diff --git a/configure.ac b/configure.ac
index 70317736ea516d075bcc27e1549ab36ec0bbb540..ec2c87cdc2fe7e21de971e55232fb682d2229bf8 100644
--- a/configure.ac
+++ b/configure.ac
@@ -126,9 +126,7 @@ AC_CHECK_FUNCS([bzero floor gethostbyname gethrtime gettimeofday \
inet_ntoa memset mkdir pathconf pow regcomp select setlocale socket \
strchr strdup strerror strrchr strstr strtol utime])
-dnl Check for exosip2
-LP_CHECK_EXOSIP2
-SFLPHONE_LIBS="$SFLPHONE_LIBS $EXOSIP_LIBS"
+SFLPHONE_LIBS="$SFLPHONE_LIBS"
dnl Check for GNU ccRTP
PKG_PROG_PKG_CONFIG
@@ -139,12 +137,6 @@ if test "x${have_libpj}" = "xfalse" ; then
AC_MSG_ERROR([PJSIP not found. http://www.pjsip.org/download.htm])
fi
-LIBOSIP2_OSIPLISTNOPOINTER_MIN_VERSION=3.0.0
-PKG_CHECK_MODULES([libosip2], [libosip2 >= ${LIBOSIP2_OSIPLISTNOPOINTER_MIN_VERSION}],
- [LIBOSIP2_CFLAGS=""], [LIBOSIP2_CFLAGS="-DLIBOSIP2_WITHPOINTER"])
-SIP_CFLAGS="$SIP_CFLAGS $LIBOSIP2_CFLAGS";
-AC_SUBST(SIP_CFLAGS)
-
LIBASOUND2_MIN_VERSION=1.0
PKG_CHECK_MODULES(alsa, alsa >= ${LIBASOUND2_MIN_VERSION})
SFLPHONE_CFLAGS="$SFLPHONE_CFLAGS $libasound2_CFLAGS"
@@ -349,26 +341,6 @@ DBUS_SERVICES_DIR="$datadir/dbus-1/services"
AC_SUBST(DBUS_SERVICES_DIR)
AC_DEFINE_UNQUOTED(DBUS_SERVICES_DIR, "$DBUS_SERVICES_DIR", [Where services dir for DBUS is])
-
-dnl QT interface
-AC_MSG_CHECKING([whether to enable sflphoneqt])
-AC_ARG_ENABLE(sflphoneqt,
- AS_HELP_STRING(
- [--enable-sflphoneqt],
- [enable sflphone-qt compilation @<:@default=no@:>@]
- ),
- [with_sflphoneqt=$enableval],
- [with_sflphoneqt=no]
-)
-AM_CONDITIONAL(WITH_QT, test "x$with_sflphoneqt" = "xyes")
-AM_CONDITIONAL(include_x11, test x$with_sflphoneqt = "xyes")
-if test "x$with_sflphoneqt" = "xno"; then
- AC_MSG_RESULT(no)
-else
- AC_MSG_RESULT(yes)
- gw_CHECK_QT
-fi
-
# Go!
AC_SUBST(SFLPHONE_CFLAGS)
AC_SUBST(SFLPHONE_LIBS)
diff --git a/src/Makefile.am b/src/Makefile.am
index 96bdd373a5244f6338fb4034a95e5110bb155193..8f178a13f104814ce45a124e3be53074cee7fb62 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -54,7 +54,7 @@ sflphoned_SOURCES = \
sflphoned_CXXFLAGS = -DPREFIX=\"$(prefix)\" -DPROGSHAREDIR=\"${datadir}/sflphone\" $(ZEROCONFFLAGS) $(IAX_FLAGS) $(SFLPHONE_CFLAGS) $(SIP_CFLAGS)
#sflphoned_LDFLAGS = -static
-sflphoned_LDADD = ./libsflphone.la $(SFLPHONE_LIBS) $(ZEROCONFLIB) $(LIB_DNSSD) $(IAX_LIBS) $(EXOSIP_LIBS) $(ALSAFLAG) $(PULSEAUDIO_LIBS) $(PJSIP_LIBS)
+sflphoned_LDADD = ./libsflphone.la $(SFLPHONE_LIBS) $(ZEROCONFLIB) $(LIB_DNSSD) $(IAX_LIBS) $(ALSAFLAG) $(PULSEAUDIO_LIBS) $(PJSIP_LIBS)
AM_CPPFLAGS = -I$(top_srcdir)/include -I$(top_srcdir)/libs $(libccext2_CFLAGS) $(libccgnu2_CFLAGS) $(IAX_CFLAGS) $(USER_INCLUDES) $(libdbuscpp_CFLAGS) \
-DCODECS_DIR=\""$(sflcodecdir)"\"
-DENABLE_TRACE
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
old mode 100755
new mode 100644
index ee874334bec661fcc020e14ac27d7f11d7f89d1a..1e69b0010fad954b4ea812664d78e0d4e3143ee4
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -698,7 +698,6 @@ ManagerImpl::callFailure(const CallID& id)
void
ManagerImpl::startVoiceMessageNotification(const AccountID& accountId, int nb_msg)
{
- _debug("cccccccccccccccccc\n");
if (_dbus) _dbus->getCallManager()->voiceMailNotify(accountId, nb_msg) ;
}
diff --git a/src/managerimpl.h b/src/managerimpl.h
old mode 100755
new mode 100644
diff --git a/src/sipcall.cpp b/src/sipcall.cpp
old mode 100755
new mode 100644
index ee6a51978fd21ce1946c4eacf270581ec9687421..451283960392d816897890a0951af3dfd62f26f1
--- a/src/sipcall.cpp
+++ b/src/sipcall.cpp
@@ -60,7 +60,7 @@ SIPCall::SIPCallInvite(pjsip_rx_data *rdata, pj_pool_t *pool)
_localSDP->origin.version = 0;
sdpAddOrigin();
- _localSDP->name = pj_str("sflphone");
+ _localSDP->name = pj_str((char*)"sflphone");
sdpAddConnectionInfo();
_localSDP->time.start = _localSDP->time.stop = 0;
sdpAddMediaDescription(pool);
@@ -101,145 +101,6 @@ SIPCall::SIPCallInvite(pjsip_rx_data *rdata, pj_pool_t *pool)
return true;
}
-bool
-SIPCall::SIPCallReinvite(eXosip_event_t *event)
-{
- if (event->cid < 1 && event->did < 1) {
- _debug("SIP Failure: Invalid cid and did\n");
- return false;
- }
-
- if (event->request == NULL) {
- _debug("SIP Failure: No request into the event\n");
- return false;
- }
-
- setCid(event->cid);
- setDid(event->did);
- setTid(event->tid);
-
- setPeerInfoFromRequest(event);
-
- sdp_message_t* remote_sdp = getRemoteSDPFromRequest(event);
- if (remote_sdp == 0) {
- return false;
- }
-
- sdp_media_t* remote_med = getRemoteMedia(event->tid, remote_sdp);
- if (remote_med == 0) {
- sdp_message_free (remote_sdp);
- return false;
- }
-
- /*if (!setRemoteAudioFromSDP(remote_med, remote_sdp)) {
- _debug("SIP Failure: unable to set IP address and port from SDP\n");
- sdp_message_free (remote_sdp);
- return false;
- }
-
- if (!setAudioCodecFromSDP(remote_med, event->tid)) {
- sdp_message_free (remote_sdp);
- return false;
- }*/
-
- osip_message_t *answer = 0;
- eXosip_lock();
- _debug("< Building Answer 200\n");
- if (0 == eXosip_call_build_answer (event->tid, 200, &answer)) {
- if ( 0 != sdp_complete_message(remote_sdp, answer)) {
- osip_message_free(answer);
- // Send 415 Unsupported media type
- eXosip_call_send_answer (event->tid, 415, NULL);
- _debug("< Sending Answer 415\n");
- } else {
-
- sdp_message_t *local_sdp = eXosip_get_sdp_info(answer);
- sdp_media_t *local_med = NULL;
- if (local_sdp != NULL) {
- local_med = eXosip_get_audio_media(local_sdp);
- }
- if (local_sdp != NULL && local_med != NULL) {
- /* search if stream is sendonly or recvonly */
- int _remote_sendrecv = sdp_analyse_attribute (remote_sdp, remote_med);
- int _local_sendrecv = sdp_analyse_attribute (local_sdp, local_med);
- _debug(" Remote SendRecv: %d\n", _remote_sendrecv);
- _debug(" Local SendRecv: %d\n", _local_sendrecv);
- if (_local_sendrecv == _SENDRECV) {
- if (_remote_sendrecv == _SENDONLY) { _local_sendrecv = _RECVONLY; }
- else if (_remote_sendrecv == _RECVONLY) { _local_sendrecv = _SENDONLY; }
- }
- _debug(" Final Local SendRecv: %d\n", _local_sendrecv);
- sdp_message_free (local_sdp);
- }
- _debug("< Sending answer 200\n");
- if (0 != eXosip_call_send_answer (event->tid, 200, answer)) {
- _debug("SipCall::newIncomingCall: cannot send 200 OK?\n");
- }
- }
- }
- eXosip_unlock ();
- sdp_message_free (remote_sdp);
- return true;
-}
-
-bool
-SIPCall::SIPCallAnswered(eXosip_event_t *event)
-{
- if (event->cid < 1 && event->did < 1) {
- _debug("SIP Failure: Invalid cid and did\n");
- return false;
- }
-
- if (event->request == NULL) {
- _debug("SIP Failure: No request into the event\n");
- return false;
- }
-
- setCid(event->cid);
- setDid(event->did);
-
- //setPeerInfoFromResponse()
-
- eXosip_lock ();
- {
- osip_message_t *ack = NULL;
- int i;
- i = eXosip_call_build_ack (event->did, &ack);
- if (i != 0) {
- _debug("SipCall::answeredCall: Cannot build ACK for call!\n");
- } else {
- sdp_message_t *local_sdp = NULL;
- sdp_message_t *remote_sdp = NULL;
-
- if (event->request != NULL && event->response != NULL) {
- local_sdp = eXosip_get_sdp_info (event->request);
- remote_sdp = eXosip_get_sdp_info (event->response);
- }
- if (local_sdp == NULL && remote_sdp != NULL) {
- /* sdp in ACK */
- i = sdp_complete_message (remote_sdp, ack);
- if (i != 0) {
- _debug("SipCall::answeredCall: Cannot complete ACK with sdp body?!\n");
- }
- }
- sdp_message_free (local_sdp);
- sdp_message_free (remote_sdp);
-
- _debug("< Send ACK\n");
- eXosip_call_send_ack (event->did, ack);
- }
- }
- eXosip_unlock ();
- return true;
-}
-
-
-bool
-SIPCall::SIPCallAnsweredWithoutHold(eXosip_event_t* event)
-{
- return true;
-}
-
bool
SIPCall::SIPCallAnsweredWithoutHold(pjsip_rx_data *rdata)
{
@@ -270,193 +131,6 @@ SIPCall::SIPCallAnsweredWithoutHold(pjsip_rx_data *rdata)
}
-int
-SIPCall::sdp_complete_message(sdp_message_t * remote_sdp, osip_message_t * msg)
-{
- // Format port to a char*
- if (remote_sdp == NULL) {
- _debug("SipCall::sdp_complete_message: No remote SDP body found for call\n");
- return -1;
- }
- if (msg == NULL) {
- _debug("SipCall::sdp_complete_message: No message to complete\n");
- return -1;
- }
-
- std::ostringstream media;
-
- // for each medias
- int iMedia = 0;
- char *tmp = NULL;
- #ifdef LIBOSIP2_WITHPOINTER
- const osip_list_t* remote_sdp_m_medias = remote_sdp->m_medias; // old abi
- #else
- const osip_list_t* remote_sdp_m_medias = &(remote_sdp->m_medias);
- #endif
- osip_list_t* remote_med_m_payloads = 0;
-
- while (!osip_list_eol(remote_sdp_m_medias, iMedia)) {
- sdp_media_t *remote_med = (sdp_media_t *)osip_list_get(remote_sdp_m_medias, iMedia);
- if (remote_med == 0) { continue; }
-
- if (0 != osip_strcasecmp (remote_med->m_media, "audio")) {
- // if this is not an "audio" media, we set it to 0
- media << "m=" << remote_med->m_media << " 0 " << remote_med->m_proto << " \r\n";
- } else {
- std::ostringstream listCodec;
- std::ostringstream listRtpMap;
-
- // search for compatible codec: foreach payload
- int iPayload = 0;
- #ifdef LIBOSIP2_WITHPOINTER
- remote_med_m_payloads = remote_med->m_payloads; // old abi
- #else
- remote_med_m_payloads = &(remote_med->m_payloads);
- #endif
-
- //while (!osip_list_eol(remote_med_m_payloads, iPayload) && iPayload < 2) {
- while (!osip_list_eol(remote_med_m_payloads, iPayload)) {
- tmp = (char *)osip_list_get(remote_med_m_payloads, iPayload);
- if (tmp!=NULL) {
- int payload = atoi(tmp);
- _debug("remote payload = %s\n", tmp);
- AudioCodecType audiocodec = (AudioCodecType)payload;
- if (audiocodec != (AudioCodecType)-1 && _codecMap.isActive(audiocodec)) {
- listCodec << payload << " ";
- //listRtpMap << "a=rtpmap:" << payload << " " << audiocodec->getCodecName() << "/" << audiocodec->getClockRate();
- listRtpMap << "a=rtpmap:" << payload << " " << _codecMap.getCodecName(audiocodec) << "/" << _codecMap.getSampleRate(audiocodec);
- if (_codecMap.getChannel(audiocodec) != 1) {
- listRtpMap << "/" << _codecMap.getChannel(audiocodec);
- }
- listRtpMap << "\r\n";
- }
- }
- iPayload++;
- }
- if (listCodec.str().empty()) {
- media << "m=" << remote_med->m_media << " 0 " << remote_med->m_proto << " \r\n";
- } else {
- // we add the media line + a=rtpmap list
- media << "m=" << remote_med->m_media << " " << getLocalExternAudioPort() << " RTP/AVP " << listCodec.str() << "\r\n";
- media << listRtpMap.str();
- }
- }
- iMedia++;
- }
- char buf[4096];
- snprintf (buf, 4096,
- "v=0\r\n"
- "o=user 0 0 IN IP4 %s\r\n"
- "s=session\r\n"
- "c=IN IP4 %s\r\n"
- "t=0 0\r\n"
- "%s\n", getLocalIp().c_str(), getLocalIp().c_str(), media.str().c_str());
-
- osip_message_set_body (msg, buf, strlen (buf));
- osip_message_set_content_type (msg, "application/sdp");
- _debug(" sdp: %s", buf);
- return 0;
-}
-
-
-
-int
-SIPCall::sdp_analyse_attribute (sdp_message_t * sdp, sdp_media_t * med)
-{
- int pos;
- int pos_media;
-
- /* test media attributes */
- pos = 0;
- #ifdef LIBOSIP2_WITHPOINTER
- const osip_list_t* med_a_attributes = med->a_attributes; // old abi
- #else
- const osip_list_t* med_a_attributes = &(med->a_attributes);
- #endif
- while (!osip_list_eol (med_a_attributes, pos)) {
- sdp_attribute_t *at;
-
- at = (sdp_attribute_t *) osip_list_get (med_a_attributes, pos);
- if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "sendonly")) {
- return _SENDONLY;
- } else if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "recvonly")) {
- return _RECVONLY;
- } else if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "sendrecv")) {
- return _SENDRECV;
- }
- pos++;
- }
-
- /* test global attributes */
- pos_media = -1;
- pos = 0;
- #ifdef LIBOSIP2_WITHPOINTER
- const osip_list_t* sdp_a_attributes = sdp->a_attributes; // old abi
- #else
- const osip_list_t* sdp_a_attributes = &(sdp->a_attributes);
- #endif
- while (!osip_list_eol (sdp_a_attributes, pos)) {
- sdp_attribute_t *at;
-
- at = (sdp_attribute_t *) osip_list_get (sdp_a_attributes, pos);
- if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "sendonly")) {
- return _SENDONLY;
- } else if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "recvonly")) {
- return _RECVONLY;
- } else if (at->a_att_field != NULL &&
- 0 == strcmp (at->a_att_field, "sendrecv")) {
- return _SENDRECV;
- }
- pos++;
- }
-
- return _SENDRECV;
-}
-
-bool
-SIPCall::setPeerInfoFromRequest(eXosip_event_t *event)
-{
- // event->request should not be NULL!
- char remote_uri[256] = "";
- std::string name("");
- std::string number("");
-
- char *tmp = NULL;
- osip_from_to_str(event->request->from, &tmp);
- if (tmp != NULL) {
- snprintf (remote_uri, 255, "%s", tmp);
- remote_uri[255] = '\0';
- osip_free (tmp);
-
- // Get the name/number
- osip_from_t *from;
- osip_from_init(&from);
- osip_from_parse(from, remote_uri);
- char *tmpname = osip_from_get_displayname(from);
- if ( tmpname != NULL ) {
- name = tmpname;
- }
- osip_uri_t* url = osip_from_get_url(from);
- if ( url != NULL && url->username != NULL) {
- number = url->username;
- }
- osip_from_free(from);
- }
-
- _debug(" Name: %s\n", name.c_str());
- _debug(" Number: %s\n", number.c_str());
- _debug(" Remote URI: %s\n", remote_uri);
-
- setPeerName(name);
- setPeerNumber(number);
- return true;
-}
-
pjmedia_sdp_session*
SIPCall::getRemoteSDPFromRequest(pjsip_rx_data *rdata)
{
@@ -472,27 +146,6 @@ SIPCall::getRemoteSDPFromRequest(pjsip_rx_data *rdata)
return sdp;
}
-sdp_media_t*
-SIPCall::getRemoteMedia(int tid, sdp_message_t* remote_sdp)
-{
- // Remote Media Port
- eXosip_lock();
- sdp_media_t *remote_med = eXosip_get_audio_media(remote_sdp);
- eXosip_unlock();
-
- if (remote_med == NULL || remote_med->m_port == NULL) {
- // no audio media proposed
- _debug("SIP Failure: unsupported media\n");
- _debug("< Sending 415 Unsupported media type\n");
- eXosip_lock();
- eXosip_call_send_answer (tid, 415, NULL);
- eXosip_unlock();
- sdp_message_free (remote_sdp);
- return 0;
- }
- return remote_med;
-}
-
bool
SIPCall::setRemoteAudioFromSDP(pjmedia_sdp_session* remote_sdp, pjmedia_sdp_media *remote_med)
{
@@ -538,9 +191,9 @@ void SIPCall::sdpAddOrigin( void )
// Use Network Time Protocol format timestamp to ensure uniqueness.
_localSDP->origin.id = tv.sec + 2208988800UL;
// The type of network ( IN for INternet )
- _localSDP->origin.net_type = pj_str("IN"); //STR_IN;
+ _localSDP->origin.net_type = pj_str((char*)"IN"); //STR_IN;
// The type of address
- _localSDP->origin.addr_type = pj_str("IP4"); //STR_IP4;
+ _localSDP->origin.addr_type = pj_str((char*)"IP4"); //STR_IP4;
// The address of the machine from which the session was created
_localSDP->origin.addr = pj_str( (char*)_ipAddr.c_str() );
}
@@ -562,10 +215,10 @@ void SIPCall::sdpAddMediaDescription(pj_pool_t* pool)
//nbMedia = getSDPMediaList().size();
_localSDP->media_count = 1;
- med->desc.media = pj_str("audio");
+ med->desc.media = pj_str((char*)"audio");
med->desc.port_count = 1;
med->desc.port = getLocalExternAudioPort();
- med->desc.transport = pj_str("RTP/AVP");
+ med->desc.transport = pj_str((char*)"RTP/AVP");
CodecsMap::iterator itr;
itr = _codecMap.getCodecsMap().begin();
@@ -642,7 +295,7 @@ bool SIPCall::createInitialOffer(pj_pool_t *pool)
_localSDP->origin.version = 0;
sdpAddOrigin();
- _localSDP->name = pj_str("sflphone");
+ _localSDP->name = pj_str((char*)"sflphone");
sdpAddConnectionInfo();
_localSDP->time.start = _localSDP->time.stop = 0;
sdpAddMediaDescription(pool);
diff --git a/src/sipcall.h b/src/sipcall.h
old mode 100755
new mode 100644
index d5b47a33aa9ac45a7b8869fca19d080ec52a7bc4..ab87d191825b4c4e25bdf1d12fccadee06f51fd3
--- a/src/sipcall.h
+++ b/src/sipcall.h
@@ -25,8 +25,6 @@
#include "audio/codecDescriptor.h"
#include "sipmanager.h"
-#include <eXosip2/eXosip.h>
-
class AudioCodec;
/**
@@ -52,37 +50,37 @@ class SIPCall : public Call
/**
* Call Identifier
- * @return int SIP call id : protected by eXosip lock
+ * @return int SIP call id
*/
int getCid() { return _cid; }
/**
* Call Identifier
- * @param cid SIP call id : protected by eXosip lock
+ * @param cid SIP call id
*/
void setCid(int cid) { _cid = cid ; }
/**
* Domain identifier
- * @return int SIP domain id : protected by eXosip lock
+ * @return int SIP domain id
*/
int getDid() { return _did; }
/**
* Domain identifier
- * @param did SIP domain id : protected by eXosip lock
+ * @param did SIP domain id
*/
void setDid(int did) { _did = did; }
/**
* Transaction identifier
- * @return int SIP transaction id : protected by eXosip lock
+ * @return int SIP transaction id
*/
int getTid() { return _tid; }
/**
* Transaction identifier
- * @param tid SIP transaction id : protected by eXosip lock
+ * @param tid SIP transaction id
*/
void setTid(int tid) { _tid = tid; }
@@ -95,31 +93,8 @@ class SIPCall : public Call
*/
bool SIPCallInvite(pjsip_rx_data *rdata, pj_pool_t *pool);
- /**
- * newReinviteCall is called when the IP-Phone user receives a change in the call
- * it's almost an newIncomingCall but we send a 200 OK
- * See: 3.7. Session with re-INVITE (IP Address Change)
- * @param event eXosip Event
- * @return bool True if ok
- */
- bool SIPCallReinvite(eXosip_event_t *event);
-
- /**
- * Peer answered to a call (on hold or not)
- * @param event eXosip Event
- * @return bool True if ok
- */
- bool SIPCallAnswered(eXosip_event_t *event);
-
- /** No longer being used */
- bool SIPCallAnsweredWithoutHold(eXosip_event_t *event);
-
bool SIPCallAnsweredWithoutHold(pjsip_rx_data *rdata);
- /** No longer being used */
- int sdp_complete_message(sdp_message_t * remote_sdp, osip_message_t * msg);
-
-
/**
* Save IP Address
* @param ip std::string
@@ -155,30 +130,13 @@ class SIPCall : public Call
pjsip_inv_session *getInvSession() {return _invSession;}
private:
-
- /** No longer being used */
- int sdp_analyse_attribute (sdp_message_t * sdp, sdp_media_t * med);
-
- /**
- * Set peer name and number with event->request->from
- * @param event eXosip event
- * @return bool False if the event is invalid
- */
- bool setPeerInfoFromRequest(eXosip_event_t *event);
-
/**
* Get a valid remote SDP or return a 400 bad request response if invalid
- * @param event eXosip event
- * @return sdp_message_t* A valid remote_sdp or 0
+ * @param
+ * @return
*/
pjmedia_sdp_session* getRemoteSDPFromRequest(pjsip_rx_data *rdata);
- /** No longer being used */
- sdp_message_t *getRemoteSDPFromRequest(eXosip_event_t*&){return NULL;}
-
- /** No longer being used */
- sdp_media_t* getRemoteMedia(int tid, sdp_message_t* remote_sdp);
-
/**
* Get a valid remote media
* @param remote_sdp pjmedia_sdp_session*
diff --git a/src/sipmanager.cpp b/src/sipmanager.cpp
old mode 100755
new mode 100644
index 27d443edf532bf4ba2f82a7255c8db97f0a66ea4..4356078c8edd8ffaa67760853b9b9cca1af47378
--- a/src/sipmanager.cpp
+++ b/src/sipmanager.cpp
@@ -622,7 +622,6 @@ void SIPManager::set_voicemail_info( AccountID account, pjsip_msg_body *body ){
// We get the notification body
msg_body = (char*)body->data;
- std::cout << " body message " << msg_body.c_str() << std::endl;
// We need the position of the first character of the string voice_str
pos_begin = msg_body.find(voice_str);
@@ -699,7 +698,6 @@ pj_bool_t SIPManager::mod_on_rx_request(pjsip_rx_data *rdata) {
method_name = "NOTIFY";
// Retrieve all the message. Should contains only the method name but ...
request = rdata->msg_info.msg->line.req.method.name.ptr;
- _debug("request = %s\n", request.c_str());
// Check if the message is a notification
if( request.find( method_name ) != -1 ) {
set_voicemail_info( account_id, rdata->msg_info.msg->body );
diff --git a/src/sipmanager.h b/src/sipmanager.h
old mode 100755
new mode 100644
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
old mode 100755
new mode 100644
index 12ea636697523a5bbf74f9ba08e4d44a704600e0..498008be267362c2d0a292d7f2d88f0b859834d8
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -27,7 +27,7 @@
#include "sipaccount.h"
#include "sipmanager.h"
#include "audio/audiortp.h"
-
+
#include "manager.h"
#include "user_cfg.h" // SIGNALISATION / PULSE #define
@@ -36,29 +36,16 @@
#define RANDOM_SIP_PORT rand() % 64000 + 1024
#define RANDOM_LOCAL_PORT ((rand() % 27250) + 5250)*2
-#define EXOSIP_ERROR_NO 0
-#define EXOSIP_ERROR_STD -1
-#define EXOSIP_ERROR_BUILDING -2
-
// 1XX responses
#define DIALOG_ESTABLISHED 101
// see: osip_const.h
-// FOR VOICE Message handling
-#define VOICE_MSG "Voice-Message"
-#define LENGTH_VOICE_MSG 15
-
// need for hold/unhold
#define INVITE_METHOD "INVITE"
-
SIPVoIPLink::SIPVoIPLink(const AccountID& accountID)
- : VoIPLink(accountID), _localExternAddress("") , eXosip_running( false )
+ : VoIPLink(accountID), _localExternAddress("")
{
- _evThread = NULL;//new EventThread(this);
-
- _nMsgVoicemail = 0;
- _eXosipRegID = EXOSIP_ERROR_STD;
_nbTryListenAddr = 2; // number of times to try to start SIP listener
_localExternPort = 0;
@@ -71,87 +58,11 @@ SIPVoIPLink::SIPVoIPLink(const AccountID& accountID)
SIPVoIPLink::~SIPVoIPLink()
{
terminate();
- delete _evThread; _evThread = 0;
}
bool
SIPVoIPLink::init()
{
- /*if( eXosip_running ){
- delete _evThread;
- _evThread=0;
- _evThread= new EventThread( this );
- eXosip_quit();
- }*/
-
- /*if (!_initDone) {
- if (0 != eXosip_init()) {
- _debug("! SIP Failure: Could not initialize eXosip\n");
- return false;
- }
-
- // Pour éviter qu'on refasse l'init sans avoir considéré l'erreur,
- // s'il y en a une ?
- _initDone = true;
- // check networking capabilities
- if ( !checkNetwork() ) {
- _debug("! SIP FAILURE: Unable to determine network capabilities\n");
- return false;
- }
-
- // if we useStun and we failed to receive something on port 5060, we try a random port
- // If use STUN server, firewall address setup
- int errExosip = 0;
- int port = DEFAULT_SIP_PORT;
-
- int iTry = 1; // try number..
-
- do {
- if (_useStun && !Manager::instance().behindNat(_stunServer, port)) {
- port = RANDOM_SIP_PORT;
- if (!Manager::instance().behindNat(_stunServer, port)) {
- _debug("! SIP Failure: Unable to check NAT setting\n");
- return false; // hoho we can't use the random sip port too...
- }
- }
-
- // second parameter, NULL is "::" for ipv6 and "0.0.0.0" for ipv4, we can put INADDR_ANY
- errExosip = eXosip_listen_addr(IPPROTO_UDP, INADDR_ANY, port, AF_INET, 0);
- if (errExosip != 0) {
- _debug("* SIP Info: [%d/%d] could not initialize SIP listener on port %d\n", iTry, _nbTryListenAddr, port);
- port = RANDOM_SIP_PORT;
- }
- } while ( errExosip != 0 && iTry < _nbTryListenAddr );
-
- if ( errExosip != 0 ) { // we didn't succeeded
- _debug("! SIP Failure: SIP failed to listen on port %d\n", port);
- return false;
- }
- _localPort = port;
- _debug(" SIP Init: listening on port %d\n", port);
-
- if (_useStun) {
- // This method is used to replace contact address with the public address of your NAT
- // it should be call after eXosip_listen_addr
- // set by last behindNat() call (ish)...
- _localExternAddress = Manager::instance().getFirewallAddress();
- _localExternPort = Manager::instance().getFirewallPort();
- eXosip_masquerade_contact(_localExternAddress.data(), _localExternPort);
- } else {
- _localExternAddress = _localIPAddress;
- _localExternPort = _localPort;
- }
-
- // Set user agent
- std::string tmp = std::string(PROGNAME_GLOBAL) + "/" + std::string(SFLPHONED_VERSION);
- eXosip_set_user_agent(tmp.data());
-
- _debug(" SIP Init: starting loop thread (SIP events)\n" );
- _evThread->start();
- }
-
- _initDone = true;
- eXosip_running = true;*/
_regc = NULL;
_initDone = true;
return true;
@@ -160,12 +71,7 @@ SIPVoIPLink::init()
void
SIPVoIPLink::terminate()
{
- //terminateSIPCall();
- // TODO The next line makes the daemon crash on
- // account delete if at least one account is registered.
- // It should called only when the last account
- // is deleted/unregistered.
- _initDone = false;
+ _initDone = false;
}
void
@@ -178,10 +84,7 @@ SIPVoIPLink::terminateSIPCall()
while( iter != _callMap.end() ) {
call = dynamic_cast<SIPCall*>(iter->second);
if (call) {
- // Release SIP stack.
- eXosip_lock();
- eXosip_call_terminate(call->getCid(), call->getDid() );
- eXosip_unlock();
+ //TODO terminate the sip call
delete call; call = 0;
}
iter++;
@@ -218,239 +121,21 @@ SIPVoIPLink::loadSIPLocalIP()
void
SIPVoIPLink::getEvent()
{
-#ifdef AAAAA
- char* tmp2;
- eXosip_event_t* event = eXosip_event_wait(0, 50);
- eXosip_lock();
- eXosip_automatic_action();
- eXosip_unlock();
-
- if ( event == NULL ) {
- return;
- }
-
-
- _debug("> SIP Event: [cdt=%4d:%4d:%4d] type=#%03d %s \n", event->cid, event->did, event->tid, event->type, event->textinfo);
- switch (event->type) {
- /* REGISTER related events */
- case EXOSIP_REGISTRATION_NEW: /** 00 < announce new registration. */
- _debugMid(" !EXOSIP_REGISTRATION_NEW event is not implemented\n");
- break;
- case EXOSIP_REGISTRATION_SUCCESS: /** 01 < user is successfully registred. */
- _debugMid(" !EXOSIP_REGISTRATION_SUCCESS ---> %s\n" , getAccountID().c_str());
- if(_eXosipRegID == EXOSIP_ERROR_STD){
- _debug("Successfully Unregister account ID = %s\n" , getAccountID().c_str());
- setRegistrationState(Unregistered);
- }
- else{
- _debug("Successfully Register account ID = %s\n" , getAccountID().c_str());
- setRegistrationState(Registered);
- }
- break;
- case EXOSIP_REGISTRATION_FAILURE: /** 02 < user is not registred. */
- SIPRegistrationFailure( event );
- _debugMid(" !EXOSIP_REGISTRATION_FAILURE\n");
- break;
- case EXOSIP_REGISTRATION_REFRESHED: /** 03 < registration has been refreshed. */
- _debugMid(" !EXOSIP_REGISTRATION_REFRESHED event is not implemented\n");
- break;
- case EXOSIP_REGISTRATION_TERMINATED: /** 04 < UA is not registred any more. */
- //setRegistrationState(Unregistered, "Registration terminated by remote host");
- setRegistrationState(Unregistered);
- _debugMid(" !EXOSIP_REGISTRATION_TERMINATED event is not implemented\n");
- break;
-
- /* INVITE related events within calls */
- case EXOSIP_CALL_INVITE: /** 05 < announce a new call */
- _debugMid(" !EXOSIP_CALL_INVITE\n");
- SIPCallInvite(event);
- break;
- case EXOSIP_CALL_REINVITE: /** 06 < announce a new INVITE within call */
- SIPCallReinvite(event);
- _debugMid(" !EXOSIP_REGISTRATION_TERMINATED event is not implemented\n");
- break;
-
- /* CALL related events */
- case EXOSIP_CALL_NOANSWER: /** 07 < announce no answer within the timeout */
- _debugMid(" !EXOSIP_CALL_NOANSWER event is not implemented\n");
- break;
- case EXOSIP_CALL_PROCEEDING: /** 08 < announce processing by a remote app */
- _debugMid(" !EXOSIP_CALL_PROCEEDING event is not implemented\n");
- break;
- case EXOSIP_CALL_RINGING: /** 09 < announce ringback */
- _debugMid(" !EXOSIP_CALL_RINGING\n");
- SIPCallRinging(event);
- break;
- case EXOSIP_CALL_ANSWERED: /** 10 < announce start of call */
- _debugMid(" !EXOSIP_CALL_ANSWERED\n");
- SIPCallAnswered(event);
- break;
- case EXOSIP_CALL_REDIRECTED: /** 11 < announce a redirection */
- _debugMid(" !EXOSIP_CALL_REDIRECTED event is not implemented\n");
- break;
- case EXOSIP_CALL_REQUESTFAILURE: /** 12 < announce a request failure */
- _debugMid(" !EXOSIP_CALL_REQUESTFAILURE");
- SIPCallRequestFailure(event);
- break;
- case EXOSIP_CALL_SERVERFAILURE: /** 13 < announce a server failure */
- _debugMid(" !EXOSIP_CALL_SERVERFAILURE");
- SIPCallServerFailure(event);
- break;
- case EXOSIP_CALL_GLOBALFAILURE: /** 14 < announce a global failure */
- _debugMid(" !EXOSIP_CALL_GLOBALFAILURE\n");
- SIPCallServerFailure(event);
- break;
- case EXOSIP_CALL_ACK: /** 15 < ACK received for 200ok to INVITE */
- _debugMid(" !EXOSIP_CALL_ACK\n");
- SIPCallAck(event);
- break;
- case EXOSIP_CALL_CANCELLED: /** 16 < announce that call has been cancelled */
- _debugMid(" !EXOSIP_CALL_CANCELLED\n");
- break;
- case EXOSIP_CALL_TIMEOUT: /** 17 < announce that call has failed */
- _debugMid(" !EXOSIP_CALL_TIMEOUT\n");
- break;
-
- /* Request related events within calls (except INVITE) */
- case EXOSIP_CALL_MESSAGE_NEW: /** 18 < announce new incoming MESSAGE. */
- _debugMid(" !EXOSIP_CALL_MESSAGE_NEW\n");
- SIPCallMessageNew(event);
- break;
- case EXOSIP_CALL_MESSAGE_PROCEEDING: /** 19 < announce a 1xx for MESSAGE. */
- _debugMid(" !EXOSIP_CALL_MESSAGE_PROCEEDING\n");
- break;
- case EXOSIP_CALL_MESSAGE_ANSWERED: /** 20 < announce a 200ok */
- // 200 OK
- _debugMid(" !EXOSIP_CALL_MESSAGE_ANSWERED\n");
- break;
- case EXOSIP_CALL_MESSAGE_REDIRECTED: /** 21 < announce a failure. */
- _debugMid(" !EXOSIP_CALL_MESSAGE_REDIRECTED\n");
- break;
- case EXOSIP_CALL_MESSAGE_REQUESTFAILURE: /** 22 < announce a failure. */
- _debugMid(" !EXOSIP_CALL_MESSAGE_REQUESTFAILURE\n");
- break;
- case EXOSIP_CALL_MESSAGE_SERVERFAILURE: /** 23 < announce a failure. */
- _debugMid(" !EXOSIP_CALL_MESSAGE_SERVERFAILURE\n");
- break;
- case EXOSIP_CALL_MESSAGE_GLOBALFAILURE: /** 24 < announce a failure. */
- _debugMid(" !EXOSIP_CALL_MESSAGE_GLOBALFAILURE\n");
- break;
-
- case EXOSIP_CALL_CLOSED: /** 25 < a BYE was received for this call */
- _debugMid(" !EXOSIP_CALL_CLOSED\n");
- SIPCallClosed(event);
- break;
-
- /* For both UAS & UAC events */
- case EXOSIP_CALL_RELEASED: /** 26 < call context is cleared. */
- _debugMid(" !EXOSIP_CALL_RELEASED\n");
- SIPCallReleased(event);
- break;
-
- /* Response received for request outside calls */
- case EXOSIP_MESSAGE_NEW: /** 27 < announce new incoming MESSAGE. */
- _debugMid(" !EXOSIP_MESSAGE_NEW\n");
- if (event->request == NULL) { break; }
- SIPMessageNew(event);
- break;
- case EXOSIP_MESSAGE_PROCEEDING: /** 28 < announce a 1xx for MESSAGE. */
- _debugMid(" !EXOSIP_MESSAGE_PROCEEDING\n");
- break;
- case EXOSIP_MESSAGE_ANSWERED: /** 29 < announce a 200ok */
- _debugMid(" !EXOSIP_MESSAGE_ANSWERED\n");
- break;
- case EXOSIP_MESSAGE_REDIRECTED: /** 30 < announce a failure. */
- _debugMid(" !EXOSIP_MESSAGE_REDIRECTED\n");
- break;
-
- case EXOSIP_MESSAGE_REQUESTFAILURE: /** 31 < announce a failure. */
- _debugMid(" !EXOSIP_MESSAGE_REQUESTFAILURE\n");
- if (event->response !=0 && event->response->status_code == SIP_METHOD_NOT_ALLOWED)
- Manager::instance().incomingMessage(getAccountID(), "Message are not allowed");
- break;
- case EXOSIP_MESSAGE_SERVERFAILURE: /** 32 < announce a failure. */
- _debugMid(" !EXOSIP_MESSAGE_SERVERFAILURE\n");
- break;
- case EXOSIP_MESSAGE_GLOBALFAILURE: /** 33 < announce a failure. */
- _debugMid(" !EXOSIP_MESSAGE_GLOBALFAILURE\n");
- break;
-
- /* Presence and Instant Messaging */
- case EXOSIP_SUBSCRIPTION_UPDATE: /** 34 < announce incoming SUBSCRIBE. */
- _debugMid(" !EXOSIP_SUBSCRIPTION_UPDATE\n");
- break;
- case EXOSIP_SUBSCRIPTION_CLOSED: /** 35 < announce end of subscription. */
- _debugMid(" !EXOSIP_SUBSCRIPTION_CLOSED\n");
- break;
-
- case EXOSIP_SUBSCRIPTION_NOANSWER: /** 37 < announce no answer */
- _debugMid(" !EXOSIP_SUBSCRIPTION_NOANSWER\n");
- break;
- case EXOSIP_SUBSCRIPTION_PROCEEDING: /** 38 < announce a 1xx */
- _debugMid(" !EXOSIP_SUBSCRIPTION_PROCEEDING\n");
- break;
- case EXOSIP_SUBSCRIPTION_ANSWERED: /** 39 < announce a 200ok */
- _debugMid(" !EXOSIP_SUBSCRIPTION_ANSWERED\n");
- eXosip_lock();
- eXosip_automatic_action();
- eXosip_unlock();
- break;
-
- case EXOSIP_SUBSCRIPTION_REDIRECTED: /** 40 < announce a redirection */
- _debugMid(" !EXOSIP_SUBSCRIPTION_REDIRECTED\n");
- break;
- case EXOSIP_SUBSCRIPTION_REQUESTFAILURE: /** 41 < announce a request failure */
- _debugMid(" !EXOSIP_SUBSCRIPTION_REQUESTFAILURE\n");
- break;
- case EXOSIP_SUBSCRIPTION_SERVERFAILURE: /** 42 < announce a server failure */
- _debugMid(" !EXOSIP_SUBSCRIPTION_REQUESTFAILURE\n");
- break;
- case EXOSIP_SUBSCRIPTION_GLOBALFAILURE: /** 43 < announce a global failure */
- _debugMid(" !EXOSIP_SUBSCRIPTION_GLOBALFAILURE\n");
- break;
- case EXOSIP_SUBSCRIPTION_NOTIFY: /** 44 < announce new NOTIFY request */
- _debugMid(" !EXOSIP_SUBSCRIPTION_NOTIFY\n");
- osip_body_t* body;
- osip_from_to_str(event->request->from, &tmp2);
- osip_message_get_body(event->request, 0, &body);
- if (body != NULL && body->body != NULL) {
-// printf("\n---------------------------------\n");
-// printf ("(%i) from: %s\n %s\n", event->tid, tmp2, body->body);
-// printf("---------------------------------\n");
- }
- osip_free(tmp2);
- break;
- case EXOSIP_SUBSCRIPTION_RELEASED: /** 45 < call context is cleared. */
- _debugMid(" !EXOSIP_SUBSCRIPTION_RELEASED\n");
- break;
-
- case EXOSIP_IN_SUBSCRIPTION_NEW: /** 46 < announce new incoming SUBSCRIBE.*/
- _debugMid(" !EXOSIP_IN_SUBSCRIPTION_NEW\n");
- break;
- case EXOSIP_IN_SUBSCRIPTION_RELEASED: /** 47 < announce end of subscription. */
- _debugMid(" !EXOSIP_IN_SUBSCRIPTION_RELEASED\n");
- break;
-
- case EXOSIP_EVENT_COUNT: /** 48 < MAX number of events */
- _debugMid(" !EXOSIP_EVENT_COUNT : SHOULD NEVER HAPPEN!!!!!\n");
- break;
- default:
- printf("received eXosip event (type, did, cid) = (%d, %d, %d)", event->type, event->did, event->cid);
- break;
- }
- eXosip_event_free(event);
-#endif
+ // Nothing anymore. PJSIP is based on asynchronous events
}
bool
SIPVoIPLink::sendRegister()
{
- AccountID id = getAccountID();
+ AccountID id;
pj_status_t status;
+ id = getAccountID();
+
if(_regc) {
- pjsip_regc_destroy(_regc);
+ status = pjsip_regc_destroy(_regc);
_regc = NULL;
+ PJ_ASSERT_RETURN( status == PJ_SUCCESS, 1 );
}
_bRegister = true;
@@ -481,11 +166,8 @@ SIPVoIPLink::sendSIPAuthentification()
/** @todo Même chose ici ? */
return false;
}
- eXosip_lock();
- int returnValue = eXosip_add_authentication_info(login.data(), login.data(), _password.data(), NULL, NULL);
- eXosip_unlock();
- return (returnValue != EXOSIP_ERROR_STD ? true : false);
+ return true;
}
bool
@@ -535,7 +217,6 @@ SIPVoIPLink::answer(const CallID& id)
if (i != 0) {
_debug("< SIP Building Error: send 400 Bad Request\n");
- //eXosip_call_send_answer (call->getTid(), SIP_BAD_REQUEST, NULL);
} else {
// use exosip, bug locked
i = 0;
@@ -559,12 +240,7 @@ SIPVoIPLink::hangup(const CallID& id)
SIPCall* call = getSIPCall(id);
if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
- //_debug("- SIP Action: Hang up call %s [cd: %3d %3d]\n", id.data(), call->getCid(), call->getDid());
- // Release SIP stack.
- //eXosip_lock();
- //eXosip_call_terminate(call->getCid(), call->getDid());
- //eXosip_unlock();
- Manager::instance().getSipManager()->hangup(call);
+ Manager::instance().getSipManager()->hangup(call);
// Release RTP thread
if (Manager::instance().isCurrentCall(id)) {
@@ -582,12 +258,9 @@ SIPVoIPLink::cancel(const CallID& id)
if (call==0) { _debug("! SIP Error: Call doesn't exist\n"); return false; }
_debug("- SIP Action: Cancel call %s [cid: %3d]\n", id.data(), call->getCid());
- // Release SIP stack.
- eXosip_lock();
- eXosip_call_terminate(call->getCid(), -1);
- eXosip_unlock();
removeCall(id);
+
return true;
}
@@ -604,65 +277,8 @@ SIPVoIPLink::onhold(const CallID& id)
_debug("* SIP Info: Stopping AudioRTP for onhold action\n");
_audiortp->closeRtpSession();
-
- //int did = call->getDid();
-
Manager::instance().getSipManager()->onhold(call);
-/*
- eXosip_lock ();
- sdp_message_t *local_sdp = eXosip_get_local_sdp(did);
- eXosip_unlock ();
-
- if (local_sdp == NULL) {
- _debug("! SIP Failure: unable to find local_sdp\n");
- return false;
- }
-
- // Build INVITE_METHOD for put call on-hold
- osip_message_t *invite = NULL;
- eXosip_lock ();
- int exosipErr = eXosip_call_build_request (did, INVITE_METHOD, &invite);
- eXosip_unlock ();
-
- if (exosipErr != 0) {
- sdp_message_free(local_sdp);
- _debug("! SIP Failure: unable to build invite method to hold call\n");
- return false;
- }
-*/
- /* add sdp body */
- /*
- {
- char *tmp = NULL;
-
- int i = sdp_hold_call(local_sdp);
- if (i != 0) {
- sdp_message_free (local_sdp);
- osip_message_free (invite);
- _debug("! SIP Failure: Unable to hold call in SDP\n");
- return false;
- }
-
- i = sdp_message_to_str(local_sdp, &tmp);
- sdp_message_free(local_sdp);
- if (i != 0) {
- osip_message_free (invite);
- osip_free (tmp);
- _debug("! SIP Failure: Unable to translate sdp message to string\n");
- return false;
- }
- osip_message_set_body (invite, tmp, strlen (tmp));
- osip_free (tmp);
- osip_message_set_content_type (invite, "application/sdp");
- }
-
- // send request
- _debug("< SIP: Send on hold request\n");
- eXosip_lock ();
- exosipErr = eXosip_call_send_request (did, invite);
- eXosip_unlock ();
- */
return true;
}
@@ -674,59 +290,6 @@ SIPVoIPLink::offhold(const CallID& id)
Manager::instance().getSipManager()->offhold(call);
-/*
- int did = call->getDid();
-
- eXosip_lock ();
- sdp_message_t *local_sdp = eXosip_get_local_sdp(did);
- eXosip_unlock ();
-
- if (local_sdp == NULL) {
- _debug("! SIP Failure: unable to find local_sdp\n");
- return false;
- }
-
- // Build INVITE_METHOD for put call off-hold
- osip_message_t *invite;
- eXosip_lock ();
- int exosipErr = eXosip_call_build_request (did, INVITE_METHOD, &invite);
- eXosip_unlock ();
-
- if (exosipErr != 0) {
- sdp_message_free(local_sdp);
- return EXOSIP_ERROR_STD;
- }
-*/
- /* add sdp body */
-/*
- {
- char *tmp = NULL;
-
- int i = sdp_off_hold_call (local_sdp);
- if (i != 0) {
- sdp_message_free (local_sdp);
- osip_message_free (invite);
- return false;
- }
-
- i = sdp_message_to_str (local_sdp, &tmp);
- sdp_message_free (local_sdp);
- if (i != 0) {
- osip_message_free (invite);
- osip_free (tmp);
- return false;
- }
- osip_message_set_body (invite, tmp, strlen (tmp));
- osip_free (tmp);
- osip_message_set_content_type (invite, "application/sdp");
- }
-
- // Send request
- _debug("< Send off hold request\n");
- eXosip_lock ();
- exosipErr = eXosip_call_send_request (did, invite);
- eXosip_unlock ();
-*/
// Enable audio
_debug("* SIP Info: Starting AudioRTP when offhold\n");
call->setState(Call::Active);
@@ -750,17 +313,7 @@ SIPVoIPLink::transfer(const CallID& id, const std::string& to)
}
_debug("In transfer, tmp_to is %s\n", tmp_to.data());
- /*osip_message_t *refer;
- eXosip_lock();
- // Build transfer request
- int exosipErr = eXosip_call_build_refer(call->getDid(), (char*)tmp_to.data(), &refer);
- if (exosipErr == 0) {
- // Send transfer request
- _debug("< SIP send transfer request to %s\n", tmp_to.data());
- exosipErr = eXosip_call_send_request(call->getDid(), refer);
- }
- eXosip_unlock();*/
-
+
Manager::instance().getSipManager()->transfer(call, tmp_to);
//_audiortp->closeRtpSession();
@@ -800,12 +353,10 @@ SIPVoIPLink::carryingDTMFdigits(const CallID& id, char code)
if (call==0) { _debug("Call doesn't exist\n"); return false; }
int duration = Manager::instance().getConfigInt(SIGNALISATION, PULSE_LENGTH);
- osip_message_t *info;
- const int body_len = 1000;
- int i;
- char *dtmf_body = new char[body_len];
+ // TODO Add DTMF with pjsip - INFO method
+ /*
eXosip_lock();
// Build info request
i = eXosip_call_build_info(call->getDid(), &info);
@@ -817,144 +368,22 @@ SIPVoIPLink::carryingDTMFdigits(const CallID& id, char code)
i = eXosip_call_send_request(call->getDid(), info);
}
eXosip_unlock();
-
- delete[] dtmf_body; dtmf_body = NULL;
+ */
+
return true;
}
bool
SIPVoIPLink::sendMessage(const std::string& to, const std::string& body)
{
- bool returnValue = false;
-
- // fast return
- if (body.empty()) {return returnValue; }
-
- osip_message_t* message = 0;
- const char* method = "MESSAGE";
-
- std::string sipFrom = getSipFrom();
- std::string sipTo = getSipTo(to);
- std::string sipRoute = getSipRoute();
-
- if (!SIPCheckUrl(sipFrom)) {
- return returnValue;
- }
- if (!SIPCheckUrl(sipTo)) {
- return returnValue;
- }
-
- int eXosipError = EXOSIP_ERROR_STD;
- eXosip_lock();
- if ( sipRoute.empty() ) {
- eXosipError = eXosip_message_build_request(&message, method, sipTo.c_str(), sipFrom.c_str(), NULL);
- } else {
- eXosipError = eXosip_message_build_request(&message, method, sipTo.c_str(), sipFrom.c_str(), sipRoute.c_str());
- }
-
- if (eXosipError == EXOSIP_ERROR_NO) {
- // add body
- // add message
- // src: http://www.atosc.org/pipermail/public/osip/2005-October/006007.html
- osip_message_set_expires(message, "120");
- osip_message_set_body(message, body.c_str(), body.length());
- osip_message_set_content_type(message, "text/plain");
-
- eXosipError = eXosip_message_send_request(message);
- if (eXosipError == EXOSIP_ERROR_NO) {
- // correctly send the message
- returnValue = true;
- }
- }
- eXosip_unlock();
- return returnValue;
+ return true;
}
// NOW
bool
SIPVoIPLink::isContactPresenceSupported()
{
- return true;
-}
-
-/*
-void
-SIPVoIPLink::subscribePresenceForContact(Contact* contact)
-{
- osip_message_t* subscription;
-
- int i;
-
- std::string to = contact->getUrl().data();
- std::ostringstream from;
-
- // Build URL of sender
- from << "sip:" << _authname.data() << "@" << getHostName().data();
-
- // Subscribe for changes on server but also polls at every 5000 interval
- i = eXosip_subscribe_build_initial_request(&subscription,
- to.data(),
- from.str().c_str(),
- NULL,
- "presence", 5000);
- if(i!=0) return;
-
- // We want to receive presence in the PIDF XML format in SIP messages
- osip_message_set_accept(subscription, "application/pidf+xml");
-
- // Send subscription
- eXosip_lock();
- i = eXosip_subscribe_send_initial_request(subscription);
- if(i!=0) _debug("Sending of subscription tp %s failed\n", to.data());
- eXosip_unlock();
-}
-*/
-
-void
-SIPVoIPLink::publishPresenceStatus(std::string status)
-{
- _debug("Publishing presence status\n");
- char buf[4096];
- int i;
- osip_message_t* publication;
-
- std::ostringstream url;
- std::string basic;
- std::string note;
-
- // Build URL of sender
- url << "sip:" << _authname.data() << "@" << getHostName().data();
-
- // TODO
- // Call function to convert status in basic and note
- // tags that are integrated in the publication
- basic = "open";
- note = "ready";
-
- snprintf(buf, 4096,
-"<?xml version=\"1.0\" encoding=\"UTF-8\"?>\n\
-<presence xmlns=\"urn:ietf:params:xml:ns:pidf\"\n\
- xmlns:es=\"urn:ietf:params:xml:ns:pidf:status:rpid-status\"\n\
- entity=\"%s\">\n\
- <tuple id=\"sg89ae\">\n\
- <status>\n\
- <basic>%s</basic>\n\
- <es:activities>\n\
- <es:activity>in-transit</es:activity>\n\
- </es:activities>\n\
- </status>\n\
- <contact priority=\"0.8\">%s</contact>\n\
- <note>%s</note>\n\
- </tuple>\n\
-</presence>"
- , url.str().c_str(), basic.data(), url.str().c_str(), note.data());
-
- // Build publish request in PIDF
- i = eXosip_build_publish(&publication, url.str().c_str(), url.str().c_str(), NULL, "presence", "1800", "application/pidf+xml", buf);
-
- eXosip_lock();
- i = eXosip_publish(publication, url.str().c_str());
- eXosip_unlock();
+ return true;
}
bool
@@ -990,87 +419,7 @@ SIPVoIPLink::SIPStartCall(SIPCall* call, const std::string& subject)
AccountID accId = getAccountID();
Manager::instance().getSipManager()->makeOutgoingCall(to, call, accId);
-
- /*osip_message_t *invite;
- eXosip_lock();
- int eXosipError = eXosip_call_build_initial_invite (&invite, (char*)to.data(),
- (char*)from.data(),
- (char*)route.data(),
- (char*)subject.data());
-
- if (eXosipError != 0) {
- eXosip_unlock();
- return false; // error when building the invite
- }
-
-
- std::ostringstream media_audio;
- std::ostringstream rtpmap_attr;
- AudioCodecType payload;
- int nbChannel;
- int iter;
-
- // Set rtpmap according to the supported codec order
- //CodecMap map = call->getCodecMap().getCodecMap();
- CodecOrder map = call->getCodecMap().getActiveCodecs();
- for(iter=0 ; iter < map.size() ; iter++){
- if(map[iter] != -1){
- payload = map[iter];
- // add each payload in the list of payload
- media_audio << payload << " ";
-
- rtpmap_attr << "a=rtpmap:" << payload << " " <<
- call->getCodecMap().getCodecName(payload) << "/" << call->getCodecMap().getSampleRate(payload);
-
- //TODO add channel infos
- nbChannel = call->getCodecMap().getChannel(payload);
- if (nbChannel!=1) {
- rtpmap_attr << "/" << nbChannel;
- }
- rtpmap_attr << "\r\n";
- }
- // go to next codec
- //*iter++;
- }*/
-
- // http://www.antisip.com/documentation/eXosip2/group__howto1__initialize.html
- // tell sip if we support SIP extension like 100rel
- // osip_message_set_supported (invite, "100rel");
-
- /* add sdp body */
- /*{
- char tmp[4096];
- snprintf (tmp, 4096,
- "v=0\r\n"
- "o=SFLphone 0 0 IN IP4 %s\r\n"
- "s=call\r\n"
- "c=IN IP4 %s\r\n"
- "t=0 0\r\n"
- "m=audio %d RTP/AVP %s\r\n"
- "%s",
- _localExternAddress.c_str(), _localExternAddress.c_str(), call->getLocalExternAudioPort(), media_audio.str().c_str(), rtpmap_attr.str().c_str());
- // media_audio should be one, two or three numbers?
- osip_message_set_body (invite, tmp, strlen (tmp));
- osip_message_set_content_type (invite, "application/sdp");
- _debug("SDP send: %s", tmp);
- }
-
- _debug("> INVITE To <%s>\n", to.data());
- int cid = eXosip_call_send_initial_invite(invite);
-
- // Keep the cid in case of cancelling
- call->setCid(cid);
-
- if (cid <= 0) {
- eXosip_unlock();
- return false ;
- } else {
- _debug("* SIP Info: Outgoing callID is %s, cid=%d\n", call->getCallId().data(), cid);
- eXosip_call_set_reference (cid, NULL);
- }
- eXosip_unlock();*/
-
return true;
}
@@ -1122,22 +471,6 @@ SIPVoIPLink::SIPToHeader(const std::string& to)
bool
SIPVoIPLink::SIPCheckUrl(const std::string& url)
{
- int i;
-
- osip_from_t *to;
- i = osip_from_init(&to);
- if (i != 0) {
- _debug("! SIP Warning: Cannot initialize osip parser\n");
- return false;
- }
- i = osip_from_parse(to, url.data());
- if (i != 0) {
- _debug("! SIP Warning: Cannot parse url %s\n", url.data());
- return false;
- }
-
- // Free memory
- osip_from_free (to);
return true;
}
@@ -1164,143 +497,6 @@ SIPVoIPLink::setCallAudioLocal(SIPCall* call)
return true;
}
-void
-SIPVoIPLink::SIPCallInvite(eXosip_event_t *event)
-{
-}
-
-void
-SIPVoIPLink::SIPCallReinvite(eXosip_event_t *event)
-{
- _debug("> REINVITE (receive)\n");
- SIPCall* call = findSIPCallWithCidDid(event->cid, event->did);
- if (call == 0) {
- _debug("! SIP Failure: unknown call\n");
- _debug("< Send 488 Not Acceptable Here");
- eXosip_lock();
- eXosip_call_send_answer(event->tid, 488, NULL);
- eXosip_unlock();
- return;
- }
- if ( call->getCallId() == Manager::instance().getCurrentCallId()) {
- // STOP tone
- Manager::instance().stopTone(true);
- // STOP old rtp session
- _debug("* SIP Info: Stopping AudioRTP when reinvite\n");
- _audiortp->closeRtpSession();
- call->setAudioStart(false);
- }
- call->SIPCallReinvite(event);
-}
-
-void
-SIPVoIPLink::SIPCallRinging(eXosip_event_t *event)
-{
-
- SIPCall* call = findSIPCallWithCid(event->cid);
- if (!call) {
- _debug("! SIP Failure: unknown call\n");
- return;
- }
- // we could set the cid/did/tid and get the FROM here...
- // but we found the call with the cid/did already, why setting it again?
- // call->ringingCall(event);
- call->setDid(event->did);
- call->setConnectionState(Call::Ringing);
- Manager::instance().peerRingingCall(call->getCallId());
-}
-
-void
-SIPVoIPLink::SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata)
-{
- //SIPCall* call = dynamic_cast<SIPCall *>(theCall);//findSIPCallWithCid(event->cid);
- if (!call) {
- _debug("! SIP Failure: unknown call\n");
- return;
- }
- //call->setDid(event->did);
-
- if (call->getConnectionState() != Call::Connected) {
- //call->SIPCallAnswered(event);
- call->SIPCallAnsweredWithoutHold(rdata);
-
- call->setConnectionState(Call::Connected);
- call->setState(Call::Active);
-
- Manager::instance().peerAnsweredCall(call->getCallId());
- if (Manager::instance().isCurrentCall(call->getCallId())) {
- _debug("* SIP Info: Starting AudioRTP when answering\n");
- if ( _audiortp->createNewSession(call) < 0) {
- _debug("RTP Failure: unable to create new session\n");
- } else {
- call->setAudioStart(true);
- }
- }
- } else {
- _debug("* SIP Info: Answering call (on/off hold to send ACK)\n");
- //call->SIPCallAnswered(event);
- }
-}
-
-void
-SIPVoIPLink::SIPCallRequestFailure(eXosip_event_t *event)
-{
- if (!event->response) { return; }
- // 404 error
- _debug(" Request Failure, receive code %d\n", event->response->status_code);
- // Handle 4XX errors
- switch (event->response->status_code) {
- case SIP_PROXY_AUTHENTICATION_REQUIRED:
- _debug("- SIP Action: Server ask required authentification: logging...\n");
- sendSIPAuthentification();
- eXosip_lock();
- eXosip_automatic_action();
- eXosip_unlock();
- break;
- case SIP_UNAUTHORIZED:
- _debug("- SIP Action: Request is unauthorized. SIP Server ask authentification: logging...\n");
- sendSIPAuthentification();
- break;
-
- case SIP_BUSY_HERE: // 486
- {
- SIPCall* call = findSIPCallWithCid(event->cid);
- if (call!=0) {
- CallID& id = call->getCallId();
- call->setConnectionState(Call::Connected);
- call->setState(Call::Busy);
- Manager::instance().callBusy(id);
- removeCall(id);
- }
- }
- break;
- case SIP_REQUEST_TERMINATED: // 487
- break;
-
- default:
- /*case SIP_BAD_REQUEST:
- case SIP_FORBIDDEN:
- case SIP_NOT_FOUND:
- case SIP_METHOD_NOT_ALLOWED:
- case SIP_406_NOT_ACCEPTABLE:
- case SIP_REQ_TIME_OUT:
- case SIP_TEMPORARILY_UNAVAILABLE:
- case SIP_ADDRESS_INCOMPLETE:
- case SIP_NOT_ACCEPTABLE_HERE: // 488 */
- // Display error on the screen phone
- {
- SIPCall* call = findSIPCallWithCid(event->cid);
- if (call!=0) {
- CallID& id = call->getCallId();
- call->setConnectionState(Call::Connected);
- call->setState(Call::Error);
- Manager::instance().callFailure(id);
- removeCall(id);
- }
- }
- }
-}
-
void
SIPVoIPLink::SIPCallServerFailure(SIPCall *call)
{
@@ -1320,75 +516,6 @@ SIPVoIPLink::SIPCallServerFailure(SIPCall *call)
//}
}
-void
-SIPVoIPLink::SIPRegistrationFailure( eXosip_event_t* event )
-{
- if(!event->response){
- setRegistrationState(ErrorHost);
- return ;
- }
-
- switch( event->response->status_code ) {
- case SIP_FORBIDDEN:
- _debug("SIP forbidden\n");
- setRegistrationState(ErrorAuth);
- break;
- case SIP_UNAUTHORIZED:
- _debug("SIP unauthorized\n");
- setRegistrationState(Error);
- break;
- default:
- setRegistrationState(ErrorAuth);
- //_debug("Unknown error: %s\n" , event->response->status_code);
- }
-}
-
-void
-SIPVoIPLink::SIPCallAck(eXosip_event_t *event)
-{
- SIPCall* call = findSIPCallWithCidDid(event->cid, event->did);
- if (!call) { return; }
- if (!call->isAudioStarted()) {
- if (Manager::instance().isCurrentCall(call->getCallId())) {
- _debug("* SIP Info: Starting AudioRTP when ack\n");
- if ( _audiortp->createNewSession(call) ) {
- call->setAudioStart(true);
- }
- }
- }
-}
-
-void
-SIPVoIPLink::SIPCallMessageNew(eXosip_event_t *event)
-{
- if (0 == event->request) return;
-
- _debug(" > SIP Event: Receive a call message\n");
-
- if (MSG_IS_INFO(event->request)) {
- _debug("* SIP Info: It's a Request Info\n");
- osip_content_type_t* c_t = event->request->content_type;
- if (c_t != 0 && c_t->type != 0 && c_t->subtype != 0 ) {
- _debug("* SIP Info: Content Type of the message: %s/%s\n", c_t->type, c_t->subtype);
- // application/dtmf-relay
- if (strcmp(c_t->type, "application") == 0 && strcmp(c_t->subtype, "dtmf-relay") == 0) {
- handleDtmfRelay(event);
- }
- }
- }
-
- osip_message_t *answerOKNewMessage;
- eXosip_lock();
- if ( 0 == eXosip_call_build_answer(event->tid, SIP_OK, &answerOKNewMessage)) {
- _debug("< SIP Sending 200 OK\n");
- eXosip_call_send_answer(event->tid, SIP_OK, answerOKNewMessage);
- } else {
- _debug("! SIP Failure: Could not sent an OK message\n");
- }
- eXosip_unlock();
-
-}
-
void
SIPVoIPLink::SIPCallClosed(SIPCall *call)
{
@@ -1425,83 +552,35 @@ SIPVoIPLink::SIPCallReleased(SIPCall *call)
}
void
-SIPVoIPLink::SIPMessageNew(eXosip_event_t *event)
+SIPVoIPLink::SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata)
{
- if (MSG_IS_OPTIONS(event->request)) {
- // old handling was
- // - send 200 OK if call id is not found
- // - send nothing if call id is found
- eXosip_lock();
- eXosip_options_send_answer (event->tid, SIP_OK, NULL);
- eXosip_unlock();
+ //SIPCall* call = dynamic_cast<SIPCall *>(theCall);//findSIPCallWithCid(event->cid);
+ if (!call) {
+ _debug("! SIP Failure: unknown call\n");
+ return;
}
- // Voice message
- else if (MSG_IS_NOTIFY(event->request)){
- int ii;
- osip_body_t *body = NULL;
- // Get the message body
- ii = osip_message_get_body(event->request, 0, &body);
- if (ii != 0) {
- _debug("! SIP Error: Cannot get body in a new EXOSIP_MESSAGE_NEW event\n");
- return;
- }
+ //call->setDid(event->did);
- // Analyse message body
- if (!body || !body->body) {
- return;
- }
- std::string str(body->body);
- std::string::size_type pos;
- std::string::size_type pos_slash;
- pos = str.find(VOICE_MSG);
-
- if (pos == std::string::npos) {
- // If the string is not found
- return;
- }
-
- pos_slash = str.find ("/");
- std::string nb_msg = str.substr(pos + LENGTH_VOICE_MSG,
- pos_slash - (pos + LENGTH_VOICE_MSG));
-
- // Set the number of voice-message
- int msgVoicemail = atoi(nb_msg.data());
- _debug(" > NOTIFY -> %i voice message for account %s\n" , msgVoicemail , getAccountID().c_str());
-
- if (msgVoicemail != 0) {
- // If there is at least one voice-message, start notification
- //Manager::instance().startVoiceMessageNotification(getAccountID(), nb_msg);
- }
- // http://www.jdrosen.net/papers/draft-ietf-simple-im-session-00.txt
- } else if (MSG_IS_MESSAGE(event->request)) {
- _debug("> MESSAGE received\n");
- // osip_content_type_t* osip_message::content_type
- osip_content_type_t* c_t = event->request->content_type;
- if (c_t != 0 && c_t->type != 0 && c_t->subtype != 0 ) {
- _debug("* SIP Info: Content Type of the message: %s/%s\n", c_t->type, c_t->subtype);
-
- osip_body_t *body = NULL;
- // Get the message body
- if (0 == osip_message_get_body(event->request, 0, &body)) {
- _debug("* SIP Info: Body length: %d\n", body->length);
- if (body->body!=0 &&
- strcmp(c_t->type,"text") == 0 &&
- strcmp(c_t->subtype,"plain") == 0
- ) {
- _debug("* SIP Info: Text body: %s\n", body->body);
- Manager::instance().incomingMessage(getAccountID(), body->body);
- }
+ if (call->getConnectionState() != Call::Connected) {
+ //call->SIPCallAnswered(event);
+ call->SIPCallAnsweredWithoutHold(rdata);
+
+ call->setConnectionState(Call::Connected);
+ call->setState(Call::Active);
+
+ Manager::instance().peerAnsweredCall(call->getCallId());
+ if (Manager::instance().isCurrentCall(call->getCallId())) {
+ _debug("* SIP Info: Starting AudioRTP when answering\n");
+ if ( _audiortp->createNewSession(call) < 0) {
+ _debug("RTP Failure: unable to create new session\n");
+ } else {
+ call->setAudioStart(true);
}
}
- osip_message_t *answerOK;
- eXosip_lock();
- if ( 0 == eXosip_message_build_answer(event->tid, SIP_OK, &answerOK)) {
- _debug("< Sending 200 OK\n");
- eXosip_message_send_answer(event->tid, SIP_OK, answerOK);
- }
- eXosip_unlock();
+ } else {
+ _debug("* SIP Info: Answering call (on/off hold to send ACK)\n");
+ //call->SIPCallAnswered(event);
}
-
}
SIPCall*
@@ -1553,7 +632,7 @@ SIPVoIPLink::getSIPCall(const CallID& id)
}
return NULL;
}
-
+/*
bool
SIPVoIPLink::handleDtmfRelay(eXosip_event_t* event) {
@@ -1588,7 +667,7 @@ SIPVoIPLink::handleDtmfRelay(eXosip_event_t* event) {
returnValue = true;
}
}
-/*
+
// we receive the duration, but we use our configuration...
posStart = dtmfBody.find("Duration=");
@@ -1602,98 +681,15 @@ SIPVoIPLink::handleDtmfRelay(eXosip_event_t* event) {
_debug("Duration value: %s\n", duration.c_str());
returnValue = true;
}
-*/
+
}
}
return returnValue;
}
-
+*/
///////////////////////////////////////////////////////////////////////////////
// Private functions
///////////////////////////////////////////////////////////////////////////////
-int
-SIPVoIPLink::sdp_hold_call (sdp_message_t * sdp)
-{
- int pos;
- int pos_media = -1;
- char *rcvsnd;
- int recv_send = -1;
-
- pos = 0;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
- while (rcvsnd != NULL) {
- if (rcvsnd != NULL && 0 == strcmp (rcvsnd, "sendonly")) {
- recv_send = 0;
- } else if (rcvsnd != NULL && (0 == strcmp (rcvsnd, "recvonly")
- || 0 == strcmp (rcvsnd, "sendrecv"))) {
- recv_send = 0;
- sprintf (rcvsnd, "sendonly");
- }
- pos++;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
- }
-
- pos_media = 0;
- while (!sdp_message_endof_media (sdp, pos_media)) {
- pos = 0;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
- while (rcvsnd != NULL) {
- if (rcvsnd != NULL && 0 == strcmp (rcvsnd, "sendonly")) {
- recv_send = 0;
- } else if (rcvsnd != NULL && (0 == strcmp (rcvsnd, "recvonly")
- || 0 == strcmp (rcvsnd, "sendrecv"))) {
- recv_send = 0;
- sprintf (rcvsnd, "sendonly");
- }
- pos++;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
- }
- pos_media++;
- }
-
- if (recv_send == -1) {
- /* we need to add a global attribute with a field set to "sendonly" */
- sdp_message_a_attribute_add (sdp, -1, osip_strdup ("sendonly"), NULL);
- }
-
- return 0;
-}
-
-int
-SIPVoIPLink::sdp_off_hold_call (sdp_message_t * sdp)
-{
- int pos;
- int pos_media = -1;
- char *rcvsnd;
-
- pos = 0;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
- while (rcvsnd != NULL) {
- if (rcvsnd != NULL && (0 == strcmp (rcvsnd, "sendonly")
- || 0 == strcmp (rcvsnd, "recvonly"))) {
- sprintf (rcvsnd, "sendrecv");
- }
- pos++;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
- }
-
- pos_media = 0;
- while (!sdp_message_endof_media (sdp, pos_media)) {
- pos = 0;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
- while (rcvsnd != NULL) {
- if (rcvsnd != NULL && (0 == strcmp (rcvsnd, "sendonly")
- || 0 == strcmp (rcvsnd, "recvonly"))) {
- sprintf (rcvsnd, "sendrecv");
- }
- pos++;
- rcvsnd = sdp_message_a_att_field_get (sdp, pos_media, pos);
- }
- pos_media++;
- }
-
- return 0;
-}
void SIPVoIPLink::setAuthName(const std::string& authname)
{
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
old mode 100755
new mode 100644
index 6ff0fc81b1feeb13a87170f1184291c932939f7d..265c6366e79c001e66ebbcedd010cb7b07373194
--- a/src/sipvoiplink.h
+++ b/src/sipvoiplink.h
@@ -23,12 +23,18 @@
#include "voiplink.h"
#include <string>
-#include <eXosip2/eXosip.h>
-#include <osip2/osip_mt.h>
-struct pjsip_regc;
-struct pj_str_t;
-struct pjmedia_sdp_session;
+#include <pjsip.h>
+#include <pjlib-util.h>
+#include <pjlib.h>
+#include <pjnath/stun_config.h>
+
+//TODO Remove this include if we don't need anything from it
+#include <pjsip_simple.h>
+
+#include <pjsip_ua.h>
+#include <pjmedia/sdp.h>
+#include <pjmedia/sdp_neg.h>
class EventThread;
class SIPCall;
@@ -54,10 +60,8 @@ class SIPVoIPLink : public VoIPLink
*/
~SIPVoIPLink();
- int eXosip_running;
-
/**
- * Try to initiate the eXosip engine/thread and set config
+ * Try to initiate the pjsip engine/thread and set config
* @return bool True if OK
*/
bool init(void);
@@ -165,10 +169,6 @@ class SIPVoIPLink : public VoIPLink
bool isContactPresenceSupported();
- //void subscribePresenceForContact(Contact* contact);
-
- void publishPresenceStatus(std::string status);
-
// TODO Not used yet
void sendMessageToContact(const CallID& id, const std::string& message);
@@ -214,7 +214,7 @@ class SIPVoIPLink : public VoIPLink
void terminateSIPCall();
/**
- * Get the local Ip by eXosip
+ * Get the local Ip
* only if the local ip address is to his default value: 127.0.0.1
* setLocalIpAdress
* @return bool false if not found
@@ -293,89 +293,33 @@ class SIPVoIPLink : public VoIPLink
*/
bool setCallAudioLocal(SIPCall* call);
- /**
- * Create a new call and send a incoming call notification to the user
- * @param event eXosip Event
- */
- void SIPCallInvite(eXosip_event_t *event);
-
- /**
- * Use a exisiting call to restart the audio
- * @param event eXosip Event
- */
- void SIPCallReinvite(eXosip_event_t *event);
-
- /**
- * Tell the user that the call is ringing
- * @param event eXosip Event
- */
- void SIPCallRinging(eXosip_event_t *event);
-
/**
* Tell the user that the call was answered
- * @param event eXosip Event
+ * @param
*/
- void SIPCallAnswered(eXosip_event_t *event);
void SIPCallAnswered(SIPCall *call, pjsip_rx_data *rdata);
- /**
- * Handling 4XX error
- * @param event eXosip Event
- */
- void SIPCallRequestFailure(eXosip_event_t *event);
-
/**
* Handling 5XX/6XX error
- * @param event eXosip Event
+ * @param
*/
void SIPCallServerFailure(SIPCall *call);
- /**
- * Handle registration failure cases ( SIP_FORBIDDEN , SIP_UNAUTHORIZED )
- * @param event eXosip event
- */
- void SIPRegistrationFailure( eXosip_event_t *event );
-
- /**
- * Handling ack (restart audio if reinvite)
- * @param event eXosip Event
- */
- void SIPCallAck(eXosip_event_t *event);
-
- /**
- * Handling message inside a call (like dtmf)
- * @param event eXosip Event
- */
- void SIPCallMessageNew(eXosip_event_t *event);
-
- /**
- * Handle an INFO with application/dtmf-relay content-type
- * @param event eXosip Event
- */
- bool handleDtmfRelay(eXosip_event_t *event);
-
/**
* Peer close the connection
- * @param event eXosip Event
+ * @param
*/
void SIPCallClosed(SIPCall *call);
/**
* The call pointer was released
* If the call was not cleared before, report an error
- * @param event eXosip Event
+ * @param
*/
void SIPCallReleased(SIPCall *call);
/**
- * Receive a new Message request
- * Option/Notify/Message
- * @param event eXosip Event
- */
- void SIPMessageNew(eXosip_event_t *event);
-
- /**
- * Find a SIPCall with cid from eXosip Event
+ * Find a SIPCall with cid
* Explication there is no DID when the dialog is not establish...
* @param cid call ID
* @return SIPCall* SIPCall pointer or 0
@@ -383,7 +327,7 @@ class SIPVoIPLink : public VoIPLink
SIPCall* findSIPCallWithCid(int cid);
/**
- * Find a SIPCall with cid and did from eXosip Event
+ * Find a SIPCall with cid and did
* @param cid call ID
* @param did domain ID
* @return SIPCall* SIPCall pointer or 0
@@ -397,24 +341,9 @@ class SIPVoIPLink : public VoIPLink
*/
SIPCall* getSIPCall(const CallID& id);
- /** To build sdp when call is on-hold */
- int sdp_hold_call (sdp_message_t * sdp);
-
- /** To build sdp when call is off-hold */
- int sdp_off_hold_call (sdp_message_t * sdp);
-
- /** EventThread get every incoming events */
- EventThread* _evThread;
-
- /** Tell if eXosip was stared (eXosip_init) */
+ /** Tell if the initialisation was done */
bool _initDone;
- /** Registration identifier, needed by unregister to build message */
- int _eXosipRegID;
-
- /** Number of voicemail */
- int _nMsgVoicemail;
-
/** when we init the listener, how many times we try to bind a port? */
int _nbTryListenAddr;
diff --git a/src/voiplink.h b/src/voiplink.h
index be69aaefbf3e520679b037a10828464c141432e2..c260cdee4ae8787f88935272e8628e59e73cfd9c 100644
--- a/src/voiplink.h
+++ b/src/voiplink.h
@@ -66,7 +66,7 @@ class VoIPLink {
/**
* Virtual method
- * Try to initiate the eXosip engine/thread and set config
+ * Try to initiate the pjsip engine/thread and set config
* @return bool True if OK
*/
virtual bool init (void) = 0;