diff --git a/sflphone-client-gnome/VERSION b/sflphone-client-gnome/VERSION
index 517317e95fb23bda5f1c305dfaa84bfa8021a824..bc440692c3b559a2d3fe749ce99f8993b879252f 100644
--- a/sflphone-client-gnome/VERSION
+++ b/sflphone-client-gnome/VERSION
@@ -1 +1 @@
-0.9.6~beta
+0.9.6~rc1
diff --git a/sflphone-common/VERSION b/sflphone-common/VERSION
index 517317e95fb23bda5f1c305dfaa84bfa8021a824..bc440692c3b559a2d3fe749ce99f8993b879252f 100644
--- a/sflphone-common/VERSION
+++ b/sflphone-common/VERSION
@@ -1 +1 @@
-0.9.6~beta
+0.9.6~rc1
diff --git a/sflphone-common/configure.ac b/sflphone-common/configure.ac
index b39dc0343faa0146a480fe1388fc95212f2c7c65..92c734e3d61ae39012220437756c563118fd5a26 100644
--- a/sflphone-common/configure.ac
+++ b/sflphone-common/configure.ac
@@ -2,7 +2,7 @@ dnl SFLPhone - configure.ac for automake 1.9 and autoconf 2.59
dnl
dnl Process this file with autoconf to produce a configure script.
AC_PREREQ(2.59)
- AC_INIT([SFLphone],[0.9.6~beta],[sflphoneteam@savoirfairelinux.com],[sflphone])
+ AC_INIT([SFLphone],[0.9.6~rc1],[sflphoneteam@savoirfairelinux.com],[sflphone])
AC_COPYRIGHT([[Copyright (c) Savoir-Faire Linux 2004-2009]])
AC_REVISION([$Revision$])
diff --git a/sflphone-common/src/config/config.cpp b/sflphone-common/src/config/config.cpp
index 9c6bb914baf03d73aaa73f94d6f6b2b5340730e5..4f8c02c8de554f762e1963133d452c67122e1654 100644
--- a/sflphone-common/src/config/config.cpp
+++ b/sflphone-common/src/config/config.cpp
@@ -22,6 +22,9 @@
#include "../global.h"
#include <fstream>
#include <cstdlib>
+#include <sys/types.h>
+#include <sys/stat.h>
+#include <errno.h>
namespace Conf
{
@@ -255,7 +258,11 @@ ConfigTree::saveConfigTree (const std::string& fileName)
}
file.close();
-
+
+ if(chmod(fileName.c_str(), S_IRUSR | S_IWUSR)) {
+ _debug("Failed to set permission on configuration file because: %s\n",strerror(errno));
+ }
+
return true;
}
@@ -267,7 +274,7 @@ int
ConfigTree::populateFromFile (const std::string& fileName)
{
bool out = false;
-
+
if (fileName.empty()) {
return 0;
}
@@ -332,7 +339,11 @@ ConfigTree::populateFromFile (const std::string& fileName)
}
file.close();
-
+
+ if(chmod(fileName.c_str(), S_IRUSR | S_IWUSR)) {
+ _debug("Failed to set permission on configuration file because: %s\n",strerror(errno));
+ }
+
return 1;
}
diff --git a/sflphone-common/src/config/config.h b/sflphone-common/src/config/config.h
index 0872889486659d2a7db64634106f36fcb58ce3a0..0fa75a24acd2e580d1ddb457179f8091507f6d3b 100644
--- a/sflphone-common/src/config/config.h
+++ b/sflphone-common/src/config/config.h
@@ -155,7 +155,7 @@ class ConfigTree
* List of sections. Each sections has an ItemList as child
*/
SectionMap _sections;
-
+
friend class ConfigTreeIterator;
public:
diff --git a/sflphone-common/src/global.h b/sflphone-common/src/global.h
index d08d9ce8524dddad65ddb2831421baf27eaa2d62..a6053980b86bb7f1947ff03ca7b9da1f55dee94c 100644
--- a/sflphone-common/src/global.h
+++ b/sflphone-common/src/global.h
@@ -32,7 +32,7 @@
#include <map>
#include <vector>
-#define SFLPHONED_VERSION "0.9.6~beta" /** Version number */
+#define SFLPHONED_VERSION "0.9.6~rc1" /** Version number */
#define HOMEDIR (getenv ("HOME")) /** Home directory */
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index 7a49b2c04a3948237400173cfd5e3859ec6f01c7..345fb7df667cac41ccbd41ddfaa2c03bea97e44a 100644
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -1293,7 +1293,7 @@ ManagerImpl::createSettingsPath (void)
{
_path = std::string (HOMEDIR) + DIR_SEPARATOR_STR + "." + PROGDIR;
- if (mkdir (_path.data(), 0755) != 0) {
+ if (mkdir (_path.data(), 0600) != 0) {
// If directory creation failed
if (errno != EEXIST) {
_debug ("Cannot create directory: %s\n", strerror (errno));
@@ -1952,31 +1952,7 @@ void ManagerImpl::setAudioManager (const int32_t& api)
switchAudioManager();
return;
-
- /*
- int manager;
-
- _debug(" ManagerImpl::setAudioManager :: %i \n",api);
-
- manager = api;
- if( manager == PULSEAUDIO )
- {
- if(app_is_running("pulseaudio") != 0)
- {
- // The pulseaudio daemon is not running
- manager = ALSA;
- notifyErrClient(PULSEAUDIO_NOT_RUNNING);
- }
- }
-
- if(manager == api)
- {
- // it means that we can change the audio manager
- setConfig( PREFERENCES , CONFIG_AUDIO , api) ;
- switchAudioManager();
- }
- */
-
+
}
int32_t
diff --git a/sflphone-common/src/sdp.cpp b/sflphone-common/src/sdp.cpp
index 9256b238b8ca8db8dd1cf6341e296940d5b1c4da..8c79cb36dd98489ccf568fc3bf27e12335a36aff 100644
--- a/sflphone-common/src/sdp.cpp
+++ b/sflphone-common/src/sdp.cpp
@@ -180,7 +180,6 @@ int Sdp::receiving_initial_offer (pjmedia_sdp_session* remote)
// pjmedia_sdp_neg_create_w_remote_offer with the remote offer, and by providing the local offer ( optional )
pj_status_t status;
- pjmedia_sdp_neg_state state;
_debug ("Receiving initial offer\n");
@@ -200,13 +199,68 @@ int Sdp::receiving_initial_offer (pjmedia_sdp_session* remote)
status = pjmedia_sdp_neg_create_w_remote_offer (_pool,
get_local_sdp_session(), remote, &_negociator);
- state = pjmedia_sdp_neg_get_state (_negociator);
-
PJ_ASSERT_RETURN (status == PJ_SUCCESS, 1);
return PJ_SUCCESS;
}
+pj_status_t Sdp::check_sdp_answer(pjsip_inv_session *inv, pjsip_rx_data *rdata)
+{
+ static const pj_str_t str_application = { "application", 11 };
+ static const pj_str_t str_sdp = { "sdp", 3 };
+ pj_status_t status;
+ pjsip_msg * message = NULL;
+ pjmedia_sdp_session * remote_sdp = NULL;
+
+ if (pjmedia_sdp_neg_get_state(inv->neg) == PJMEDIA_SDP_NEG_STATE_LOCAL_OFFER) {
+
+ message = rdata->msg_info.msg;
+
+ if(message == NULL) {
+ _debug("No message");
+ return PJMEDIA_SDP_EINSDP;
+ }
+
+ if (message->body == NULL) {
+ _debug("Empty message body\n");
+ return PJMEDIA_SDP_EINSDP;
+ }
+
+ if (pj_stricmp(&message->body->content_type.type, &str_application) || pj_stricmp(&message->body->content_type.subtype, &str_sdp)) {
+ _debug("Incoming Message does not contain SDP\n");
+ return PJMEDIA_SDP_EINSDP;
+ }
+
+ // Parse the SDP body.
+ status = pjmedia_sdp_parse(rdata->tp_info.pool, (char*)message->body->data, message->body->len, &remote_sdp);
+ if (status == PJ_SUCCESS) {
+ status = pjmedia_sdp_validate(remote_sdp);
+ }
+
+ if (status != PJ_SUCCESS) {
+ _debug("SDP cannot be validated\n");
+ return PJMEDIA_SDP_EINSDP;
+ }
+
+ // This is an answer
+ _debug("Got SDP answer %s\n", pjsip_rx_data_get_info(rdata));
+ status = pjmedia_sdp_neg_set_remote_answer(inv->pool, inv->neg, remote_sdp);
+
+ if (status != PJ_SUCCESS) {
+ _debug("An error occured while processing remote answer %s\n", pjsip_rx_data_get_info(rdata));
+ return PJMEDIA_SDP_EINSDP;
+ }
+
+ // Prefer our codecs to remote when possible
+ pjmedia_sdp_neg_set_prefer_remote_codec_order(inv->neg, 0);
+
+ status = pjmedia_sdp_neg_negotiate(inv->pool, inv->neg, 0);
+ _debug("Negotiation returned with status %d PJ_SUCCESS being %d\n", status, PJ_SUCCESS);
+ }
+
+ return status;
+}
+
void Sdp::sdp_add_protocol (void)
{
this->_local_offer->origin.version = 0;
@@ -298,7 +352,7 @@ void Sdp::clean_session_media()
_session_media.clear();
}
-void Sdp::set_negociated_offer (const pjmedia_sdp_session *sdp)
+void Sdp::set_negotiated_sdp (const pjmedia_sdp_session *sdp)
{
int nb_media, nb_codecs;
@@ -308,7 +362,6 @@ void Sdp::set_negociated_offer (const pjmedia_sdp_session *sdp)
std::string type, dir;
CodecsMap codecs_list;
CodecsMap::iterator iter;
- AudioCodec *codec_to_add;
pjmedia_sdp_attr *attribute;
pjmedia_sdp_rtpmap *rtpmap;
diff --git a/sflphone-common/src/sdp.h b/sflphone-common/src/sdp.h
index ec3178955b57c5370872d43ada2c92302f9f18af..fcd28db15eabc1aabfc0b0d91292829037572204 100644
--- a/sflphone-common/src/sdp.h
+++ b/sflphone-common/src/sdp.h
@@ -25,6 +25,8 @@
#include <pjmedia/sdp_neg.h>
#include <pjsip/sip_transport.h>
#include <pjlib.h>
+#include <pjsip_ua.h>
+#include <pjmedia/errno.h>
#include <pj/pool.h>
#include <pj/assert.h>
@@ -97,6 +99,17 @@ class Sdp {
*/
int receiving_initial_offer( pjmedia_sdp_session* remote );
+ /*
+ * On receiving a message, check if it contains SDP and negotiate. Should be used for
+ * SDP answer and offer but currently is only used for answer.
+ * SDP negociator instance with the remote offer.
+ *
+ * @param inv The the invitation
+ * @param rdata The remote data
+ */
+
+ pj_status_t check_sdp_answer(pjsip_inv_session *inv, pjsip_rx_data *rdata);
+
/*
* Remove all media in the session media vector.
*/
@@ -138,7 +151,7 @@ class Sdp {
*
* @param sdp the negociated offer
*/
- void set_negociated_offer( const pjmedia_sdp_session *sdp );
+ void set_negotiated_sdp ( const pjmedia_sdp_session *sdp );
/*
* Attribute the specified port to every medias provided
diff --git a/sflphone-common/src/sipvoiplink.cpp b/sflphone-common/src/sipvoiplink.cpp
index af41b6d268c760870fcca500204e488df9331432..7d402327517b2d17505b39c2e501abd055f32681 100644
--- a/sflphone-common/src/sipvoiplink.cpp
+++ b/sflphone-common/src/sipvoiplink.cpp
@@ -503,7 +503,6 @@ SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
SIPCall* call = new SIPCall (id, Call::Outgoing, _pool);
-
if (call) {
account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (Manager::instance().getAccountFromCall (id)));
@@ -520,7 +519,6 @@ SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
setCallAudioLocal (call, getLocalIPAddress(), useStun(), getStunServer());
-
try {
_debug ("CREATE NEW RTP SESSION FROM NEWOUTGOINGCALL\n");
_audiortp->createNewSession (call);
@@ -528,9 +526,6 @@ SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
_debug ("Failed to create rtp thread from newOutGoingCall\n");
}
-
-
-
call->initRecFileName();
_debug ("Try to make a call to: %s with call ID: %s\n", toUrl.data(), id.data());
@@ -560,8 +555,6 @@ SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
bool
SIPVoIPLink::answer (const CallID& id)
{
-
- int i;
SIPCall *call;
pj_status_t status;
pjsip_tx_data *tdata;
@@ -1415,7 +1408,6 @@ bool get_dns_server_addresses (std::vector<std::string> *servers)
nameservers.push_back (inet_ntoa (address));
}
- //nameservers.push_back ("192.168.50.3");
*servers = nameservers;
return true;
@@ -1427,8 +1419,6 @@ pj_status_t SIPVoIPLink::enable_dns_srv_resolver (pjsip_endpoint *endpt, pj_dns_
pj_status_t status;
pj_dns_resolver *resv;
std::vector <std::string> dns_servers;
- pj_uint16_t port = 5353;
- pjsip_resolver_t *res;
int scount, i;
// Create the DNS resolver instance
@@ -1706,13 +1696,10 @@ int SIPVoIPLink::createUDPServer (void)
pj_sockaddr_in bound_addr;
pjsip_host_port a_name;
char tmpIP[32];
- pj_sock_t sock;
-
// Init bound address to ANY
pj_memset (&bound_addr, 0, sizeof (bound_addr));
-
bound_addr.sin_addr.s_addr = pj_htonl (PJ_INADDR_ANY);
bound_addr.sin_port = pj_htons ( (pj_uint16_t) _localPort);
bound_addr.sin_family = PJ_AF_INET;
@@ -1875,22 +1862,17 @@ void SIPVoIPLink::handle_reinvite (SIPCall *call)
}
}
-
-/*******************************/
-/* CALLBACKS IMPLEMENTATION */
-/*******************************/
-
+// This callback is called when the invite session state has changed
void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
{
-
_debug ("--------------------- call_on_state_changed --------------------- %i\n", inv->state);
SIPCall *call;
AccountID accId;
SIPVoIPLink *link;
pjsip_rx_data *rdata;
-
-
+ pj_status_t status;
+
/* Retrieve the call information */
call = reinterpret_cast<SIPCall*> (inv->mod_data[_mod_ua.id]);
@@ -1900,7 +1882,7 @@ void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
//Retrieve the body message
rdata = e->body.tsx_state.src.rdata;
-
+
/* If this is an outgoing INVITE that was created because of
* REFER/transfer, send NOTIFY to transferer.
*/
@@ -1961,84 +1943,79 @@ void call_on_state_changed (pjsip_inv_session *inv, pjsip_event *e)
}
}
}
- } else {
+
+ return;
+ }
+
+ // The call is ringing - We need to handle this case only on outgoing call
+ if (inv->state == PJSIP_INV_STATE_EARLY && e->body.tsx_state.tsx->role == PJSIP_ROLE_UAC) {
+ call->setConnectionState (Call::Ringing);
+ Manager::instance().peerRingingCall (call->getCallId());
+ }
+ // After 2xx is sent/received.
+ else if (inv->state == PJSIP_INV_STATE_CONNECTING) {
+ status = call->getLocalSDP()->check_sdp_answer (inv, rdata);
+ if (status != PJ_SUCCESS) {
+ _debug("Failed to check_incoming_sdp in call_on_state_changed\n");
+ return;
+ }
+ }
+ // After we sent or received a ACK - The connection is established
+ else if (inv->state == PJSIP_INV_STATE_CONFIRMED) {
- // The call is ringing - We need to handle this case only on outgoing call
- if (inv->state == PJSIP_INV_STATE_EARLY && e->body.tsx_state.tsx->role == PJSIP_ROLE_UAC) {
- call->setConnectionState (Call::Ringing);
- Manager::instance().peerRingingCall (call->getCallId());
+ /* If the call is a direct IP-to-IP call */
+ if (call->getCallConfiguration () == Call::IPtoIP) {
+ link = SIPVoIPLink::instance ("");
+ } else {
+ accId = Manager::instance().getAccountFromCall (call->getCallId());
+ link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accId));
}
- // We receive a ACK - The connection is established
- else if (inv->state == PJSIP_INV_STATE_CONFIRMED) {
+ if (link)
+ link->SIPCallAnswered (call, rdata);
+ }
+ else if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
+ _debug ("------------------- Call disconnected ---------------------\n");
+ _debug ("State: %i, Disconnection cause: %i\n", inv->state, inv->cause);
- /* If the call is a direct IP-to-IP call */
- if (call->getCallConfiguration () == Call::IPtoIP) {
- link = SIPVoIPLink::instance ("");
- } else {
+ switch (inv->cause) {
+ /* The call terminates normally - BYE / CANCEL */
+ case PJSIP_SC_OK:
+ case PJSIP_SC_DECLINE:
+ case PJSIP_SC_REQUEST_TERMINATED:
accId = Manager::instance().getAccountFromCall (call->getCallId());
link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accId));
- }
-
- if (link)
- link->SIPCallAnswered (call, rdata);
- }
-
- else if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
- int count = 0;
- _debug ("------------------- Call disconnected ---------------------\n");
- _debug ("State: %i, Disconnection cause: %i\n", inv->state, inv->cause);
-
- switch (inv->cause) {
- /* The call terminates normally - BYE / CANCEL */
-
- case PJSIP_SC_OK:
-
- case PJSIP_SC_DECLINE:
-
- case PJSIP_SC_REQUEST_TERMINATED:
-
- accId = Manager::instance().getAccountFromCall (call->getCallId());
- link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accId));
-
- if (link) {
- link->SIPCallClosed (call);
- }
-
- break;
-
- /* The call connection failed */
-
- case PJSIP_SC_NOT_FOUND: /* peer not found */
-
- case PJSIP_SC_REQUEST_TIMEOUT: /* request timeout */
-
- case PJSIP_SC_NOT_ACCEPTABLE_HERE: /* no compatible codecs */
- case PJSIP_SC_NOT_ACCEPTABLE_ANYWHERE:
-
- case PJSIP_SC_UNSUPPORTED_MEDIA_TYPE:
-
- case PJSIP_SC_UNAUTHORIZED:
-
- case PJSIP_SC_REQUEST_PENDING:
- accId = Manager::instance().getAccountFromCall (call->getCallId());
- link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accId));
+ if (link) {
+ link->SIPCallClosed (call);
+ }
- if (link) {
- link->SIPCallServerFailure (call);
- }
+ break;
+ /* The call connection failed */
+ case PJSIP_SC_NOT_FOUND: /* peer not found */
+ case PJSIP_SC_REQUEST_TIMEOUT: /* request timeout */
+ case PJSIP_SC_NOT_ACCEPTABLE_HERE: /* no compatible codecs */
+ case PJSIP_SC_NOT_ACCEPTABLE_ANYWHERE:
+ case PJSIP_SC_UNSUPPORTED_MEDIA_TYPE:
+ case PJSIP_SC_UNAUTHORIZED:
+ case PJSIP_SC_REQUEST_PENDING:
+ accId = Manager::instance().getAccountFromCall (call->getCallId());
+ link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink (accId));
- break;
+ if (link) {
+ link->SIPCallServerFailure (call);
+ }
- default:
- _debug ("sipvoiplink.cpp - line %d : Unhandled call state. This is probably a bug.\n", __LINE__);
- break;
- }
+ break;
+ default:
+ _debug ("sipvoiplink.cpp - line %d : Unhandled call state. This is probably a bug.\n", __LINE__);
+ break;
}
}
+
}
+// This callback is called after SDP offer/answer session has completed.
void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
{
_debug ("--------------------- call_on_media_update --------------------- \n");
@@ -2048,38 +2025,42 @@ void call_on_media_update (pjsip_inv_session *inv, pj_status_t status)
SIPVoIPLink * link = NULL;
SIPCall * call;
-
- if (status != PJ_SUCCESS) {
- _debug ("Error while negociating the offer\n");
- return;
- }
-
- // Get the new sdp, result of the negociation
- pjmedia_sdp_neg_get_active_local (inv->neg, &local_sdp);
- pjmedia_sdp_neg_get_active_remote(inv->neg, &remote_sdp);
-
+
call = reinterpret_cast<SIPCall *> (inv->mod_data[getModId() ]);
if (!call) {
_debug ("Call declined by peer, SDP negociation stopped\n");
return;
}
-
- // Clean the resulting sdp offer to create a new one (in case of a reinvite)
- call->getLocalSDP()->clean_session_media();
-
- // Set the fresh negociated one
- call->getLocalSDP()->set_negociated_offer (local_sdp);
- // Set remote ip / port
- call->getLocalSDP()->set_media_transport_info_from_remote_sdp (remote_sdp);
-
link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(AccountNULL));
if(link == NULL) {
_debug ("Failed to get sip link\n");
return;
}
+ if (status != PJ_SUCCESS) {
+ _debug ("Error while negotiating the offer\n");
+ link->hangup(call->getCallId());
+ Manager::instance().callFailure(call->getCallId());
+ return;
+ }
+
+ // Get the new sdp, result of the negotiation
+ pjmedia_sdp_neg_get_active_local (inv->neg, &local_sdp);
+ pjmedia_sdp_neg_get_active_remote (inv->neg, &remote_sdp);
+
+ // Clean the resulting sdp offer to create a new one (in case of a reinvite)
+ call->getLocalSDP()->clean_session_media();
+
+ // Set the fresh negotiated one, no matter if that was an offer or answer.
+ // The local sdp is updated in case of an answer, even if the remote sdp
+ // is kept internally.
+ call->getLocalSDP()->set_negotiated_sdp (local_sdp);
+
+ // Set remote ip / port
+ call->getLocalSDP()->set_media_transport_info_from_remote_sdp (remote_sdp);
+
try {
call->setAudioStart (true);
link->getAudioRtp()->start();
@@ -2107,11 +2088,8 @@ void call_on_tsx_changed (pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_
void regc_cb (struct pjsip_regc_cbparam *param)
{
-
- //AccountID *id = static_cast<AccountID *> (param->token);
SIPAccount *account;
- //_debug("UserAgent: Account ID is %s, Register result: %d, Status: %d\n", id->data(), param->status, param->code);
account = static_cast<SIPAccount *> (param->token);
if (!account)
@@ -2167,11 +2145,10 @@ void regc_cb (struct pjsip_regc_cbparam *param)
}
+// Optional function to be called to process incoming request message.
pj_bool_t
mod_on_rx_request (pjsip_rx_data *rdata)
{
-
-
pj_status_t status;
pj_str_t reason;
unsigned options = 0;
@@ -2325,7 +2302,6 @@ mod_on_rx_request (pjsip_rx_data *rdata)
get_remote_sdp_from_offer (rdata, &r_sdp);
-// _debug("r_sdp = %s\n", r_sdp);
status = call->getLocalSDP()->receiving_initial_offer (r_sdp);
if (status!=PJ_SUCCESS) {
@@ -2334,7 +2310,6 @@ mod_on_rx_request (pjsip_rx_data *rdata)
return false;
}
-
call->setConnectionState (Call::Progressing);
call->setPeerNumber (peerNumber);
@@ -2364,7 +2339,6 @@ mod_on_rx_request (pjsip_rx_data *rdata)
return true;
}
-
// Specify media capability during invite session creation
status = pjsip_inv_create_uas (dialog, rdata, call->getLocalSDP()->get_local_sdp_session(), 0, &inv);