata->msg->type != PJSIP_RESPONSE_MSG ||
tdata->msg->line.status.code/10 != 18)
{
return;
}
/* Retransmit 18x */
++call->trickle_ice.retrans18x_count;
PJ_LOG(4,(THIS_FILE,
"Call %d: ICE trickle retransmitting 18x (retrans #%d)",
call->index, call->trickle_ice.retrans18x_count));
pjsip_tx_data_add_ref(tdata);
pjsip_tsx_retransmit_no_state(call->inv->invite_tsx, tdata);
/* Schedule next retransmission */
if (call->trickle_ice.retrans18x_count < 6) {
pj_uint32_t tmp;
tmp = (1 << call->trickle_ice.retrans18x_count) * pjsip_cfg()->tsx.t1;
delay.sec = 0;
delay.msec = tmp;
pj_time_val_normalize(&delay);
} else {
delay.sec = 1;
delay.msec = 500;
}
pjsua_schedule_timer(te, &delay);
}
static void trickle_ice_recv_sip_info(pjsua_call *call, pjsip_rx_data *rdata)
{
pjsip_media_type med_type;
pjsip_rdata_sdp_info *sdp_info;
pj_status_t status;
unsigned i, j, med_cnt;
pj_bool_t use_med_prov;
pjsip_media_type_init2(&med_type, "application", "trickle-ice-sdpfrag");
/* Parse the SDP */
sdp_info = pjsip_rdata_get_sdp_info2(rdata, &med_type);
if (!sdp_info->sdp) {
pj_status_t err = sdp_info->body.ptr? sdp_info->sdp_err:PJ_ENOTFOUND;
pjsua_perror(THIS_FILE, "Failed to parse trickle ICE SDP in "
"incoming INFO", err);
return;
}
PJSUA_LOCK();
/* Retrieve the candidates from the SDP */
use_med_prov = call->med_prov_cnt > call->med_cnt;
med_cnt = use_med_prov? call->med_prov_cnt : call->med_cnt;
for (i = 0; i < sdp_info->sdp->media_count; ++i) {
pjmedia_transport *tp = NULL;
pj_str_t mid, ufrag, pwd;
unsigned cand_cnt = PJ_ICE_ST_MAX_CAND;
pj_ice_sess_cand cand[PJ_ICE_ST_MAX_CAND];
pj_bool_t end_of_cand;
status = pjmedia_ice_trickle_decode_sdp(sdp_info->sdp, i, &mid,
&ufrag, &pwd,
&cand_cnt, cand,
&end_of_cand);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Failed to retrive ICE candidates from "
"SDP in incoming INFO", status);
continue;
}
for (j = 0; j < med_cnt; ++j) {
pjsua_call_media *cm = use_med_prov? &call->media_prov[j] :
&call->media[j];
tp = cm->tp_orig;
if (tp && tp->type == PJMEDIA_TRANSPORT_TYPE_ICE &&
pj_strcmp(&cm->rem_mid, &mid) == 0)
{
break;
}
}
if (j == med_cnt) {
pjsua_perror(THIS_FILE, "Cannot add remote candidates from SDP in "
"incoming INFO because media ID (SDP a=mid) is not "
"recognized",
PJ_EIGNORED);
continue;
}
/* Update ICE checklist */
status = pjmedia_ice_trickle_update(tp, &ufrag, &pwd, cand_cnt, cand,
end_of_cand);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Failed to update ICE checklist from "
"incoming INFO", status);
}
}
PJSUA_UNLOCK();
}
static void trickle_ice_send_sip_info(pj_timer_heap_t *th,
struct pj_timer_entry *te)
{
pjsua_call *call = (pjsua_call*)te->user_data;
pj_pool_t *tmp_pool = NULL;
pj_bool_t all_end_of_cand, use_med_prov;
pjmedia_sdp_session *sdp;
unsigned i, med_cnt;
pjsua_msg_data msg_data;
pjsip_generic_string_hdr hdr1, hdr2;
pj_status_t status = PJ_SUCCESS;
pj_bool_t forced, need_send = PJ_FALSE;
pj_sockaddr orig_addr;
pj_str_t SIP_INFO = {"INFO", 4};
pj_str_t CONTENT_DISP_STR = {"Content-Disposition", 19};
pj_str_t INFO_PKG_STR = {"Info-Package", 12};
pj_str_t TRICKLE_ICE_STR = {"trickle-ice", 11};
PJ_UNUSED_ARG(th);
PJSUA_LOCK();
/* Check provisional media or active media to use */
use_med_prov = call->med_prov_cnt > call->med_cnt;
med_cnt = use_med_prov? call->med_prov_cnt : call->med_cnt;
/* Check if any pending INFO already */
if (call->trickle_ice.pending_info)
goto on_return;
/* Check if any new candidate, if not forced */
forced = (te->id == 2);
if (!forced) {
for (i = 0; i < med_cnt; ++i) {
pjsua_call_media *cm = use_med_prov? &call->media_prov[i] :
&call->media[i];
pjmedia_transport *tp = cm->tp_orig;
if (!tp || tp->type != PJMEDIA_TRANSPORT_TYPE_ICE)
continue;
if (pjmedia_ice_trickle_has_new_cand(tp))
break;
}
/* No new local candidate */
if (i == med_cnt)
goto on_return;
}
PJ_LOG(4,(THIS_FILE, "Call %d: ICE trickle sending SIP INFO%s",
call->index, (forced? " (forced)":"")));
/* Create temporary pool */
tmp_pool = pjsua_pool_create("tmp_ice", 128, 128);
/* Create empty SDP */
pj_sockaddr_init(pj_AF_INET(), &orig_addr, NULL, 0);
status = pjmedia_endpt_create_base_sdp(pjsua_var.med_endpt, tmp_pool,
NULL, &orig_addr, &sdp);
if (status != PJ_SUCCESS)
goto on_return;
/* Generate SDP for SIP INFO */
all_end_of_cand = PJ_TRUE;
for (i = 0; i < med_cnt; ++i) {
pjsua_call_media *cm = use_med_prov? &call->media_prov[i] :
&call->media[i];
pjmedia_transport *tp = cm->tp_orig;
pj_bool_t end_of_cand = PJ_FALSE;
if (!tp || tp->type != PJMEDIA_TRANSPORT_TYPE_ICE)
continue;
status = pjmedia_ice_trickle_send_local_cand(tp, tmp_pool, sdp,
&end_of_cand);
if (status != PJ_SUCCESS || !end_of_cand)
all_end_of_cand = PJ_FALSE;
need_send |= (status==PJ_SUCCESS);
}
if (!need_send)
goto on_return;
/* Generate and send SIP INFO */
pjsua_msg_data_init(&msg_data);
pjsip_generic_string_hdr_init2(&hdr1, &INFO_PKG_STR, &TRICKLE_ICE_STR);
pj_list_push_back(&msg_data.hdr_list, &hdr1);
pjsip_generic_string_hdr_init2(&hdr2, &CONTENT_DISP_STR, &INFO_PKG_STR);
pj_list_push_back(&msg_data.hdr_list, &hdr2);
msg_data.content_type = pj_str("application/trickle-ice-sdpfrag");
msg_data.msg_body.ptr = pj_pool_alloc(tmp_pool, PJSIP_MAX_PKT_LEN);
msg_data.msg_body.slen = pjmedia_sdp_print(sdp, msg_data.msg_body.ptr,
PJSIP_MAX_PKT_LEN);
if (msg_data.msg_body.slen == -1) {
PJ_LOG(3,(THIS_FILE,
"Warning! Call %d: ICE trickle failed to print SDP for "
"SIP INFO due to insufficient buffer", call->index));
goto on_return;
}
status = pjsua_call_send_request(call->index, &SIP_INFO, &msg_data);
if (status != PJ_SUCCESS)
goto on_return;
/* Set flag for pending SIP INFO */
call->trickle_ice.pending_info = PJ_TRUE;
/* Stop trickling if local candidate gathering for all media is done */
if (all_end_of_cand) {
PJ_LOG(4,(THIS_FILE, "Call %d: ICE trickle stopped trickling "
"as local candidate gathering completed",
call->index));
call->trickle_ice.trickling = PJSUA_OP_STATE_DONE;
}
/* Update ICE checklist after conveying local candidates. */
for (i = 0; i < med_cnt; ++i) {
pjsua_call_media *cm = use_med_prov? &call->media_prov[i] :
&call->media[i];
pjmedia_transport *tp = cm->tp_orig;
if (!tp || tp->type != PJMEDIA_TRANSPORT_TYPE_ICE)
continue;
pjmedia_ice_trickle_update(tp, NULL, NULL, 0, NULL, PJ_FALSE);
}
on_return:
if (tmp_pool)
pj_pool_release(tmp_pool);
/* Reschedule if we are trickling */
if (call->trickle_ice.trickling == PJSUA_OP_STATE_RUNNING) {
pj_time_val delay = {0, PJSUA_TRICKLE_ICE_NEW_CAND_CHECK_INTERVAL};
/* Reset forced mode after successfully sending forced SIP INFO */
te->id = (status==PJ_SUCCESS? 0 : 2);
pj_time_val_normalize(&delay);
pjsua_schedule_timer(te, &delay);
}
PJSUA_UNLOCK();
}
/* Before sending INFO can be started, UA needs to confirm these:
* 1. dialog is established (perhaps early) at both sides,
* 2. trickle ICE is supported by peer.
*
* This function needs to be called when:
* - UAS sending 18x, to start 18x retrans
* - UAC receiving 18x, to forcefully send SIP INFO & start trickling
* - UAS receiving INFO, to cease 18x retrans & start trickling
* - UAS receiving PRACK, to start trickling
* - UAC/UAS receiving remote SDP (and check for trickle ICE support),
* to start trickling.
*/
void pjsua_ice_check_start_trickling(pjsua_call *call,
pj_bool_t forceful,
pjsip_event *e)
{
pjsip_inv_session *inv = call->inv;
/* Make sure trickling/sending-INFO has not been started */
if (!forceful && call->trickle_ice.trickling >= PJSUA_OP_STATE_RUNNING)
return;
/* Make sure trickle ICE is enabled */
if (!call->trickle_ice.enabled)
return;
/* Make sure the dialog state is established */
if (!inv || inv->dlg->state != PJSIP_DIALOG_STATE_ESTABLISHED)
return;
/* First, make sure remote dialog is also established. */
if (inv->state == PJSIP_INV_STATE_CONFIRMED) {
/* Set flag indicating remote dialog is established */
call->trickle_ice.remote_dlg_est = PJ_TRUE;
} else if (inv->state > PJSIP_INV_STATE_CONFIRMED) {
/* Call is terminating/terminated (just trying to be safe) */
call->trickle_ice.remote_dlg_est = PJ_FALSE;
} else if (!call->trickle_ice.remote_dlg_est && e) {
/* Call is being initialized */
pjsip_msg *msg = NULL;
pjsip_rx_data *rdata = NULL;
pjsip_tx_data *tdata = NULL;
pj_bool_t has_100rel = (inv->options & PJSIP_INV_REQUIRE_100REL);
pj_timer_entry *te = &call->trickle_ice.timer;
if (e->type == PJSIP_EVENT_TSX_STATE &&
e->body.tsx_state.type == PJSIP_EVENT_RX_MSG)
{
rdata = e->body.tsx_state.src.rdata;
} else if (e->type == PJSIP_EVENT_TSX_STATE &&
e->body.tsx_state.type == PJSIP_EVENT_TX_MSG)
{
tdata = e->body.tsx_state.src.tdata;
} else {
return;
}
/* UAC must have received 18x at this point, so dialog must have been
* established at the remote side.
*/
if (inv->role == PJSIP_ROLE_UAC) {
/* UAC needs to send SIP INFO when receiving 18x and 100rel is not
* active.
* Note that 18x may not have SDP (so we don't know if remote
* supports trickle ICE), but we should send INFO anyway, as the
* draft allows start trickling without answer.
*/
if (!has_100rel && rdata &&
rdata->msg_info.msg->type == PJSIP_RESPONSE_MSG &&
rdata->msg_info.msg->line.status.code/10 == 18)
{
pjsip_rdata_sdp_info *sdp_info;
sdp_info = pjsip_rdata_get_sdp_info(rdata);
if (sdp_info->sdp) {
unsigned i;
for (i = 0; i < sdp_info->sdp->media_count; ++i) {
if (pjmedia_ice_sdp_has_trickle(sdp_info->sdp, i)) {
call->trickle_ice.remote_sup = PJ_TRUE;
break;
}
}
} else {
/* Start sending SIP INFO forcefully */
forceful = PJ_TRUE;
}
if (forceful || call->trickle_ice.remote_sup) {
PJ_LOG(4,(THIS_FILE,
"Call %d: ICE trickle started after UAC "
"receiving 18x (with%s SDP)",
call->index, sdp_info->sdp?"":"out"));
}
}
}
/* But if we are the UAS, we need to wait for SIP PRACK or INFO to
* confirm dialog state at remote. And while waiting, 18x needs to be
* retransmitted.
*/
else {
if (tdata && e->body.tsx_state.tsx == inv->invite_tsx &&
call->trickle_ice.retrans18x_count == 0)
{
/* Ignite 18x retransmission */
msg = tdata->msg;
if (msg->type == PJSIP_RESPONSE_MSG &&
msg->line.status.code/10 == 18)
{
pj_time_val delay;
delay.sec = pjsip_cfg()->tsx.t1 / 1000;
delay.msec = pjsip_cfg()->tsx.t1 % 1000;
pj_assert(!pj_timer_entry_running(te));
te->cb = &trickle_ice_retrans_18x;
pjsua_schedule_timer(te, &delay);
PJ_LOG(4,(THIS_FILE,
"Call %d: ICE trickle start retransmitting 18x",
call->index));
}
return;
}
/* Check for incoming PRACK or INFO to stop 18x retransmission */
if (!rdata)
return;
msg = rdata->msg_info.msg;
if (has_100rel) {
/* With 100rel, has received PRACK? */
if (msg->type != PJSIP_REQUEST_MSG ||
pjsip_method_cmp(&msg->line.req.method,
pjsip_get_prack_method()))
{
return;
}
} else {
pj_str_t INFO_PKG_STR = {"Info-Package", 12};
pjsip_generic_string_hdr *hdr;
/* Without 100rel, has received INFO? */
if (msg->type != PJSIP_REQUEST_MSG ||
pjsip_method_cmp(&msg->line.req.method,
&pjsip_info_method))
{
return;
}
/* With Info-Package header containing 'trickle-ice' */
hdr = (pjsip_generic_string_hdr*)
pjsip_msg_find_hdr_by_name(msg, &INFO_PKG_STR, NULL);
if (!hdr || pj_strcmp2(&hdr->hvalue, "trickle-ice"))
return;
/* Set the flag indicating remote supports trickle ICE */
call->trickle_ice.remote_sup = PJ_TRUE;
}
PJ_LOG(4,(THIS_FILE,
"Call %d: ICE trickle stop retransmitting 18x after "
"receiving %s",
call->index, (has_100rel?"PRACK":"INFO")));
}
/* Set flag indicating remote dialog is established.
* Any 18x retransmission should be ceased automatically.
*/
call->trickle_ice.remote_dlg_est = PJ_TRUE;
}
/* Check if ICE trickling can be started */
if (!forceful &&
(!call->trickle_ice.remote_dlg_est || !call->trickle_ice.remote_sup))
{
return;
}
/* Let's start trickling (or sending SIP INFO) */
if (forceful || call->trickle_ice.trickling < PJSUA_OP_STATE_RUNNING)
{
pj_timer_entry *te = &call->trickle_ice.timer;
pj_time_val delay = {0,0};
if (call->trickle_ice.trickling < PJSUA_OP_STATE_RUNNING)
call->trickle_ice.trickling = PJSUA_OP_STATE_RUNNING;
pjsua_cancel_timer(te);
te->id = forceful? 2 : 0;
te->cb = &trickle_ice_send_sip_info;
pjsua_schedule_timer(te, &delay);
PJ_LOG(4,(THIS_FILE,
"Call %d: ICE trickle start trickling",
call->index));
}
}
/*
* 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;
unsigned num_locks = 0;
pj_log_push_indent();
call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
if (!call) {
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:
if (call->hanging_up) {
/* This can happen if there is a crossover between
* our CANCEL request and the remote's 200 response.
* So we send BYE here.
*/
call_inv_end_session(call, 200, NULL, NULL);
return;
}
pj_gettimeofday(&call->conn_time);
if (call->trickle_ice.enabled) {
call->trickle_ice.remote_dlg_est = PJ_TRUE;
pjsua_ice_check_start_trickling(call, PJ_FALSE, NULL);
}
/* See if auto reinvite was pended as media update was done in the
* EARLY state and remote does not support UPDATE.
*/
if (call->reinv_pending) {
call->reinv_pending = PJ_FALSE;
pjsua_call_schedule_reinvite_check(call, 0);
}
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 reinvite timer, if it is active */
if (call->reinv_timer.id) {
pjsua_cancel_timer(&call->reinv_timer);
call->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);
}
}
}
}
/* Destroy media session when invite session is disconnected. */
if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
PJSUA_LOCK();
if (!call->hanging_up)
pjsua_media_channel_deinit(call->index);
PJSUA_UNLOCK();
}
/* Release locks before calling callbacks, to avoid deadlock. */
while (PJSUA_LOCK_IS_LOCKED()) {
num_locks++;
PJSUA_UNLOCK();
}
/* Ticket #1627: Invoke on_call_tsx_state() when call is disconnected. */
if (inv->state == PJSIP_INV_STATE_DISCONNECTED &&
e->type == PJSIP_EVENT_TSX_STATE &&
!call->hanging_up && call->inv &&
pjsua_var.ua_cfg.cb.on_call_tsx_state)
{
(*pjsua_var.ua_cfg.cb.on_call_tsx_state)(call->index,
e->body.tsx_state.tsx, e);
}
if (!call->hanging_up && pjsua_var.ua_cfg.cb.on_call_state)
(*pjsua_var.ua_cfg.cb.on_call_state)(call->index, e);
/* Re-acquire the locks. */
for (;num_locks > 0; num_locks--)
PJSUA_LOCK();
/* call->inv may be NULL now */
/* Finally, free call when invite session is disconnected. */
if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
PJSUA_LOCK();
/* Free call */
call->inv = NULL;
pj_assert(pjsua_var.call_cnt > 0);
--pjsua_var.call_cnt;
/* Reset call */
reset_call(call->index);
pjsua_check_snd_dev_idle();
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_get_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 )
{
pjsip_tx_data *tdata;
pj_status_t status;
status = pjsip_inv_end_session(inv, code, NULL, &tdata);
if (status != PJ_SUCCESS || !tdata)
return;
#if DISABLED_FOR_TICKET_1185
pjsua_call *call;
/* Add SDP in 488 status */
call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
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();
call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
if (call->hanging_up)
goto on_return;
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "SDP negotiation has failed", status);
/* Revert back provisional media. */
pjsua_media_prov_revert(call->index);
/* 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_EARLY &&
inv->state != PJSIP_INV_STATE_CONFIRMED) ||
(inv->state == PJSIP_INV_STATE_EARLY && call->med_cnt == 0))
{
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;
}
call->med_update_success = (status == PJ_SUCCESS);
/* Trickle ICE tasks:
* - Check remote SDP for trickle ICE support & start sending SIP INFO.
*/
{
unsigned i;
for (i = 0; i < remote_sdp->media_count; ++i) {
if (pjmedia_ice_sdp_has_trickle(remote_sdp, i))
break;
}
call->trickle_ice.remote_sup = (i < remote_sdp->media_count);
if (call->trickle_ice.remote_sup)
pjsua_ice_check_start_trickling(call, PJ_FALSE, NULL);
}
/* 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 this is not the initial INVITE, don't disconnect call due to
* no media after SDP negotiation.
*/
if (status == PJMEDIA_SDPNEG_ENOMEDIA &&
call->inv->state == PJSIP_INV_STATE_CONFIRMED)
{
status = PJ_SUCCESS;
}
/* Disconnect call after failure in media channel update */
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. */
pjsua_call_schedule_reinvite_check(call, 0);
/* Call application callback, if any */
if (!call->hanging_up && pjsua_var.ua_cfg.cb.on_call_media_state)
pjsua_var.ua_cfg.cb.on_call_media_state(call->index);
on_return:
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,
pj_bool_t as_answerer)
{
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)
*/
/* https://github.com/pjsip/pjproject/issues/880
if (call->dir != PJMEDIA_DIR_ENCODING) {
*/
/* https://github.com/pjsip/pjproject/issues/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");
/* When as answerer, just simply set dir to "sendonly", note that
* if the offer uses "sendonly" or "inactive", the SDP negotiator
* will change our answer dir to "inactive".
*/
if (as_answerer || (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, PJ_FALSE);
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,
struct pjsip_inv_on_rx_offer_cb_param *param)
{
pjsua_call *call;
pjmedia_sdp_session *answer;
unsigned i;
pj_status_t status;
const pjmedia_sdp_session *offer = param->offer;
pjsua_call_setting opt;
pj_bool_t async = PJ_FALSE;
call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
if (call->hanging_up)
return;
/* Supply candidate answer */
PJ_LOG(4,(THIS_FILE, "Call %d: received updated media offer",
call->index));
pj_log_push_indent();
if (pjsua_call_media_is_changing(call)) {
PJ_LOG(1,(THIS_FILE, "Unable to process offer" ERR_MEDIA_CHANGING));
goto on_return;
}
pjsua_call_cleanup_flag(&call->opt);
opt = call->opt;
if (pjsua_var.ua_cfg.cb.on_call_rx_reinvite &&
param->rdata->msg_info.msg->type == PJSIP_REQUEST_MSG &&
param->rdata->msg_info.msg->line.req.method.id == PJSIP_INVITE_METHOD)
{
pjsip_status_code code = PJSIP_SC_OK;
/* If on_call_rx_reinvite() callback is implemented,
* call it first.
*/
(*pjsua_var.ua_cfg.cb.on_call_rx_reinvite)(
call->index, offer,
(pjsip_rx_data *)param->rdata,
NULL, &async, &code, &opt);
if (async) {
pjsip_tx_data *response;
status = pjsip_inv_initial_answer(inv,
(pjsip_rx_data *)param->rdata,
100, NULL, NULL, &response);
if (status != PJ_SUCCESS) {
PJ_PERROR(3, (THIS_FILE, status,
"Failed to create initial answer"));
goto on_return;
}
status = pjsip_inv_send_msg(inv, response);
if (status != PJ_SUCCESS) {
PJ_PERROR(3, (THIS_FILE, status,
"Failed to send initial answer"));
goto on_return;
}
PJ_LOG(4,(THIS_FILE, "App will manually answer the re-INVITE "
"on call %d", call->index));
}
if (code != PJSIP_SC_OK) {
PJ_LOG(4,(THIS_FILE, "Rejecting re-INVITE updated media offer "
"on call %d", call->index));
goto on_return;
}
call->opt = opt;
}
if (pjsua_var.ua_cfg.cb.on_call_rx_offer && !async) {
pjsip_status_code code = PJSIP_SC_OK;
(*pjsua_var.ua_cfg.cb.on_call_rx_offer)(call->index, offer, NULL,
&code, &opt);
if (code != PJSIP_SC_OK) {
PJ_LOG(4,(THIS_FILE, "Rejecting updated media offer on call %d",
call->index));
goto on_return;
}
call->opt = opt;
}
/* Re-init media for the new remote offer before creating SDP */
status = apply_call_setting(call, &call->opt, offer);
if (status != PJ_SUCCESS)
goto on_return;
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;
}
if (async) {
call->rx_reinv_async = async;
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, PJ_TRUE);
}
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:
pj_log_pop_indent();
}
/*
* Called when receiving re-INVITE.
*/
static pj_status_t pjsua_call_on_rx_reinvite(pjsip_inv_session *inv,
const pjmedia_sdp_session *offer,
pjsip_rx_data *rdata)
{
pjsua_call *call;
pj_bool_t async;
PJ_UNUSED_ARG(offer);
PJ_UNUSED_ARG(rdata);
call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
async = call->rx_reinv_async;
call->rx_reinv_async = PJ_FALSE;
return (async? PJ_SUCCESS: !PJ_SUCCESS);
}
/*
* 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;
unsigned mi;
pj_log_push_indent();
call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
if (call->hanging_up || pjsua_call_media_is_changing(call)) {
*offer = NULL;
PJ_LOG(1,(THIS_FILE, "Unable to create offer%s",
call->hanging_up? ", call hanging up":
ERR_MEDIA_CHANGING));
goto on_return;
}
#if RESTART_ICE_ON_REINVITE
/* Ticket #1783, RFC 5245 section 12.5:
* If an agent receives a mid-dialog re-INVITE that contains no offer,
* it MUST restart ICE for each media stream and go through the process
* of gathering new candidates.
*/
for (mi=0; mi<call->med_cnt; ++mi) {
pjsua_call_media *call_med = &call->media[mi];
pjmedia_transport_info tpinfo;
pjmedia_ice_transport_info *ice_info;
/* Check if the media is using ICE */
pjmedia_transport_info_init(&tpinfo);
pjmedia_transport_get_info(call_med->tp, &tpinfo);
ice_info = (pjmedia_ice_transport_info*)
pjmedia_transport_info_get_spc_info(
&tpinfo, PJMEDIA_TRANSPORT_TYPE_ICE);
if (!ice_info)
continue;
/* Stop and re-init ICE stream transport.
* According to RFC 5245 section 9.1.1.1, during ICE restart,
* media can continue to be sent to the previously validated pair.
*/
pjmedia_transport_media_stop(call_med->tp);
pjmedia_transport_media_create(call_med->tp, call->inv->pool_prov,
(call_med->enable_rtcp_mux?
PJMEDIA_TPMED_RTCP_MUX: 0),
NULL, mi);
PJ_LOG(4, (THIS_FILE, "Restarting ICE for media %d", mi));
}
#endif
pjsua_call_cleanup_flag(&call->opt);
if (pjsua_var.ua_cfg.cb.on_call_tx_offer) {
(*pjsua_var.ua_cfg.cb.on_call_tx_offer)(call->index, NULL,
&call->opt);
}
/* We may need to re-initialize media before creating SDP */
if (call->med_prov_cnt == 0 || pjsua_var.ua_cfg.cb.on_call_tx_offer) {
status = apply_call_setting(call, &call->opt, NULL);
if (status != PJ_SUCCESS)
goto on_return;
}
/* 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:
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
* transferred successfully.
*/
if (call && !call->hanging_up &&
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 && !call->hanging_up &&
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 or an error response, 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 || call->hanging_up || !event ||
!pjsua_var.ua_cfg.cb.on_call_transfer_status)
{
/* Application is not interested with call progress status */
goto on_return;
}
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;
msg = rdata->msg_info.msg;
/* This better be a NOTIFY request */
if (pjsip_method_cmp(&msg->line.req.method,
pjsip_get_notify_method()) == 0)
{
/* Check if there's body */
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 = msg->line.status.code;
status_line.reason = msg->line.status.reason;
}
} 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_get_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_transferred( 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;
pjsua_call_setting call_opt;
pj_log_push_indent();
existing_call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
if (existing_call->hanging_up) {
pjsip_dlg_respond( inv->dlg, rdata, 487, NULL, NULL, NULL);
goto on_return;
}
/* 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);
}
pjsua_call_cleanup_flag(&existing_call->opt);
call_opt = existing_call->opt;
if (pjsua_var.ua_cfg.cb.on_call_transfer_request2) {
(*pjsua_var.ua_cfg.cb.on_call_transfer_request2)(existing_call->index,
&refer_to->hvalue,
&code,
&call_opt);
}
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 transferred 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 *tdata2;
const pj_str_t str_false = { "false", 5};
pjsip_hdr *hdr;
status = pjsip_dlg_create_response(inv->dlg, rdata, code, NULL,
&tdata2);
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(tdata2->pool, &str_refer_sub,
&str_false);
pjsip_msg_add_hdr(tdata2->msg, hdr);
/* Send answer */
status = pjsip_dlg_send_response(inv->dlg, pjsip_rdata_get_tsx(rdata),
tdata2);
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.
* Note that the user_data of the new call is initialized to the
* original call, it is needed by PJSUA2 to update its states.
* While PJSUA app can always override it anytime.
*/
tmp = pj_str(uri);
status = pjsua_call_make_call(existing_call->acc_id, &tmp, &call_opt,
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)
{
/* Incoming INFO request for media control, DTMF, trickle ICE, etc. */
const pj_str_t STR_APPLICATION = { "application", 11};
const pj_str_t STR_MEDIA_CONTROL_XML = { "media_control+xml", 17 };
const pj_str_t STR_DTMF_RELAY = { "dtmf-relay", 10 };
const pj_str_t STR_TRICKLE_ICE_SDP = { "trickle-ice-sdpfrag", 19 };
pjsua_call *call;
pj_log_push_indent();
call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
if (call == NULL)
goto on_return;
if (call->inv == NULL || call->hanging_up) {
/* Call has been disconnected. */
goto on_return;
}
/* https://github.com/pjsip/pjproject/issues/1452:
* If a request is retried due to 401/407 challenge, don't process the
* transaction first but wait until we've retried it.
*/
if (tsx->role == PJSIP_ROLE_UAC &&
(tsx->status_code==401 || tsx->status_code==407) &&
tsx->last_tx && tsx->last_tx->auth_retry)
{
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_transferred(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_accept_hdr *accept_hdr;
rdata = e->body.tsx_state.src.rdata;
/* 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.
*/
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 (e->type == PJSIP_EVENT_TSX_STATE &&
tsx->role == PJSIP_ROLE_UAC &&
pjsip_method_cmp(&tsx->method, &pjsip_message_method)==0 &&
(tsx->state == PJSIP_TSX_STATE_COMPLETED ||
(tsx->state == PJSIP_TSX_STATE_TERMINATED &&
e->body.tsx_state.prev_state != PJSIP_TSX_STATE_COMPLETED)))
{
/* 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) {
pj_str_t im_body = im_data->body;
if (im_body.slen==0) {
pjsip_msg_body *body = tsx->last_tx->msg->body;
pj_strset(&im_body, body->data, body->len);
}
if (pjsua_var.ua_cfg.cb.on_pager_status) {
pjsua_var.ua_cfg.cb.on_pager_status(im_data->call_id,
&im_data->to,
&im_body,
im_data->user_data,
(pjsip_status_code)
tsx->status_code,
&tsx->status_text);
}
if (pjsua_var.ua_cfg.cb.on_pager_status2) {
pjsip_rx_data* rdata;
if (e->body.tsx_state.type == PJSIP_EVENT_RX_MSG)
rdata = e->body.tsx_state.src.rdata;
else
rdata = NULL;
pjsua_var.ua_cfg.cb.on_pager_status2(im_data->call_id,
&im_data->to,
&im_body,
im_data->user_data,
(pjsip_status_code)
tsx->status_code,
&tsx->status_text,
tsx->last_tx,
rdata, im_data->acc_id);
}
}
}
} else if (tsx->role == PJSIP_ROLE_UAC &&
pjsip_method_cmp(&tsx->method, pjsip_get_invite_method())==0 &&
tsx->state >= PJSIP_TSX_STATE_COMPLETED &&
e->body.tsx_state.prev_state < PJSIP_TSX_STATE_COMPLETED &&
(!PJSIP_IS_STATUS_IN_CLASS(tsx->status_code, 300) &&
tsx->status_code!=401 && tsx->status_code!=407 &&
tsx->status_code!=422))
{
if (tsx->status_code/100 == 2) {
/* If we have sent CANCEL and the original INVITE returns a 2xx,
* we then send BYE.
*/
if (call->hanging_up) {
PJ_LOG(3,(THIS_FILE, "Unsuccessful in cancelling the original "
"INVITE for call %d due to %d response, sending BYE "
"instead", call->index, tsx->status_code));
call_disconnect(call->inv, PJSIP_SC_OK);
}
} else {
/* Monitor the status of call hold/unhold request */
if (tsx->last_tx == (pjsip_tx_data*)call->hold_msg) {
/* Outgoing call hold failed */
call->local_hold = PJ_FALSE;
PJ_LOG(3,(THIS_FILE, "Error putting call %d on hold "
"(reason=%d)", call->index, tsx->status_code));
} else if (call->opt.flag & PJSUA_CALL_UNHOLD) {
/* Call unhold failed */
call->local_hold = PJ_TRUE;
PJ_LOG(3,(THIS_FILE, "Error releasing hold on call %d "
"(reason=%d)", call->index, tsx->status_code));
}
}
if (tsx->last_tx == (pjsip_tx_data*)call->hold_msg) {
call->hold_msg = NULL;
}
if (tsx->last_tx->msg->body &&
(tsx->status_code/100 != 2 || !call->med_update_success))
{
/* Either we get non-2xx or media update failed,
* revert back provisional media.
*/
pjsua_media_prov_revert(call->index);
}
} else if (tsx->role == PJSIP_ROLE_UAC &&
pjsip_method_cmp(&tsx->method, &pjsip_update_method)==0 &&
tsx->state >= PJSIP_TSX_STATE_COMPLETED &&
e->body.tsx_state.prev_state < PJSIP_TSX_STATE_COMPLETED &&
(!PJSIP_IS_STATUS_IN_CLASS(tsx->status_code, 300) &&
tsx->status_code!=401 && tsx->status_code!=407 &&
tsx->status_code!=422))
{
if (tsx->last_tx->msg->body &&
(tsx->status_code/100 != 2 || !call->med_update_success))
{
/* Either we get non-2xx or media update failed,
* revert back provisional media.
*/
pjsua_media_prov_revert(call->index);
}
} else if (tsx->role==PJSIP_ROLE_UAS &&
tsx->state==PJSIP_TSX_STATE_TRYING &&
pjsip_method_cmp(&tsx->method, &pjsip_info_method)==0)
{
pjsip_rx_data *rdata = e->body.tsx_state.src.rdata;
pjsip_msg_body *body = rdata->msg_info.msg->body;
/* Check Media Control content in the INFO message */
if (body && body->len &&
pj_stricmp(&body->content_type.type, &STR_APPLICATION)==0 &&
pj_stricmp(&body->content_type.subtype, &STR_MEDIA_CONTROL_XML)==0)
{
pjsip_tx_data *tdata;
pj_str_t control_st;
pj_status_t status;
/* Apply and answer the INFO request */
pj_strset(&control_st, (char*)body->data, body->len);
status = pjsua_media_apply_xml_control(call->index, &control_st);
if (status == PJ_SUCCESS) {
status = pjsip_endpt_create_response(tsx->endpt, rdata,
200, NULL, &tdata);
if (status == PJ_SUCCESS)
status = pjsip_tsx_send_msg(tsx, tdata);
} else {
status = pjsip_endpt_create_response(tsx->endpt, rdata,
400, NULL, &tdata);
if (status == PJ_SUCCESS)
status = pjsip_tsx_send_msg(tsx, tdata);
}
}
/* Check DTMF content in the INFO message */
else if (body && body->len &&
pj_stricmp(&body->content_type.type, &STR_APPLICATION)==0 &&
pj_stricmp(&body->content_type.subtype, &STR_DTMF_RELAY)==0)
{
pjsip_tx_data *tdata;
pj_status_t status;
pj_bool_t is_handled = PJ_FALSE;
if (pjsua_var.ua_cfg.cb.on_dtmf_digit2 ||
pjsua_var.ua_cfg.cb.on_dtmf_event)
{
pjsua_dtmf_info info = {0};
pj_str_t delim, token, input;
pj_ssize_t found_idx;
delim = pj_str("\r\n");
input = pj_str(rdata->msg_info.msg->body->data);
found_idx = pj_strtok(&input, &delim, &token, 0);
if (found_idx != input.slen) {
/* Get signal/digit */
const pj_str_t STR_SIGNAL = { "Signal", 6 };
const pj_str_t STR_DURATION = { "Duration", 8 };
char *val;
pj_ssize_t count_equal_sign;
val = pj_strstr(&input, &STR_SIGNAL);
if (val) {
char* p = val + STR_SIGNAL.slen;
count_equal_sign = 0;
while ((p - input.ptr < input.slen) && (*p == ' ' || *p == '=')) {
if(*p == '=')
count_equal_sign++;
++p;
}
if (count_equal_sign == 1 && (p - input.ptr < input.slen)) {
info.digit = *p;
is_handled = PJ_TRUE;
} else {
PJ_LOG(2, (THIS_FILE, "Invalid dtmf-relay format"));
}
/* Get duration */
input.ptr += token.slen + 2;
input.slen -= (token.slen + 2);
val = pj_strstr(&input, &STR_DURATION);
if (val && is_handled) {
pj_str_t val_str;
char* ptr = val + STR_DURATION.slen;
count_equal_sign = 0;
while ((ptr - input.ptr < input.slen) &&
(*ptr == ' ' || *ptr == '='))
{
if (*ptr == '=')
count_equal_sign++;
++ptr;
}
if ((count_equal_sign == 1) &&
(ptr - input.ptr < input.slen))
{
val_str.ptr = ptr;
val_str.slen = input.slen - (ptr - input.ptr);
info.duration = pj_strtoul(&val_str);
} else {
info.duration = PJSUA_UNKNOWN_DTMF_DURATION;
is_handled = PJ_FALSE;
PJ_LOG(2, (THIS_FILE,
"Invalid dtmf-relay format"));
}
}
if (is_handled) {
info.method = PJSUA_DTMF_METHOD_SIP_INFO;
if (pjsua_var.ua_cfg.cb.on_dtmf_event) {
pjsua_dtmf_event evt;
pj_timestamp begin_of_time, timestamp;
/* Use the current instant as the events start
* time.
*/
begin_of_time.u64 = 0;
pj_get_timestamp(&timestamp);
evt.method = info.method;
evt.timestamp = pj_elapsed_msec(&begin_of_time,
&timestamp);
evt.digit = info.digit;
evt.duration = info.duration;
/* There is only one message indicating the full
* duration of the digit.
*/
evt.flags = PJMEDIA_STREAM_DTMF_IS_END;
(*pjsua_var.ua_cfg.cb.on_dtmf_event)(call->index,
&evt);
} else {
(*pjsua_var.ua_cfg.cb.on_dtmf_digit2)(call->index,
&info);
}
status = pjsip_endpt_create_response(tsx->endpt, rdata,
200, NULL, &tdata);
if (status == PJ_SUCCESS)
status = pjsip_tsx_send_msg(tsx, tdata);
}
}
}
}
if (!is_handled) {
status = pjsip_endpt_create_response(tsx->endpt, rdata,
400, NULL, &tdata);
if (status == PJ_SUCCESS)
status = pjsip_tsx_send_msg(tsx, tdata);
}
}
/* Check Trickle ICE content in the INFO message */
else if (body && body->len &&
pj_stricmp(&body->content_type.type, &STR_APPLICATION)==0 &&
pj_stricmp(&body->content_type.subtype,
&STR_TRICKLE_ICE_SDP)==0)
{
pjsip_tx_data *tdata;
pj_status_t status;
/* Trickle ICE tasks:
* - UAS receiving INFO, cease 18x retrans & start trickling
*/
if (call->trickle_ice.enabled) {
pjsua_ice_check_start_trickling(call, PJ_FALSE, e);
/* Process the SIP INFO content */
trickle_ice_recv_sip_info(call, rdata);
/* Send 200 response, regardless */
status = pjsip_endpt_create_response(tsx->endpt, rdata,
200, NULL, &tdata);
} else {
/* Trickle ICE not enabled, send 400 response */
status = pjsip_endpt_create_response(tsx->endpt, rdata,
400, NULL, &tdata);
}
if (status == PJ_SUCCESS)
status = pjsip_tsx_send_msg(tsx, tdata);
}
} else if (tsx->role == PJSIP_ROLE_UAC &&
pjsip_method_cmp(&tsx->method, &pjsip_info_method)==0 &&
(tsx->state == PJSIP_TSX_STATE_COMPLETED ||
(tsx->state == PJSIP_TSX_STATE_TERMINATED &&
e->body.tsx_state.prev_state != PJSIP_TSX_STATE_COMPLETED)))
{
pjsip_msg_body *body = NULL;
if (e->body.tsx_state.type == PJSIP_EVENT_TX_MSG)
body = e->body.tsx_state.src.tdata->msg->body;
else
body = e->body.tsx_state.tsx->last_tx->msg->body;
/* Check DTMF content in the INFO message */
if (body && body->len &&
pj_stricmp(&body->content_type.type, &STR_APPLICATION)==0 &&
pj_stricmp(&body->content_type.subtype, &STR_DTMF_RELAY)==0)
{
/* Status of outgoing INFO request */
if (tsx->status_code >= 200 && tsx->status_code < 300) {
PJ_LOG(4,(THIS_FILE,
"Call %d: DTMF sent successfully with INFO",
call->index));
} else if (tsx->status_code >= 300) {
PJ_LOG(4,(THIS_FILE,
"Call %d: Failed to send DTMF with INFO: %d/%.*s",
call->index,
tsx->status_code,
(int)tsx->status_text.slen,
tsx->status_text.ptr));
}
}
/* Check Trickle ICE content in the INFO message */
else if (body && body->len &&
pj_stricmp(&body->content_type.type, &STR_APPLICATION)==0 &&
pj_stricmp(&body->content_type.subtype,
&STR_TRICKLE_ICE_SDP)==0)
{
/* Reset pending SIP INFO for Trickle ICE */
call->trickle_ice.pending_info = PJ_FALSE;
}
} else if (inv->state < PJSIP_INV_STATE_CONFIRMED &&
pjsip_method_cmp(&tsx->method, pjsip_get_invite_method())==0 &&
tsx->state == PJSIP_TSX_STATE_PROCEEDING &&
tsx->status_code/10 == 18)
{
/* Trickle ICE tasks:
* - UAS sending 18x, start 18x retrans
* - UAC receiving 18x, forcefully send SIP INFO & start trickling
*/
pj_bool_t force = call->trickle_ice.trickling<PJSUA_OP_STATE_RUNNING;
pjsua_ice_check_start_trickling(call, force, e);
} else if (tsx->role == PJSIP_ROLE_UAS &&
pjsip_method_cmp(&tsx->method, pjsip_get_prack_method())==0 &&
tsx->state==PJSIP_TSX_STATE_TRYING)
{
/* Trickle ICE tasks:
* - UAS receiving PRACK, start trickling
*/
pjsua_ice_check_start_trickling(call, PJ_FALSE, e);
}
on_return:
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();
if (!call->hanging_up && pjsua_var.ua_cfg.cb.on_call_redirected) {
op = (*pjsua_var.ua_cfg.cb.on_call_redirected)(call->index,
target, e);
} else {
if (!call->hanging_up) {
PJ_LOG(4,(THIS_FILE, "Unhandled redirection for call %d "
"(callback not implemented by application). "
"Disconnecting call.",
call->index));
}
op = PJSIP_REDIRECT_STOP;
}
pj_log_pop_indent();
return op;
}