diff --git a/AUTHORS b/AUTHORS
index c160be3ad66aa39d814cc1e031a1215d39998add..9c394b8203b62d4774d04c9291838f97bd86fe8d 100644
--- a/AUTHORS
+++ b/AUTHORS
@@ -12,3 +12,6 @@ Sherry Yang (syangs04 at yahoo dot com)
Imran Akbar (imr at stanford dot edu)
- Working on Win32 port
+
+Contributors:
+Mikael Magnusson
\ No newline at end of file
diff --git a/CHANGES b/CHANGES
index 46a79d0dd4fac35c9b9ca16363b0a90cfc952752..cdbf50b138cdd0d0ab17cab5f1c050bf7efd3bbc 100644
--- a/CHANGES
+++ b/CHANGES
@@ -2,6 +2,8 @@
SFLphone (0.4.1-pre2) / now
* rearranged utilspp use
* Bug fix when trying to use more than 6 lines.
+ * Now, we send a final response to reINVITEs
+ (thanks to Mikael Magnusson)
SFLphone (0.4.1-pre1) / 2005-08-11
* Use libeXosip2
diff --git a/ChangeLog b/ChangeLog
index c1435fbd31487c9e881c81eafa40a4bc5a5de09f..3a7eecc9b0c4a7d86db55f143306a381ee8d51ac 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,7 @@
+Jean-Philippe Barette-LaPierre (30 April 2005) version 0.4.1-pre1
+- Now, we send a final response to reINVITEs
+ (thanks to Mikael Magnusson)
+
Jean-Philippe Barette-LaPierre (29 April 2005) version 0.4.1-pre1
- Bug fix when trying to use more than 6 lines.
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index a26cf65cf82e5b9040c866955dec851878e1f9e9..78602dd790e277741ee4ce7ee90c81a75e9f8e19 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -55,419 +55,417 @@ using namespace std;
#define VOICE_MSG "Voice-Message"
#define LENGTH_VOICE_MSG 15
-SipVoIPLink::SipVoIPLink (short id) : VoIPLink (id)
+SipVoIPLink::SipVoIPLink (short id)
+ : VoIPLink (id)
{
- setId(id);
- _localPort = 0;
- _cid = 0;
- _reg_id = -1;
- _nMsgVoicemail = 0;
- _evThread = new EventThread (this);
- _sipcallVector = new SipCallVector();
- _audiortp = new AudioRtp();
+ setId(id);
+ _localPort = 0;
+ _cid = 0;
+ _reg_id = -1;
+ _nMsgVoicemail = 0;
+ _evThread = new EventThread(this);
+ _sipcallVector = new SipCallVector();
+ _audiortp = new AudioRtp();
}
SipVoIPLink::~SipVoIPLink (void)
{
- if (_evThread != NULL) {
- delete _evThread;
- _evThread = NULL;
- }
- delete _sipcallVector;
- delete _audiortp;
+ delete _sipcallVector;
+ delete _audiortp;
}
bool
SipVoIPLink::checkNetwork (void)
{
- // Set IP address
- if (getLocalIp() == -1) {
- // If no network
- return false;
- } else {
- return true;
- }
+ // Set IP address
+ if (getLocalIp() == -1) {
+ // If no network
+ return false;
+ } else {
+ return true;
+ }
}
int
SipVoIPLink::init (void)
{
- string tmp;
- int i;
+ string tmp;
+ int i;
- tmp = string(PROGNAME) + "/" + string(VERSION);
+ tmp = string(PROGNAME) + "/" + string(VERSION);
- i = eXosip_init ();
- if (i != 0) {
- _debug("Could not initialize eXosip\n");
- exit (0);
- }
+ i = eXosip_init ();
+ if (i != 0) {
+ _debug("Could not initialize eXosip\n");
+ exit (0);
+ }
- srand (time(NULL));
- 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;
- }
- }
+ srand (time(NULL));
+ 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());
- if (behindNat() != 1) {
- return 0;
- }
+ // If use STUN server, firewall address setup
+ if (Manager::instance().useStun()) {
+ eXosip_set_user_agent(tmp.data());
+ if (behindNat() != 1) {
+ return 0;
+ }
- eXosip_masquerade_contact((Manager::instance().getFirewallAddress()).data(),
- Manager::instance().getFirewallPort());
+ eXosip_masquerade_contact((Manager::instance().getFirewallAddress()).data(),
+ Manager::instance().getFirewallPort());
- }
+ }
- // Set user agent
- eXosip_set_user_agent(tmp.data());
+ // Set user agent
+ eXosip_set_user_agent(tmp.data());
- _evThread->start();
- return 1;
+ _evThread->start();
+ return 1;
}
bool
SipVoIPLink::isInRtpmap (int index, int payload, CodecDescriptorVector* cdv) {
- for (int i = 0; i < index; i++) {
- if (cdv->at(i)->getPayload() == payload) {
- return true;
- }
+ for (int i = 0; i < index; i++) {
+ if (cdv->at(i)->getPayload() == payload) {
+ return true;
}
- return false;
+ }
+ return false;
}
void
-SipVoIPLink::quit(void)
+SipVoIPLink::terminate(void)
{
- eXosip_quit();
+ delete _evThread;
+ eXosip_quit();
}
int
SipVoIPLink::setRegister (void)
{
- int i;
- osip_message_t *reg = NULL;
+ int i;
+ osip_message_t *reg = NULL;
- string proxy = "sip:" + get_config_fields_str(SIGNALISATION, PROXY);
+ string proxy = "sip:" + get_config_fields_str(SIGNALISATION, PROXY);
- string hostname = "sip:" + get_config_fields_str(SIGNALISATION, HOST_PART);
+ string hostname = "sip:" + get_config_fields_str(SIGNALISATION, HOST_PART);
- string from = fromHeader(get_config_fields_str(SIGNALISATION, USER_PART),
- get_config_fields_str(SIGNALISATION, HOST_PART));
+ string from = fromHeader(get_config_fields_str(SIGNALISATION, USER_PART),
+ get_config_fields_str(SIGNALISATION, HOST_PART));
- if (get_config_fields_str(SIGNALISATION, HOST_PART).empty()) {
- Manager::instance().error()->errorName(HOST_PART_FIELD_EMPTY);
- return -1;
- }
- if (get_config_fields_str(SIGNALISATION, USER_PART).empty()) {
- Manager::instance().error()->errorName(USER_PART_FIELD_EMPTY);
- return -1;
- }
+ if (get_config_fields_str(SIGNALISATION, HOST_PART).empty()) {
+ Manager::instance().error()->errorName(HOST_PART_FIELD_EMPTY);
+ return -1;
+ }
+ if (get_config_fields_str(SIGNALISATION, USER_PART).empty()) {
+ Manager::instance().error()->errorName(USER_PART_FIELD_EMPTY);
+ return -1;
+ }
- eXosip_lock();
- if (setAuthentication() == -1) {
- _debug("No authentication\n");
- eXosip_unlock();
- return -1;
- }
+ eXosip_lock();
+ if (setAuthentication() == -1) {
+ _debug("No authentication\n");
+ eXosip_unlock();
+ return -1;
+ }
- _debug("REGISTER From: %s\n", from.data());
- if (!get_config_fields_str(SIGNALISATION, PROXY).empty()) {
- _reg_id = eXosip_register_build_initial_register ((char*)from.data(),
- (char*)proxy.data(), NULL, EXPIRES_VALUE, ®);
- } else {
- _reg_id = eXosip_register_build_initial_register ((char*)from.data(),
- (char*)hostname.data(), NULL, EXPIRES_VALUE, ®);
- }
- if (_reg_id < 0) {
- eXosip_unlock();
- return -1;
- }
-
- i = eXosip_register_send_register (_reg_id, reg);
- if (i == -2) {
- _debug("cannot build registration, check the setup\n");
- eXosip_unlock();
- return -1;
- }
- if (i == -1) {
- _debug("Registration Failed\n");
- eXosip_unlock();
- return -1;
- }
+ _debug("REGISTER From: %s\n", from.data());
+ if (!get_config_fields_str(SIGNALISATION, PROXY).empty()) {
+ _reg_id = eXosip_register_build_initial_register ((char*)from.data(),
+ (char*)proxy.data(), NULL, EXPIRES_VALUE, ®);
+ } else {
+ _reg_id = eXosip_register_build_initial_register ((char*)from.data(),
+ (char*)hostname.data(), NULL, EXPIRES_VALUE, ®);
+ }
+ if (_reg_id < 0) {
+ eXosip_unlock();
+ return -1;
+ }
+
+ i = eXosip_register_send_register (_reg_id, reg);
+ if (i == -2) {
+ _debug("cannot build registration, check the setup\n");
+ eXosip_unlock();
+ return -1;
+ }
+ if (i == -1) {
+ _debug("Registration Failed\n");
+ eXosip_unlock();
+ return -1;
+ }
- eXosip_unlock();
+ eXosip_unlock();
- Manager::instance().error()->setError(0);
+ Manager::instance().error()->setError(0);
- return i;
+ return i;
}
int
SipVoIPLink::setUnregister (void)
{
- int i;
-// int reg_id = -1;
- osip_message_t *reg = NULL;
+ int i;
+ // int reg_id = -1;
+ osip_message_t *reg = NULL;
- eXosip_lock();
+ eXosip_lock();
- if (_reg_id > 0) {
- _debug("UNREGISTER\n");
- i = eXosip_register_build_register (_reg_id, 0, ®);
- }
+ if (_reg_id > 0) {
+ _debug("UNREGISTER\n");
+ i = eXosip_register_build_register (_reg_id, 0, ®);
+ }
- if (_reg_id < 0) {
- eXosip_unlock();
- return -1;
- }
+ if (_reg_id < 0) {
+ eXosip_unlock();
+ return -1;
+ }
- i = eXosip_register_send_register (_reg_id, reg);
- if (i == -2) {
- _debug("cannot build registration, check the setup\n");
- eXosip_unlock();
- return -1;
- }
- if (i == -1) {
- _debug("Registration Failed\n");
- eXosip_unlock();
- return -1;
- }
+ i = eXosip_register_send_register (_reg_id, reg);
+ if (i == -2) {
+ _debug("cannot build registration, check the setup\n");
+ eXosip_unlock();
+ return -1;
+ }
+ if (i == -1) {
+ _debug("Registration Failed\n");
+ eXosip_unlock();
+ return -1;
+ }
- eXosip_unlock();
+ eXosip_unlock();
- Manager::instance().error()->setError(0);
- return i;
+ Manager::instance().error()->setError(0);
+ return i;
}
int
SipVoIPLink::outgoingInvite (short id, const string& to_url)
{
- string from;
- string to;
+ string from;
+ string to;
- // Form the From header field basis on configuration panel
- from = fromHeader(get_config_fields_str(SIGNALISATION, USER_PART),
- get_config_fields_str(SIGNALISATION, HOST_PART));
+ // Form the From header field basis on configuration panel
+ from = fromHeader(get_config_fields_str(SIGNALISATION, USER_PART),
+ get_config_fields_str(SIGNALISATION, HOST_PART));
- to = toHeader(to_url);
+ to = toHeader(to_url);
- if (to.find("@") == string::npos and
- get_config_fields_int(SIGNALISATION, AUTO_REGISTER) == YES) {
- to = to + "@" + get_config_fields_str(SIGNALISATION, HOST_PART);
- }
+ if (to.find("@") == string::npos and
+ get_config_fields_int(SIGNALISATION, AUTO_REGISTER) == YES) {
+ to = to + "@" + get_config_fields_str(SIGNALISATION, HOST_PART);
+ }
- _debug("From: %s\n", from.data());
- _debug("To: %s\n", to.data());
-
- if (get_config_fields_str(SIGNALISATION, PROXY).empty()) {
- // If no SIP proxy setting for direct call with only IP address
- if (checkNetwork()) {
- if (startCall(id, from, to, "", "") <= 0) {
- _debug("Warning SipVoIPLink: call not started\n");
- return -1;
- }
- } else {
- Manager::instance().displayErrorText(id, "No network found\n");
- return -1;
- }
- return 0;
- } else {
- // If SIP proxy setting
- string route = "<sip:" +
- get_config_fields_str(SIGNALISATION, PROXY) + ";lr>";
- if (checkNetwork()) {
- if (startCall(id, from, to, "", route) <= 0) {
- _debug("Warning SipVoIPLink: call not started\n");
- return -1;
- }
- } else {
- Manager::instance().displayErrorText(id, "No network found\n");
- return -1;
- }
- return 0;
- }
+ _debug("From: %s\n", from.data());
+ _debug("To: %s\n", to.data());
+
+ if (get_config_fields_str(SIGNALISATION, PROXY).empty()) {
+ // If no SIP proxy setting for direct call with only IP address
+ if (checkNetwork()) {
+ if (startCall(id, from, to, "", "") <= 0) {
+ _debug("Warning SipVoIPLink: call not started\n");
+ return -1;
+ }
+ } else {
+ Manager::instance().displayErrorText(id, "No network found\n");
+ return -1;
+ }
+ return 0;
+ } else {
+ // If SIP proxy setting
+ string route = "<sip:" +
+ get_config_fields_str(SIGNALISATION, PROXY) + ";lr>";
+ if (checkNetwork()) {
+ if (startCall(id, from, to, "", route) <= 0) {
+ _debug("Warning SipVoIPLink: call not started\n");
+ return -1;
+ }
+ } else {
+ Manager::instance().displayErrorText(id, "No network found\n");
+ return -1;
+ }
+ return 0;
+ }
}
int
SipVoIPLink::answer (short id)
{
- int i;
- int port;
- char tmpbuf[64];
- bzero (tmpbuf, 64);
- // Get port
- snprintf (tmpbuf, 63, "%d", getSipCall(id)->getLocalAudioPort());
+ int i;
+ int port;
+ char tmpbuf[64];
+ bzero (tmpbuf, 64);
+ // Get port
+ snprintf (tmpbuf, 63, "%d", getSipCall(id)->getLocalAudioPort());
- _debug("Answer call [id = %d, cid = %d, did = %d]\n",
- id, getSipCall(id)->getCid(), getSipCall(id)->getDid());
- port = getSipCall(id)->getLocalAudioPort();
- _debug("Local audio port: %d\n", port);
+ _debug("Answer call [id = %d, cid = %d, did = %d]\n",
+ id, getSipCall(id)->getCid(), getSipCall(id)->getDid());
+ 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_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.
- if (_audiortp->createNewSession (getSipCall(id)) < 0) {
- _debug("FATAL: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
- exit(1);
- }
- return i;
+ 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_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.
+ if (_audiortp->createNewSession (getSipCall(id)) < 0) {
+ _debug("FATAL: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
+ exit(1);
+ }
+ return i;
}
int
SipVoIPLink::hangup (short id)
{
- int i = 0;
- if (!Manager::instance().getbCongestion()) {
- _debug("Hang up call [id = %d, cid = %d, did = %d]\n",
- id, getSipCall(id)->getCid(), getSipCall(id)->getDid());
- // Release SIP stack.
- eXosip_lock();
- i = eXosip_call_terminate (getSipCall(id)->getCid(),
- getSipCall(id)->getDid());
- eXosip_unlock();
-
- // Release RTP channels
- _audiortp->closeRtpSession(getSipCall(id));
- }
+ int i = 0;
+ if (!Manager::instance().getbCongestion()) {
+ _debug("Hang up call [id = %d, cid = %d, did = %d]\n",
+ id, getSipCall(id)->getCid(), getSipCall(id)->getDid());
+ // Release SIP stack.
+ eXosip_lock();
+ i = eXosip_call_terminate (getSipCall(id)->getCid(),
+ getSipCall(id)->getDid());
+ eXosip_unlock();
+
+ // Release RTP channels
+ _audiortp->closeRtpSession(getSipCall(id));
+ }
- deleteSipCall(id);
- return i;
+ deleteSipCall(id);
+ return i;
}
int
SipVoIPLink::cancel (short id)
{
- int i = 0;
- if (!Manager::instance().getbCongestion()) {
- _debug("Cancel call [id = %d, cid = %d]\n", id, getCid());
- // Release SIP stack.
- eXosip_lock();
- i = eXosip_call_terminate (getCid(), -1);
- eXosip_unlock();
- }
- deleteSipCall(id);
- return i;
+ int i = 0;
+ if (!Manager::instance().getbCongestion()) {
+ _debug("Cancel call [id = %d, cid = %d]\n", id, getCid());
+ // Release SIP stack.
+ eXosip_lock();
+ i = eXosip_call_terminate (getCid(), -1);
+ eXosip_unlock();
+ }
+ deleteSipCall(id);
+ return i;
}
int
SipVoIPLink::onhold (short id)
{
- osip_message_t *invite;
- int i;
- int did;
+ osip_message_t *invite;
+ int i;
+ int did;
- sdp_message_t *local_sdp = NULL;
+ sdp_message_t *local_sdp = NULL;
- did = getSipCall(id)->getDid();
+ did = getSipCall(id)->getDid();
- eXosip_lock ();
- local_sdp = eXosip_get_local_sdp (did);
- eXosip_unlock ();
+ eXosip_lock ();
+ local_sdp = eXosip_get_local_sdp (did);
+ eXosip_unlock ();
- if (local_sdp == NULL) {
- return -1;
- }
+ if (local_sdp == NULL) {
+ return -1;
+ }
- eXosip_lock ();
- // Build INVITE_METHOD for put call on-hold
- i = eXosip_call_build_request (did, INVITE_METHOD, &invite);
- eXosip_unlock ();
+ 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;
- }
+ if (i != 0) {
+ sdp_message_free(local_sdp);
+ return -1;
+ }
- /* add sdp body */
+ /* 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;
+ 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_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 ();
+ eXosip_lock ();
+ // Send request
+ i = eXosip_call_send_request (did, invite);
+ eXosip_unlock ();
- // Disable audio
- _audiortp->closeRtpSession(getSipCall(id));
- return i;
+ // Disable audio
+ _audiortp->closeRtpSession(getSipCall(id));
+ return i;
}
int
SipVoIPLink::offhold (short id)
{
- 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;
- }
+ 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;
+ }
- eXosip_lock ();
- // Build INVITE_METHOD for put call off-hold
- i = eXosip_call_build_request (did, INVITE_METHOD, &invite);
- eXosip_unlock ();
+ 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;
- }
+ if (i != 0) {
+ sdp_message_free(local_sdp);
+ return -1;
+ }
/* add sdp body */
{
@@ -475,458 +473,462 @@ SipVoIPLink::offhold (short id)
i = sdp_off_hold_call (local_sdp);
if (i != 0) {
- sdp_message_free (local_sdp);
- osip_message_free (invite);
- return -1;
+ 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_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 ();
+ 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__);
- exit(1);
- }
- return i;
+ // Enable audio
+ if (_audiortp->createNewSession (getSipCall(id)) < 0) {
+ _debug("FATAL: Unable to start sound (%s:%d)\n", __FILE__, __LINE__);
+ exit(1);
+ }
+ return i;
}
int
SipVoIPLink::transfer (short id, const string& to)
{
- osip_message_t *refer;
- int i;
- string tmp_to;
- tmp_to = toHeader(to);
- if (tmp_to.find("@") == string::npos) {
- tmp_to = tmp_to + "@" + get_config_fields_str(SIGNALISATION, HOST_PART);
- }
+ osip_message_t *refer;
+ int i;
+ string tmp_to;
+ tmp_to = toHeader(to);
+ if (tmp_to.find("@") == string::npos) {
+ tmp_to = tmp_to + "@" + get_config_fields_str(SIGNALISATION, HOST_PART);
+ }
- eXosip_lock();
- // 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;
+ eXosip_lock();
+ // 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;
}
int
SipVoIPLink::refuse (short id)
{
- int i;
- char tmpbuf[64];
- bzero (tmpbuf, 64);
- // Get local port
- snprintf (tmpbuf, 63, "%d", getSipCall(id)->getLocalAudioPort());
+ int i;
+ char tmpbuf[64];
+ bzero (tmpbuf, 64);
+ // Get local port
+ snprintf (tmpbuf, 63, "%d", getSipCall(id)->getLocalAudioPort());
- osip_message_t *answer = NULL;
- eXosip_lock();
- 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;
+ osip_message_t *answer = NULL;
+ eXosip_lock();
+ 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;
}
int
SipVoIPLink::getEvent (void)
{
- eXosip_event_t *event;
- short id;
- char *name;
-
- 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_INVITE: //
- // Set local random port for incoming call
- if (!Manager::instance().useStun()) {
- setLocalPort(RANDOM_LOCAL_PORT);
- } else {
- // If there is a firewall
- if (behindNat() != 0) {
- setLocalPort(Manager::instance().getFirewallPort());
- } else {
- return -1;
- }
- }
+ eXosip_event_t *event;
+ short id;
+ char *name;
+
+ 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_INVITE: //
+ // Set local random port for incoming call
+ if (!Manager::instance().useStun()) {
+ setLocalPort(RANDOM_LOCAL_PORT);
+ } else {
+ // If there is a firewall
+ if (behindNat() != 0) {
+ setLocalPort(Manager::instance().getFirewallPort());
+ } else {
+ return -1;
+ }
+ }
- // Generate id
- id = Manager::instance().generateNewCallId();
- Manager::instance().pushBackNewCall(id, Incoming);
- _debug("Incoming Call with identifiant %d [cid = %d, did = %d]\n",
- id, event->cid, event->did);
- _debug("Local audio port: %d\n", _localPort);
-
- // Display the callerId-name
- osip_from_t *from;
- osip_from_init(&from);
-
- 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) {
- Manager::instance().getCall(id)->setCallerIdName(name);
- } else {
- return -1;
- }
- _debug("From: %s\n", name);
- osip_from_free(from);
+ // Generate id
+ id = Manager::instance().generateNewCallId();
+ Manager::instance().pushBackNewCall(id, Incoming);
+ _debug("Incoming Call with identifiant %d [cid = %d, did = %d]\n",
+ id, event->cid, event->did);
+ _debug("Local audio port: %d\n", _localPort);
+
+ // Display the callerId-name
+ osip_from_t *from;
+ osip_from_init(&from);
+
+ 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) {
+ Manager::instance().getCall(id)->setCallerIdName(name);
+ } else {
+ return -1;
+ }
+ _debug("From: %s\n", name);
+ osip_from_free(from);
- // Associate an audio port with a call
- getSipCall(id)->setLocalAudioPort(_localPort);
+ // Associate an audio port with a call
+ getSipCall(id)->setLocalAudioPort(_localPort);
- getSipCall(id)->newIncomingCall(event);
- if (Manager::instance().incomingCall(id) < 0) {
- Manager::instance().displayErrorText(id, "Incoming call failed");
- return -1;
- }
+ getSipCall(id)->newIncomingCall(event);
+ if (Manager::instance().incomingCall(id) < 0) {
+ Manager::instance().displayErrorText(id, "Incoming call failed");
+ return -1;
+ }
- break;
-
- // The peer-user answers
- case EXOSIP_CALL_ANSWERED:
- id = findCallId(event);
- if (id == 0) {
- id = findCallIdWhenRinging();
- }
- _debug("Call is answered [id = %d, cid = %d, did = %d], localport=%d\n",
- id, event->cid, event->did,getSipCall(id)->getLocalAudioPort());
+ break;
+
+ case EXOSIP_CALL_REINVITE:
+ eXosip_call_send_answer(event->tid, 403, NULL);
+ break;
+
+ // The peer-user answers
+ case EXOSIP_CALL_ANSWERED:
+ id = findCallId(event);
+ if (id == 0) {
+ id = findCallIdWhenRinging();
+ }
+ _debug("Call is answered [id = %d, cid = %d, did = %d], localport=%d\n",
+ id, event->cid, event->did,getSipCall(id)->getLocalAudioPort());
- // Answer
- if (id > 0 and !Manager::instance().getCall(id)->isOnHold()
- 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.
- if (_audiortp->createNewSession (getSipCall(id)) < 0) {
- _debug("FATAL: Unable to start sound (%s:%d)\n",
- __FILE__, __LINE__);
- exit(1);
- }
- }
- } else {
- // Answer to on/off hold to send ACK
- if (id > 0) {
- getSipCall(id)->answeredCall(event);
- _debug("-----------------------\n");
- }
- }
- break;
+ // Answer
+ if (id > 0 and !Manager::instance().getCall(id)->isOnHold()
+ 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.
+ if (_audiortp->createNewSession (getSipCall(id)) < 0) {
+ _debug("FATAL: Unable to start sound (%s:%d)\n",
+ __FILE__, __LINE__);
+ 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();
+ case EXOSIP_CALL_RINGING: //peer call is ringing
+ id = findCallIdWhenRinging();
- _debug("Call is ringing [id = %d, cid = %d, did = %d]\n",
- id, event->cid, event->did);
+ _debug("Call is ringing [id = %d, cid = %d, did = %d]\n",
+ id, event->cid, event->did);
- if (id > 0) {
- getSipCall(id)->ringingCall(event);
- Manager::instance().peerRingingCall(id);
- } else {
- return -1;
- }
- break;
-
- 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;
+ if (id > 0) {
+ getSipCall(id)->ringingCall(event);
+ Manager::instance().peerRingingCall(id);
+ } else {
+ return -1;
+ }
+ break;
+
+ 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);
- _debug("Call is closed [id = %d, cid = %d, did = %d]\n",
- id, event->cid, event->did);
+ // The peer-user closed the phone call(we received BYE).
+ case EXOSIP_CALL_CLOSED:
+ id = findCallId(event);
+ _debug("Call is closed [id = %d, cid = %d, did = %d]\n",
+ id, event->cid, event->did);
- if (id > 0) {
- if (!Manager::instance().getCall(id)->isProgressing()) {
- _audiortp->closeRtpSession(getSipCall(id));
- }
- Manager::instance().peerHungupCall(id);
- deleteSipCall(id);
- } else {
- return -1;
- }
- break;
-
- case EXOSIP_CALL_REQUESTFAILURE:
- id = findCallId(event);
-
- // Handle 4XX errors
- switch (event->response->status_code) {
- case AUTH_REQUIRED:
- eXosip_lock();
- eXosip_automatic_action();
- eXosip_unlock();
- break;
- case UNAUTHORIZED:
- setAuthentication();
- break;
-
- case BAD_REQ:
- case FORBIDDEN:
- case NOT_FOUND:
- case NOT_ALLOWED:
- case NOT_ACCEPTABLE:
- case REQ_TIMEOUT:
- case TEMP_UNAVAILABLE:
- case ADDR_INCOMPLETE:
- case BUSY_HERE:
- // Display error on the screen phone
- Manager::instance().displayError(event->response->reason_phrase);
- Manager::instance().congestion(true);
- break;
- case REQ_TERMINATED:
- break;
- default:
- break;
- }
- break;
-
- case EXOSIP_CALL_SERVERFAILURE:
- // Handle 5XX errors
- switch (event->response->status_code) {
- case SERVICE_UNAVAILABLE:
- Manager::instance().ringback(false);
- Manager::instance().congestion(true);
- break;
- default:
- break;
- }
- break;
-
- case EXOSIP_CALL_GLOBALFAILURE:
- // Handle 6XX errors
- switch (event->response->status_code) {
- case BUSY_EVERYWHERE:
- case DECLINE:
- Manager::instance().ringback(false);
- Manager::instance().congestion(true);
- break;
- default:
- break;
- }
- break;
-
- case EXOSIP_REGISTRATION_SUCCESS:
- Manager::instance().displayStatus(LOGGED_IN_STATUS);
- break;
-
- case EXOSIP_REGISTRATION_FAILURE:
- break;
-
- case EXOSIP_MESSAGE_NEW:
- unsigned int k;
+ if (id > 0) {
+ if (!Manager::instance().getCall(id)->isProgressing()) {
+ _audiortp->closeRtpSession(getSipCall(id));
+ }
+ Manager::instance().peerHungupCall(id);
+ deleteSipCall(id);
+ } else {
+ return -1;
+ }
+ break;
+
+ case EXOSIP_CALL_REQUESTFAILURE:
+ id = findCallId(event);
+
+ // Handle 4XX errors
+ switch (event->response->status_code) {
+ case AUTH_REQUIRED:
+ eXosip_lock();
+ eXosip_automatic_action();
+ eXosip_unlock();
+ break;
+ case UNAUTHORIZED:
+ setAuthentication();
+ break;
+
+ case BAD_REQ:
+ case FORBIDDEN:
+ case NOT_FOUND:
+ case NOT_ALLOWED:
+ case NOT_ACCEPTABLE:
+ case REQ_TIMEOUT:
+ case TEMP_UNAVAILABLE:
+ case ADDR_INCOMPLETE:
+ case BUSY_HERE:
+ // Display error on the screen phone
+ Manager::instance().displayError(event->response->reason_phrase);
+ Manager::instance().congestion(true);
+ break;
+ case REQ_TERMINATED:
+ break;
+ default:
+ break;
+ }
+ break;
+
+ case EXOSIP_CALL_SERVERFAILURE:
+ // Handle 5XX errors
+ switch (event->response->status_code) {
+ case SERVICE_UNAVAILABLE:
+ Manager::instance().ringback(false);
+ Manager::instance().congestion(true);
+ break;
+ default:
+ break;
+ }
+ break;
+
+ case EXOSIP_CALL_GLOBALFAILURE:
+ // Handle 6XX errors
+ switch (event->response->status_code) {
+ case BUSY_EVERYWHERE:
+ case DECLINE:
+ Manager::instance().ringback(false);
+ Manager::instance().congestion(true);
+ break;
+ default:
+ break;
+ }
+ break;
+
+ case EXOSIP_REGISTRATION_SUCCESS:
+ Manager::instance().displayStatus(LOGGED_IN_STATUS);
+ break;
+
+ case EXOSIP_REGISTRATION_FAILURE:
+ break;
+
+ case EXOSIP_MESSAGE_NEW:
+ unsigned int k;
- if (event->request != NULL && MSG_IS_OPTIONS(event->request)) {
- 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 (k == _sipcallVector->size()) {
- /* answer 200 ok */
- eXosip_options_send_answer (event->tid, OK, NULL);
- } else if (_sipcallVector->at(k)->getCid() == event->cid) {
- /* already answered! */
- } else {
- /* answer 486 ok */
- eXosip_options_send_answer (event->tid, BUSY_HERE, NULL);
- }
- eXosip_unlock();
- }
+ // TODO: Que faire si rien trouve??
+ eXosip_lock();
+ if (k == _sipcallVector->size()) {
+ /* answer 200 ok */
+ eXosip_options_send_answer (event->tid, OK, NULL);
+ } else if (_sipcallVector->at(k)->getCid() == event->cid) {
+ /* already answered! */
+ } else {
+ /* answer 486 ok */
+ eXosip_options_send_answer (event->tid, BUSY_HERE, NULL);
+ }
+ eXosip_unlock();
+ }
- // Voice message
- else if (event->request != NULL && MSG_IS_NOTIFY(event->request)){
- int ii;
- unsigned int pos;
- unsigned int pos_slash;
- string *str;
- string nb_msg;
- osip_body_t *body;
-
- // Get the message body
- ii = osip_message_get_body(event->request, 0, &body);
- if (ii != 0) {
- _debug("Cannot get body\n");
- return -1;
- }
+ // Voice message
+ else if (event->request != NULL && MSG_IS_NOTIFY(event->request)){
+ int ii;
+ unsigned int pos;
+ unsigned int pos_slash;
+ string *str;
+ string nb_msg;
+ osip_body_t *body;
+
+ // Get the message body
+ ii = osip_message_get_body(event->request, 0, &body);
+ if (ii != 0) {
+ _debug("Cannot get body\n");
+ return -1;
+ }
- // Analyse message body
- str = new string(body->body);
- pos = str->find (VOICE_MSG);
+ // Analyse message body
+ str = new string(body->body);
+ pos = str->find (VOICE_MSG);
- if (pos == string::npos) {
- // If the string is not found
- return -1;
- }
+ if (pos == string::npos) {
+ // If the string is not found
+ return -1;
+ }
- pos_slash = str->find ("/");
- nb_msg = str->substr(pos + LENGTH_VOICE_MSG,
- pos_slash - (pos + LENGTH_VOICE_MSG));
-
- // Set the number of voice-message
- setMsgVoicemail(atoi(nb_msg.data()));
-
- if (getMsgVoicemail() != 0) {
- // If there is at least one voice-message, start notification
- Manager::instance().startVoiceMessageNotification();
- } else {
- // Stop notification when there is 0 voice message
- Manager::instance().stopVoiceMessageNotification();
- }
- delete str;
- }
- break;
-
- default:
- return -1;
- break;
- }
- eXosip_event_free(event);
+ pos_slash = str->find ("/");
+ nb_msg = str->substr(pos + LENGTH_VOICE_MSG,
+ pos_slash - (pos + LENGTH_VOICE_MSG));
+
+ // Set the number of voice-message
+ setMsgVoicemail(atoi(nb_msg.data()));
+
+ if (getMsgVoicemail() != 0) {
+ // If there is at least one voice-message, start notification
+ Manager::instance().startVoiceMessageNotification();
+ } else {
+ // Stop notification when there is 0 voice message
+ Manager::instance().stopVoiceMessageNotification();
+ }
+ delete str;
+ }
+ break;
+
+ default:
+ return -1;
+ break;
+ }
+ eXosip_event_free(event);
- return 0;
+ return 0;
}
int
SipVoIPLink::getLocalPort (void)
{
- return _localPort;
+ return _localPort;
}
void
SipVoIPLink::setLocalPort (int port)
{
- _localPort = port;
+ _localPort = port;
}
void
SipVoIPLink::carryingDTMFdigits (short id, char code) {
- int duration = get_config_fields_int(SIGNALISATION, PULSE_LENGTH);
- osip_message_t *info;
- const int body_len = 1000;
- int i;
-
- char *dtmf_body = new char[body_len];
-
- eXosip_lock();
- // 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();
+ int duration = get_config_fields_int(SIGNALISATION, PULSE_LENGTH);
+ osip_message_t *info;
+ const int body_len = 1000;
+ int i;
+
+ char *dtmf_body = new char[body_len];
+
+ eXosip_lock();
+ // 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;
+ delete[] dtmf_body;
}
void
SipVoIPLink::newOutgoingCall (short callid)
{
- _sipcallVector->push_back(new SipCall(callid,
- Manager::instance().getCodecDescVector()));
- if (getSipCall(callid) != NULL) {
- getSipCall(callid)->setStandBy(true);
- }
+ _sipcallVector->push_back(new SipCall(callid,
+ Manager::instance().getCodecDescVector()));
+ if (getSipCall(callid) != NULL) {
+ getSipCall(callid)->setStandBy(true);
+ }
}
void
SipVoIPLink::newIncomingCall (short callid)
{
- SipCall* sipcall = new SipCall(callid, Manager::instance().getCodecDescVector());
- _sipcallVector->push_back(sipcall);
+ SipCall* sipcall = new SipCall(callid, Manager::instance().getCodecDescVector());
+ _sipcallVector->push_back(sipcall);
}
void
SipVoIPLink::deleteSipCall (short callid)
{
- unsigned int i = 0;
- while (i < _sipcallVector->size()) {
- if (_sipcallVector->at(i)->getId() == callid) {
- _sipcallVector->erase(_sipcallVector->begin()+i);
- return;
- } else {
- i++;
- }
- }
+ unsigned int i = 0;
+ while (i < _sipcallVector->size()) {
+ if (_sipcallVector->at(i)->getId() == callid) {
+ _sipcallVector->erase(_sipcallVector->begin()+i);
+ return;
+ } else {
+ i++;
+ }
+ }
}
SipCall*
SipVoIPLink::getSipCall (short callid)
{
- for (unsigned int i = 0; i < _sipcallVector->size(); i++) {
- if (_sipcallVector->at(i)->getId() == callid) {
- return _sipcallVector->at(i);
- }
- }
- return NULL;
+ for (unsigned int i = 0; i < _sipcallVector->size(); i++) {
+ if (_sipcallVector->at(i)->getId() == callid) {
+ return _sipcallVector->at(i);
+ }
+ }
+ return NULL;
}
AudioCodec*
SipVoIPLink::getAudioCodec (short callid)
{
- if (getSipCall(callid)) {
- return getSipCall(callid)->getAudioCodec();
- } else {
- return NULL;
- }
+ if (getSipCall(callid)) {
+ return getSipCall(callid)->getAudioCodec();
+ } else {
+ return NULL;
+ }
}
///////////////////////////////////////////////////////////////////////////////
// Private functions
@@ -934,348 +936,348 @@ SipVoIPLink::getAudioCodec (short callid)
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);
+ 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++;
+ 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;
+ 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);
+ 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++;
+ 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;
+ 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? */
- }
+ 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++;
+ 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, " ");
}
-
- osip_message_set_body (answer, buf, strlen (buf));
- osip_message_set_content_type (answer, "application/sdp");
+ 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 0;
+ 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)
{
- StunAddress4 stunSvrAddr;
- stunSvrAddr.addr = 0;
+ StunAddress4 stunSvrAddr;
+ stunSvrAddr.addr = 0;
- // Stun server
- string svr = get_config_fields_str(SIGNALISATION, STUN_SERVER);
+ // Stun server
+ string svr = get_config_fields_str(SIGNALISATION, STUN_SERVER);
- // Convert char* to StunAddress4 structure
- bool ret = stunParseServerName ((char*)svr.data(), stunSvrAddr);
- if (!ret) {
- _debug("SIP: Stun server address not valid\n");
- return 0;
- }
+ // Convert char* to StunAddress4 structure
+ bool ret = stunParseServerName ((char*)svr.data(), stunSvrAddr);
+ if (!ret) {
+ _debug("SIP: Stun server address not valid\n");
+ return 0;
+ }
- // Firewall address
- _debug("STUN server: %s\n", svr.data());
- Manager::instance().getStunInfo(stunSvrAddr);
+ // Firewall address
+ _debug("STUN server: %s\n", svr.data());
+ Manager::instance().getStunInfo(stunSvrAddr);
- return 1;
+ return 1;
}
int
SipVoIPLink::getLocalIp (void)
{
- int ret = 0;
- char* myIPAddress = new char[65];
- ret = eXosip_guess_localip (2, myIPAddress, 64);
- setLocalIpAddress(string(myIPAddress));
+ int ret = 0;
+ char* myIPAddress = new char[65];
+ ret = eXosip_guess_localip (2, myIPAddress, 64);
+ setLocalIpAddress(string(myIPAddress));
- return ret;
+ return ret;
}
int
SipVoIPLink::checkUrl (const string& url)
{
- int i;
+ int i;
- osip_from_t *to;
- i = osip_from_init(&to);
- if (i != 0) {
- _debug("Warning: Cannot initialize\n");
- return -1;
- }
- i = osip_from_parse(to, url.data());
- if (i != 0) {
- _debug("Warning: Cannot parse url\n");
- return -1;
- }
+ osip_from_t *to;
+ i = osip_from_init(&to);
+ if (i != 0) {
+ _debug("Warning: Cannot initialize\n");
+ return -1;
+ }
+ i = osip_from_parse(to, url.data());
+ if (i != 0) {
+ _debug("Warning: Cannot parse url\n");
+ return -1;
+ }
- // Free memory
- osip_from_free (to);
- return 0;
+ // Free memory
+ osip_from_free (to);
+ return 0;
}
int
SipVoIPLink::setAuthentication (void)
{
- string login, pass, realm;
- login = get_config_fields_str(SIGNALISATION, AUTH_USER_NAME);
- if (login.empty()) {
- login = get_config_fields_str(SIGNALISATION, USER_PART);
- }
- pass = get_config_fields_str(SIGNALISATION, PASSWORD);
- if (pass.empty()) {
- Manager::instance().error()->errorName(PASSWD_FIELD_EMPTY);
- return -1;
- }
+ string login, pass, realm;
+ login = get_config_fields_str(SIGNALISATION, AUTH_USER_NAME);
+ if (login.empty()) {
+ login = get_config_fields_str(SIGNALISATION, USER_PART);
+ }
+ pass = get_config_fields_str(SIGNALISATION, PASSWORD);
+ if (pass.empty()) {
+ Manager::instance().error()->errorName(PASSWD_FIELD_EMPTY);
+ return -1;
+ }
- if (eXosip_add_authentication_info(login.data(), login.data(),
- pass.data(), NULL, NULL) != 0) {
- return -1;
- }
- return 0;
+ if (eXosip_add_authentication_info(login.data(), login.data(),
+ pass.data(), NULL, NULL) != 0) {
+ return -1;
+ }
+ return 0;
}
string
SipVoIPLink::fromHeader (const string& user, const string& host)
{
- string displayname = get_config_fields_str(SIGNALISATION, FULL_NAME);
- return ("\"" + displayname + "\"" + " <sip:" + user + "@" + host + ">");
+ string displayname = get_config_fields_str(SIGNALISATION, FULL_NAME);
+ return ("\"" + displayname + "\"" + " <sip:" + user + "@" + host + ">");
}
string
SipVoIPLink::toHeader(const string& to)
{
- if (to.find("sip:") == string::npos) {
- return ("sip:" + to );
- } else {
- return to;
- }
+ if (to.find("sip:") == string::npos) {
+ return ("sip:" + to );
+ } else {
+ return to;
+ }
}
int
SipVoIPLink::startCall (short id, const string& from, const string& to,
- const string& subject, const string& route)
+ const string& subject, const string& route)
{
- osip_message_t *invite;
- int i;
-
- if (checkUrl(from) != 0) {
- Manager::instance().error()->errorName(FROM_ERROR);
- return -1;
- }
- if (checkUrl(to) != 0) {
- Manager::instance().error()->errorName(TO_ERROR);
- return -1;
- }
+ osip_message_t *invite;
+ int i;
+
+ if (checkUrl(from) != 0) {
+ Manager::instance().error()->errorName(FROM_ERROR);
+ return -1;
+ }
+ if (checkUrl(to) != 0) {
+ Manager::instance().error()->errorName(TO_ERROR);
+ return -1;
+ }
- char port[64];
- if (!Manager::instance().useStun()) {
- // Set random port for outgoing call if no firewall
- setLocalPort(RANDOM_LOCAL_PORT);
- _debug("Local audio port: %d\n",_localPort);
- } else {
- // If use Stun server
- if (behindNat() != 0) {
- _debug("sip invite: firewall port = %d\n",Manager::instance().getFirewallPort());
- setLocalPort(Manager::instance().getFirewallPort());
- } else {
- return -1;
- }
- }
+ char port[64];
+ if (!Manager::instance().useStun()) {
+ // Set random port for outgoing call if no firewall
+ setLocalPort(RANDOM_LOCAL_PORT);
+ _debug("Local audio port: %d\n",_localPort);
+ } else {
+ // If use Stun server
+ if (behindNat() != 0) {
+ _debug("sip invite: firewall port = %d\n",Manager::instance().getFirewallPort());
+ setLocalPort(Manager::instance().getFirewallPort());
+ } else {
+ return -1;
+ }
+ }
- // Set local audio port for sipcall(id)
- if (getSipCall(id) != NULL) {
- getSipCall(id)->setLocalAudioPort(_localPort);
- } else {
- return -1;
- }
+ // Set local audio port for sipcall(id)
+ if (getSipCall(id) != NULL) {
+ getSipCall(id)->setLocalAudioPort(_localPort);
+ } else {
+ return -1;
+ }
- bzero (port, 64);
- snprintf (port, 63, "%d", getLocalPort());
+ bzero (port, 64);
+ snprintf (port, 63, "%d", getLocalPort());
- i = eXosip_call_build_initial_invite (&invite, (char*)to.data(),
+ i = eXosip_call_build_initial_invite (&invite, (char*)to.data(),
(char*)from.data(),
(char*)route.data(),
(char*)subject.data());
- if (i != 0) {
- return -1;
- }
+ 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);
+ 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);
+ // 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);
- }
- }
+ 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 */
{
@@ -1290,60 +1292,60 @@ SipVoIPLink::startCall (short id, const string& from, const string& to,
"c=IN IP4 %s\r\n"
"t=0 0\r\n"
"m=audio %s RTP/AVP %s\r\n"
- "%s",
+ "%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();
+ eXosip_lock();
- i = eXosip_call_send_initial_invite (invite);
+ i = eXosip_call_send_initial_invite (invite);
- if (i <= 0) {
- eXosip_unlock();
- return -1;
- } else {
- eXosip_call_set_reference (i, NULL);
- }
+ if (i <= 0) {
+ eXosip_unlock();
+ return -1;
+ } else {
+ eXosip_call_set_reference (i, NULL);
+ }
- eXosip_unlock();
+ eXosip_unlock();
- // Keep the cid in case of cancelling
- setCid(i);
+ // Keep the cid in case of cancelling
+ setCid(i);
- return i;
+ return i;
}
short
SipVoIPLink::findCallId (eXosip_event_t *e)
{
- unsigned int k;
+ unsigned int k;
- for (k = 0; k < _sipcallVector->size(); k++) {
- if (_sipcallVector->at(k)->getCid() == e->cid and
- _sipcallVector->at(k)->getDid() == e->did) {
- return _sipcallVector->at(k)->getId();
- }
+ for (k = 0; k < _sipcallVector->size(); k++) {
+ if (_sipcallVector->at(k)->getCid() == e->cid and
+ _sipcallVector->at(k)->getDid() == e->did) {
+ return _sipcallVector->at(k)->getId();
}
- return 0;
+ }
+ return 0;
}
short
SipVoIPLink::findCallIdWhenRinging (void)
{
- unsigned int k;
- int i = Manager::instance().selectedCall();
+ unsigned int k;
+ int i = Manager::instance().selectedCall();
- if (i != -1) {
- return i;
- } else {
- for (k = 0; k < _sipcallVector->size(); k++) {
- if (_sipcallVector->at(k)->getStandBy()) {
- return _sipcallVector->at(k)->getId();
- }
- }
- }
- return 0;
+ if (i != -1) {
+ return i;
+ } else {
+ for (k = 0; k < _sipcallVector->size(); k++) {
+ if (_sipcallVector->at(k)->getStandBy()) {
+ return _sipcallVector->at(k)->getId();
+ }
+ }
+ }
+ return 0;
}