diff --git a/daemon/src/audio/audiortp/AudioRtpFactory.cpp b/daemon/src/audio/audiortp/AudioRtpFactory.cpp
index d2dcf09d9843954f2d36b2c950a1538896f4b69c..f443908ac03d458d1fceaffb3eb084d5a24ad148 100644
--- a/daemon/src/audio/audiortp/AudioRtpFactory.cpp
+++ b/daemon/src/audio/audiortp/AudioRtpFactory.cpp
@@ -57,47 +57,29 @@ AudioRtpFactory::~AudioRtpFactory()
void AudioRtpFactory::initAudioRtpConfig ()
{
- if (_rtpSession != NULL) {
- _debug ("An audio rtp thread was already created but not" \
- "destroyed. Forcing it before continuing.");
+ if (_rtpSession != NULL)
stop();
- }
-
- std::string accountId(Manager::instance().getAccountFromCall (ca_->getCallId()));
-
- _debug ("AudioRtpFactory: Init rtp session for account %s", accountId.c_str());
-
- // Manager::instance().getAccountLink (accountId);
- Account *account = Manager::instance().getAccount (accountId);
- if (!account)
- _error ("AudioRtpFactory: Error no account found");
+ std::string accountId(Manager::instance().getAccountFromCall(ca_->getCallId()));
- registerAccount(account, accountId);
-}
-
-void AudioRtpFactory::registerAccount(Account * /*account*/, const std::string & /* id */)
-{
- _srtpEnabled = false;
- _keyExchangeProtocol = Symmetric;
- _helloHashEnabled = false;
-}
+ SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount (accountId));
+ if (account) {
+ _srtpEnabled = account->getSrtpEnabled();
+ std::string key(account->getSrtpKeyExchange());
+ if (key == "sdes")
+ _keyExchangeProtocol = Sdes;
+ else if (key == "zrtp")
+ _keyExchangeProtocol = Zrtp;
+ else
+ _keyExchangeProtocol = Symmetric;
-void AudioRtpFactory::registerAccount(SIPAccount *sipaccount, const std::string& accountId)
-{
- _srtpEnabled = sipaccount->getSrtpEnabled();
- std::string tempkey(sipaccount->getSrtpKeyExchange());
-
- if (tempkey == "sdes")
- _keyExchangeProtocol = Sdes;
- else if (tempkey == "zrtp")
- _keyExchangeProtocol = Zrtp;
- else
+ _helloHashEnabled = account->getZrtpHelloHash();
+ } else {
+ _srtpEnabled = false;
_keyExchangeProtocol = Symmetric;
-
- _debug ("AudioRtpFactory: Registered account %s profile selected with key exchange protocol number %d", accountId.c_str(), _keyExchangeProtocol);
- _helloHashEnabled = sipaccount->getZrtpHelloHash();
+ _helloHashEnabled = false;
+ }
}
void AudioRtpFactory::initAudioSymmetricRtpSession ()
@@ -113,24 +95,20 @@ void AudioRtpFactory::initAudioSymmetricRtpSession ()
case Zrtp:
_rtpSession = new AudioZrtpSession (ca_, zidFilename);
-
if (_helloHashEnabled) {
// TODO: be careful with that. The hello hash is computed asynchronously. Maybe it's
// not even available at that point.
ca_->getLocalSDP()->setZrtpHash (static_cast<AudioZrtpSession *> (_rtpSession)->getHelloHash());
_debug ("AudioRtpFactory: Zrtp hello hash fed to SDP");
}
-
break;
case Sdes:
-
_rtpSession = new AudioSrtpSession (ca_);
break;
default:
- _debug ("AudioRtpFactory: Unsupported Rtp Session Exception Type!");
- throw UnsupportedRtpSessionType();
+ throw UnsupportedRtpSessionType("Unsupported Rtp Session Exception Type!");
}
} else {
_rtpSession = new AudioSymmetricRtpSession (ca_);
diff --git a/daemon/src/audio/audiortp/AudioRtpFactory.h b/daemon/src/audio/audiortp/AudioRtpFactory.h
index 78fb34d2234730f41feac72f41e7c38b7602e1fd..8aa38a83b2cb6a41e7f9b9c846cc55b6655a58da 100644
--- a/daemon/src/audio/audiortp/AudioRtpFactory.h
+++ b/daemon/src/audio/audiortp/AudioRtpFactory.h
@@ -152,8 +152,6 @@ class AudioRtpFactory
void sendDtmfDigit (int digit);
private:
- void registerAccount(Account *account, const std::string &id);
- void registerAccount(SIPAccount *account, const std::string &id);
AudioRtpSession *_rtpSession;
ost::Mutex _audioRtpThreadMutex;
diff --git a/daemon/src/iax/iaxvoiplink.cpp b/daemon/src/iax/iaxvoiplink.cpp
index 4ba351cbd82671516759d047410d5d645da2eba2..742ea4b254aae73cbf04450d3df2a17859908709 100644
--- a/daemon/src/iax/iaxvoiplink.cpp
+++ b/daemon/src/iax/iaxvoiplink.cpp
@@ -165,10 +165,9 @@ IAXVoIPLink::terminateIAXCall()
void
IAXVoIPLink::getEvent()
{
- iax_event* event = NULL;
-
{
ost::MutexLock m(mutexIAX_);
+ iax_event *event;
while ((event = iax_get_event (IAX_NONBLOCKING)) != NULL) {
// If we received an 'ACK', libiax2 tells apps to ignore them.
if (event->etype == IAX_EVENT_NULL)
@@ -263,8 +262,6 @@ IAXVoIPLink::getIAXCall (const std::string& id)
void
IAXVoIPLink::sendRegister (Account *a)
{
- _debug ("IAX: Sending registration");
-
IAXAccount *account = dynamic_cast<IAXAccount*>(a);
if (account->getHostname().empty())
@@ -273,43 +270,28 @@ IAXVoIPLink::sendRegister (Account *a)
if (account->getUsername().empty())
throw VoipLinkException("Account username is empty");
- // lock
ost::MutexLock m(mutexIAX_);
- // Always use a brand new session
if (regSession_)
iax_destroy(regSession_);
regSession_ = iax_session_new();
- if (!regSession_)
- _debug ("IAX: Error when generating new session for register");
- else {
- _debug ("IAX: Sending registration to %s with user %s", account->getHostname().c_str() , account->getUsername().c_str());
- int val = iax_register (regSession_, account->getHostname().data(), account->getUsername().data(), account->getPassword().data(), 120);
- _debug ("IAX: Return value: %d", val);
- // set the time-out to 15 seconds, after that, resend a registration request.
- // until we unregister.
+ if (regSession_) {
+ iax_register (regSession_, account->getHostname().data(), account->getUsername().data(), account->getPassword().data(), 120);
nextRefreshStamp_ = time (NULL) + 10;
-
- account->setRegistrationState (Trying);
+ account->setRegistrationState(Trying);
}
}
void
IAXVoIPLink::sendUnregister (Account *a)
{
- _debug ("IAXVoipLink: Send unregister");
-
- {
- ost::MutexLock m(mutexIAX_);
- if (regSession_) {
- iax_destroy (regSession_);
- regSession_ = NULL;
- }
- }
-
- mutexIAX_.leaveMutex();
+ ost::MutexLock m(mutexIAX_);
+ if (regSession_) {
+ iax_destroy (regSession_);
+ regSession_ = NULL;
+ }
nextRefreshStamp_ = 0;
diff --git a/daemon/src/managerimpl.cpp b/daemon/src/managerimpl.cpp
index 6c01fa5f4155cb9d0cdef8877b042f43cf034538..b455abb83a9771cafd08668b9b9f1bb64e90bb35 100644
--- a/daemon/src/managerimpl.cpp
+++ b/daemon/src/managerimpl.cpp
@@ -2755,10 +2755,6 @@ void ManagerImpl::loadAccountMap(Conf::YamlParser *parser)
// IP to IP settings (most likely using port 5060)
link->createDefaultSipUdpTransport();
- // Call this method to create TLS listener
- link->createDefaultSipTlsListener();
-
-
// Force IP2IP settings to be loaded to be loaded
// No registration in the sense of the REGISTER method is performed.
ip2ip->registerVoIPLink();
diff --git a/daemon/src/sip/sipaccount.cpp b/daemon/src/sip/sipaccount.cpp
index 505e8ed90173724fd5958d50b2b94cc7c789d443..ad7b3fe5fb489dd7b8d5ed3e19ae9bba7ba85b9c 100644
--- a/daemon/src/sip/sipaccount.cpp
+++ b/daemon/src/sip/sipaccount.cpp
@@ -535,7 +535,7 @@ void SIPAccount::registerVoIPLink()
return;
try {
- link_->sendRegister (this);
+ link_->sendRegister(this);
}
catch (const VoipLinkException &e) {
_error("SIPAccount: %s", e.what());
diff --git a/daemon/src/sip/sipvoiplink.cpp b/daemon/src/sip/sipvoiplink.cpp
index 11a87bc470ce1556ad09a0129418e597d98f542e..a9a3b3d9f3eab5a8a2ed6bdd90a78cfbdd14c383 100644
--- a/daemon/src/sip/sipvoiplink.cpp
+++ b/daemon/src/sip/sipvoiplink.cpp
@@ -57,7 +57,6 @@
#include "pjsip/sip_endpoint.h"
#include "pjsip/sip_transport_tls.h"
-#include "pjsip/sip_transport_tls.h"
#include "pjsip/sip_uri.h"
#include <pjnath.h>
@@ -80,9 +79,6 @@ namespace {
/** The default transport (5060) */
static pjsip_transport *_localUDPTransport = NULL;
-/** The local tls listener */
-static pjsip_tpfactory *_localTlsListener = NULL;
-
/** A map to retreive SFLphone internal call id
* Given a SIP call ID (usefull for transaction sucha as transfer)*/
static std::map<std::string, std::string> transferCallID;
@@ -138,6 +134,33 @@ std::string loadSIPLocalIP()
return pj_inet_ntoa (ip_addr.ipv4.sin_addr);
return "";
}
+
+pjsip_route_hdr *createRouteSet(const std::string &route, pj_pool_t *hdr_pool)
+{
+ int port = 0;
+ std::string host;
+
+ size_t found = route.find(":");
+ if (found != std::string::npos) {
+ host = route.substr(0, found);
+ port = atoi(route.substr(found + 1, route.length()).c_str());
+ }
+ else {
+ host = route;
+ }
+
+ pjsip_route_hdr *route_set = pjsip_route_hdr_create (hdr_pool);
+ pjsip_route_hdr *routing = pjsip_route_hdr_create (hdr_pool);
+ pjsip_sip_uri *url = pjsip_sip_uri_create (hdr_pool, 0);
+ routing->name_addr.uri = (pjsip_uri*) url;
+ pj_strdup2(hdr_pool, &url->host, host.c_str());
+ url->port = port;
+
+ pj_list_push_back(route_set, pjsip_hdr_clone (hdr_pool, routing));
+
+ return route_set;
+}
+
} // end anonymous namespace
/*************************************************************************************************/
@@ -251,26 +274,34 @@ SIPVoIPLink::getEvent()
pj_time_val timeout = {0, 10};
pjsip_endpt_handle_events(_endpt, &timeout);
-
}
void SIPVoIPLink::sendRegister (Account *a)
{
SIPAccount *account = static_cast<SIPAccount*>(a);
- acquireTransport(account);
+ if (account->transport)
+ pjsip_transport_dec_ref(account->transport);
- ost::MutexLock m(mutexSIP_);
+ createSipTransport(account);
+ if (account->transport)
+ return;
- pjsip_regc *regc = account->getRegistrationInfo();
- account->setRegister(true);
+ // Could not create new transport, this transport may already exists
+ account->transport = transportMap_[account->getLocalPort()];
+ if (account->transport) {
+ pjsip_transport_add_ref(account->transport);
+ } else {
+ account->transport = _localUDPTransport;
+ account->setLocalPort(_localUDPTransport->local_name.port);
+ }
- int expire_value = atoi(account->getRegistrationExpire().c_str());
- if (!expire_value)
- expire_value = PJSIP_REGC_EXPIRATION_NOT_SPECIFIED;
+ ost::MutexLock m(mutexSIP_);
+ account->setRegister(true);
account->setRegistrationState (Trying);
+ pjsip_regc *regc = account->getRegistrationInfo();
if (pjsip_regc_create(_endpt, (void *) account, ®istration_cb, ®c) != PJ_SUCCESS)
throw VoipLinkException("UserAgent: Unable to create regc structure.");
@@ -285,55 +316,49 @@ void SIPVoIPLink::sendRegister (Account *a)
pj_str_t pjContact = pj_str((char*)contact.c_str());
pj_str_t pjSrv = pj_str((char*)srvUri.c_str());
+ int expire_value = atoi(account->getRegistrationExpire().c_str());
+ if (!expire_value)
+ expire_value = PJSIP_REGC_EXPIRATION_NOT_SPECIFIED;
+
+
if (pjsip_regc_init (regc, &pjSrv, &pjFrom, &pjFrom, 1, &pjContact, expire_value) != PJ_SUCCESS)
throw VoipLinkException("Unable to initialize account registration structure");
- if (!account->getServiceRoute().empty()) {
- pjsip_route_hdr *route_set = createRouteSet(account, _pool);
- pjsip_regc_set_route_set (regc, route_set);
- }
+ if (!account->getServiceRoute().empty())
+ pjsip_regc_set_route_set (regc, createRouteSet(account->getServiceRoute(), _pool));
+
+ pjsip_regc_set_credentials (regc, account->getCredentialCount(), account->getCredInfo());
- unsigned count = account->getCredentialCount();
- pjsip_cred_info *info = account->getCredInfo();
- pjsip_regc_set_credentials (regc, count, info);
- // Add User-Agent Header
pjsip_hdr hdr_list;
pj_list_init (&hdr_list);
-
- pj_str_t useragent = pj_str ((char*)account->getUserAgentName().c_str());
+ std::string useragent(account->getUserAgentName());
+ pj_str_t pJuseragent = pj_str ((char*)useragent.c_str());
const pj_str_t STR_USER_AGENT = { (char*) "User-Agent", 10 };
- pjsip_generic_string_hdr *h = pjsip_generic_string_hdr_create (_pool, &STR_USER_AGENT, &useragent);
+ pjsip_generic_string_hdr *h = pjsip_generic_string_hdr_create(_pool, &STR_USER_AGENT, &pJuseragent);
pj_list_push_back (&hdr_list, (pjsip_hdr*) h);
pjsip_regc_add_headers (regc, &hdr_list);
+
pjsip_tx_data *tdata;
if (pjsip_regc_register (regc, PJ_TRUE, &tdata) != PJ_SUCCESS)
throw VoipLinkException("Unable to initialize transaction data for account registration");
- pjsip_tpselector *tp = initTransportSelector (account->transport, _pool);
-
- pj_status_t status = pjsip_regc_set_transport (regc, tp);
+ if (pjsip_regc_set_transport (regc, initTransportSelector (account->transport, _pool)) != PJ_SUCCESS)
+ throw VoipLinkException("Unable to set transport");
// decrease transport's ref count, counter incrementation is managed when acquiring transport
- if (account->transport)
- pjsip_transport_dec_ref(account->transport);
-
- if (status != PJ_SUCCESS)
- throw VoipLinkException("Unable to set transport");
+ pjsip_transport_dec_ref(account->transport);
// pjsip_regc_send increment the transport ref count by one,
- status = pjsip_regc_send(regc, tdata);
+ if (pjsip_regc_send(regc, tdata) != PJ_SUCCESS)
+ throw VoipLinkException("Unable to send account registration request");
// Decrease transport's ref count, since coresponding reference counter decrementation
// is performed in pjsip_regc_destroy. This function is never called in SFLphone as the
// regc data structure is permanently associated to the account at first registration.
- if (account->transport)
- pjsip_transport_dec_ref (account->transport);
-
- if (status != PJ_SUCCESS)
- throw VoipLinkException("Unable to send account registration request");
+ pjsip_transport_dec_ref (account->transport);
account->setRegistrationInfo (regc);
}
@@ -342,15 +367,6 @@ void SIPVoIPLink::sendUnregister (Account *a)
{
SIPAccount *account = dynamic_cast<SIPAccount *>(a);
- // If an transport is attached to this account, detach it and decrease reference counter
- if (account->transport) {
- _debug ("Sent account unregistration using transport: %s %s (refcnt=%d)",
- account->transport->obj_name,
- account->transport->info,
- (int) pj_atomic_get (account->transport->ref_cnt));
-
- }
-
// This may occurs if account failed to register and is in state INVALID
if (!account->isRegister()) {
account->setRegistrationState (Unregistered);
@@ -373,19 +389,11 @@ void SIPVoIPLink::sendUnregister (Account *a)
Call *SIPVoIPLink::newOutgoingCall (const std::string& id, const std::string& toUrl)
{
- // Create a new SIP call
- SIPCall* call = new SIPCall (id, Call::Outgoing, _cp);
-
- // Find the account associated to this call
SIPAccount *account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (Manager::instance().getAccountFromCall (id)));
- if (account == NULL) {
- _error ("UserAgent: Error: Could not retrieving account to make call with");
- call->setConnectionState (Call::Disconnected);
- call->setState (Call::Error);
- delete call;
- // TODO: We should investigate how we could get rid of this error and create a IP2IP call instead
+ if (account == NULL) // TODO: We should investigate how we could get rid of this error and create a IP2IP call instead
throw VoipLinkException("Could not get account for this call");
- }
+
+ SIPCall* call = new SIPCall (id, Call::Outgoing, _cp);
// If toUri is not a well formated sip URI, use account information to process it
std::string toUri;
@@ -393,13 +401,10 @@ Call *SIPVoIPLink::newOutgoingCall (const std::string& id, const std::string& to
toUrl.find("sips:") != std::string::npos)
toUri = toUrl;
else
- toUri = account->getToUri (toUrl);
+ toUri = account->getToUri(toUrl);
call->setPeerNumber (toUri);
- _debug ("UserAgent: New outgoing call %s to %s", id.c_str(), toUri.c_str());
-
std::string localAddr(getInterfaceAddrFromName (account->getLocalInterface ()));
- _debug ("UserAgent: Local address for thi call: %s", localAddr.c_str());
if (localAddr == "0.0.0.0")
localAddr = loadSIPLocalIP();
@@ -419,27 +424,22 @@ Call *SIPVoIPLink::newOutgoingCall (const std::string& id, const std::string& to
// the session initialization is completed
sfl::Codec* audiocodec = Manager::instance().audioCodecFactory.instantiateCodec (PAYLOAD_CODEC_ULAW);
if (audiocodec == NULL) {
- _error ("UserAgent: Could not instantiate codec");
delete call;
throw VoipLinkException ("Could not instantiate codec for early media");
}
try {
- _info ("UserAgent: Creating new rtp session");
call->getAudioRtp()->initAudioRtpConfig ();
call->getAudioRtp()->initAudioSymmetricRtpSession ();
call->getAudioRtp()->initLocalCryptoInfo ();
- _info ("UserAgent: Start audio rtp session");
call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
} catch (...) {
delete call;
throw VoipLinkException ("Could not start rtp session for early media");
}
- // init file name according to peer phone number
- call->initRecFileName (toUrl);
+ call->initRecFileName(toUrl);
- // Building the local SDP offer
call->getLocalSDP()->setLocalIP (addrSdp);
call->getLocalSDP()->createOffer(account->getActiveCodecs());
@@ -454,25 +454,19 @@ Call *SIPVoIPLink::newOutgoingCall (const std::string& id, const std::string& to
void
SIPVoIPLink::answer (Call *c)
{
- pjsip_tx_data *tdata;
-
- _debug ("UserAgent: Answering call");
-
SIPCall *call = dynamic_cast<SIPCall*>(c);
- if (call != NULL) {
- pjsip_inv_session *inv_session = call->inv;
+ if (!call)
+ return;
- _debug ("UserAgent: SDP negotiation success! : call %s ", call->getCallId().c_str());
- // Create and send a 200(OK) response
- if (pjsip_inv_answer (inv_session, PJSIP_SC_OK, NULL, NULL, &tdata) != PJ_SUCCESS)
- throw VoipLinkException("Could not init invite request answer (200 OK)");
-
- if (pjsip_inv_send_msg (inv_session, tdata) != PJ_SUCCESS)
- throw VoipLinkException("Could not send invite request answer (200 OK)");
+ pjsip_tx_data *tdata;
+ if (pjsip_inv_answer (call->inv, PJSIP_SC_OK, NULL, NULL, &tdata) != PJ_SUCCESS)
+ throw VoipLinkException("Could not init invite request answer (200 OK)");
- call->setConnectionState (Call::Connected);
- call->setState (Call::Active);
- }
+ if (pjsip_inv_send_msg (call->inv, tdata) != PJ_SUCCESS)
+ throw VoipLinkException("Could not send invite request answer (200 OK)");
+
+ call->setConnectionState (Call::Connected);
+ call->setState (Call::Active);
}
void
@@ -491,7 +485,7 @@ SIPVoIPLink::hangup (const std::string& id)
// Looks for sip routes
if (not (account->getServiceRoute().empty())) {
- pjsip_route_hdr *route_set = createRouteSet(account, inv->pool);
+ pjsip_route_hdr *route_set = createRouteSet(account->getServiceRoute(), inv->pool);
pjsip_dlg_set_route_set (inv->dlg, route_set);
}
@@ -562,18 +556,10 @@ SIPVoIPLink::offhold (const std::string& id)
throw VoipLinkException("Could not find sdp session");
try {
- // Retreive previously selected codec
- int pl;
+ int pl = PAYLOAD_CODEC_ULAW;
sfl::Codec *sessionMedia = sdpSession->getSessionMedia();
- if (sessionMedia == NULL) {
- _warn("UserAgent: Session media not yet initialized, using default (ULAW)");
- pl = PAYLOAD_CODEC_ULAW;
- }
- else
- pl = (int) sessionMedia->getPayloadType();
-
- _debug ("UserAgent: Payload from session media %d", pl);
-
+ if (sessionMedia)
+ pl = sessionMedia->getPayloadType();
// Create a new instance for this codec
sfl::Codec* audiocodec = Manager::instance().audioCodecFactory.instantiateCodec (pl);
@@ -740,14 +726,11 @@ bool
SIPVoIPLink::refuse (const std::string& id)
{
SIPCall *call = getSIPCall (id);
-
- // can't refuse outgoing call or connected
if (!call->isIncoming() or call->getConnectionState() == Call::Connected)
return false;
call->getAudioRtp()->stop();
- // User refuse current call. Notify peer
pjsip_tx_data *tdata;
if (pjsip_inv_end_session (call->inv, PJSIP_SC_DECLINE, NULL, &tdata) != PJ_SUCCESS)
return false;
@@ -835,6 +818,9 @@ SIPVoIPLink::SIPStartCall(SIPCall *call)
pj_str_t pjContact = pj_str((char*)contact.c_str());
pj_str_t pjTo = pj_str((char*)toUri.c_str());
+
+
+
pjsip_dialog *dialog;
if (pjsip_dlg_create_uac (pjsip_ua_instance(), &pjFrom, &pjContact, &pjTo, NULL, &dialog) != PJ_SUCCESS)
return false;
@@ -843,7 +829,7 @@ SIPVoIPLink::SIPStartCall(SIPCall *call)
return false;
if (not account->getServiceRoute().empty())
- pjsip_dlg_set_route_set(dialog, createRouteSet(account, call->inv->pool));
+ pjsip_dlg_set_route_set(dialog, createRouteSet(account->getServiceRoute(), call->inv->pool));
pjsip_auth_clt_set_credentials (&dialog->auth_sess, account->getCredentialCount(), account->getCredInfo());
@@ -902,7 +888,7 @@ SIPCall*
SIPVoIPLink::getSIPCall (const std::string& id)
{
SIPCall *result = dynamic_cast<SIPCall*> (getCall (id));
- if (result == 0)
+ if (result == NULL)
throw VoipLinkException("Could not find SIPCall " + id);
return result;
}
@@ -951,15 +937,13 @@ bool SIPVoIPLink::SIPNewIpToIpCall (const std::string& id, const std::string& to
return false;
}
- if (createTlsTransport (account, remoteAddr) != PJ_SUCCESS) {
+ createTlsTransport(account, remoteAddr);
+ if (!account->transport) {
delete call;
return false;
}
}
- if (account->transport == NULL)
- account->transport = _localUDPTransport;
-
if (!SIPStartCall(call)) {
delete call;
return false;
@@ -986,11 +970,10 @@ static pj_bool_t stun_sock_on_rx_data_cb (pj_stun_sock *stun_sock UNUSED, void *
pj_status_t SIPVoIPLink::stunServerResolve (SIPAccount *account)
{
- // Initialize STUN configuration
pj_stun_config stunCfg;
pj_stun_config_init (&stunCfg, &_cp->factory, 0, pjsip_endpt_get_ioqueue (_endpt), pjsip_endpt_get_timer_heap (_endpt));
- const pj_stun_sock_cb stun_sock_cb = {
+ static const pj_stun_sock_cb stun_sock_cb = {
stun_sock_on_rx_data_cb,
NULL,
stun_sock_on_status_cb
@@ -1019,46 +1002,17 @@ pj_status_t SIPVoIPLink::stunServerResolve (SIPAccount *account)
return status;
}
-void SIPVoIPLink::acquireTransport (SIPAccount *account)
-{
- if (account->transport)
- pjsip_transport_dec_ref(account->transport);
-
- if (createSipTransport(account))
- return;
-
- // Could not create new transport, this transport may already exists
- account->transport = transportMap_[account->getLocalPort()];
- if (account->transport) {
- pjsip_transport_add_ref(account->transport);
- } else {
- account->transport = _localUDPTransport;
- account->setLocalPort(_localUDPTransport->local_name.port);
- }
-}
-
void SIPVoIPLink::createDefaultSipUdpTransport()
{
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(IP2IP_PROFILE));
- pjsip_transport *transport = createUdpTransport(account, true);
- if (transport) {
- transportMap_[account->getLocalPort()] = transport;
- _localUDPTransport = transport;
- account->transport = transport;
- }
-}
-
+ createUdpTransport(account);
+ assert(account->transport);
-void SIPVoIPLink::createDefaultSipTlsListener()
-{
- SIPAccount *account = dynamic_cast<SIPAccount*>(Manager::instance().getAccount(IP2IP_PROFILE));
- if (account->isTlsEnabled())
- createTlsListener(account);
+ _localUDPTransport = account->transport;
}
-
-void SIPVoIPLink::createTlsListener (SIPAccount *account)
+void SIPVoIPLink::createTlsListener (SIPAccount *account, pjsip_tpfactory **listener)
{
pj_sockaddr_in local_addr;
pj_sockaddr_in_init (&local_addr, 0, 0);
@@ -1070,106 +1024,157 @@ void SIPVoIPLink::createTlsListener (SIPAccount *account)
pjsip_host_port a_name = {
pj_str((char*)loadSIPLocalIP().c_str()),
- account->getTlsListenerPort()
+ local_addr.sin_port
};
- pjsip_tls_transport_start(_endpt, account->getTlsSetting(), &local_addr, &a_name, 1, &_localTlsListener);
+ pjsip_tls_transport_start(_endpt, account->getTlsSetting(), &local_addr, &a_name, 1, listener);
}
-bool SIPVoIPLink::createSipTransport (SIPAccount *account)
+void SIPVoIPLink::createTlsTransport (SIPAccount *account, std::string remoteAddr)
{
- if (account->isTlsEnabled()) {
- if (_localTlsListener == NULL)
- createTlsListener (account);
+ account->transport = NULL;
+ pj_str_t remote;
+ pj_cstr (&remote, remoteAddr.c_str());
+
+ pj_sockaddr_in rem_addr;
+ pj_sockaddr_in_init (&rem_addr, &remote, (pj_uint16_t) 5061);
+
+ static pjsip_tpfactory *localTlsListener = NULL; /** The local tls listener */
+ if (localTlsListener == NULL)
+ createTlsListener(account, &localTlsListener);
+ pjsip_endpt_acquire_transport(_endpt, PJSIP_TRANSPORT_TLS, &rem_addr, sizeof (rem_addr), NULL, &account->transport);
+}
+
+
+void SIPVoIPLink::createSipTransport (SIPAccount *account)
+{
+ if (account->isTlsEnabled()) {
std::string remoteSipUri(account->getServerUri());
size_t sips = remoteSipUri.find ("<sips:") + 6;
size_t trns = remoteSipUri.find (";transport");
std::string remoteAddr(remoteSipUri.substr (sips, trns-sips));
- // Nothing to do, TLS listener already created at pjsip's startup and TLS connection
- // is automatically handled in pjsip when sending registration messages.
- return createTlsTransport (account, remoteAddr) == PJ_SUCCESS;
- } else {
- // Launch a new UDP listener/transport, using the published address
- if (account->isStunEnabled ()) {
- if (createAlternateUdpTransport (account) != PJ_SUCCESS)
- return false;
- } else {
- pjsip_transport *transport = createUdpTransport (account, false);
- if (!transport)
- return false;
-
- transportMap_[account->getLocalPort()] = transport;
- account->transport = transport;
- }
- }
-
- return true;
+ createTlsTransport(account, remoteAddr);
+ } else if (account->isStunEnabled ())
+ createStunTransport(account);
+ else
+ createUdpTransport(account);
}
-pjsip_transport *SIPVoIPLink::createUdpTransport (SIPAccount *account, bool local)
+void SIPVoIPLink::createUdpTransport (SIPAccount *account)
{
- std::string listeningAddress(loadSIPLocalIP());
- if (listeningAddress.empty())
- return NULL;
-
- // We are trying to initialize a UDP transport available for all local accounts and direct IP calls
- if (account->getLocalInterface () != "default")
- listeningAddress = getInterfaceAddrFromName (account->getLocalInterface());
-
- int listeningPort = account->getLocalPort ();
+ account->transport = NULL;
+ std::string listeningAddress;
+ pj_uint16_t listeningPort = account->getLocalPort();
pj_sockaddr_in bound_addr;
- pj_memset (&bound_addr, 0, sizeof (bound_addr));
+ pj_bzero(&bound_addr, sizeof (bound_addr));
+ bound_addr.sin_port = pj_htons(listeningPort);
+ bound_addr.sin_family = PJ_AF_INET;
if (account->getLocalInterface () == "default") {
- // Init bound address to ANY
- bound_addr.sin_addr.s_addr = pj_htonl (PJ_INADDR_ANY);
listeningAddress = loadSIPLocalIP ();
+ bound_addr.sin_addr.s_addr = pj_htonl (PJ_INADDR_ANY);
} else {
- // bind this account to a specific interface
- pj_str_t temporary_address;
- pj_strdup2 (_pool, &temporary_address, listeningAddress.c_str());
- bound_addr.sin_addr = pj_inet_addr (&temporary_address);
+ listeningAddress = getInterfaceAddrFromName(account->getLocalInterface());
+ bound_addr.sin_addr = pj_inet_addr2(listeningAddress.c_str());
}
- bound_addr.sin_port = pj_htons ( (pj_uint16_t) listeningPort);
- bound_addr.sin_family = PJ_AF_INET;
- pj_bzero (bound_addr.sin_zero, sizeof (bound_addr.sin_zero));
-
- // Create UDP-Server (default port: 5060)
- // Use here either the local information or the published address
- if (!account->getPublishedSameasLocal ()) {
- listeningAddress = account->getPublishedAddress ();
- listeningPort = account->getPublishedPort ();
- _debug ("UserAgent: Creating UDP transport published %s:%i", listeningAddress.c_str (), listeningPort);
+ if (!account->getPublishedSameasLocal()) {
+ listeningAddress = account->getPublishedAddress();
+ listeningPort = account->getPublishedPort();
}
- // We must specify this here to avoid the IP2IP_PROFILE
- // to create a transport with name 0.0.0.0 to appear in the via header
- if (local)
+ // We must specify this here to avoid the IP2IP_PROFILE to create a transport with name 0.0.0.0 to appear in the via header
+ if (account->getAccountID() == IP2IP_PROFILE)
listeningAddress = loadSIPLocalIP();
if (listeningAddress.empty() || listeningPort == 0)
- return NULL;
+ return;
+
+ const pjsip_host_port a_name = {
+ pj_str((char*)listeningAddress.c_str()),
+ listeningPort
+ };
+
+ pjsip_udp_transport_start(_endpt, &bound_addr, &a_name, 1, &account->transport);
+ pjsip_tpmgr_dump_transports(pjsip_endpt_get_tpmgr(_endpt)); // dump debug information to stdout
+
+ if (account->transport)
+ transportMap_[account->getLocalPort()] = account->transport;
+
+}
+
+pjsip_tpselector *SIPVoIPLink::initTransportSelector (pjsip_transport *transport, pj_pool_t *tp_pool)
+{
+ assert(transport);
+ pjsip_tpselector *tp = (pjsip_tpselector *) pj_pool_zalloc (tp_pool, sizeof (pjsip_tpselector));
+ tp->type = PJSIP_TPSELECTOR_TRANSPORT;
+ tp->u.transport = transport;
+ return tp;
+}
+
+
+
+void SIPVoIPLink::createStunTransport (SIPAccount *account)
+{
+ account->transport = NULL;
+ pj_str_t stunServer = account->getStunServerName ();
+ pj_uint16_t stunPort = account->getStunPort ();
+
+ if (stunServerResolve(account) != PJ_SUCCESS) {
+ _error("Can't resolve STUN server");
+ return;
+ }
+
+ pj_sock_t sock = PJ_INVALID_SOCKET;
+
+ pj_sockaddr_in boundAddr;
+ if (pj_sockaddr_in_init (&boundAddr, &stunServer, 0) != PJ_SUCCESS) {
+ _error("Can't initialize IPv4 socket on %*s:%i", stunServer.slen, stunServer.ptr, stunPort);
+ return;
+ }
- pjsip_host_port a_name;
- pj_bzero (&a_name, sizeof (pjsip_host_port));
- pj_cstr (&a_name.host, listeningAddress.c_str());
- a_name.port = listeningPort;
+ if (pj_sock_socket (pj_AF_INET(), pj_SOCK_DGRAM(), 0, &sock) != PJ_SUCCESS) {
+ _error("Can't create or bind socket");
+ return;
+ }
- pjsip_transport *transport;
- if (pjsip_udp_transport_start (_endpt, &bound_addr, &a_name, 1, &transport) != PJ_SUCCESS)
- transport = NULL;
+ // Query the mapped IP address and port on the 'outside' of the NAT
+ pj_sockaddr_in pub_addr;
+ if (pjstun_get_mapped_addr (&_cp->factory, 1, &sock, &stunServer, stunPort, &stunServer, stunPort, &pub_addr) != PJ_SUCCESS) {
+ _error("Can't contact STUN server");
+ pj_sock_close (sock);
+ return;
+ }
- pjsip_tpmgr * tpmgr = pjsip_endpt_get_tpmgr (_endpt);
- pjsip_tpmgr_dump_transports (tpmgr);
+ pjsip_host_port a_name = {
+ pj_str (pj_inet_ntoa (pub_addr.sin_addr)),
+ pj_ntohs (pub_addr.sin_port)
+ };
- return transport;
+ std::string listeningAddress = std::string (a_name.host.ptr, a_name.host.slen);
+
+ account->setPublishedAddress (listeningAddress);
+ account->setPublishedPort (a_name.port);
+
+ pjsip_udp_transport_attach2 (_endpt, PJSIP_TRANSPORT_UDP, sock, &a_name, 1, &account->transport);
+
+ pjsip_tpmgr_dump_transports (pjsip_endpt_get_tpmgr (_endpt));
+}
+
+
+void SIPVoIPLink::shutdownSipTransport (SIPAccount *account)
+{
+ if (account->transport) {
+ pjsip_transport_dec_ref(account->transport);
+ account->transport = NULL;
+ }
}
+
void SIPVoIPLink::findLocalAddressFromUri (const std::string& uri, pjsip_transport *transport, std::string& addr, std::string &port)
{
std::stringstream ss;
@@ -1225,132 +1230,6 @@ void SIPVoIPLink::findLocalAddressFromUri (const std::string& uri, pjsip_transpo
}
-pjsip_tpselector *SIPVoIPLink::initTransportSelector (pjsip_transport *transport, pj_pool_t *tp_pool)
-{
- assert(transport);
- pjsip_tpselector *tp = (pjsip_tpselector *) pj_pool_zalloc (tp_pool, sizeof (pjsip_tpselector));
- tp->type = PJSIP_TPSELECTOR_TRANSPORT;
- tp->u.transport = transport;
- return tp;
-}
-
-
-
-pj_status_t SIPVoIPLink::createTlsTransport (SIPAccount *account, std::string remoteAddr)
-{
- pj_str_t remote;
-
- pj_cstr (&remote, remoteAddr.c_str());
-
- pj_sockaddr_in rem_addr;
- pj_sockaddr_in_init (&rem_addr, &remote, (pj_uint16_t) 5061);
-
- // Update TLS settings for account registration using the default listeners
- // Pjsip does not allow to create multiple listener
- // pjsip_tpmgr *mgr = pjsip_endpt_get_tpmgr(_endpt);
- // pjsip_tls_listener_update_settings(_endpt, _pool, mgr, _localTlsListener, account->getTlsSetting());
-
- // Create a new TLS connection from TLS listener
- pjsip_transport *tls;
- pj_status_t success = pjsip_endpt_acquire_transport (_endpt, PJSIP_TRANSPORT_TLS, &rem_addr, sizeof (rem_addr), NULL, &tls);
-
- if (success != PJ_SUCCESS)
- _debug ("UserAgent: Error could not create TLS transport");
- else
- account->transport = tls;
-
- return success;
-}
-
-pj_status_t SIPVoIPLink::createAlternateUdpTransport (SIPAccount *account)
-{
- pj_str_t stunServer = account->getStunServerName ();
- pj_uint16_t stunPort = account->getStunPort ();
-
- pj_status_t status = stunServerResolve (account);
- if (status != PJ_SUCCESS) {
- _error ("UserAgent: Error: Resolving STUN server: %i", status);
- return status;
- }
-
- pj_sock_t sock = PJ_INVALID_SOCKET;
-
- _debug ("UserAgent: Initializing IPv4 socket on %s:%i", stunServer.ptr, stunPort);
- pj_sockaddr_in boundAddr;
- status = pj_sockaddr_in_init (&boundAddr, &stunServer, 0);
- if (status != PJ_SUCCESS) {
- _debug ("UserAgent: Error: Initializing IPv4 socket on %s:%i", stunServer.ptr, stunPort);
- return status;
- }
-
- // Create and bind the socket
- status = pj_sock_socket (pj_AF_INET(), pj_SOCK_DGRAM(), 0, &sock);
- if (status != PJ_SUCCESS) {
- _debug ("UserAgent: Error: Unable to create or bind socket (%d)", status);
- return status;
- }
-
- // Query the mapped IP address and port on the 'outside' of the NAT
- pj_sockaddr_in pub_addr;
- status = pjstun_get_mapped_addr (&_cp->factory, 1, &sock, &stunServer, stunPort, &stunServer, stunPort, &pub_addr);
- if (status != PJ_SUCCESS) {
- _debug ("UserAgwent: Error: Contacting STUN server (%d)", status);
- pj_sock_close (sock);
- return status;
- }
-
- _debug ("UserAgent: Firewall address : %s:%d", pj_inet_ntoa (pub_addr.sin_addr), pj_ntohs (pub_addr.sin_port));
-
- pjsip_host_port a_name;
- a_name.host = pj_str (pj_inet_ntoa (pub_addr.sin_addr));
- a_name.port = pj_ntohs (pub_addr.sin_port);
-
- std::string listeningAddress = std::string (a_name.host.ptr, a_name.host.slen);
- int listeningPort = a_name.port;
-
- // Set the address to be used in SDP
- account->setPublishedAddress (listeningAddress);
- account->setPublishedPort (listeningPort);
-
- // Create the UDP transport
- pjsip_transport *transport;
- status = pjsip_udp_transport_attach2 (_endpt, PJSIP_TRANSPORT_UDP, sock, &a_name, 1, &transport);
- if (status != PJ_SUCCESS) {
- _debug ("UserAgent: Error: Creating alternate SIP UDP listener (%d)", status);
- return status;
- }
-
- account->transport = transport;
- if (transport) {
- _debug ("UserAgent: Initial ref count: %s %s (refcnt=%i)", transport->obj_name, transport->info,
- (int) pj_atomic_get (transport->ref_cnt));
-
- pj_sockaddr *addr = (pj_sockaddr*) & (transport->key.rem_addr);
-
- static char str[PJ_INET6_ADDRSTRLEN];
- pj_inet_ntop ( ( (const pj_sockaddr*) addr)->addr.sa_family, pj_sockaddr_get_addr (addr), str, sizeof (str));
-
- _debug ("UserAgent: KEY: %s:%d",str, pj_sockaddr_get_port ( (const pj_sockaddr*) & (transport->key.rem_addr)));
-
- }
-
- pjsip_tpmgr_dump_transports (pjsip_endpt_get_tpmgr (_endpt));
-
- return PJ_SUCCESS;
-}
-
-
-void SIPVoIPLink::shutdownSipTransport (SIPAccount *account)
-{
- _debug ("UserAgent: Shutdown Sip Transport");
-
- pjsip_transport *tr = account->transport;
- if (tr) {
- pjsip_transport_dec_ref(tr);
- account->transport = NULL;
- }
-}
-
namespace {
std::string parseDisplayName(const char * buffer)
{
@@ -1382,40 +1261,7 @@ void stripSipUriPrefix(std::string& sipUri)
if (found != std::string::npos)
sipUri.erase(found);
}
-} // end anonymous namespace
-
-pjsip_route_hdr *SIPVoIPLink::createRouteSet(Account *account, pj_pool_t *hdr_pool)
-{
- SIPAccount *sipaccount = dynamic_cast<SIPAccount *>(account);
- std::string route = sipaccount->getServiceRoute();
- _debug ("UserAgent: Set Service-Route with %s", route.c_str());
- size_t found = route.find(":");
- std::string port;
- std::string host;
- if (found != std::string::npos) {
- host = route.substr(0, found);
- port = route.substr(found + 1, route.length());
- }
- else {
- host = route;
- port = "0";
- }
-
- pjsip_route_hdr *route_set = pjsip_route_hdr_create (hdr_pool);
- pjsip_route_hdr *routing = pjsip_route_hdr_create (hdr_pool);
- pjsip_sip_uri *url = pjsip_sip_uri_create (hdr_pool, 0);
- routing->name_addr.uri = (pjsip_uri*) url;
- pj_strdup2 (hdr_pool, &url->host, host.c_str());
- url->port = atoi(port.c_str());
-
- pj_list_push_back (route_set, pjsip_hdr_clone (hdr_pool, routing));
-
- return route_set;
-
-}
-
-namespace {
int SIPSessionReinvite (SIPCall *call)
{
pjsip_tx_data *tdata;
@@ -1441,13 +1287,11 @@ void invite_session_state_changed_cb (pjsip_inv_session *inv, pjsip_event *e)
int statusCode = tsx ? tsx->status_code : 404;
if (statusCode) {
const pj_str_t * description = pjsip_get_status_text (statusCode);
- // test wether or not dbus manager is instantiated, if not no need to notify the client
Manager::instance().getDbusManager()->getCallManager()->sipCallStateChanged (call->getCallId(), std::string (description->ptr, description->slen), statusCode);
}
}
if (inv->state == PJSIP_INV_STATE_EARLY and e->body.tsx_state.tsx->role == PJSIP_ROLE_UAC) {
- // The call is ringing - We need to handle this case only on outgoing call
call->setConnectionState (Call::Ringing);
Manager::instance().peerRingingCall (call->getCallId());
} else if (inv->state == PJSIP_INV_STATE_CONFIRMED) {
@@ -1479,13 +1323,10 @@ void invite_session_state_changed_cb (pjsip_inv_session *inv, pjsip_event *e)
break;
}
}
-
}
void sdp_request_offer_cb (pjsip_inv_session *inv, const pjmedia_sdp_session *offer)
{
- _debug ("UserAgent: %s (%d): on_rx_offer REINVITE", __FILE__, __LINE__);
-
SIPCall *call = (SIPCall*) inv->mod_data[_mod_ua.id ];
if (!call)
return;
@@ -1571,12 +1412,8 @@ void sdp_media_update_cb (pjsip_inv_session *inv, pj_status_t status)
// Update internal field for
sdpSession->setMediaTransportInfoFromRemoteSdp();
- try {
- call->getAudioRtp()->updateDestinationIpAddress();
- call->getAudioRtp()->setDtmfPayloadType(sdpSession->getTelephoneEventType());
- } catch (...) {
-
- }
+ call->getAudioRtp()->updateDestinationIpAddress();
+ call->getAudioRtp()->setDtmfPayloadType(sdpSession->getTelephoneEventType());
// Get the crypto attribute containing srtp's cryptographic context (keys, cipher)
CryptoOffer crypto_offer;
diff --git a/daemon/src/sip/sipvoiplink.h b/daemon/src/sip/sipvoiplink.h
index 57cf84de38e8dee8ee7e448116ececa0d30296d3..4b64720bd596b6e88c8610b0432796b22e362e87 100644
--- a/daemon/src/sip/sipvoiplink.h
+++ b/daemon/src/sip/sipvoiplink.h
@@ -257,16 +257,7 @@ class SIPVoIPLink : public VoIPLink
pjsip_tpselector *initTransportSelector (pjsip_transport *, pj_pool_t *);
/**
- * Helper function for creating a route set from information
- * stored in configuration file.
- */
- pjsip_route_hdr *createRouteSet(Account *account, pj_pool_t *pool);
-
- /**
- * This function unset the transport for a given account. It tests wether the
- * associated transport is used by other accounts. If not, it shutdown the transport
- * putting its reference counter to zero. PJSIP assumes transport destruction since
- * this action can be delayed by ongoing SIP transactions.
+ * This function unset the transport for a given account.
*/
void shutdownSipTransport (SIPAccount *account);
@@ -285,11 +276,6 @@ class SIPVoIPLink : public VoIPLink
*/
void createDefaultSipUdpTransport();
- /**
- * Create the default TLS litener using IP2IP_PROFILE settings
- */
- void createDefaultSipTlsListener();
-
private:
/**
* Start a SIP Call
@@ -313,49 +299,35 @@ class SIPVoIPLink : public VoIPLink
*/
pj_status_t stunServerResolve (SIPAccount *);
-
- /**
- * Function used to create a new sip transport or get an existing one from the map.
- * The SIP transport is "acquired" according to account's current settings.
- * This function should be called before registering an account
- * @param account An account for which transport is to be set
- *
- */
- void acquireTransport (SIPAccount *account);
-
/**
- * Create the default TLS litener according to account settings.
+ * Create the default TLS listener.
*/
- void createTlsListener (SIPAccount*);
+ void createTlsListener (SIPAccount*, pjsip_tpfactory **listener);
/**
* General Sip transport creation method according to the
* transport type specified in account settings
* @param account The account for which a transport must be created.
*/
- bool createSipTransport (SIPAccount *account);
+ void createSipTransport (SIPAccount *account);
/**
* Create SIP UDP transport from account's setting
* @param account The account for which a transport must be created.
- * @param local True if the account is IP2IP
- * @return the transport
*/
- pjsip_transport * createUdpTransport (SIPAccount *account, bool local);
+ void createUdpTransport (SIPAccount *account);
/**
* Create a TLS transport from the default TLS listener from
* @param account The account for which a transport must be created.
- * @return pj_status_t PJ_SUCCESS on success
*/
- pj_status_t createTlsTransport (SIPAccount *, std::string remoteAddr);
+ void createTlsTransport (SIPAccount *, std::string remoteAddr);
/**
* Create a UDP transport using stun server to resove public address
* @param account The account for which a transport must be created.
- * @return pj_status_t PJ_SUCCESS on success
*/
- pj_status_t createAlternateUdpTransport (SIPAccount *account);
+ void createStunTransport (SIPAccount *account);
/**
* Get the correct address to use (ie advertised) from