/**
* Copyright (C) 2004-2005 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author : Laurielle Lea <laurielle.lea@savoirfairelinux.com>
*
* Portions Copyright (C) 2002,2003 Aymeric Moizard <jack@atosc.org>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include <eXosip2/eXosip.h>
#include <osip2/osip.h>
#include "sipvoiplink.h"
#include "global.h"
#include "audio/codecDescriptor.h"
#include "manager.h"
#include "sipcall.h"
#include "user_cfg.h"
#include "eventthread.h"
#define DEFAULT_SIP_PORT 5060
#define RANDOM_SIP_PORT rand() % 64000 + 1024
#define DEFAULT_LOCAL_PORT 10500
#define RANDOM_LOCAL_PORT ((rand() % 27250) + 5250)*2
#define VOICE_MSG "Voice-Message"
#define LENGTH_VOICE_MSG 15
SipVoIPLink::SipVoIPLink() : VoIPLink()
{
// default _audioRTP object initialization
_evThread = new EventThread(this);
_localPort = 0;
_nMsgVoicemail = 0;
_reg_id = -1;
// defautlt _sipcallVector object initialization
_registrationSend = false;
_started = false;
}
SipVoIPLink::~SipVoIPLink(void) {
endSipCalls();
delete _evThread; _evThread = NULL;
if (_started) {
eXosip_quit();
}
}
// for voIPLink interface
void
SipVoIPLink::terminate(void)
{
}
bool
SipVoIPLink::checkNetwork (void)
{
// Set IP address
return getSipLocalIp();
}
int
SipVoIPLink::init (void)
{
std::string tmp;
tmp = std::string(PROGNAME) + "/" + std::string(SFLPHONED_VERSION);
if (0 != eXosip_init()) {
_debug("Could not initialize eXosip\n");
exit (0);
}
_started = true;
srand (time(NULL));
//should be NULL or INADDR_ANY for the second parameter?
int i;
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;
}
// This method is used to replace contact address with the public address of your NAT.
eXosip_masquerade_contact((Manager::instance().getFirewallAddress()).data(), Manager::instance().getFirewallPort());
} else {
// Set user agent
eXosip_set_user_agent(tmp.data());
}
_debug("SIP VoIP Link: listen to SIP Events\n");
_evThread->start();
return 1;
}
/**
* Subscibe to message-summary notify
* It allows eXosip to not send ' 481 Subcription Does Not Exist ' response
*/
void
SipVoIPLink::subscribeMessageSummary()
{
osip_message_t *subscribe;
const char *route= NULL;
// from/to
ManagerImpl& manager = Manager::instance();
std::string from = fromHeader(manager.getConfigString(SIGNALISATION, USER_PART), manager.getConfigString(SIGNALISATION, HOST_PART));
// to
std::string to;
to = manager.getConfigString(SIGNALISATION, PROXY);
if (!to.empty()) {
to = toHeader(manager.getConfigString(SIGNALISATION, USER_PART)) + "@" + to;
} else {
to = from;
}
// like in http://www.faqs.org/rfcs/rfc3842.html
const char *event="message-summary";
int expires = 86400;
// return 0 if no error
// the first from is the to... but we send the same
int error = eXosip_subscribe_build_initial_request(&subscribe, to.c_str(), from.c_str(), route, event, expires);
if (error == 0) {
// Accept: application/simple-message-summary
osip_message_set_header (subscribe, "Accept", "application/simple-message-summary");
_debug("Sending Message-summary subscription");
eXosip_lock();
// return 0 if ok
error = eXosip_subscribe_send_initial_request (subscribe);
_debug(" and return %d\n", error);
eXosip_unlock();
}
}
bool
SipVoIPLink::isInRtpmap (int index, int payload, CodecDescriptorVector* cdv) {
for (int i = 0; i < index; i++) {
if (cdv->at(i)->getPayload() == payload) {
return true;
}
}
return false;
}
int
SipVoIPLink::setRegister (void)
{
ManagerImpl& manager = Manager::instance();
// all this will be inside the profil associate with the voip link
std::string proxy = "sip:" + manager.getConfigString(SIGNALISATION, PROXY);
std::string hostname = "sip:" + manager.getConfigString(SIGNALISATION, HOST_PART);
std::string from = fromHeader(manager.getConfigString(SIGNALISATION, USER_PART), manager.getConfigString(SIGNALISATION, HOST_PART));
if (manager.getConfigString(SIGNALISATION, HOST_PART).empty()) {
manager.displayConfigError("Fill host part field");
return -1;
}
if (manager.getConfigString(SIGNALISATION, USER_PART).empty()) {
manager.displayConfigError("Fill user part field");
return -1;
}
if (setAuthentication() == -1) {
_debug("No authentication\n");
return -1;
}
_debug("REGISTER From: %s\n", from.data());
osip_message_t *reg = NULL;
eXosip_lock();
if (!manager.getConfigString(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;
}
osip_message_set_header (reg, "Event", "Registration");
osip_message_set_header (reg, "Allow-Events", "presence");
int 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();
// subscribe to message one time?
// subscribeMessageSummary();
_registrationSend = true;
return i;
}
/**
* setUnregister
* unregister if we already send the first registration
* @return -1 if there is an error
*/
int
SipVoIPLink::setUnregister (void)
{
if ( _registrationSend ) {
int i = 0;
osip_message_t *reg = NULL;
eXosip_lock();
if (_reg_id > 0) {
_debug("UNREGISTER\n");
i = eXosip_register_build_register (_reg_id, 0, ®);
}
if (_reg_id < 0) {
eXosip_unlock();
return -1;
}
i = eXosip_register_send_register (_reg_id, reg);
if (i == -2) {
_debug("(unregister) Cannot build registration, check the setup\n");
eXosip_unlock();
return -1;
}
if (i == -1) {
_debug("(unregister) Registration Failed\n");
eXosip_unlock();
return -1;
}
eXosip_unlock();
return i;
} else {
// no registration send before
return -1;
}
}
int
SipVoIPLink::outgoingInvite (CALLID id, const std::string& to_url)
{
std::string from;
std::string to;
// TODO: should be inside account settings
ManagerImpl& manager = Manager::instance();
// Form the From header field basis on configuration panel
std::string host = manager.getConfigString(SIGNALISATION, HOST_PART);
if ( host.empty() ) {
host = getLocalIpAddress();
}
from = fromHeader(manager.getConfigString(SIGNALISATION, USER_PART), host);
to = toHeader(to_url);
if (to.find("@") == std::string::npos and
manager.getConfigInt(SIGNALISATION, AUTO_REGISTER)) {
to = to + "@" + manager.getConfigString(SIGNALISATION, HOST_PART);
}
_debug("From: %s\n", from.data());
_debug("To: %s\n", to.data());
if (manager.getConfigString(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.displayErrorText(id, "No network found\n");
return -1;
}
return 0;
} else {
// If SIP proxy setting
std::string route = "<sip:" +
manager.getConfigString(SIGNALISATION, PROXY) + ";lr>";
if (checkNetwork()) {
if (startCall(id, from, to, "", route) <= 0) {
_debug("Warning SipVoIPLink: call not started\n");
return -1;
}
} else {
manager.displayErrorText(id, "No network found\n");
return -1;
}
return 0;
}
}
/**
* @return 0 is good, -1 is bad
*/
int
SipVoIPLink::answer (CALLID id)
{
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);
osip_message_t *answerMessage = 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, &answerMessage);
if (i != 0) {
// Send 400 BAD_REQUEST
eXosip_call_send_answer (ca->getTid(), BAD_REQ, NULL);
} else {
i = sdp_complete_200ok (ca->getDid(), answerMessage, port);
if (i != 0) {
osip_message_free (answerMessage);
// Send 415 UNSUPPORTED_MEDIA_TYPE
eXosip_call_send_answer (ca->getTid(), UNSUP_MEDIA_TYPE, NULL);
} else {
eXosip_call_send_answer (ca->getTid(), OK, answerMessage);
}
}
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__);
i = -1;
}
return i;
}
/**
* @return > 0 is good, -1 is bad
*/
int
SipVoIPLink::hangup (CALLID id)
{
int i = 0;
SipCall* sipcall = getSipCall(id);
if (sipcall == NULL) { return -1; }
_debug("Hang up call [id = %d, cid = %d, did = %d]\n",
id, sipcall->getCid(), sipcall->getDid());
// Release SIP stack.
eXosip_lock();
i = eXosip_call_terminate (sipcall->getCid(), sipcall->getDid());
eXosip_unlock();
// Release RTP channels
_audiortp.closeRtpSession();
deleteSipCall(id);
return i;
}
int
SipVoIPLink::cancel (CALLID id)
{
int i = 0;
SipCall* sipcall = getSipCall(id);
_debug("Cancel call [id = %d, cid = %d]\n", id, sipcall->getCid());
// Release SIP stack.
eXosip_lock();
i = eXosip_call_terminate (sipcall->getCid(), -1);
eXosip_unlock();
deleteSipCall(id);
return i;
}
/*
* @return -1 = sipcall not present
*/
int
SipVoIPLink::onhold (CALLID id)
{
osip_message_t *invite;
int i;
int did;
sdp_message_t *local_sdp = NULL;
SipCall *sipcall = getSipCall(id);
if ( sipcall == NULL ) { return -1; }
did = sipcall->getDid();
eXosip_lock ();
local_sdp = eXosip_get_local_sdp (did);
eXosip_unlock ();
if (local_sdp == NULL) {
return -1;
}
// Build INVITE_METHOD for put call on-hold
eXosip_lock ();
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);
osip_free (tmp);
return -1;
}
osip_message_set_body (invite, tmp, strlen (tmp));
osip_free (tmp);
osip_message_set_content_type (invite, "application/sdp");
}
// Send request
_audiortp.closeRtpSession();
eXosip_lock ();
i = eXosip_call_send_request (did, invite);
eXosip_unlock ();
// Disable audio
return i;
}
/**
* @return 0 is good, -1 is bad
*/
int
SipVoIPLink::offhold (CALLID 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;
}
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);
osip_free (tmp);
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__);
i = -1;
}
return i;
}
int
SipVoIPLink::transfer (CALLID id, const std::string& to)
{
osip_message_t *refer;
int i;
std::string tmp_to;
tmp_to = toHeader(to);
if (tmp_to.find("@") == std::string::npos) {
tmp_to = tmp_to + "@" + Manager::instance().getConfigString(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;
}
int
SipVoIPLink::refuse (CALLID id)
{
int i;
char tmpbuf[64];
bzero (tmpbuf, 64);
// Get local port
snprintf (tmpbuf, 63, "%d", getSipCall(id)->getLocalAudioPort());
osip_message_t *answerMessage = NULL;
eXosip_lock();
// not BUSY.. where decline the invitation!
i = eXosip_call_build_answer (getSipCall(id)->getTid(), SIP_DECLINE, &answerMessage);
if (i == 0) {
i = eXosip_call_send_answer (getSipCall(id)->getTid(), SIP_DECLINE, answerMessage);
}
eXosip_unlock();
return i;
}
int
SipVoIPLink::getEvent (void)
{
// wait for 0 s, 50 ms
eXosip_event_t* event = eXosip_event_wait (0, 50);
eXosip_lock();
eXosip_automatic_action();
eXosip_unlock();
if (event == NULL) {
return -1;
}
SipCall* sipcall = NULL;
CALLID id = 0;
int returnValue = 0;
_debug("Receive SipEvent #%d %s\n", event->type, event->textinfo);
switch (event->type) {
// IP-Phone user receives a new call
case EXOSIP_CALL_INVITE: //
_debug("> INVITE (receive)\n");
checkNetwork();
// 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 {
returnValue = -1;
break;
}
}
// Generate id
id = Manager::instance().generateNewCallId();
Manager::instance().pushBackNewCall(id, Incoming);
_debug(" ID: %d [cid = %d, did = %d]\n",id, event->cid, event->did);
// Associate an audio port with a call
sipcall = getSipCall(id);
sipcall->setLocalAudioPort(_localPort);
sipcall->setLocalIp(getLocalIpAddress());
_debug(" Local listening port: %d\n", _localPort);
_debug(" Local listening IP: %s\n", getLocalIpAddress().c_str());
sipcall->newIncomingCall(event);
if (Manager::instance().incomingCall(id, sipcall->getName(), sipcall->getNumber()) < 0) {
Manager::instance().displayErrorText(id, " Incoming Call Failed");
}
break;
case EXOSIP_CALL_REINVITE:
_debug("!!! EXOSIP_CALL_REINVITE: Should reinvite? !!!\n");
//eXosip_call_send_answer(event->tid, 403, NULL);
//488 as http://www.atosc.org/pipermail/public/osip/2005-June/005385.html
id = findCallId(event);
if (id != 0) {
sipcall = getSipCall(id);
if (sipcall != NULL) {
_debug("Call reinvite : [id = %d, cid = %d, did = %d], localport=%d\n", id, event->cid, event->did,sipcall->getLocalAudioPort());
_audiortp.closeRtpSession();
sipcall->newIncomingCall(event);
_audiortp.createNewSession(sipcall);
}
} else {
eXosip_call_send_answer(event->tid, 488, NULL);
}
break;
case EXOSIP_CALL_PROCEEDING: // 8
// proceeding call...
break;
case EXOSIP_CALL_RINGING: // 9 peer call is ringing
id = findCallIdInitial(event);
_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 {
returnValue = -1;
}
break;
// The peer-user answers
case EXOSIP_CALL_ANSWERED: // 10
{
id = findCallIdInitial(event);
if ( id != 0) {
sipcall = getSipCall(id);
if ( sipcall ) {
_debug("Call is answered [id = %d, cid = %d, did = %d], localport=%d\n",
id, event->cid, event->did,sipcall->getLocalAudioPort());
}
// Answer
if (Manager::instance().callCanBeAnswered(id)) {
sipcall->setStandBy(false);
_debug("Answering call first time\n");
if (sipcall->answeredCall(event) != -1) {
sipcall->answeredCall_without_hold(event);
Manager::instance().peerAnsweredCall(id);
if(!Manager::instance().callIsOnHold(id)) {
// Outgoing call is answered, start the sound channel.
_debug("Starting AudioRTP\n");
if (_audiortp.createNewSession (sipcall) < 0) {
_debug("FATAL: Unable to start sound (%s:%d)\n",
__FILE__, __LINE__);
returnValue = -1;
break;
}
}
}
} else {
// Answer to on/off hold to send ACK
_debug("Answering call\n");
sipcall->answeredCall(event);
}
break;
} else {
returnValue = -1;
}
}
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 {
returnValue = -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);
if (id != 0) {
if (Manager::instance().callCanBeClosed(id)) {
sipcall = getSipCall(id);
_audiortp.closeRtpSession();
}
Manager::instance().peerHungupCall(id);
deleteSipCall(id);
} else {
returnValue = -1;
}
break;
case EXOSIP_CALL_RELEASED:
//id = findCallId(event);
//if (id!=0) {
//Manager::instance().peerHungupCall(id);
//deleteSipCall(id);
//}
break;
case EXOSIP_CALL_REQUESTFAILURE:
id = findCallId(event);
// Handle 4XX errors
switch (event->response->status_code) {
case AUTH_REQUIRED:
_debug("SIP Server ask required authentification: logging...\n");
setAuthentication();
eXosip_lock();
eXosip_automatic_action();
eXosip_unlock();
break;
case UNAUTHORIZED:
_debug("Request is unauthorized. SIP Server ask authentification: logging...\n");
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 NOT_ACCEPTABLE_HERE: // 488
// Display error on the screen phone
//Manager::instance().displayError(event->response->reason_phrase);
Manager::instance().displayErrorText(id, event->response->reason_phrase);
Manager::instance().callFailure(id);
break;
case BUSY_HERE:
Manager::instance().displayErrorText(id, event->response->reason_phrase);
Manager::instance().callBusy(id);
break;
case REQ_TERMINATED:
break;
default:
break;
}
break;
case EXOSIP_CALL_SERVERFAILURE:
// Handle 5XX errors
switch (event->response->status_code) {
case SERVICE_UNAVAILABLE:
id = findCallId(event);
Manager::instance().callFailure(id);
break;
default:
break;
}
break;
case EXOSIP_CALL_GLOBALFAILURE:
// Handle 6XX errors
switch (event->response->status_code) {
case BUSY_EVERYWHERE:
case DECLINE:
id = findCallId(event);
Manager::instance().callFailure(id);
break;
default:
break;
}
break;
case EXOSIP_CALL_MESSAGE_NEW: // 18
// TODO:
break;
case EXOSIP_REGISTRATION_SUCCESS: // 1
// Manager::instance().displayStatus(LOGGED_IN_STATUS);
break;
case EXOSIP_REGISTRATION_FAILURE: // 2
//Manager::instance().displayError("getEvent : 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;
}
}
// 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;
std::string nb_msg;
osip_body_t *body = NULL;
// Get the message body
ii = osip_message_get_body(event->request, 0, &body);
if (ii != 0) {
_debug("Cannot get body in a new EXOSIP_MESSAGE_NEW event\n");
returnValue = -1;
break;
}
// Analyse message body
if (!body || !body->body) {
returnValue = -1;
break;
}
std::string str(body->body);
pos = str.find(VOICE_MSG);
if (pos == std::string::npos) {
// If the string is not found
returnValue = -1;
break;
}
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(nb_msg);
} else {
// Stop notification when there is 0 voice message
Manager::instance().stopVoiceMessageNotification();
}
}
break;
case EXOSIP_SUBSCRIPTION_ANSWERED: // 38
eXosip_lock();
eXosip_automatic_action();
eXosip_unlock();
break;
case EXOSIP_SUBSCRIPTION_REQUESTFAILURE: //40
break;
default:
//Manager::instance().displayErrorText(event->type, "getEvent:default");
returnValue = -1;
break;
}
//_debug(" : end event : %d / %d\n", event->type, returnValue);
eXosip_event_free(event);
return returnValue;
}
int
SipVoIPLink::getLocalPort (void)
{
return _localPort;
}
void
SipVoIPLink::setLocalPort (int port)
{
_localPort = port;
}
void
SipVoIPLink::carryingDTMFdigits (CALLID id, char code) {
int duration = Manager::instance().getConfigInt(SIGNALISATION, PULSE_LENGTH);
osip_message_t *info;
const int body_len = 1000;
int i;
char *dtmf_body = new char[body_len];
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; dtmf_body = NULL;
}
void
SipVoIPLink::newOutgoingCall (CALLID id)
{
SipCall* sipcall = new SipCall(id, Manager::instance().getCodecDescVector());
if (sipcall != NULL) {
_sipcallVector.push_back(sipcall);
sipcall->setStandBy(true);
}
}
void
SipVoIPLink::newIncomingCall (CALLID id)
{
SipCall* sipcall = new SipCall(id, Manager::instance().getCodecDescVector());
if (sipcall != NULL) {
_sipcallVector.push_back(sipcall);
}
}
void
SipVoIPLink::deleteSipCall (CALLID id)
{
std::vector< SipCall* >::iterator iter = _sipcallVector.begin();
while(iter != _sipcallVector.end()) {
if (*iter && (*iter)->getId() == id) {
delete *iter; *iter = NULL;
_sipcallVector.erase(iter);
return;
}
iter++;
}
}
void
SipVoIPLink::endSipCalls()
{
std::vector< SipCall* >::iterator iter = _sipcallVector.begin();
while(iter != _sipcallVector.end()) {
if ( *iter ) {
// Release SIP stack.
eXosip_lock();
eXosip_call_terminate ((*iter)->getCid(), (*iter)->getDid());
eXosip_unlock();
// Release RTP channels
_audiortp.closeRtpSession();
delete *iter; *iter = NULL;
}
iter++;
}
_sipcallVector.clear();
}
SipCall*
SipVoIPLink::getSipCall (CALLID id)
{
SipCall* sipcall = NULL;
for (unsigned int i = 0; i < _sipcallVector.size(); i++) {
sipcall = _sipcallVector.at(i);
if (sipcall && sipcall->getId() == id) {
return sipcall;
}
}
return NULL;
}
AudioCodec*
SipVoIPLink::getAudioCodec (CALLID id)
{
SipCall* sipcall = getSipCall(id);
if (sipcall != NULL) {
return sipcall->getAudioCodec();
} else {
return NULL;
}
}
///////////////////////////////////////////////////////////////////////////////
// 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 * answerMessage, 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 (answerMessage, buf, strlen (buf));
osip_message_set_content_type (answerMessage, "application/sdp");
sdp_message_free (remote_sdp);
return 0;
}
int
SipVoIPLink::behindNat (void)
{
StunAddress4 stunSvrAddr;
stunSvrAddr.addr = 0;
// Stun server
std::string svr = Manager::instance().getConfigString(SIGNALISATION, STUN_SERVER);
// Convert char* to StunAddress4 structure
bool ret = stunParseServerName ((char*)svr.data(), stunSvrAddr);
if (!ret) {
_debug("SIP: Stun server address (%s) is not valid\n", svr.data());
return 0;
}
// Firewall address
//_debug("STUN server: %s\n", svr.data());
Manager::instance().getStunInfo(stunSvrAddr);
return 1;
}
/**
* Get the local Ip by eXosip
* only if the local ip address is to his default value: 127.0.0.1
* setLocalIpAdress
* @return false if not found
*/
bool
SipVoIPLink::getSipLocalIp (void)
{
bool returnValue = true;
if (getLocalIpAddress() == "127.0.0.1") {
char* myIPAddress = new char[65];
if (eXosip_guess_localip (AF_INET, myIPAddress, 64) == -1) {
returnValue = false;
}
setLocalIpAddress(std::string(myIPAddress));
delete [] myIPAddress; myIPAddress = NULL;
}
return returnValue;
}
int
SipVoIPLink::checkUrl (const std::string& url)
{
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;
}
// Free memory
osip_from_free (to);
return 0;
}
int
SipVoIPLink::setAuthentication (void)
{
ManagerImpl& manager = Manager::instance();
std::string login, pass, realm;
login = manager.getConfigString(SIGNALISATION, AUTH_USER_NAME);
if (login.empty()) {
login = manager.getConfigString(SIGNALISATION, USER_PART);
}
pass = manager.getConfigString(SIGNALISATION, PASSWORD);
if (pass.empty()) {
manager.displayConfigError("Fill password field");
return -1;
}
int returnValue = 0;
eXosip_lock();
if (eXosip_add_authentication_info(login.data(), login.data(),
pass.data(), NULL, NULL) != 0) {
returnValue = -1;
}
eXosip_unlock();
return returnValue;
}
std::string
SipVoIPLink::fromHeader (const std::string& user, const std::string& host)
{
std::string displayname = Manager::instance().getConfigString(SIGNALISATION,
FULL_NAME);
return ("\"" + displayname + "\"" + " <sip:" + user + "@" + host + ">");
}
std::string
SipVoIPLink::toHeader(const std::string& to)
{
if (to.find("sip:") == std::string::npos) {
return ("sip:" + to );
} else {
return to;
}
}
int
SipVoIPLink::startCall (CALLID id, const std::string& from, const std::string& to, const std::string& subject, const std::string& route)
{
SipCall* sipcall = getSipCall(id);
if ( sipcall == NULL) {
return -1; // error, we can't find the sipcall
}
osip_message_t *invite;
if (checkUrl(from) != 0) {
Manager::instance().displayConfigError("Error in source address");
return -1;
}
if (checkUrl(to) != 0) {
Manager::instance().displayErrorText(id, "Error in destination address");
return -1;
}
if (!Manager::instance().useStun()) {
// Set random port for outgoing call if no firewall
setLocalPort(RANDOM_LOCAL_PORT);
_debug("SipVoIPLink::startCall: Local audio port: %d\n",_localPort);
} else {
// If use Stun server
if (behindNat() != 0) {
_debug("SipVoIPLink::startCall: sip invite: firewall port = %d\n",Manager::instance().getFirewallPort());
setLocalPort(Manager::instance().getFirewallPort());
} else {
return -1;
}
}
// Set local audio port for sipcall(id)
sipcall->setLocalAudioPort(_localPort);
sipcall->setLocalIp(getLocalIpAddress());
int i = eXosip_call_build_initial_invite (&invite, (char*)to.data(),
(char*)from.data(),
(char*)route.data(),
(char*)subject.data());
if (i != 0) {
return -1; // error when building the invite
}
int payload;
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
CodecDescriptorVector* cdv = Manager::instance().getCodecDescVector();
unsigned int nb = cdv->size();
for (unsigned int iCodec = 0; iCodec < nb; iCodec++) {
payload = cdv->at(iCodec)->getPayload();
// Add payload to rtpmap if it is not already added
if (!isInRtpmap(iCodec, payload, cdv)) {
snprintf(media, 63, "%d ", payload);
strcat (media_audio, media);
snprintf(rtpmap, 127, "a=rtpmap: %d %s/%d\r\n", payload,
cdv->at(iCodec)->rtpmapPayload(payload).data(), SAMPLING_RATE);
strcat(rtpmap_attr, rtpmap);
}
}
// http://www.antisip.com/documentation/eXosip2/group__howto1__initialize.html
// tell sip if we support SIP extension like 100rel
// osip_message_set_supported (invite, "100rel");
/* add sdp body */
{
char tmp[4096];
char localip[128];
eXosip_guess_localip (AF_INET, localip, 128);
char port[64];
bzero (port, 64);
snprintf (port, 63, "%d", getLocalPort());
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);
// media_audio should be one, two or three numbers?
osip_message_set_body (invite, tmp, strlen (tmp));
osip_message_set_content_type (invite, "application/sdp");
}
eXosip_lock();
// this is the cid (call id from exosip)
int cid = eXosip_call_send_initial_invite (invite);
// Keep the cid in case of cancelling
sipcall->setCid(cid);
if (cid <= 0) {
eXosip_unlock();
return -1;
} else {
eXosip_call_set_reference (cid, NULL);
}
eXosip_unlock();
return cid; // this is the Cid
}
CALLID
SipVoIPLink::findCallId (eXosip_event_t *e)
{
for (unsigned int k = 0; k < _sipcallVector.size(); k++) {
SipCall* sipcall = _sipcallVector.at(k);
if (sipcall && sipcall->getCid() == e->cid &&
sipcall->getDid() == e->did) {
return sipcall->getId();
}
}
return 0;
}
/**
* This function is used when findCallId failed (return 0)
* ie: the dialog id change
* can be use when anwsering a new call or
* when cancelling a call
*/
CALLID
SipVoIPLink::findCallIdInitial (eXosip_event_t *e)
{
for (unsigned int k = 0; k < _sipcallVector.size(); k++) {
SipCall* sipcall = _sipcallVector.at(k);
// the dialog id is not set when you do a new call
// so you can't check it when you want to retreive it
// for the first call anwser
if (sipcall && sipcall->getCid() == e->cid) {
return sipcall->getId();
}
}
return 0;
}