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[#1722] Acquire TLS transport for Ip to IP calls

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This commit is contained in:
Alexandre Savard
2010-01-20 13:15:49 -05:00
parent 8485cc8847
commit 138b000173
1 changed files with 29 additions and 8 deletions
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37
sflphone-common/src/sip/sipvoiplink.cpp
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@ -1562,9 +1562,9 @@ bool SIPVoIPLink::new_ip_to_ip_call (const CallID& id, const std::string& to)
addrSdp = localAddress;
}
// Set local address for RTP media
setCallAudioLocal (call, localAddress);
_debug ("toUri received in new_ip_to_ip call %s", to.c_str());
std::string toUri = account->getToUri (to);
call->setPeerNumber (toUri);
_debug ("toUri in new_ip_to_ip call %s", toUri.c_str());
@ -1582,7 +1582,28 @@ bool SIPVoIPLink::new_ip_to_ip_call (const CallID& id, const std::string& to)
call->getLocalSDP()->set_ip_address (addrSdp);
call->getLocalSDP()->create_initial_offer();
// If no account already set, use the default one created at pjsip initialization
// Init TLS transport if enabled
if(account->isTlsEnabled()) {
_debug("TLS enabled for ip-to-ip calls, acquire TLS transport from pjsip's manager");
pj_sockaddr_in rem_addr;
pj_str_t remote;
int at = toUri.find("@");
std::string remotestr = toUri.substr(at+1, toUri.size()-1);
pj_cstr(&remote, remotestr.c_str());
pj_sockaddr_in_init(&rem_addr, &remote, (pj_uint16_t)5060);
pjsip_transport *tcp;
pjsip_endpt_acquire_transport(_endpt, PJSIP_TRANSPORT_TLS, &rem_addr, sizeof(rem_addr),
NULL, &tcp);
account->setAccountTransport(tcp);
}
// If no transport already set, use the default one created at pjsip initialization
if (account->getAccountTransport() == NULL) {
_debug ("No transport for this account, using the default one");
account->setAccountTransport (_localUDPTransport);
@ -1604,11 +1625,8 @@ bool SIPVoIPLink::new_ip_to_ip_call (const CallID& id, const std::string& to)
int port = findLocalPortFromUri (toUri, account->getAccountTransport());
std::stringstream ss;
std::string portStr;
ss << port;
ss >> portStr;
contactUri = account->getContactHeader (address, portStr);
@ -1880,6 +1898,7 @@ bool SIPVoIPLink::pjsip_init()
}
if (directIpCallsTlsEnabled) {
_debug("*********************Tls IP to IP call enabled Create IT");
errPjsip = createTlsTransportRetryOnFailure (IP2IP_PROFILE);
}
@ -2770,6 +2789,9 @@ pj_status_t SIPVoIPLink::createTlsTransport (AccountID id)
if (status != PJ_SUCCESS) {
_debug ("Error creating SIP TLS listener (%d)", status);
}
else {
}
return PJ_SUCCESS;
}
@ -2970,9 +2992,8 @@ void SIPVoIPLink::handle_reinvite (SIPCall *call)
_debug ("! SIP Failure: Unable to create RTP Session (%s:%d)", __FILE__, __LINE__);
}
*/
_debug("******************************************");
_debug("* handle_reinvite *");
_debug("******************************************");
_debug("Handle reINVITE");
call->getAudioRtp()->updateDestinationIpAddress();
}