diff --git a/ChangeLog b/ChangeLog
index 6a49b5bae99a4a4c8f9ceb4de6e9c8d8e7aee99f..49d5fd6420f8fa64e22d34ff0ebcb6e687da0247 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,7 @@
+Laurielle LEA (27 July 2005) version 0.4
+- Migrate from libeXoSIP 0.9.0 to libeXosip2-1.9.1-pre15
+(http://www.antisip.com/download/)
+
Laurielle LEA (21 July 2005) version 0.4
- Change README.
diff --git a/DEPS b/DEPS
index d1f4c99b5cf5983d5c1f8044e2f1893dd034cab7..e924e675273db15a1e6a888060a69e2d03f010b6 100644
--- a/DEPS
+++ b/DEPS
@@ -1,5 +1,5 @@
PortAudio: http://portaudio.com/archives/pa_snapshot_v19.tar.gz
Common C++ 2 1.3.6: http://sourceforge.net/projects/cplusplus/
ccRTP 1.3.0: http://sourceforge.net/projects/cplusplus/
-libosip 2.2.0: http://savannah.gnu.org/projects/osip/
-libeXoSIP 0.9.0: http://savannah.nongnu.org/projects/exosip/
+libosip 2.2.1: http://savannah.gnu.org/projects/osip/
+libeXosip2-1.9.1-pre14: http://www.antisip.com/download/
diff --git a/config.h.in b/config.h.in
index 35f2516284c6792300322b539bec03d3de744ae5..af06bd0e0a6d50960f55781f1e29427775e75048 100644
--- a/config.h.in
+++ b/config.h.in
@@ -15,8 +15,8 @@
/* Define to 1 if you have the <dlfcn.h> header file. */
#undef HAVE_DLFCN_H
-/* Define to 1 if you have the <eXosip/eXosip.h> header file. */
-#undef HAVE_EXOSIP_EXOSIP_H
+/* Define to 1 if you have the <eXosip2/eXosip.h> header file. */
+#undef HAVE_EXOSIP2_EXOSIP_H
/* Define to 1 if you have the <inttypes.h> header file. */
#undef HAVE_INTTYPES_H
diff --git a/configure.ac b/configure.ac
index 741fd3d5da43cb114984e12eec201dc80ee0022a..a2556030f891c215e1bbd7d08fc648205323f4c8 100644
--- a/configure.ac
+++ b/configure.ac
@@ -54,7 +54,7 @@ dnl Checks for header files.
AC_HEADER_STDC
AC_CHECK_HEADERS( \
ostream \
-eXosip/eXosip.h \
+eXosip2/eXosip.h \
portaudio.h \
)
@@ -78,11 +78,11 @@ PKG_CHECK_MODULES(libosip2, libosip2 >= ${LIBOSIP2_MIN_VERSION})
SFLPHONE_CXXFLAGS="$SFLPHONE_CXXFLAGS $libosip2_CFLAGS"
SFLPHONE_LIBS="$SFLPHONE_LIBS $libosip2_LIBS"
-if test $ac_cv_header_eXosip_eXosip_h = no; then
- AC_MSG_ERROR([*** missing eXipsip/eXosip.h. You need a working eXosip installation. See http://savannah.nongnu.org/projects/exosip/])
+if test $ac_cv_header_eXosip2_eXosip_h = no; then
+ AC_MSG_ERROR([*** missing eXosip2/eXosip.h. You need a working eXosip2 installation. See http://www.antisip.com/download/])
fi
-libexosip_LIBS="-leXosip "
-SFLPHONE_LIBS="$SFLPHONE_LIBS $libexosip_LIBS"
+libexosip2_LIBS="-leXosip2 "
+SFLPHONE_LIBS="$SFLPHONE_LIBS $libexosip2_LIBS"
AC_SUBST(LIBQT)
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index b593b9bf13bd009a6656d9ea32920a156bb34425..5c43810e56db0742905eedb6926df8289f893fa9 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -189,6 +189,7 @@ AudioRtpRTX::initAudioRtpSession (void)
setCancel(cancelImmediate);
}
}
+ _debug("-----------------------\n");
}
void
diff --git a/src/gui/qt/qtGUImainwindow.cpp b/src/gui/qt/qtGUImainwindow.cpp
index 1b380ef9f764b7f40958d02770a902317c2ac705..8eb23f2ae41ca0a67ef34fd24ba448bc224a33b5 100644
--- a/src/gui/qt/qtGUImainwindow.cpp
+++ b/src/gui/qt/qtGUImainwindow.cpp
@@ -707,6 +707,7 @@ QtGUIMainWindow::callIsOnHold(int id, int line, int busyLine)
int
QtGUIMainWindow::callIsIncoming (int id, int line, int busyLine)
{
+ _TabIncomingCalls[line] = -1;
changeLineStatePixmap(line, BUSY);
putOnHoldBusyLine(busyLine);
if (qt_answerCall(id) != 1) {
@@ -1218,6 +1219,7 @@ QtGUIMainWindow::hangupLine (void)
Manager::instance().displayErrorText("Hangup call failed !\n");
}
} else if ((i = isThereIncomingCall()) > 0){
+ _debug("&&&&&&&&&& i = %d\n", i);
// To refuse new incoming call
_debug("Refuse call %d\n", id);
if (qt_refuseCall(i)) {
diff --git a/src/gui/qt/qtGUImainwindow.h b/src/gui/qt/qtGUImainwindow.h
index a47f746526600417919f632d7527a34661fb1190..e9e58964e65c63bfb01eef2e71289c40615b2bfa 100644
--- a/src/gui/qt/qtGUImainwindow.h
+++ b/src/gui/qt/qtGUImainwindow.h
@@ -286,7 +286,7 @@ private:
inline void setPrevLine(int line) { _prevLine = line; }
inline int getPrevLine(void) { return _prevLine; }
- // Array of incoming calls
+ // Array of incoming calls, contains call-id
int _TabIncomingCalls[NUMBER_OF_LINES];
// The current phoneline
diff --git a/src/sipcall.cpp b/src/sipcall.cpp
index 334c86b8b94d050daf765ceed0cace610727566a..90eb0b6b0765aeef8e1f4990c582301052a23d07 100644
--- a/src/sipcall.cpp
+++ b/src/sipcall.cpp
@@ -24,6 +24,12 @@
#include <iostream>
+// For AF_INET
+#include <sys/socket.h>
+#include <sys/types.h>
+#include <netinet/in.h>
+//
+
#include "global.h"
#include "audio/audiocodec.h"
#include "audio/codecDescriptor.h"
@@ -95,6 +101,18 @@ SipCall::getCid (void)
return _cid;
}
+void
+SipCall::setTid (int tid)
+{
+ _tid = tid;
+}
+
+int
+SipCall::getTid (void)
+{
+ return _tid;
+}
+
int
SipCall::getRemoteSdpAudioPort (void)
{
@@ -124,187 +142,634 @@ int
SipCall::newIncomingCall (eXosip_event_t *event) {
SipCall *ca = this;
+ sdp_message_t *remote_sdp = NULL;
+
_cid = event->cid;
_did = event->did;
+ _tid = event->tid;
if (_did < 1 && _cid < 1) {
+ exit(0);
return -1; /* not enough information for this event?? */
}
osip_strncpy (_textinfo, event->textinfo, 255);
- osip_strncpy (_req_uri, event->req_uri, 255);
- osip_strncpy (_local_uri, event->local_uri, 255);
- osip_strncpy (_local_uri, event->remote_uri, 255);
- osip_strncpy (_subject, event->subject, 255);
-
- osip_strncpy (ca->_remote_sdp_audio_ip, event->remote_sdp_audio_ip, 49);
- ca->_remote_sdp_audio_port = event->remote_sdp_audio_port;
-
- ca->payload = event->payload;
- _debug("For incoming ca->_payload = %d\n", ca->payload);
- osip_strncpy (ca->_payload_name, event->payload_name, 49);
- setAudioCodec(_cdv->at(0)->alloc(ca->payload, ca->_payload_name));
-
- osip_strncpy (ca->_sdp_body, event->sdp_body, 1000);
- osip_strncpy (ca->_sdp_body, event->sdp_body, 1000);
- _debug("\n%s\n", ca->_sdp_body);
-
- if (event->reason_phrase[0] != '\0') {
- osip_strncpy(this->_reason_phrase, event->reason_phrase, 49);
- this->_status_code = event->status_code;
+
+ if (event->response != NULL) {
+ _status_code = event->response->status_code;
+ snprintf (_reason_phrase, 49, "%s", event->response->reason_phrase);
}
-
+
+ if (event->request != NULL) {
+ char *tmp = NULL;
+
+ osip_from_to_str (event->request->from, &tmp);
+ if (tmp != NULL) {
+ snprintf (_remote_uri, 255, "%s", tmp);
+ osip_free (tmp);
+ }
+ }
+
+ /* negotiate payloads */
+ if (event->request != NULL) {
+ remote_sdp = eXosip_get_sdp_info (event->request);
+ }
+
+ if (remote_sdp == NULL) {
+ _debug("missing SDP in INVITE request\n");
+ }
+
+ if (remote_sdp != NULL) { /* TODO: else build an offer */
+ sdp_connection_t *conn;
+ sdp_media_t *remote_med;
+ char *tmp = NULL;
+
+ if (remote_sdp == NULL) {
+ _debug("No remote SDP body found for call\n");
+ // Send 400 BAD REQUEST
+ eXosip_call_send_answer (_tid, 400, NULL);
+ sdp_message_free (remote_sdp);
+ return 0;
+ }
+
+ conn = eXosip_get_audio_connection (remote_sdp);
+ if (conn != NULL && conn->c_addr != NULL) {
+ snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
+ }
+ remote_med = eXosip_get_audio_media (remote_sdp);
+
+ if (remote_med == NULL || remote_med->m_port == NULL) {
+ /* no audio media proposed */
+ // Send 415 Unsupported media type
+ eXosip_call_send_answer (_tid, 415, NULL);
+ sdp_message_free (remote_sdp);
+ return 0;
+ }
+
+ _remote_sdp_audio_port = atoi (remote_med->m_port);
+
+ if (_remote_sdp_audio_port > 0 && _remote_sdp_audio_ip[0] != '\0') {
+ int pos;
+ pos = 0;
+ while (!osip_list_eol (remote_med->m_payloads, pos)) {
+ tmp = (char *) osip_list_get (remote_med->m_payloads, pos);
+ if (tmp != NULL &&
+ (0 == osip_strcasecmp (tmp, "0")
+ || 0 == osip_strcasecmp (tmp, "8"))) {
+ break;
+ }
+ tmp = NULL;
+ pos++;
+ }
+ }
+ if (tmp != NULL) {
+ ca->payload = atoi (tmp);
+ } else {
+ // Send 415 Unsupported media type
+ eXosip_call_send_answer (_tid, 415, NULL);
+ sdp_message_free (remote_sdp);
+ return 0;
+ }
+
+ if (tmp != NULL && (ca->payload == 0 || ca->payload == 8)
+ && _remote_sdp_audio_port > 0 && _remote_sdp_audio_ip[0] != '\0') {
+ }
+ }
+
+ if (remote_sdp != NULL) { /* TODO: else build an offer */
+ osip_message_t *answer;
+ int i;
+
+ eXosip_lock ();
+ i = eXosip_call_build_answer (_tid, 183, &answer);
+ if (i == 0) {
+ i = sdp_complete_message (remote_sdp, answer);
+ if (i != 0) {
+ osip_message_free (answer);
+ // Send 415 Unsupported media type
+ eXosip_call_send_answer (_tid, 415, NULL);
+ } else {
+ /* start sending audio */
+ if (ca->enable_audio > 0) {
+ ca->enable_audio = -1;
+ }
+ if (ca->enable_audio != 1) /* audio is started */ {
+ sdp_message_t *local_sdp;
+ local_sdp = eXosip_get_sdp_info (answer);
+ if (remote_sdp != NULL && local_sdp != NULL) {
+ sdp_connection_t *conn;
+ sdp_media_t *local_med;
+ sdp_media_t *remote_med;
+ char *tmp = NULL;
+ int audio_port = 0;
+
+ conn = eXosip_get_audio_connection (remote_sdp);
+ if (conn != NULL && conn->c_addr != NULL) {
+ snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
+ }
+ remote_med = eXosip_get_audio_media (remote_sdp);
+ if (remote_med != NULL && remote_med->m_port != NULL) {
+ _remote_sdp_audio_port = atoi (remote_med->m_port);
+ }
+ local_med = eXosip_get_audio_media (local_sdp);
+ if (local_med != NULL && local_med->m_port != NULL) {
+ audio_port = atoi (local_med->m_port);
+ }
+
+ if (_remote_sdp_audio_port > 0
+ && _remote_sdp_audio_ip[0] != '\0'
+ && local_med != NULL) {
+ tmp = (char *) osip_list_get (local_med->m_payloads, 0);
+ }
+ if (tmp != NULL) {
+ ca->payload = atoi (tmp);
+ _debug("For incoming _payload = %d\n", ca->payload);
+ setAudioCodec(_cdv->at(0)->alloc(ca->payload, ""));
+ }
+ if (tmp != NULL
+ && audio_port > 0
+ && _remote_sdp_audio_port > 0
+ && _remote_sdp_audio_ip[0] != '\0') {
+
+ /* search if stream is sendonly or recvonly */
+ _remote_sendrecv =
+ sdp_analyse_attribute (remote_sdp, remote_med);
+ _local_sendrecv =
+ sdp_analyse_attribute (local_sdp, local_med);
+ if (_local_sendrecv == _SENDRECV) {
+ if (_remote_sendrecv == _SENDONLY)
+ _local_sendrecv = _RECVONLY;
+ else if (_remote_sendrecv == _RECVONLY)
+ _local_sendrecv = _SENDONLY;
+ }
+ }
+ }
+ sdp_message_free (local_sdp);
+ }
+
+ i = eXosip_call_send_answer (_tid, 183, answer);
+ }
+
+ if (i != 0) {
+ _debug("cannot send 183 progress?\n");
+ }
+ }
+ eXosip_unlock ();
+ }
+
this->_state = event->type;
+ sdp_message_free (remote_sdp);
return 0;
}
int
SipCall::ringingCall (eXosip_event_t *event) {
- SipCall *ca = this;
+ SipCall* ca = this;
this->_cid = event->cid;
this->_did = event->did;
+ this->_tid = event->tid;
if (this->_did < 1 && this->_cid < 1) {
return -1;
}
- osip_strncpy(this->_textinfo, event->textinfo, 255);
- osip_strncpy(this->_req_uri, event->req_uri, 255);
- osip_strncpy(this->_local_uri, event->local_uri, 255);
- osip_strncpy(this->_remote_uri, event->remote_uri, 255);
- osip_strncpy(this->_subject, event->subject, 255);
+ osip_strncpy (_textinfo, event->textinfo, 255);
- osip_strncpy(ca->_remote_sdp_audio_ip, event->remote_sdp_audio_ip, 49);
- ca->_remote_sdp_audio_port = event->remote_sdp_audio_port;
- ca->payload = event->payload;
- osip_strncpy(ca->_payload_name, event->payload_name, 49);
-
- if (event->reason_phrase[0]!='\0') {
- osip_strncpy(this->_reason_phrase, event->reason_phrase, 49);
- this->_status_code = event->status_code;
+ if (event->response != NULL) {
+ _status_code = event->response->status_code;
+ snprintf (_reason_phrase, 49, "%s", event->response->reason_phrase);
+ }
+
+ if (event->request != NULL) {
+ char *tmp = NULL;
+
+ osip_from_to_str (event->request->from, &tmp);
+ if (tmp != NULL) {
+ snprintf (_remote_uri, 255, "%s", tmp);
+ osip_free (tmp);
+ }
+ }
+
+ sdp_message_t *remote_sdp;
+ sdp_message_t *local_sdp;
+
+ local_sdp = eXosip_get_sdp_info (event->request);
+ remote_sdp = eXosip_get_sdp_info (event->response);
+ if (remote_sdp == NULL) {
+ _debug("No remote SDP body found for call\n");
+ /* TODO: remote_sdp = retreive from ack above */
+ }
+ if (local_sdp == NULL) {
+ _debug("SDP body was probably in the ACK (TODO)\n");
}
+ if (remote_sdp != NULL && local_sdp != NULL) {
+ sdp_connection_t *conn;
+ sdp_media_t *local_med;
+ sdp_media_t *remote_med;
+ char *tmp = NULL;
+ int audio_port = 0;
+
+ conn = eXosip_get_audio_connection (remote_sdp);
+ if (conn != NULL && conn->c_addr != NULL) {
+ snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
+ }
+ remote_med = eXosip_get_audio_media (remote_sdp);
+ if (remote_med != NULL && remote_med->m_port != NULL) {
+ _remote_sdp_audio_port = atoi (remote_med->m_port);
+ }
+ local_med = eXosip_get_audio_media (local_sdp);
+ if (local_med != NULL && local_med->m_port != NULL) {
+ audio_port = atoi (local_med->m_port);
+ }
+ if (_remote_sdp_audio_port > 0 && _remote_sdp_audio_ip[0] != '\0'
+ && remote_med != NULL) {
+ tmp = (char *) osip_list_get (remote_med->m_payloads, 0);
+ }
+ if (tmp != NULL) {
+ ca->payload = atoi (tmp);
+ }
+ if (tmp != NULL
+ && audio_port > 0
+ && _remote_sdp_audio_port > 0
+ && _remote_sdp_audio_ip[0] != '\0') {
+ /* search if stream is sendonly or recvonly */
+ _remote_sendrecv =
+ sdp_analyse_attribute (remote_sdp, remote_med);
+ _local_sendrecv = sdp_analyse_attribute (local_sdp, local_med);
+ if (_local_sendrecv == _SENDRECV) {
+ if (_remote_sendrecv == _SENDONLY)
+ _local_sendrecv = _RECVONLY;
+ else if (_remote_sendrecv == _RECVONLY)
+ _local_sendrecv = _SENDONLY;
+ }
+ }
+ }
+ sdp_message_free (local_sdp);
+ sdp_message_free (remote_sdp);
+
this->_state = event->type;;
return 0;
}
+int
+SipCall::receivedAck (eXosip_event_t *event)
+{
+ SipCall *ca = this;
+
+ _cid = event->cid;
+ _did = event->did;
+
+
+ if (event->ack != NULL) {
+ sdp_message_t *remote_sdp;
+ remote_sdp = eXosip_get_sdp_info (event->ack);
+ if (remote_sdp != NULL) {
+ _debug("SDP detected in ACK!\n");
+ } else {
+ _debug("no SDP detected in ACK!\n");
+ }
+ }
+
+ if (ca->enable_audio != 1) { /* audio is started */
+ sdp_message_t *remote_sdp;
+ sdp_message_t *local_sdp;
+
+ remote_sdp = eXosip_get_remote_sdp (_did);
+ local_sdp = eXosip_get_local_sdp (_did);
+ if (remote_sdp == NULL) {
+ _debug("No remote SDP body found for call\n");
+ }
+ if (remote_sdp != NULL && local_sdp != NULL) {
+ sdp_connection_t *conn;
+ sdp_media_t *local_med;
+ sdp_media_t *remote_med;
+ char *tmp = NULL;
+ int audio_port = 0;
+
+ conn = eXosip_get_audio_connection (remote_sdp);
+ if (conn != NULL && conn->c_addr != NULL) {
+ snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
+ }
+ remote_med = eXosip_get_audio_media (remote_sdp);
+ if (remote_med != NULL && remote_med->m_port != NULL) {
+ _remote_sdp_audio_port = atoi (remote_med->m_port);
+ }
+ local_med = eXosip_get_audio_media (local_sdp);
+ if (local_med != NULL && local_med->m_port != NULL) {
+ audio_port = atoi (local_med->m_port);
+ }
+
+ if (_remote_sdp_audio_port > 0
+ && _remote_sdp_audio_ip[0] != '\0' && local_med != NULL) {
+ tmp = (char *) osip_list_get (local_med->m_payloads, 0);
+ }
+ if (tmp != NULL) {
+ ca->payload = atoi (tmp);
+ }
+ if (tmp != NULL
+ && audio_port > 0
+ && _remote_sdp_audio_port > 0
+ && _remote_sdp_audio_ip[0] != '\0') {
+
+ /* search if stream is sendonly or recvonly */
+ _remote_sendrecv =
+ sdp_analyse_attribute (remote_sdp, remote_med);
+ _local_sendrecv = sdp_analyse_attribute (local_sdp, local_med);
+ if (_local_sendrecv == _SENDRECV) {
+ if (_remote_sendrecv == _SENDONLY)
+ _local_sendrecv = _RECVONLY;
+ else if (_remote_sendrecv == _RECVONLY)
+ _local_sendrecv = _SENDONLY;
+ }
+ }
+ }
+ sdp_message_free (local_sdp);
+ sdp_message_free (remote_sdp);
+ }
+
+ _state = event->type;
+ return 0;
+}
int
SipCall::answeredCall(eXosip_event_t *event) {
- SipCall *ca = this;
-
_cid = event->cid;
_did = event->did;
if (_did < 1 && _cid < 1) {
- return -1; /* not enough information for this event?? */
+ exit(0);
+ return -1; /* not enough information for this event?? */
}
-
osip_strncpy(this->_textinfo, event->textinfo, 255);
- osip_strncpy(this->_req_uri, event->req_uri, 255);
- osip_strncpy(this->_local_uri, event->local_uri, 255);
- osip_strncpy(this->_remote_uri, event->remote_uri, 255);
- osip_strncpy(this->_subject, event->subject, 255);
-
- osip_strncpy(ca->_remote_sdp_audio_ip, event->remote_sdp_audio_ip, 49);
- ca->_remote_sdp_audio_port = event->remote_sdp_audio_port;
- osip_strncpy(ca->_payload_name, event->payload_name, 49);
- osip_strncpy (ca->_sdp_body, event->sdp_body, 1000);
- _debug("\n%s\n", ca->_sdp_body);
-
- // For outgoing calls, find the first payload of the remote user
- int i, size;
- char temp[64];
- bzero(temp, 64);
- size = _cdv->size();
- // Codec array in common with the 2 parts
- int m_audio[size];
- for (int a = 0; a < size; a++)
- m_audio[a] = -1;
-
- sdp_message_t *sdp;
- i = sdp_message_init(&sdp);
- if (i != 0) _debug("Cannot allocate a SDP packet\n");
- i = sdp_message_parse(sdp, ca->_sdp_body);
- if (i != 0) _debug("Cannot parse the SDP body\n");
- int pos = 0;
- if (sdp == NULL) _debug("SDP = NULL\n");
-/*********/
- while (!sdp_message_endof_media (sdp, pos)) {
- int k = 0;
- char *tmp = sdp_message_m_media_get (sdp, pos);
-
- if (0 == osip_strncasecmp (tmp, "audio", 5)) {
- char *payload = NULL;
- int c = 0;
- do {
- payload = sdp_message_m_payload_get (sdp, pos, k);
- for (int j = 0; j < size; j++) {
- snprintf(temp, 63, "%d", _cdv->at(j)->getPayload());
- if (payload != NULL and strcmp(temp, payload) == 0) {
- m_audio[c] = _cdv->at(j)->getPayload();
- c++;
- break;
- }
- }
- k++;
- } while (payload != NULL);
- }
- pos++;
- }
-/***********/
- if (m_audio[0] == -1) {
- noSupportedCodec();
- } else {
- ca->payload = m_audio[0];
- setAudioCodec(_cdv->at(0)->alloc(ca->payload, ""));
- }
- _debug("For outgoing call: ca->_payload = %d\n", ca->payload);
-
- sdp_message_free(sdp);
+ if (event->response != NULL) {
+ _status_code = event->response->status_code;
+ snprintf (_reason_phrase, 49, "%s", event->response->reason_phrase);
+ }
+
+ if (event->request != NULL) {
+ char *tmp = NULL;
- if (event->reason_phrase[0]!='\0') {
- osip_strncpy(this->_reason_phrase, event->reason_phrase, 49);
- this->_status_code = event->status_code;
+ osip_from_to_str (event->request->from, &tmp);
+ if (tmp != NULL) {
+ snprintf (_remote_uri, 255, "%s", tmp);
+ osip_free (tmp);
+ }
}
+
+ eXosip_lock ();
+ {
+ osip_message_t *ack = NULL;
+ int i;
+
+ i = eXosip_call_build_ack (_did, &ack);
+ if (i != 0) {
+ _debug("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("Cannot complete ACK with sdp body?!\n");
+ }
+ }
+ sdp_message_free (local_sdp);
+ sdp_message_free (remote_sdp);
+
+ eXosip_call_send_ack (_did, ack);
+ }
+ }
+ eXosip_unlock ();
+
this->_state = event->type;
return 0;
}
-int
-SipCall::onholdCall (eXosip_event_t *event) {
+void
+SipCall::answeredCall_without_hold (eXosip_event_t *event)
+{
SipCall *ca = this;
- osip_strncpy(ca->_textinfo, event->textinfo, 255);
-
- osip_strncpy(ca->_remote_sdp_audio_ip, event->remote_sdp_audio_ip, 49);
- ca->_remote_sdp_audio_port = event->remote_sdp_audio_port;
+ if (ca->enable_audio == 1 && event->response != NULL) {
+ sdp_message_t *sdp = eXosip_get_sdp_info (event->response);
+ if (sdp != NULL) {
+ /* audio is started and session has just been modified */
+ ca->enable_audio = -1;
+ sdp_message_free (sdp);
+ }
+ }
- osip_strncpy(ca->_reason_phrase, event->reason_phrase, 49);
- ca->_status_code = event->status_code;
-
- ca->_state = event->type;
- return 0;
+ if (ca->enable_audio != 1) { /* audio is started */
+ sdp_message_t *remote_sdp;
+ sdp_message_t *local_sdp;
+
+ local_sdp = eXosip_get_sdp_info (event->request);
+ remote_sdp = eXosip_get_sdp_info (event->response);
+ if (remote_sdp == NULL) {
+ _debug("No remote SDP body found for call\n");
+ /* TODO: remote_sdp = retreive from ack above */
+ }
+ if (local_sdp == NULL) {
+ _debug("SDP body was probably in the ACK (TODO)\n");
+ }
+
+ if (remote_sdp != NULL && local_sdp != NULL) {
+ sdp_connection_t *conn;
+ sdp_media_t *local_med;
+ sdp_media_t *remote_med;
+ char *tmp = NULL;
+ int audio_port = 0;
+
+ conn = eXosip_get_audio_connection (remote_sdp);
+ if (conn != NULL && conn->c_addr != NULL) {
+ snprintf (_remote_sdp_audio_ip, 49, "%s", conn->c_addr);
+ }
+ remote_med = eXosip_get_audio_media (remote_sdp);
+ if (remote_med != NULL && remote_med->m_port != NULL) {
+ _remote_sdp_audio_port = atoi (remote_med->m_port);
+ }
+ local_med = eXosip_get_audio_media (local_sdp);
+ if (local_med != NULL && local_med->m_port != NULL) {
+ audio_port = atoi (local_med->m_port);
+ }
+
+ if (_remote_sdp_audio_port > 0
+ && _remote_sdp_audio_ip[0] != '\0' && remote_med != NULL) {
+ tmp = (char *) osip_list_get (remote_med->m_payloads, 0);
+ }
+ if (tmp != NULL) {
+ ca->payload = atoi (tmp);
+ _debug("For outgoing call: ca->_payload = %d\n", ca->payload);
+ setAudioCodec(_cdv->at(0)->alloc(ca->payload, ""));
+ }
+ if (tmp != NULL
+ && audio_port > 0
+ && _remote_sdp_audio_port > 0
+ && _remote_sdp_audio_ip[0] != '\0') {
+
+ /* search if stream is sendonly or recvonly */
+ _remote_sendrecv =
+ sdp_analyse_attribute (remote_sdp, remote_med);
+ _local_sendrecv = sdp_analyse_attribute (local_sdp, local_med);
+ if (_local_sendrecv == _SENDRECV) {
+ if (_remote_sendrecv == _SENDONLY)
+ _local_sendrecv = _RECVONLY;
+ else if (_remote_sendrecv == _RECVONLY)
+ _local_sendrecv = _SENDONLY;
+ }
+ }
+ }
+ sdp_message_free (local_sdp);
+ sdp_message_free (remote_sdp);
+ }
}
int
-SipCall::offholdCall (eXosip_event_t *event) {
- SipCall *ca = this;
+SipCall::sdp_complete_message(sdp_message_t * remote_sdp,
+ osip_message_t * msg)
+{
+ sdp_media_t *remote_med;
+ char *tmp = NULL;
+ char buf[4096];
+ int pos;
- osip_strncpy(ca->_textinfo, event->textinfo, 255);
+ char localip[128];
- osip_strncpy(ca->_remote_sdp_audio_ip, event->remote_sdp_audio_ip, 49);
- ca->_remote_sdp_audio_port = event->remote_sdp_audio_port;
+ // Format port to a char*
+ char port_tmp[64];
+ bzero(port_tmp, 64);
+ snprintf(port_tmp, 63, "%d", _local_audio_port);
+
+ if (remote_sdp == NULL) {
+ _debug("No remote SDP body found for call\n");
+ return -1;
+ }
+ if (msg == NULL) {
+ _debug("No message to complete\n");
+ return -1;
+ }
+
+ eXosip_guess_localip (AF_INET, localip, 128);
+ 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", localip, localip);
+
+ pos = 0;
+ while (!osip_list_eol (remote_sdp->m_medias, pos)) {
+ char payloads[128];
+ int pos2;
+
+ memset (payloads, '\0', sizeof (payloads));
+ remote_med = (sdp_media_t *) osip_list_get (remote_sdp->m_medias, pos);
+
+ if (0 == osip_strcasecmp (remote_med->m_media, "audio")) {
+ pos2 = 0;
+ while (!osip_list_eol (remote_med->m_payloads, pos2)) {
+ tmp = (char *) osip_list_get (remote_med->m_payloads, pos2);
+ if (tmp != NULL &&
+ (0 == osip_strcasecmp (tmp, "0")
+ || 0 == osip_strcasecmp (tmp, "8")
+ || 0 == osip_strcasecmp (tmp, "3"))) {
+ strcat (payloads, tmp);
+ strcat (payloads, " ");
+ }
+ pos2++;
+ }
+ strcat (buf, "m=");
+ strcat (buf, remote_med->m_media);
+ if (pos2 == 0 || payloads[0] == '\0') {
+ strcat (buf, " 0 RTP/AVP \r\n");
+ return -1; /* refuse anyway */
+ } else {
+ strcat (buf, " ");
+ strcat (buf, port_tmp);
+ strcat (buf, " RTP/AVP ");
+ strcat (buf, payloads);
+ strcat (buf, "\r\n");
+
+ if (NULL != strstr (payloads, " 0 ")
+ || (payloads[0] == '0' && payloads[1] == ' '))
+ strcat (buf, "a=rtpmap:0 PCMU/8000\r\n");
+ if (NULL != strstr (payloads, " 8 ")
+ || (payloads[0] == '8' && payloads[1] == ' '))
+ strcat (buf, "a=rtpmap:8 PCMA/8000\r\n");
+ if (NULL != strstr (payloads, " 3 ")
+ || (payloads[0] == '8' && payloads[1] == ' '))
+ strcat (buf, "a=rtpmap:8 GSM/8000\r\n");
+ }
+ } else {
+ strcat (buf, "m=");
+ strcat (buf, remote_med->m_media);
+ strcat (buf, " 0 ");
+ strcat (buf, remote_med->m_proto);
+ strcat (buf, " \r\n");
+ }
+ pos++;
+ }
- osip_strncpy(ca->_reason_phrase, event->reason_phrase, 49);
- ca->_status_code = event->status_code;
-
- ca->_state = event->type;
+ osip_message_set_body (msg, buf, strlen (buf));
+ osip_message_set_content_type (msg, "application/sdp");
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;
+ 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;
+ 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;
+}
+
void
SipCall::alloc(void) {
this->_reason_phrase = new char[50];
diff --git a/src/sipcall.h b/src/sipcall.h
index 2b37a5d77b895ca5ad59d43b0c4663b16a15b1e6..6a8a38ba077b11747e1741440ff43154b6b9eb19 100644
--- a/src/sipcall.h
+++ b/src/sipcall.h
@@ -22,13 +22,16 @@
#ifndef __SIP_CALL_H__
#define __SIP_CALL_H__
-#include <eXosip/eXosip.h>
+#include <eXosip2/eXosip.h>
#include <vector>
class CodecDescriptor;
class AudioCodec;
#define NOT_USED 0
+#define _SENDRECV 0
+#define _SENDONLY 1
+#define _RECVONLY 2
using namespace std;
typedef vector<CodecDescriptor*, allocator<CodecDescriptor*> > CodecDescriptorVector;
@@ -43,9 +46,9 @@ public:
int newIncomingCall (eXosip_event_t *);
int answeredCall (eXosip_event_t *);
+ void answeredCall_without_hold (eXosip_event_t *);
int ringingCall (eXosip_event_t *);
- int onholdCall (eXosip_event_t *);
- int offholdCall (eXosip_event_t *);
+ int receivedAck (eXosip_event_t *);
void setLocalAudioPort (int);
int getLocalAudioPort (void);
@@ -55,11 +58,15 @@ public:
int getDid (void);
void setCid (int cid);
int getCid (void);
+ void setTid (int tid);
+ int getTid (void);
int getRemoteSdpAudioPort (void);
char* getRemoteSdpAudioIp (void);
AudioCodec* getAudioCodec (void);
void setAudioCodec (AudioCodec* ac);
+ inline char* getRemoteUri (void) { return _remote_uri; }
+
inline void setStandBy (bool standby) { _standby = standby; }
inline bool getStandBy (void) { return _standby; }
@@ -67,6 +74,9 @@ private:
void alloc (void);
void dealloc (void);
void noSupportedCodec(void);
+
+ int sdp_complete_message(sdp_message_t * remote_sdp, osip_message_t * msg);
+ int sdp_analyse_attribute (sdp_message_t * sdp, sdp_media_t * med);
CodecDescriptorVector* _cdv;
AudioCodec* _audiocodec;
@@ -74,6 +84,7 @@ private:
short _id;
int _cid; // call id
int _did; // dialog id
+ int _tid; // transaction id
bool _standby; // wait for a cid and did when outgoing call is made
int _status_code;
@@ -88,9 +99,12 @@ private:
char* _remote_sdp_audio_ip;
char* _payload_name;
char* _sdp_body;
+ int _payload;
int _state;
int _local_audio_port;
int _remote_sdp_audio_port;
+ int _local_sendrecv; /* _SENDRECV, _SENDONLY, _RECVONLY */
+ int _remote_sendrecv; /* _SENDRECV, _SENDONLY, _RECVONLY */
};
#endif // __SIP_CALL_H__
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index 300affeae41570f29b52e41a471fab828f1283ff..1594eaf1617a0ab7dcf25ef85b4236c7aa2d84dc 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -18,7 +18,7 @@
*/
#include <sys/time.h>
-#include <eXosip/eXosip.h>
+#include <eXosip2/eXosip.h>
#include <osip2/osip.h>
#include <osipparser2/osip_const.h>
#include <osipparser2/osip_headers.h>
@@ -87,16 +87,26 @@ int
SipVoIPLink::init (void)
{
string tmp;
+ int i;
tmp = string(PROGNAME) + "/" + string(VERSION);
+ i = eXosip_init ();
+ if (i != 0) {
+ _debug("Could not initialize eXosip\n");
+ exit (0);
+ }
+
srand (time(NULL));
- if (eXosip_init (NULL, NULL, DEFAULT_SIP_PORT) != 0) {
- if (eXosip_init (NULL, NULL, RANDOM_SIP_PORT) != 0) {
- _debug("Cannot init eXosip\n");
+ i = eXosip_listen_addr(IPPROTO_UDP, NULL, DEFAULT_SIP_PORT, AF_INET, 0);
+ if (i != 0) {
+ i = eXosip_listen_addr(IPPROTO_UDP, NULL, RANDOM_SIP_PORT, AF_INET, 0);
+ if (i != 0) {
+ _debug("Could not initialize transport layer\n");
return -1;
}
}
+
// If use STUN server, firewall address setup
if (Manager::instance().useStun()) {
eXosip_set_user_agent(tmp.data());
@@ -104,11 +114,13 @@ SipVoIPLink::init (void)
return 0;
}
- eXosip_set_firewallip((Manager::instance().getFirewallAddress()).data());
+ eXosip_masquerade_contact((Manager::instance().getFirewallAddress()).data(),
+ Manager::instance().getFirewallPort());
+
}
eXosip_set_user_agent(tmp.data());
- initRtpmapCodec();
+// initRtpmapCodec();
_evThread->start();
return 1;
}
@@ -126,33 +138,7 @@ SipVoIPLink::isInRtpmap (int index, int payload, CodecDescriptorVector* cdv) {
void
SipVoIPLink::initRtpmapCodec (void)
{
- int payload;
- unsigned int nb;
- char rtpmap[128];
- char tmp[64];
- bzero(rtpmap, 128);
- bzero(tmp, 64);
-
- /* reset all payload to fit application capabilities */
- eXosip_sdp_negotiation_remove_audio_payloads();
-
- // Set rtpmap according to the supported codec order
- nb = Manager::instance().getNumberOfCodecs();
- for (unsigned int i = 0; i < nb; i++) {
- payload = Manager::instance().getCodecDescVector()->at(i)->getPayload();
-
- // Add payload to rtpmap if it is not already added
- if (!isInRtpmap(i, payload, Manager::instance().getCodecDescVector())) {
- snprintf(rtpmap, 127, "%d %s/%d", payload,
- Manager::instance().getCodecDescVector()->at(i)->rtpmapPayload(payload).data(), SAMPLING_RATE);
- snprintf(tmp, 63, "%d", payload);
-
- eXosip_sdp_negotiation_add_codec( osip_strdup(tmp), NULL,
- osip_strdup("RTP/AVP"), NULL, NULL, NULL, NULL,NULL,
- osip_strdup(rtpmap));
- }
- }
}
void
@@ -164,7 +150,9 @@ SipVoIPLink::quit(void)
int
SipVoIPLink::setRegister (void)
{
+ int i;
int reg_id = -1;
+ osip_message_t *reg = NULL;
string proxy = "sip:" + get_config_fields_str(SIGNALISATION, PROXY);
@@ -189,21 +177,20 @@ SipVoIPLink::setRegister (void)
return -1;
}
- _debug("register From: %s\n", from.data());
+ _debug("REGISTER From: %s\n", from.data());
if (!get_config_fields_str(SIGNALISATION, PROXY).empty()) {
- reg_id = eXosip_register_init((char*)from.data(),
- (char*)proxy.data(),NULL);
+ reg_id = eXosip_register_build_initial_register ((char*)from.data(),
+ (char*)proxy.data(), NULL, EXPIRES_VALUE, ®);
} else {
- reg_id = eXosip_register_init((char*)from.data(),
- (char*)hostname.data(), NULL);
+ reg_id = eXosip_register_build_initial_register ((char*)from.data(),
+ (char*)hostname.data(), NULL, EXPIRES_VALUE, ®);
}
if (reg_id < 0) {
eXosip_unlock();
return -1;
}
-
- int i = eXosip_register(reg_id, EXPIRES_VALUE);
-
+
+ i = eXosip_register_send_register (reg_id, reg);
if (i == -2) {
_debug("cannot build registration, check the setup\n");
eXosip_unlock();
@@ -216,7 +203,7 @@ SipVoIPLink::setRegister (void)
}
eXosip_unlock();
- return 0;
+ return i;
}
int
SipVoIPLink::outgoingInvite (short id, const string& to_url)
@@ -271,6 +258,7 @@ int
SipVoIPLink::answer (short id)
{
int i;
+ int port;
char tmpbuf[64];
bzero (tmpbuf, 64);
// Get port
@@ -278,10 +266,34 @@ SipVoIPLink::answer (short id)
_debug("Answer call [id = %d, cid = %d, did = %d]\n",
id, getSipCall(id)->getCid(), getSipCall(id)->getDid());
- _debug("Local audio port: %d\n", getSipCall(id)->getLocalAudioPort());
+ port = getSipCall(id)->getLocalAudioPort();
+ _debug("Local audio port: %d\n", port);
+
+ osip_message_t *answer = NULL;
+ SipCall* ca = getSipCall(id);
+
+ // Send 180 RINGING
+ eXosip_lock ();
+ eXosip_call_send_answer (ca->getTid(), RINGING, NULL);
+ eXosip_unlock ();
+
+ // Send 200 OK
eXosip_lock();
- i = eXosip_answer_call(getSipCall(id)->getDid(), 200, tmpbuf);
+ i = eXosip_call_build_answer (ca->getTid(), OK, &answer);
+ if (i != 0) {
+ // Send 400 BAD_REQUEST
+ eXosip_call_send_answer (ca->getTid(), BAD_REQ, NULL);
+ } else {
+ i = sdp_complete_200ok (ca->getDid(), answer, port);
+ if (i != 0) {
+ osip_message_free (answer);
+ // Send 415 UNSUPPORTED_MEDIA_TYPE
+ eXosip_call_send_answer (ca->getTid(), UNSUP_MEDIA_TYPE, NULL);
+ } else {
+ eXosip_call_send_answer (ca->getTid(), OK, answer);
+ }
+ }
eXosip_unlock();
// Incoming call is answered, start the sound channel.
@@ -301,7 +313,7 @@ SipVoIPLink::hangup (short id)
id, getSipCall(id)->getCid(), getSipCall(id)->getDid());
// Release SIP stack.
eXosip_lock();
- i = eXosip_terminate_call (getSipCall(id)->getCid(),
+ i = eXosip_call_terminate (getSipCall(id)->getCid(),
getSipCall(id)->getDid());
eXosip_unlock();
@@ -321,7 +333,7 @@ SipVoIPLink::cancel (short id)
_debug("Cancel call [id = %d, cid = %d]\n", id, getCid());
// Release SIP stack.
eXosip_lock();
- i = eXosip_terminate_call (getCid(), -1);
+ i = eXosip_call_terminate (getCid(), -1);
eXosip_unlock();
}
deleteSipCall(id);
@@ -331,12 +343,64 @@ SipVoIPLink::cancel (short id)
int
SipVoIPLink::onhold (short id)
{
- int i;
+ osip_message_t *invite;
+ int i;
+ int did;
+
+ sdp_message_t *local_sdp = NULL;
+
+ did = getSipCall(id)->getDid();
- eXosip_lock();
- i = eXosip_on_hold_call(getSipCall(id)->getDid());
- eXosip_unlock();
+ eXosip_lock ();
+ local_sdp = eXosip_get_local_sdp (did);
+ eXosip_unlock ();
+
+ if (local_sdp == NULL) {
+ return -1;
+ }
+
+/*
+ char port[10];
+ snprintf (port, 10, "%d", getSipCall(id)->getLocalAudioPort());
+ */
+ eXosip_lock ();
+ // Build INVITE_METHOD for put call on-hold
+ i = eXosip_call_build_request (did, INVITE_METHOD, &invite);
+ eXosip_unlock ();
+
+ if (i != 0) {
+ sdp_message_free(local_sdp);
+ return -1;
+ }
+
+ /* add sdp body */
+ {
+ char *tmp = NULL;
+
+ i = sdp_hold_call (local_sdp);
+ if (i != 0) {
+ sdp_message_free (local_sdp);
+ osip_message_free (invite);
+ return -1;
+ }
+
+ i = sdp_message_to_str (local_sdp, &tmp);
+ sdp_message_free (local_sdp);
+ if (i != 0) {
+ osip_message_free (invite);
+ return -1;
+ }
+ osip_message_set_body (invite, tmp, strlen (tmp));
+ osip_free (tmp);
+ osip_message_set_content_type (invite, "application/sdp");
+ }
+
+ eXosip_lock ();
+ // Send request
+ i = eXosip_call_send_request (did, invite);
+ eXosip_unlock ();
+
// Disable audio
_audiortp->closeRtpSession(getSipCall(id));
return i;
@@ -345,12 +409,62 @@ SipVoIPLink::onhold (short id)
int
SipVoIPLink::offhold (short id)
{
- int i;
-
- eXosip_lock();
- i = eXosip_off_hold_call(getSipCall(id)->getDid(), NULL, 0);
- eXosip_unlock();
+ osip_message_t *invite;
+ int i;
+ int did;
+
+ sdp_message_t *local_sdp = NULL;
+
+ did = getSipCall(id)->getDid();
+ eXosip_lock ();
+ local_sdp = eXosip_get_local_sdp (did);
+ eXosip_unlock ();
+ if (local_sdp == NULL) {
+ return -1;
+ }
+
+ /*
+ char port[10];
+ snprintf (port, 10, "%d", getSipCall(id)->getLocalAudioPort());
+ */
+ eXosip_lock ();
+ // Build INVITE_METHOD for put call off-hold
+ i = eXosip_call_build_request (did, INVITE_METHOD, &invite);
+ eXosip_unlock ();
+
+ if (i != 0) {
+ sdp_message_free(local_sdp);
+ return -1;
+ }
+
+ /* add sdp body */
+ {
+ char *tmp = NULL;
+
+ i = sdp_off_hold_call (local_sdp);
+ if (i != 0) {
+ sdp_message_free (local_sdp);
+ osip_message_free (invite);
+ return -1;
+ }
+
+ i = sdp_message_to_str (local_sdp, &tmp);
+ sdp_message_free (local_sdp);
+ if (i != 0) {
+ osip_message_free (invite);
+ return -1;
+ }
+ osip_message_set_body (invite, tmp, strlen (tmp));
+ osip_free (tmp);
+ osip_message_set_content_type (invite, "application/sdp");
+ }
+
+ eXosip_lock ();
+ // Send request
+ i = eXosip_call_send_request (did, invite);
+ eXosip_unlock ();
+
// Enable audio
if (_audiortp->createNewSession (getSipCall(id)) < 0) {
_debug("FATAL: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
@@ -362,6 +476,7 @@ SipVoIPLink::offhold (short id)
int
SipVoIPLink::transfer (short id, const string& to)
{
+ osip_message_t *refer;
int i;
string tmp_to;
tmp_to = toHeader(to);
@@ -370,7 +485,13 @@ SipVoIPLink::transfer (short id, const string& to)
}
eXosip_lock();
- i = eXosip_transfer_call(getSipCall(id)->getDid(), (char*)tmp_to.data());
+ // Build transfer request
+ i = eXosip_call_build_refer (getSipCall(id)->getDid(), (char*)tmp_to.data(),
+ &refer);
+ if (i == 0) {
+ // Send transfer request
+ i = eXosip_call_send_request (getSipCall(id)->getDid(), refer);
+ }
eXosip_unlock();
return i;
}
@@ -384,8 +505,12 @@ SipVoIPLink::refuse (short id)
// Get local port
snprintf (tmpbuf, 63, "%d", getSipCall(id)->getLocalAudioPort());
+ osip_message_t *answer = NULL;
eXosip_lock();
- i = eXosip_answer_call(getSipCall(id)->getDid(), BUSY_HERE, tmpbuf);
+ i = eXosip_call_build_answer (getSipCall(id)->getTid(), BUSY_HERE, &answer);
+ if (i == 0) {
+ i = eXosip_call_send_answer (getSipCall(id)->getTid(), BUSY_HERE, answer);
+ }
eXosip_unlock();
return i;
}
@@ -396,16 +521,18 @@ SipVoIPLink::getEvent (void)
eXosip_event_t *event;
short id;
char *name;
- static int countReg = 0;
- eXosip_automatic_refresh();
event = eXosip_event_wait (0, 50);
+ eXosip_lock();
+ eXosip_automatic_action();
+ eXosip_unlock();
+
if (event == NULL) {
return -1;
}
switch (event->type) {
// IP-Phone user receives a new call
- case EXOSIP_CALL_NEW: //
+ case EXOSIP_CALL_INVITE: //
// Set local random port for incoming call
if (!Manager::instance().useStun()) {
setLocalPort(RANDOM_LOCAL_PORT);
@@ -418,6 +545,7 @@ SipVoIPLink::getEvent (void)
}
}
+ // Generate id
id = Manager::instance().generateNewCallId();
Manager::instance().pushBackNewCall(id, Incoming);
_debug("Incoming Call with identifiant %d [cid = %d, did = %d]\n",
@@ -427,7 +555,17 @@ SipVoIPLink::getEvent (void)
// Display the callerId-name
osip_from_t *from;
osip_from_init(&from);
- osip_from_parse(from, event->remote_uri);
+
+ if (event->request != NULL) {
+ char *tmp = NULL;
+
+ osip_from_to_str (event->request->from, &tmp);
+ if (tmp != NULL) {
+ snprintf (getSipCall(id)->getRemoteUri(), 256, "%s", tmp);
+ osip_free (tmp);
+ }
+ }
+ osip_from_parse(from, getSipCall(id)->getRemoteUri());
name = osip_from_get_displayname(from);
Manager::instance().displayTextMessage(id, name);
if (Manager::instance().getCall(id) != NULL) {
@@ -438,15 +576,16 @@ SipVoIPLink::getEvent (void)
_debug("From: %s\n", name);
osip_from_free(from);
+ // Associate an audio port with a call
+ getSipCall(id)->setLocalAudioPort(_localPort);
+
+
getSipCall(id)->newIncomingCall(event);
if (Manager::instance().incomingCall(id) < 0) {
Manager::instance().displayErrorText("Incoming call failed");
return -1;
}
- // Associate an audio port with a call
- getSipCall(id)->setLocalAudioPort(_localPort);
-
break;
// The peer-user answers
@@ -463,6 +602,7 @@ SipVoIPLink::getEvent (void)
and !Manager::instance().getCall(id)->isOffHold()) {
getSipCall(id)->setStandBy(false);
if (getSipCall(id)->answeredCall(event) != -1) {
+ getSipCall(id)->answeredCall_without_hold(event);
Manager::instance().peerAnsweredCall(id);
// Outgoing call is answered, start the sound channel.
@@ -472,9 +612,15 @@ SipVoIPLink::getEvent (void)
exit(1);
}
}
+ } else {
+ // Answer to on/off hold to send ACK
+ if (id > 0) {
+ getSipCall(id)->answeredCall(event);
+ _debug("-----------------------\n");
+ }
}
break;
-
+
case EXOSIP_CALL_RINGING: //peer call is ringing
id = findCallIdWhenRinging();
@@ -492,6 +638,17 @@ SipVoIPLink::getEvent (void)
case EXOSIP_CALL_REDIRECTED:
break;
+ case EXOSIP_CALL_ACK:
+ id = findCallId(event);
+ _debug("ACK received [id = %d, cid = %d, did = %d]\n",
+ id, event->cid, event->did);
+ if (id > 0) {
+ getSipCall(id)->receivedAck(event);
+ } else {
+ return -1;
+ }
+ break;
+
// The peer-user closed the phone call(we received BYE).
case EXOSIP_CALL_CLOSED:
id = findCallId(event);
@@ -509,35 +666,14 @@ SipVoIPLink::getEvent (void)
}
break;
- case EXOSIP_CALL_HOLD:
- id = findCallId(event);
- if (id > 0) {
- getSipCall(id)->onholdCall(event);
- } else {
- return -1;
- }
- break;
-
- case EXOSIP_CALL_OFFHOLD:
- id = findCallId(event);
- if (id > 0) {
- getSipCall(id)->offholdCall(event);
- } else {
- return -1;
- }
- break;
-
case EXOSIP_CALL_REQUESTFAILURE:
id = findCallId(event);
// Handle 4XX errors
- switch (event->status_code) {
+ switch (event->response->status_code) {
case AUTH_REQUIRED:
- if (setAuthentication() == -1) {
- break;
- }
eXosip_lock();
- eXosip_retry_call (event->cid);
+ eXosip_automatic_action();
eXosip_unlock();
break;
case BAD_REQ:
@@ -551,7 +687,7 @@ SipVoIPLink::getEvent (void)
case ADDR_INCOMPLETE:
case BUSY_HERE:
// Display error on the screen phone
- Manager::instance().displayError(event->reason_phrase);
+ Manager::instance().displayError(event->response->reason_phrase);
Manager::instance().congestion(true);
break;
case REQ_TERMINATED:
@@ -563,7 +699,7 @@ SipVoIPLink::getEvent (void)
case EXOSIP_CALL_SERVERFAILURE:
// Handle 5XX errors
- switch (event->status_code) {
+ switch (event->response->status_code) {
case SERVICE_UNAVAILABLE:
Manager::instance().ringback(false);
Manager::instance().congestion(true);
@@ -575,7 +711,7 @@ SipVoIPLink::getEvent (void)
case EXOSIP_CALL_GLOBALFAILURE:
// Handle 6XX errors
- switch (event->status_code) {
+ switch (event->response->status_code) {
case BUSY_EVERYWHERE:
case DECLINE:
Manager::instance().ringback(false);
@@ -587,64 +723,36 @@ SipVoIPLink::getEvent (void)
break;
case EXOSIP_REGISTRATION_SUCCESS:
- _debug("-- Registration succeeded --\n");
Manager::instance().displayStatus(LOGGED_IN_STATUS);
break;
case EXOSIP_REGISTRATION_FAILURE:
- _debug("-- Registration failed --\n");
- if (countReg <= 3) {
- setRegister();
- countReg++;
- }
-
break;
- case EXOSIP_OPTIONS_NEW:
- /* answer the OPTIONS method */
- /* 1: search for an existing call */
+ case EXOSIP_MESSAGE_NEW:
unsigned int k;
- for (k = 0; k < _sipcallVector->size(); k++) {
- if (_sipcallVector->at(k)->getCid() == event->cid) {
- break;
+ if (event->request != NULL && MSG_IS_OPTIONS(event->request)) {
+ for (k = 0; k < _sipcallVector->size(); k++) {
+ if (_sipcallVector->at(k)->getCid() == event->cid) {
+ break;
+ }
}
+
+ // TODO: Que faire si rien trouve??
+ eXosip_lock();
+ if (_sipcallVector->at(k)->getCid() == event->cid) {
+ /* already answered! */
+ }
+ else if (k == _sipcallVector->size()) {
+ /* answer 200 ok */
+ eXosip_options_send_answer (event->tid, OK, NULL);
+ } else {
+ /* answer 486 ok */
+ eXosip_options_send_answer (event->tid, BUSY_HERE, NULL);
+ }
+ eXosip_unlock();
}
-
- // TODO: Que faire si rien trouve??
- eXosip_lock();
- if (_sipcallVector->at(k)->getCid() == event->cid) {
- /* already answered! */
- }
- else if (k == _sipcallVector->size()) {
- /* answer 200 ok */
- eXosip_answer_options (event->cid, event->did, 200);
- } else {
- /* answer 486 ok */
- eXosip_answer_options (event->cid, event->did, 486);
- }
- eXosip_unlock();
- break;
-
- case EXOSIP_OPTIONS_ANSWERED:
- break;
-
- case EXOSIP_OPTIONS_PROCEEDING:
- break;
-
- case EXOSIP_OPTIONS_REDIRECTED:
- break;
-
- case EXOSIP_OPTIONS_REQUESTFAILURE:
- break;
-
- case EXOSIP_OPTIONS_SERVERFAILURE:
- break;
-
- case EXOSIP_OPTIONS_GLOBALFAILURE:
- break;
-
- case EXOSIP_SUBSCRIPTION_NOTIFY:
break;
default:
@@ -671,17 +779,23 @@ SipVoIPLink::setLocalPort (int port)
void
SipVoIPLink::carryingDTMFdigits (short id, char code) {
int duration = get_config_fields_int(SIGNALISATION, PULSE_LENGTH);
-
- static const int body_len = 128;
+ osip_message_t *info;
+ const int body_len = 1000;
+ int i;
char *dtmf_body = new char[body_len];
- snprintf(dtmf_body, body_len - 1,
- "Signal=%c\r\nDuration=%d\r\n",
- code, duration);
eXosip_lock();
- eXosip_info_call(getSipCall(id)->getDid(), "application/dtmf-relay",
- dtmf_body);
+ // Build info request
+ i = eXosip_call_build_info (getSipCall(id)->getDid(), &info);
+ if (i == 0) {
+ snprintf(dtmf_body, body_len - 1, "Signal=%c\r\nDuration=%d\r\n",
+ code, duration);
+ osip_message_set_content_type (info, "application/dtmf-relay");
+ osip_message_set_body (info, dtmf_body, strlen (dtmf_body));
+ // Send info request
+ i = eXosip_call_send_request (getSipCall(id)->getDid(), info);
+ }
eXosip_unlock();
delete[] dtmf_body;
@@ -741,6 +855,178 @@ SipVoIPLink::getAudioCodec (short callid)
///////////////////////////////////////////////////////////////////////////////
// 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;
+}
+
+int
+SipVoIPLink::sdp_complete_200ok (int did, osip_message_t * answer, int port)
+{
+ sdp_message_t *remote_sdp;
+ sdp_media_t *remote_med;
+ char *tmp = NULL;
+ char buf[4096];
+ char port_tmp[64];
+ int pos;
+
+ char localip[128];
+
+ // Format port to a char*
+ bzero(port_tmp, 64);
+ snprintf(port_tmp, 63, "%d", port);
+
+ remote_sdp = eXosip_get_remote_sdp (did);
+ if (remote_sdp == NULL) {
+ return -1; /* no existing body? */
+ }
+
+ eXosip_guess_localip (AF_INET, localip, 128);
+ 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", localip,localip);
+
+ pos = 0;
+ while (!osip_list_eol (remote_sdp->m_medias, pos)) {
+ char payloads[128];
+ int pos2;
+
+ memset (payloads, '\0', sizeof (payloads));
+ remote_med = (sdp_media_t *) osip_list_get (remote_sdp->m_medias, pos);
+
+ if (0 == osip_strcasecmp (remote_med->m_media, "audio")) {
+ pos2 = 0;
+ while (!osip_list_eol (remote_med->m_payloads, pos2)) {
+ tmp = (char *) osip_list_get (remote_med->m_payloads, pos2);
+ if (tmp != NULL && (0 == osip_strcasecmp (tmp, "0")
+ || 0 == osip_strcasecmp (tmp, "8")
+ || 0 == osip_strcasecmp (tmp, "3"))) {
+ strcat (payloads, tmp);
+ strcat (payloads, " ");
+ }
+ pos2++;
+ }
+ strcat (buf, "m=");
+ strcat (buf, remote_med->m_media);
+ if (pos2 == 0 || payloads[0] == '\0') {
+ strcat (buf, " 0 RTP/AVP \r\n");
+ sdp_message_free (remote_sdp);
+ return -1; /* refuse anyway */
+ } else {
+ strcat (buf, " ");
+ strcat (buf, port_tmp);
+ strcat (buf, " RTP/AVP ");
+ strcat (buf, payloads);
+ strcat (buf, "\r\n");
+
+ if (NULL != strstr (payloads, " 0 ")
+ || (payloads[0] == '0' && payloads[1] == ' ')) {
+ strcat (buf, "a=rtpmap:0 PCMU/8000\r\n");
+ }
+ if (NULL != strstr (payloads, " 8 ")
+ || (payloads[0] == '8' && payloads[1] == ' ')) {
+ strcat (buf, "a=rtpmap:8 PCMA/8000\r\n");
+ }
+ if (NULL != strstr (payloads, " 3")
+ || (payloads[0] == '3' && payloads[1] == ' ')) {
+ strcat (buf, "a=rtpmap:3 GSM/8000\r\n");
+ }
+ }
+ } else {
+ strcat (buf, "m=");
+ strcat (buf, remote_med->m_media);
+ strcat (buf, " 0 ");
+ strcat (buf, remote_med->m_proto);
+ strcat (buf, " \r\n");
+ }
+ pos++;
+ }
+
+ osip_message_set_body (answer, buf, strlen (buf));
+ osip_message_set_content_type (answer, "application/sdp");
+ sdp_message_free (remote_sdp);
+ return 0;
+}
+
int
SipVoIPLink::behindNat (void)
@@ -852,18 +1138,10 @@ SipVoIPLink::startCall (short id, const string& from, const string& to,
Manager::instance().error()->errorName(TO_ERROR);
return -1;
}
-
- i = eXosip_build_initial_invite(&invite,(char*)to.data(),(char*)from.data(),
- (char*)route.data(), (char*)subject.data());
- if (i != 0) {
- return -1;
- }
-
- eXosip_lock();
char port[64];
if (!Manager::instance().useStun()) {
- // Set random port for outgoing call
+ // Set random port for outgoing call if no firewall
setLocalPort(RANDOM_LOCAL_PORT);
_debug("Local audio port: %d\n",_localPort);
} else {
@@ -876,6 +1154,7 @@ SipVoIPLink::startCall (short id, const string& from, const string& to,
}
}
+ // Set local audio port for sipcall(id)
if (getSipCall(id) != NULL) {
getSipCall(id)->setLocalAudioPort(_localPort);
} else {
@@ -884,12 +1163,73 @@ SipVoIPLink::startCall (short id, const string& from, const string& to,
bzero (port, 64);
snprintf (port, 63, "%d", getLocalPort());
-
- i = eXosip_initiate_call(invite, NULL, NULL, port);
+
+ i = eXosip_call_build_initial_invite (&invite, (char*)to.data(),
+ (char*)from.data(),
+ (char*)route.data(),
+ (char*)subject.data());
+ if (i != 0) {
+ return -1;
+ }
+
+
+ int payload;
+ unsigned int nb;
+ char rtpmap[128];
+ char rtpmap_attr[2048];
+ char media[64];
+ char media_audio[64];
+
+ bzero(rtpmap, 128);
+ bzero(rtpmap_attr, 2048);
+ bzero(media, 64);
+ bzero(media_audio, 64);
+
+ // Set rtpmap according to the supported codec order
+ nb = Manager::instance().getNumberOfCodecs();
+ for (unsigned int i = 0; i < nb; i++) {
+ payload = Manager::instance().getCodecDescVector()->at(i)->getPayload();
+
+ // Add payload to rtpmap if it is not already added
+ if (!isInRtpmap(i, payload, Manager::instance().getCodecDescVector())) {
+ snprintf(media, 63, "%d ", payload);
+ strcat (media_audio, media);
+
+ snprintf(rtpmap, 127, "a=rtpmap: %d %s/%d\r\n", payload,
+ Manager::instance().getCodecDescVector()->at(i)->rtpmapPayload(payload).data(), SAMPLING_RATE);
+ strcat(rtpmap_attr, rtpmap);
+ }
+ }
+
+ /* add sdp body */
+ {
+ char tmp[4096];
+ char localip[128];
+
+ eXosip_guess_localip (AF_INET, localip, 128);
+ 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 %s RTP/AVP %s\r\n"
+ "%s",
+ localip, localip, port, media_audio, rtpmap_attr);
+
+ osip_message_set_body (invite, tmp, strlen (tmp));
+ osip_message_set_content_type (invite, "application/sdp");
+ }
+
+ eXosip_lock();
+
+ i = eXosip_call_send_initial_invite (invite);
if (i <= 0) {
eXosip_unlock();
return -1;
+ } else {
+ eXosip_call_set_reference (i, NULL);
}
eXosip_unlock();
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
index ae34952765d451170a34e07bf43b51e8f2637c04..fbcbf7d9c3f3469e67918f9ecf37323fcfdf97c4 100644
--- a/src/sipvoiplink.h
+++ b/src/sipvoiplink.h
@@ -20,7 +20,8 @@
#ifndef __SIP_VOIP_LINK_H__
#define __SIP_VOIP_LINK_H__
-#include <eXosip/eXosip.h>
+#include <eXosip2/eXosip.h>
+#include <osipparser2/sdp_message.h>
#include <string>
#include <vector>
@@ -30,8 +31,12 @@
using namespace std;
#define EXPIRES_VALUE 180
+#define INVITE_METHOD "INVITE"
// 1XX responses
#define DIALOG_ESTABLISHED 101
+#define RINGING 180
+// 2XX
+#define OK 200
// 4XX Errors
#define BAD_REQ 400
#define UNAUTHORIZED 401
@@ -41,6 +46,7 @@ using namespace std;
#define NOT_ACCEPTABLE 406
#define AUTH_REQUIRED 407
#define REQ_TIMEOUT 408
+#define UNSUP_MEDIA_TYPE 415
#define TEMP_UNAVAILABLE 480
#define ADDR_INCOMPLETE 484
#define BUSY_HERE 486
@@ -122,6 +128,21 @@ private:
short findCallId (eXosip_event_t *e);
short findCallIdWhenRinging (void);
bool isInRtpmap (int index, int payload, CodecDescriptorVector* cdv);
+
+ /*
+ * To build sdp with 200 OK when answered call
+ */
+ int sdp_complete_200ok(int did, osip_message_t * answer, int port);
+
+ /*
+ * 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* _evThread;
SipCallVector* _sipcallVector;