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sipvoiplink.cpp
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sipvoiplink.cpp 63.60 KiB
/*
* Copyright (C) 2004, 2005, 2006, 2008, 2009, 2010 Savoir-Faire Linux Inc.
*
* Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
* Author: Yun Liu <yun.liu@savoirfairelinux.com>
* Author: Pierre-Luc Bacon <pierre-luc.bacon@savoirfairelinux.com>
* Author: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
*
* 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 3 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.
*
* Additional permission under GNU GPL version 3 section 7:
*
* If you modify this program, or any covered work, by linking or
* combining it with the OpenSSL project's OpenSSL library (or a
* modified version of that library), containing parts covered by the
* terms of the OpenSSL or SSLeay licenses, Savoir-Faire Linux Inc.
* grants you additional permission to convey the resulting work.
* Corresponding Source for a non-source form of such a combination
* shall include the source code for the parts of OpenSSL used as well
* as that of the covered work.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "sipvoiplink.h"
#include "manager.h"
#include "sip/sdp.h"
#include "sipcall.h"
#include "sipaccount.h"
#include "eventthread.h"
#include "SdesNegotiator.h"
#include "dbus/dbusmanager.h"
#include "dbus/callmanager.h"
#include "hooks/urlhook.h"
#include "im/InstantMessaging.h"
#include "audio/audiolayer.h"
#include "pjsip/sip_endpoint.h"
#include "pjsip/sip_transport_tls.h"
#include "pjsip/sip_uri.h"
#include <pjnath.h>
#include <netinet/in.h>
#include <arpa/nameser.h>
#include <resolv.h>
#include <istream>
#include <utility> // for std::pair
#include <sys/types.h>
#include <sys/socket.h>
#include <sys/ioctl.h>#include <linux/if.h>
#include <map>
using namespace sfl;
namespace {
static pjsip_transport *_localUDPTransport = NULL; /** The default transport (5060) */
/** 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;
/**************** EXTERN VARIABLES AND FUNCTIONS (callbacks) **************************/
/**
* Set audio (SDP) configuration for a call
* localport, localip, localexternalport
* @param call a SIPCall valid pointer
*/
void setCallMediaLocal (SIPCall* call, const std::string &localIP);
/**
* Helper function to parser header from incoming sip messages
*/
std::string fetchHeaderValue (pjsip_msg *msg, std::string field);
static pj_caching_pool pool_cache, *_cp = &pool_cache;
static pj_pool_t *_pool;
static pjsip_endpoint *_endpt;
static pjsip_module _mod_ua;
static pj_thread_t *thread;
static void sdp_media_update_cb (pjsip_inv_session *inv, pj_status_t status UNUSED);
static void sdp_request_offer_cb (pjsip_inv_session *inv, const pjmedia_sdp_session *offer);
static void sdp_create_offer_cb (pjsip_inv_session *inv, pjmedia_sdp_session **p_offer);
static void invite_session_state_changed_cb (pjsip_inv_session *inv, pjsip_event *e);
static void outgoing_request_forked_cb (pjsip_inv_session *inv, pjsip_event *e);
static void transaction_state_changed_cb (pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e);
static void registration_cb (struct pjsip_regc_cbparam *param);
static pj_bool_t transaction_request_cb (pjsip_rx_data *rdata);
static pj_bool_t transaction_response_cb (pjsip_rx_data *rdata UNUSED) ;
static void transfer_client_cb (pjsip_evsub *sub, pjsip_event *event);
/**
* Send a reINVITE inside an active dialog to modify its state
* Local SDP session should be modified before calling this method
* @param sip call
*/
int SIPSessionReinvite(SIPCall *);
/**
* Helper function to process refer function on call transfer
*/
void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata);
std::string loadSIPLocalIP()
{
pj_sockaddr ip_addr;
if (pj_gethostip (pj_AF_INET(), &ip_addr) == PJ_SUCCESS)
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
/*************************************************************************************************/
SIPVoIPLink::SIPVoIPLink() : evThread_(new EventThread(this))
{
#define TRY(ret) do { \
if (ret != PJ_SUCCESS) \
throw VoipLinkException(#ret " failed"); \
} while(0)
srand(time(NULL)); // to get random number for RANDOM_PORT
TRY(pj_init());
TRY(pjlib_util_init());
pj_log_set_level(6); // From 0 (min) to 6 (max)
TRY(pjnath_init());
pj_caching_pool_init (_cp, &pj_pool_factory_default_policy, 0);
_pool = pj_pool_create (&_cp->factory, "sflphone", 4000, 4000, NULL);
if (!_pool)
throw VoipLinkException("UserAgent: Could not initialize memory pool");
TRY(pjsip_endpt_create (&_cp->factory, pj_gethostname()->ptr, &_endpt));
if (loadSIPLocalIP().empty())
throw VoipLinkException("UserAgent: Unable to determine network capabilities");
TRY(pjsip_tsx_layer_init_module(_endpt));
TRY(pjsip_ua_init_module(_endpt, NULL));
TRY(pjsip_replaces_init_module(_endpt)); // See the Replaces specification in RFC 3891
TRY(pjsip_100rel_init_module(_endpt));
// Initialize and register sflphone module
_mod_ua.name = pj_str ( (char*) PACKAGE);
_mod_ua.id = -1;
_mod_ua.priority = PJSIP_MOD_PRIORITY_APPLICATION;
_mod_ua.on_rx_request = &transaction_request_cb;
_mod_ua.on_rx_response = &transaction_response_cb;
TRY(pjsip_endpt_register_module(_endpt, &_mod_ua));
TRY(pjsip_evsub_init_module(_endpt));
TRY(pjsip_xfer_init_module(_endpt));
static const pjsip_inv_callback inv_cb = {
invite_session_state_changed_cb,
outgoing_request_forked_cb,
transaction_state_changed_cb,
sdp_request_offer_cb, sdp_create_offer_cb,
sdp_media_update_cb,
NULL,
NULL,
};
TRY(pjsip_inv_usage_init(_endpt, &inv_cb));
static const pj_str_t allowed[] = { { (char*) "INFO", 4}, { (char*) "REGISTER", 8}, { (char*) "OPTIONS", 7}, { (char*) "MESSAGE", 7 } }; // //{"INVITE", 6}, {"ACK",3}, {"BYE",3}, {"CANCEL",6}
pjsip_endpt_add_capability (_endpt, &_mod_ua, PJSIP_H_ALLOW, NULL, PJ_ARRAY_SIZE (allowed), allowed);
static const pj_str_t text_plain = { (char*) "text/plain", 10 };
pjsip_endpt_add_capability (_endpt, &_mod_ua, PJSIP_H_ACCEPT, NULL, 1, &text_plain);
static const pj_str_t accepted = { (char*) "application/sdp", 15 };
pjsip_endpt_add_capability (_endpt, &_mod_ua, PJSIP_H_ACCEPT, NULL, 1, &accepted);
_debug ("UserAgent: pjsip version %s for %s initialized", pj_get_version(), PJ_OS_NAME);
TRY(pjsip_replaces_init_module(_endpt));
evThread_->start();
}
SIPVoIPLink::~SIPVoIPLink()
{
delete evThread_;
pj_thread_join (thread);
pj_thread_destroy (thread);
const pj_time_val tv = {0, 10};
pjsip_endpt_handle_events(_endpt, &tv);
pjsip_endpt_destroy (_endpt);
pj_pool_release (_pool);
pj_caching_pool_destroy (_cp);
pj_shutdown();
}
SIPVoIPLink* SIPVoIPLink::instance()
{
static SIPVoIPLink* instance = NULL;
if (!instance)
instance = new SIPVoIPLink;
return instance;
}
void SIPVoIPLink::init() {}
void SIPVoIPLink::terminate() {}
void
SIPVoIPLink::getEvent()
{
static pj_thread_desc desc;
// We have to register the external thread so it could access the pjsip frameworks
if (!pj_thread_is_registered())
pj_thread_register(NULL, desc, &thread);
static const pj_time_val timeout = {0, 10};
pjsip_endpt_handle_events(_endpt, &timeout);
}
void SIPVoIPLink::sendRegister (Account *a)
{
SIPAccount *account = static_cast<SIPAccount*>(a);
createSipTransport(account);
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.");
std::string srvUri(account->getServerUri());
std::string address, port;
findLocalAddressFromUri(srvUri, account->transport, address, port);
std::string from(account->getFromUri());
pj_str_t pjFrom = pj_str((char*)from.c_str());
std::string contact(account->getContactHeader(address, port));
pj_str_t pjContact = pj_str((char*)contact.c_str());
pj_str_t pjSrv = pj_str((char*)srvUri.c_str());
if (pjsip_regc_init (regc, &pjSrv, &pjFrom, &pjFrom, 1, &pjContact, account->getRegistrationExpire()) != PJ_SUCCESS)
throw VoipLinkException("Unable to initialize account registration structure");
if (!account->getServiceRoute().empty())
pjsip_regc_set_route_set (regc, createRouteSet(account->getServiceRoute(), _pool));
pjsip_regc_set_credentials (regc, account->getCredentialCount(), account->getCredInfo());
pjsip_hdr hdr_list;
pj_list_init (&hdr_list);
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, &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");
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
pjsip_transport_dec_ref(account->transport);
// pjsip_regc_send increment the transport ref count by one,
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.
pjsip_transport_dec_ref (account->transport);
account->setRegistrationInfo (regc);
}
void SIPVoIPLink::sendUnregister (Account *a)
{
SIPAccount *account = dynamic_cast<SIPAccount *>(a);
// This may occurs if account failed to register and is in state INVALID
if (!account->isRegister()) {
account->setRegistrationState (Unregistered);
return;
}
pjsip_regc *regc = account->getRegistrationInfo();
if (!regc)
throw VoipLinkException("Registration structure is NULL");
pjsip_tx_data *tdata = NULL;
if (pjsip_regc_unregister (regc, &tdata) != PJ_SUCCESS)
throw VoipLinkException("Unable to unregister sip account");
if (pjsip_regc_send (regc, tdata) != PJ_SUCCESS)
throw VoipLinkException("Unable to send request to unregister sip account");
account->setRegister (false);
}
Call *SIPVoIPLink::newOutgoingCall (const std::string& id, const std::string& toUrl)
{
SIPAccount *account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (Manager::instance().getAccountFromCall (id)));
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;
if((toUrl.find("sip:") != std::string::npos) or
toUrl.find("sips:") != std::string::npos)
toUri = toUrl;
else
toUri = account->getToUri(toUrl);
call->setPeerNumber (toUri);
std::string localAddr(getInterfaceAddrFromName (account->getLocalInterface ()));
if (localAddr == "0.0.0.0")
localAddr = loadSIPLocalIP();
setCallMediaLocal (call, localAddr);
// May use the published address as well
std::string addrSdp = account->isStunEnabled() ?
account->getPublishedAddress() :
getInterfaceAddrFromName(account->getLocalInterface());
if (addrSdp == "0.0.0.0")
addrSdp = loadSIPLocalIP();
// Initialize the session using ULAW as default codec in case of early media
// The session should be ready to receive media once the first INVITE is sent, before
// the session initialization is completed
sfl::Codec* audiocodec = Manager::instance().audioCodecFactory.instantiateCodec (PAYLOAD_CODEC_ULAW);
if (audiocodec == NULL) {
delete call;
throw VoipLinkException ("Could not instantiate codec for early media");
}
try {
call->getAudioRtp()->initAudioRtpConfig ();
call->getAudioRtp()->initAudioSymmetricRtpSession ();
call->getAudioRtp()->initLocalCryptoInfo ();
call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
} catch (...) {
delete call;
throw VoipLinkException ("Could not start rtp session for early media");
}
call->initRecFileName(toUrl);
call->getLocalSDP()->setLocalIP (addrSdp);
call->getLocalSDP()->createOffer(account->getActiveCodecs());
if (!SIPStartCall(call)) {
delete call;
throw VoipLinkException("Could not send outgoing INVITE request for new call");
}
return call;
}
void
SIPVoIPLink::answer (Call *c)
{
SIPCall *call = dynamic_cast<SIPCall*>(c);
if (!call)
return;
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)");
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
SIPVoIPLink::hangup (const std::string& id)
{
SIPCall* call = getSIPCall(id);
std::string account_id(Manager::instance().getAccountFromCall(id));
SIPAccount *account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (account_id));
if (account == NULL)
throw VoipLinkException("Could not find account for this call");
pjsip_inv_session *inv = call->inv;
if (inv == NULL)
throw VoipLinkException("No invite session for this call");
// Looks for sip routes
if (not (account->getServiceRoute().empty())) {
pjsip_route_hdr *route_set = createRouteSet(account->getServiceRoute(), inv->pool);
pjsip_dlg_set_route_set (inv->dlg, route_set);
}
pjsip_tx_data *tdata = NULL;
// User hangup current call. Notify peer
if (pjsip_inv_end_session (inv, 404, NULL, &tdata) != PJ_SUCCESS || !tdata)
return;
if (pjsip_inv_send_msg (inv, tdata) != PJ_SUCCESS)
return;
// Make sure user data is NULL in callbacks
inv->mod_data[_mod_ua.id] = NULL;
if (Manager::instance().isCurrentCall (id))
call->getAudioRtp()->stop();
removeCall (id);
}
void
SIPVoIPLink::peerHungup (const std::string& id)
{
SIPCall* call = getSIPCall(id);
// User hangup current call. Notify peer
pjsip_tx_data *tdata = NULL; if (pjsip_inv_end_session (call->inv, 404, NULL, &tdata) != PJ_SUCCESS || !tdata)
return;
if (pjsip_inv_send_msg (call->inv, tdata) != PJ_SUCCESS)
return;
// Make sure user data is NULL in callbacks
call->inv->mod_data[_mod_ua.id ] = NULL;
if (Manager::instance().isCurrentCall (id))
call->getAudioRtp()->stop();
removeCall (id);
}
void
SIPVoIPLink::onhold (const std::string& id)
{
SIPCall *call = getSIPCall(id);
call->setState (Call::Hold);
call->getAudioRtp()->stop();
Sdp *sdpSession = call->getLocalSDP();
if (!sdpSession)
throw VoipLinkException("Could not find sdp session");
sdpSession->removeAttributeFromLocalAudioMedia("sendrecv");
sdpSession->removeAttributeFromLocalAudioMedia("sendonly");
sdpSession->addAttributeToLocalAudioMedia("sendonly");
SIPSessionReinvite (call);
}
void
SIPVoIPLink::offhold (const std::string& id)
{
SIPCall *call = getSIPCall (id);
Sdp *sdpSession = call->getLocalSDP();
if (sdpSession == NULL)
throw VoipLinkException("Could not find sdp session");
try {
int pl = PAYLOAD_CODEC_ULAW;
sfl::Codec *sessionMedia = sdpSession->getSessionMedia();
if (sessionMedia)
pl = sessionMedia->getPayloadType();
// Create a new instance for this codec
sfl::Codec* audiocodec = Manager::instance().audioCodecFactory.instantiateCodec (pl);
if (audiocodec == NULL)
throw VoipLinkException("Could not instantiate codec");
call->getAudioRtp()->initAudioRtpConfig ();
call->getAudioRtp()->initAudioSymmetricRtpSession ();
call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
}
catch (const SdpException &e) {
_error("UserAgent: Exception: %s", e.what());
}
catch (...) {
throw VoipLinkException("Could not create audio rtp session");
}
sdpSession->removeAttributeFromLocalAudioMedia("sendrecv");
sdpSession->removeAttributeFromLocalAudioMedia("sendonly");
sdpSession->addAttributeToLocalAudioMedia("sendrecv");
if (SIPSessionReinvite(call) == PJ_SUCCESS) call->setState (Call::Active);
}
void
SIPVoIPLink::sendTextMessage (sfl::InstantMessaging *module, const std::string& callID, const std::string& message, const std::string& from)
{
SIPCall *call;
try {
call = getSIPCall (callID);
}
catch (const VoipLinkException &e) {
return;
}
/* Send IM message */
sfl::InstantMessaging::UriList list;
sfl::InstantMessaging::UriEntry entry;
entry[sfl::IM_XML_URI] = std::string ("\"" + from + "\""); // add double quotes for xml formating
list.push_front (entry);
module->send_sip_message (call->inv, callID, module->appendUriList (message, list));
}
bool
SIPVoIPLink::transferCommon(SIPCall *call, pj_str_t *dst)
{
pjsip_evsub_user xfer_cb;
pj_bzero (&xfer_cb, sizeof (xfer_cb));
xfer_cb.on_evsub_state = &transfer_client_cb;
pjsip_evsub *sub;
if (pjsip_xfer_create_uac (call->inv->dlg, &xfer_cb, &sub) != PJ_SUCCESS)
return false;
/* Associate this voiplink of call with the client subscription
* We can not just associate call with the client subscription
* because after this function, we can no find the cooresponding
* voiplink from the call any more. But the voiplink is useful!
*/
pjsip_evsub_set_mod_data (sub, _mod_ua.id, this);
/*
* Create REFER request.
*/
pjsip_tx_data *tdata;
if (pjsip_xfer_initiate (sub, dst, &tdata) != PJ_SUCCESS)
return false;
// Put SIP call id in map in order to retrieve call during transfer callback
std::string callidtransfer(call->inv->dlg->call_id->id.ptr, call->inv->dlg->call_id->id.slen);
transferCallID[callidtransfer] = call->getCallId();
/* Send. */
if (pjsip_xfer_send_request (sub, tdata) != PJ_SUCCESS)
return false;
return true;
}
void
SIPVoIPLink::transfer (const std::string& id, const std::string& to)
{
SIPCall *call = getSIPCall(id);
call->stopRecording();
std::string account_id(Manager::instance().getAccountFromCall(id));
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount (account_id));
if (account == NULL) throw VoipLinkException("Could not find account");
std::string toUri;
pj_str_t dst = { NULL, 0 };
if (to.find ("@") == std::string::npos) {
toUri = account->getToUri(to);
pj_cstr (&dst, toUri.c_str());
}
if (!transferCommon(getSIPCall(id), &dst))
throw VoipLinkException("Couldn't transfer");
}
bool SIPVoIPLink::attendedTransfer(const std::string& id, const std::string& to)
{
pjsip_dialog *target_dlg = getSIPCall(to)->inv->dlg;
pjsip_uri *uri = (pjsip_uri*) pjsip_uri_get_uri(target_dlg->remote.info->uri);
char str_dest_buf[PJSIP_MAX_URL_SIZE*2] = { '<' };
pj_str_t dst = { str_dest_buf, 1 };
dst.slen += pjsip_uri_print(PJSIP_URI_IN_REQ_URI, uri, str_dest_buf+1, sizeof(str_dest_buf)-1);
dst.slen += pj_ansi_snprintf(str_dest_buf + dst.slen,
sizeof(str_dest_buf) - dst.slen,
"?"
"Replaces=%.*s"
"%%3Bto-tag%%3D%.*s"
"%%3Bfrom-tag%%3D%.*s>",
(int)target_dlg->call_id->id.slen,
target_dlg->call_id->id.ptr,
(int)target_dlg->remote.info->tag.slen,
target_dlg->remote.info->tag.ptr,
(int)target_dlg->local.info->tag.slen,
target_dlg->local.info->tag.ptr);
return transferCommon(getSIPCall(id), &dst);
}
bool
SIPVoIPLink::refuse (const std::string& id)
{
SIPCall *call = getSIPCall (id);
if (!call->isIncoming() or call->getConnectionState() == Call::Connected)
return false;
call->getAudioRtp()->stop();
pjsip_tx_data *tdata;
if (pjsip_inv_end_session (call->inv, PJSIP_SC_DECLINE, NULL, &tdata) != PJ_SUCCESS)
return false;
if (pjsip_inv_send_msg (call->inv, tdata) != PJ_SUCCESS)
return false;
// Make sure the pointer is NULL in callbacks
call->inv->mod_data[_mod_ua.id] = NULL;
removeCall (id);
return true;
}
std::string
SIPVoIPLink::getCurrentCodecName(Call *call) const
{
return dynamic_cast<SIPCall*>(call)->getLocalSDP()->getCodecName();
}
voidSIPVoIPLink::carryingDTMFdigits (const std::string& id, char code)
{
std::string accountID(Manager::instance().getAccountFromCall(id));
SIPAccount *account = static_cast<SIPAccount*>(Manager::instance().getAccount(accountID));
if (account) try {
dtmfSend(getSIPCall(id), code, account->getDtmfType());
} catch (VoipLinkException) {
// don't do anything if call doesn't exist
}
}
void
SIPVoIPLink::dtmfSend (SIPCall *call, char code, DtmfType dtmf)
{
if (dtmf == OVERRTP) {
call->getAudioRtp()->sendDtmfDigit(code - '0');
return;
}
// else : dtmf == SIPINFO
pj_str_t methodName = pj_str((char*)"INFO");
pjsip_method method;
pjsip_method_init_np (&method, &methodName);
/* Create request message. */
pjsip_tx_data *tdata;
if (pjsip_dlg_create_request (call->inv->dlg, &method, -1, &tdata) != PJ_SUCCESS)
return;
int duration = Manager::instance().voipPreferences.getPulseLength();
char dtmf_body[1000];
snprintf(dtmf_body, sizeof dtmf_body - 1, "Signal=%c\r\nDuration=%d\r\n", code, duration);
/* Create "application/dtmf-relay" message body. */
pj_str_t content = pj_str(dtmf_body);
pj_str_t type = pj_str((char*)"application");
pj_str_t subtype = pj_str((char*)"dtmf-relay");
tdata->msg->body = pjsip_msg_body_create (tdata->pool, &type, &subtype, &content);
if (tdata->msg->body == NULL)
pjsip_tx_data_dec_ref (tdata);
else
pjsip_dlg_send_request (call->inv->dlg, tdata, _mod_ua.id, NULL);
}
bool
SIPVoIPLink::SIPStartCall(SIPCall *call)
{
std::string id(Manager::instance().getAccountFromCall(call->getCallId()));
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(id));
if (account == NULL)
return false;
std::string toUri(call->getPeerNumber()); // expecting a fully well formed sip uri
std::string address, port;
findLocalAddressFromUri(toUri, account->transport, address, port);
std::string from(account->getFromUri());
pj_str_t pjFrom = pj_str((char*)from.c_str());
std::string contact(account->getContactHeader(address, port));
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;
if (pjsip_inv_create_uac(dialog, call->getLocalSDP()->getLocalSdpSession(), 0, &call->inv) != PJ_SUCCESS)
return false;
if (not account->getServiceRoute().empty())
pjsip_dlg_set_route_set(dialog, createRouteSet(account->getServiceRoute(), call->inv->pool));
pjsip_auth_clt_set_credentials (&dialog->auth_sess, account->getCredentialCount(), account->getCredInfo());
call->inv->mod_data[_mod_ua.id] = call;
pjsip_tx_data *tdata;
if (pjsip_inv_invite(call->inv, &tdata) != PJ_SUCCESS)
return false;
pjsip_tpselector *tp = initTransportSelector(account->transport, call->inv->pool);
if (pjsip_dlg_set_transport (dialog, tp) != PJ_SUCCESS)
return false;
if (pjsip_inv_send_msg(call->inv, tdata) != PJ_SUCCESS)
return false;
call->setConnectionState(Call::Progressing);
call->setState(Call::Active);
addCall(call);
return true;
}
void
SIPVoIPLink::SIPCallServerFailure (SIPCall *call)
{
std::string id(call->getCallId());
Manager::instance().callFailure(id);
removeCall(id);
}
void
SIPVoIPLink::SIPCallClosed (SIPCall *call)
{
std::string id(call->getCallId());
if (Manager::instance().isCurrentCall(id))
call->getAudioRtp()->stop();
Manager::instance().peerHungupCall(id);
removeCall(id);
}
void
SIPVoIPLink::SIPCallAnswered (SIPCall *call, pjsip_rx_data *rdata UNUSED)
{
if (call->getConnectionState() != Call::Connected) {
call->setConnectionState (Call::Connected);
call->setState (Call::Active);
Manager::instance().peerAnsweredCall (call->getCallId());
}
}
SIPCall*
SIPVoIPLink::getSIPCall (const std::string& id)
{
SIPCall *result = dynamic_cast<SIPCall*> (getCall (id));
if (result == NULL)
throw VoipLinkException("Could not find SIPCall " + id);
return result;
}
bool SIPVoIPLink::SIPNewIpToIpCall (const std::string& id, const std::string& to)
{
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(IP2IP_PROFILE));
if (!account)
return false;
SIPCall *call = new SIPCall(id, Call::Outgoing, _cp);
call->setCallConfiguration(Call::IPtoIP);
call->initRecFileName(to);
std::string localAddress(getInterfaceAddrFromName (account->getLocalInterface ()));
if (localAddress == "0.0.0.0")
localAddress = loadSIPLocalIP();
setCallMediaLocal (call, localAddress);
std::string toUri = account->getToUri (to);
call->setPeerNumber (toUri);
sfl::Codec* audiocodec = Manager::instance().audioCodecFactory.instantiateCodec (PAYLOAD_CODEC_ULAW);
// Audio Rtp Session must be initialized before creating initial offer in SDP session
// since SDES require crypto attribute.
call->getAudioRtp()->initAudioRtpConfig ();
call->getAudioRtp()->initAudioSymmetricRtpSession ();
call->getAudioRtp()->initLocalCryptoInfo ();
call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
// Building the local SDP offer
call->getLocalSDP()->setLocalIP (localAddress);
call->getLocalSDP()->createOffer (account->getActiveCodecs ());
// Init TLS transport if enabled
if (account->isTlsEnabled()) {
int at = toUri.find ("@");
int trns = toUri.find (";transport");
std::string remoteAddr = toUri.substr (at+1, trns-at-1);
if (toUri.find ("sips:") != 1) {
_debug ("UserAgent: Error \"sips\" scheme required for TLS call");
delete call;
return false;
}
shutdownSipTransport(account);
createTlsTransport(account, remoteAddr);
if (!account->transport) {
delete call;
return false;
}
}
if (!SIPStartCall(call)) {
delete call;
return false;
}
return true;
}
///////////////////////////////////////////////////////////////////////////////
// Private functions
///////////////////////////////////////////////////////////////////////////////
static pj_bool_t stun_sock_on_status_cb (pj_stun_sock *stun_sock UNUSED, pj_stun_sock_op op UNUSED, pj_status_t status)
{
return status == PJ_SUCCESS;
}
static pj_bool_t stun_sock_on_rx_data_cb (pj_stun_sock *stun_sock UNUSED, void *pkt UNUSED, unsigned pkt_len UNUSED, const pj_sockaddr_t *src_addr UNUSED, unsigned addr_len UNUSED)
{
return PJ_TRUE;
}
pj_status_t SIPVoIPLink::stunServerResolve (SIPAccount *account)
{
pj_stun_config stunCfg;
pj_stun_config_init (&stunCfg, &_cp->factory, 0, pjsip_endpt_get_ioqueue (_endpt), pjsip_endpt_get_timer_heap (_endpt));
static const pj_stun_sock_cb stun_sock_cb = {
stun_sock_on_rx_data_cb,
NULL,
stun_sock_on_status_cb
};
pj_stun_sock *stun_sock;
pj_status_t status = pj_stun_sock_create (&stunCfg, "stunresolve", pj_AF_INET(), &stun_sock_cb, NULL, NULL, &stun_sock);
pj_str_t stunServer = account->getStunServerName();
if (status != PJ_SUCCESS) {
char errmsg[PJ_ERR_MSG_SIZE];
pj_strerror (status, errmsg, sizeof (errmsg));
_debug ("Error creating STUN socket for %.*s: %s", (int) stunServer.slen, stunServer.ptr, errmsg);
return status;
}
status = pj_stun_sock_start (stun_sock, &stunServer, account->getStunPort (), NULL);
if (status != PJ_SUCCESS) {
char errmsg[PJ_ERR_MSG_SIZE];
pj_strerror (status, errmsg, sizeof (errmsg));
_debug ("Error starting STUN socket for %.*s: %s", (int) stunServer.slen, stunServer.ptr, errmsg);
pj_stun_sock_destroy (stun_sock);
}
return status;
}
void SIPVoIPLink::createDefaultSipUdpTransport()
{
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(IP2IP_PROFILE));
createUdpTransport(account);
assert(account->transport);
_localUDPTransport = account->transport;
}
void SIPVoIPLink::createTlsListener (SIPAccount *account, pjsip_tpfactory **listener)
{
pj_sockaddr_in local_addr;
pj_sockaddr_in_init (&local_addr, 0, 0);
local_addr.sin_port = pj_htons(account->getTlsListenerPort());
pj_str_t pjAddress;
pj_cstr (&pjAddress, PJ_INADDR_ANY);
pj_sockaddr_in_set_str_addr (&local_addr, &pjAddress);
pjsip_host_port a_name = {
pj_str((char*)loadSIPLocalIP().c_str()),
local_addr.sin_port
};
pjsip_tls_transport_start(_endpt, account->getTlsSetting(), &local_addr, &a_name, 1, listener);
}
void 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);
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)
{
shutdownSipTransport(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));
createTlsTransport(account, remoteAddr);
} else if (account->isStunEnabled ())
createStunTransport(account);
else
createUdpTransport(account);
if (!account->transport) {
// 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::createUdpTransport (SIPAccount *account)
{
std::string listeningAddress;
pj_uint16_t listeningPort = account->getLocalPort();
pj_sockaddr_in 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") {
listeningAddress = loadSIPLocalIP ();
bound_addr.sin_addr.s_addr = pj_htonl (PJ_INADDR_ANY);
} else {
listeningAddress = getInterfaceAddrFromName(account->getLocalInterface());
bound_addr.sin_addr = pj_inet_addr2(listeningAddress.c_str());
}
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 (account->getAccountID() == IP2IP_PROFILE)
listeningAddress = loadSIPLocalIP();
if (listeningAddress.empty() || listeningPort == 0)
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)
{
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;
}
if (pj_sock_socket (pj_AF_INET(), pj_SOCK_DGRAM(), 0, &sock) != PJ_SUCCESS) {
_error("Can't create or bind socket");
return;
}
// 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_host_port a_name = {
pj_str (pj_inet_ntoa (pub_addr.sin_addr)),
pj_ntohs (pub_addr.sin_port)
};
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;
ss << DEFAULT_SIP_PORT;
port = ss.str();
pjsip_uri *genericUri = pjsip_parse_uri(_pool, (char*)uri.data(), uri.size(), 0);
const pj_str_t *pjMachineName = pj_gethostname();
addr = std::string(pjMachineName->ptr, pjMachineName->slen);
if (genericUri == NULL)
return;
pjsip_sip_uri *sip_uri = (pjsip_sip_uri*) pjsip_uri_get_uri (genericUri);
if (sip_uri == NULL)
return;
pjsip_transport_type_e transportType;
if (PJSIP_URI_SCHEME_IS_SIPS (sip_uri)) {
transportType = PJSIP_TRANSPORT_TLS;
ss.str("");
ss << DEFAULT_SIP_TLS_PORT;
port = ss.str();
} else {
if (transport == NULL)
transport = _localUDPTransport;
transportType = PJSIP_TRANSPORT_UDP;
}
pjsip_tpmgr *tpmgr = pjsip_endpt_get_tpmgr(_endpt);
if (!tpmgr)
return;
pjsip_tpselector *tp_sel = NULL;
if (transportType == PJSIP_TRANSPORT_UDP && transport)
tp_sel = initTransportSelector (transport, _pool);
pj_str_t localAddress;
int i_port;
if (pjsip_tpmgr_find_local_addr(tpmgr, _pool, transportType, tp_sel, &localAddress, &i_port) != PJ_SUCCESS)
return;
addr = std::string(localAddress.ptr, localAddress.slen);
if (addr == "0.0.0.0")
addr = loadSIPLocalIP();
ss.str("");
ss << i_port;
port = ss.str();
}
namespace {
std::string parseDisplayName(const char * buffer)
{ // Parse the display name from "From" header
const char* from_header = strstr (buffer, "From: ");
if (!from_header)
return "";
std::string temp(from_header);
size_t begin_displayName = temp.find ("\"") + 1;
size_t end_displayName = temp.rfind ("\"");
std::string displayName(temp.substr(begin_displayName, end_displayName - begin_displayName));
if (displayName.size() > 25)
return "";
return displayName;
}
void stripSipUriPrefix(std::string& sipUri)
{
//Remove sip: prefix
size_t found = sipUri.find("sip:");
if (found != std::string::npos)
sipUri.erase (found, found + 4);
found = sipUri.find ("@");
if (found != std::string::npos)
sipUri.erase(found);
}
int SIPSessionReinvite (SIPCall *call)
{
pjsip_tx_data *tdata;
pjmedia_sdp_session *local_sdp = call->getLocalSDP()->getLocalSdpSession();
if (local_sdp && pjsip_inv_reinvite(call->inv, NULL, local_sdp, &tdata) == PJ_SUCCESS)
return pjsip_inv_send_msg (call->inv, tdata);
return !PJ_SUCCESS;
}
void invite_session_state_changed_cb (pjsip_inv_session *inv, pjsip_event *e)
{
SIPCall *call = reinterpret_cast<SIPCall*> (inv->mod_data[_mod_ua.id]);
if (call == NULL)
return;
SIPVoIPLink *link = SIPVoIPLink::instance();
if (inv->state != PJSIP_INV_STATE_CONFIRMED) {
// Update UI with the current status code and description
pjsip_transaction * tsx = e->body.tsx_state.tsx;
int statusCode = tsx ? tsx->status_code : 404;
if (statusCode) {
const pj_str_t * description = pjsip_get_status_text (statusCode);
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) {
call->setConnectionState (Call::Ringing);
Manager::instance().peerRingingCall (call->getCallId());
} else if (inv->state == PJSIP_INV_STATE_CONFIRMED) {
// After we sent or received a ACK - The connection is established
link->SIPCallAnswered (call, e->body.tsx_state.src.rdata);
} else if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
std::string accId(Manager::instance().getAccountFromCall (call->getCallId()));
switch (inv->cause) {
// The call terminates normally - BYE / CANCEL
case PJSIP_SC_OK: case PJSIP_SC_REQUEST_TERMINATED:
link->SIPCallClosed (call);
break;
case PJSIP_SC_DECLINE:
if (inv->role != PJSIP_ROLE_UAC)
break;
case PJSIP_SC_NOT_FOUND:
case PJSIP_SC_REQUEST_TIMEOUT:
case PJSIP_SC_NOT_ACCEPTABLE_HERE: /* no compatible codecs */
case PJSIP_SC_NOT_ACCEPTABLE_ANYWHERE:
case PJSIP_SC_UNSUPPORTED_MEDIA_TYPE:
case PJSIP_SC_UNAUTHORIZED:
case PJSIP_SC_FORBIDDEN:
case PJSIP_SC_REQUEST_PENDING:
case PJSIP_SC_ADDRESS_INCOMPLETE:
default:
link->SIPCallServerFailure (call);
break;
}
}
}
void sdp_request_offer_cb (pjsip_inv_session *inv, const pjmedia_sdp_session *offer)
{
SIPCall *call = (SIPCall*) inv->mod_data[_mod_ua.id ];
if (!call)
return;
std::string accId(Manager::instance().getAccountFromCall(call->getCallId()));
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(accId));
call->getLocalSDP()->receiveOffer (offer, account->getActiveCodecs());
call->getLocalSDP()->startNegotiation();
pjsip_inv_set_sdp_answer (call->inv, call->getLocalSDP()->getLocalSdpSession());
}
static void sdp_create_offer_cb (pjsip_inv_session *inv, pjmedia_sdp_session **p_offer)
{
SIPCall *call = reinterpret_cast<SIPCall*>(inv->mod_data[_mod_ua.id]);
std::string accountid(Manager::instance().getAccountFromCall(call->getCallId()));
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(accountid));
std::string localAddress(SIPVoIPLink::instance()->getInterfaceAddrFromName(account->getLocalInterface ()));
std::string addrSdp(localAddress);
if (localAddress == "0.0.0.0")
localAddress = loadSIPLocalIP();
if (addrSdp == "0.0.0.0")
addrSdp = localAddress;
setCallMediaLocal (call, localAddress);
call->getLocalSDP()->setLocalIP(addrSdp);
call->getLocalSDP()->createOffer(account->getActiveCodecs());
*p_offer = call->getLocalSDP()->getLocalSdpSession();
}
// This callback is called after SDP offer/answer session has completed.
void sdp_media_update_cb (pjsip_inv_session *inv, pj_status_t status)
{
const pjmedia_sdp_session *remote_sdp;
const pjmedia_sdp_session *local_sdp;
SIPCall *call = reinterpret_cast<SIPCall *> (inv->mod_data[_mod_ua.id]);
if (call == NULL) {
_debug ("UserAgent: Call declined by peer, SDP negotiation stopped"); return;
}
if (status != PJ_SUCCESS) {
_warn ("UserAgent: Error: while negotiating the offer");
SIPVoIPLink::instance()->hangup (call->getCallId());
Manager::instance().callFailure (call->getCallId());
return;
}
if (!inv->neg) {
_warn ("UserAgent: Error: no negotiator for this session");
return;
}
// Retreive SDP session for this call
Sdp *sdpSession = call->getLocalSDP();
// Get active session sessions
pjmedia_sdp_neg_get_active_remote (inv->neg, &remote_sdp);
pjmedia_sdp_neg_get_active_local (inv->neg, &local_sdp);
// Print SDP session
char buffer[1000];
memset(buffer, 0, sizeof buffer);
pjmedia_sdp_print(remote_sdp, buffer, 1000);
_debug("SDP: Remote active SDP Session:\n%s", buffer);
memset(buffer, 0, 1000);
pjmedia_sdp_print(local_sdp, buffer, 1000);
_debug("SDP: Local active SDP Session:\n%s", buffer);
// Set active SDP sessions
sdpSession->setActiveRemoteSdpSession(remote_sdp);
sdpSession->setActiveLocalSdpSession(local_sdp);
// Update internal field for
sdpSession->setMediaTransportInfoFromRemoteSdp();
call->getAudioRtp()->updateDestinationIpAddress();
call->getAudioRtp()->setDtmfPayloadType(sdpSession->getTelephoneEventType());
// Get the crypto attribute containing srtp's cryptographic context (keys, cipher)
CryptoOffer crypto_offer;
call->getLocalSDP()->getRemoteSdpCryptoFromOffer (remote_sdp, crypto_offer);
bool nego_success = false;
if (!crypto_offer.empty()) {
std::vector<sfl::CryptoSuiteDefinition>localCapabilities;
for (int i = 0; i < 3; i++)
localCapabilities.push_back (sfl::CryptoSuites[i]);
sfl::SdesNegotiator sdesnego(localCapabilities, crypto_offer);
if (sdesnego.negotiate()) {
_debug ("UserAgent: SDES negotiation successfull");
nego_success = true;
try {
call->getAudioRtp()->setRemoteCryptoInfo (sdesnego);
} catch (...) {}
Manager::instance().getDbusManager()->getCallManager()->secureSdesOn (call->getCallId());
} else {
Manager::instance().getDbusManager()->getCallManager()->secureSdesOff (call->getCallId());
}
}
// We did not found any crypto context for this media, RTP fallback
if (!nego_success && call->getAudioRtp()->isSdesEnabled()) {
call->getAudioRtp()->stop();
call->getAudioRtp()->setSrtpEnabled (false);
std::string accountID = Manager::instance().getAccountFromCall(call->getCallId());
if (((SIPAccount *) Manager::instance().getAccount(accountID))->getSrtpFallback())
call->getAudioRtp()->initAudioSymmetricRtpSession ();
}
if (!sdpSession)
return;
sfl::AudioCodec *sessionMedia = sdpSession->getSessionMedia();
if (!sessionMedia)
return;
try {
Manager::instance().audioLayerMutexLock();
Manager::instance().getAudioDriver()->startStream();
Manager::instance().audioLayerMutexUnlock();
int pl = sessionMedia->getPayloadType();
if (pl != call->getAudioRtp()->getSessionMedia()) {
sfl::Codec* audiocodec = Manager::instance().audioCodecFactory.instantiateCodec(pl);
call->getAudioRtp()->updateSessionMedia(static_cast<sfl::AudioCodec *>(audiocodec));
}
}
catch (const SdpException &e) {
_error("UserAgent: Exception: %s", e.what());
}
catch (const std::exception& rtpException) {
_error ("UserAgent: Exception: %s", rtpException.what());
}
}
void outgoing_request_forked_cb (pjsip_inv_session *inv UNUSED, pjsip_event *e UNUSED)
{
}
void transaction_state_changed_cb (pjsip_inv_session *inv UNUSED, pjsip_transaction *tsx, pjsip_event *e)
{
assert (tsx);
assert(e);
if (tsx->role != PJSIP_ROLE_UAS || tsx->state != PJSIP_TSX_STATE_TRYING)
return;
if (pjsip_method_cmp (&tsx->method, &pjsip_refer_method) ==0) {
onCallTransfered (inv, e->body.tsx_state.src.rdata); /** Handle the refer method **/
return;
}
pjsip_tx_data* t_data;
if (e->body.rx_msg.rdata) {
pjsip_rx_data *r_data = e->body.rx_msg.rdata;
if (r_data && r_data->msg_info.msg->line.req.method.id == PJSIP_OTHER_METHOD) {
std::string request = pjsip_rx_data_get_info (r_data);
_debug ("UserAgent: %s", request.c_str());
if (request.find ("NOTIFY") == std::string::npos && request.find ("INFO") != std::string::npos) {
pjsip_dlg_create_response (inv->dlg, r_data, PJSIP_SC_OK, NULL, &t_data);
pjsip_dlg_send_response (inv->dlg, tsx, t_data);
return;
}
}
}
if (!e->body.tsx_state.src.rdata)
return;
// Incoming TEXT message
// Get the message inside the transaction
pjsip_rx_data *r_data = e->body.tsx_state.src.rdata;
std::string formatedMessage = (char*) r_data->msg_info.msg->body->data;
// Try to determine who is the recipient of the message
SIPCall *call = reinterpret_cast<SIPCall *> (inv->mod_data[_mod_ua.id]);
if (!call)
return;
// Respond with a 200/OK
pjsip_dlg_create_response (inv->dlg, r_data, PJSIP_SC_OK, NULL, &t_data);
pjsip_dlg_send_response (inv->dlg, tsx, t_data);
sfl::InstantMessaging *module = Manager::instance().getInstantMessageModule();
try {
// retreive the recipient-list of this message
std::string urilist = module->findTextUriList (formatedMessage);
sfl::InstantMessaging::UriList list = module->parseXmlUriList (urilist);
// If no item present in the list, peer is considered as the sender
std::string from;
if (list.empty()) {
from = call->getPeerNumber ();
} else {
from = list.front()[IM_XML_URI];
if (from == "Me")
from = call->getPeerNumber ();
}
// strip < and > characters in case of an IP address
if (from[0] == '<' && from[from.size()-1] == '>')
from = from.substr (1, from.size()-2);
Manager::instance().incomingMessage(call->getCallId(), from, module->findTextMessage (formatedMessage));
} catch (sfl::InstantMessageException &e) {
_error ("SipVoipLink: %s", e.what());
}
}
void registration_cb (struct pjsip_regc_cbparam *param)
{
SIPAccount *account = static_cast<SIPAccount *>(param->token);
if (account == NULL)
return;
const pj_str_t *description = pjsip_get_status_text (param->code);
if (param->code && description) {
std::string state(description->ptr, description->slen);
Manager::instance().getDbusManager()->getCallManager()->registrationStateChanged (account->getAccountID(), state, param->code);
std::pair<int, std::string> details (param->code, state);
// TODO: there id a race condition for this ressource when closing the application
account->setRegistrationStateDetailed (details);
}
if (param->status != PJ_SUCCESS) {
account->setRegistrationState (ErrorAuth);
account->setRegister (false);
SIPVoIPLink::instance()->shutdownSipTransport (account);
return;
}
if (param->code < 0 || param->code >= 300) {
switch (param->code) {
case 606:
account->setRegistrationState (ErrorConfStun);
break;
case 503:
case 408:
account->setRegistrationState (ErrorHost);
break;
case 401:
case 403:
case 404:
account->setRegistrationState (ErrorAuth);
break;
case 423: { // Expiration Interval Too Brief
account->doubleRegistrationExpire();
account->registerVoIPLink();
}
break;
default:
account->setRegistrationState (Error);
break;
}
account->setRegister (false);
SIPVoIPLink::instance ()->shutdownSipTransport (account);
} else {
if (account->isRegister())
account->setRegistrationState (Registered);
else {
account->setRegistrationState (Unregistered);
SIPVoIPLink::instance ()->shutdownSipTransport (account);
}
}
}
static void handleIncomingOptions (pjsip_rx_data *rdata)
{
pjsip_tx_data *tdata;
if (pjsip_endpt_create_response (_endpt, rdata, PJSIP_SC_OK, NULL, &tdata) != PJ_SUCCESS)
return;
#define ADD_HDR(hdr) do { \
const pjsip_hdr *cap_hdr = hdr; \
if (cap_hdr) \
pjsip_msg_add_hdr (tdata->msg, (pjsip_hdr*) pjsip_hdr_clone (tdata->pool, cap_hdr)); \
} while(0)
#define ADD_CAP(cap) ADD_HDR(pjsip_endpt_get_capability(_endpt, cap, NULL));
ADD_CAP(PJSIP_H_ALLOW);
ADD_CAP(PJSIP_H_ACCEPT);
ADD_CAP(PJSIP_H_SUPPORTED);
ADD_HDR(pjsip_evsub_get_allow_events_hdr(NULL));
pjsip_response_addr res_addr;
pjsip_get_response_addr (tdata->pool, rdata, &res_addr);
if (pjsip_endpt_send_response (_endpt, &res_addr, tdata, NULL, NULL) != PJ_SUCCESS)
pjsip_tx_data_dec_ref (tdata);
}
static pj_bool_t transaction_request_cb (pjsip_rx_data *rdata)
{
pjsip_method *method = &rdata->msg_info.msg->line.req.method;
if (method->id == PJSIP_ACK_METHOD && pjsip_rdata_get_dlg (rdata))
return true;
pjsip_sip_uri *sip_to_uri = (pjsip_sip_uri *) pjsip_uri_get_uri (rdata->msg_info.to->uri);
pjsip_sip_uri *sip_from_uri = (pjsip_sip_uri *) pjsip_uri_get_uri (rdata->msg_info.from->uri);
std::string userName(sip_to_uri->user.ptr, sip_to_uri->user.slen);
std::string server(sip_from_uri->host.ptr, sip_from_uri->host.slen);
std::string account_id(Manager::instance().getAccountIdFromNameAndServer (userName, server));
std::string displayName(parseDisplayName(rdata->msg_info.msg_buf));
if (method->id == PJSIP_OTHER_METHOD) {
pj_str_t *str = &method->name;
std::string request(str->ptr, str->slen);
if (request.find ("NOTIFY") != (size_t)-1) {
int voicemail;
if (sscanf((const char*)rdata->msg_info.msg->body->data, "Voice-Message: %d/", &voicemail) == 1 && voicemail != 0)
Manager::instance().startVoiceMessageNotification (account_id, voicemail);
}
pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_OK, NULL, NULL, NULL);
return true;
} else if (method->id == PJSIP_OPTIONS_METHOD) {
handleIncomingOptions (rdata);
return true;
} else if (method->id != PJSIP_INVITE_METHOD && method->id != PJSIP_ACK_METHOD) {
pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_METHOD_NOT_ALLOWED, NULL, NULL, NULL);
return true;
}
SIPAccount *account = dynamic_cast<SIPAccount *>(Manager::instance().getAccount(account_id));
pjmedia_sdp_session *r_sdp;
pjsip_msg_body *body = rdata->msg_info.msg->body;
if (!body || pjmedia_sdp_parse (rdata->tp_info.pool, (char*) body->data, body->len, &r_sdp) != PJ_SUCCESS)
r_sdp = NULL;
if (account->getActiveCodecs().empty()) {
pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_NOT_ACCEPTABLE_HERE, NULL, NULL, NULL);
return false;
}
// Verify that we can handle the request
unsigned options = 0;
if (pjsip_inv_verify_request (rdata, &options, NULL, NULL, _endpt, NULL) != PJ_SUCCESS) {
pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_METHOD_NOT_ALLOWED, NULL, NULL, NULL);
return true;
}
if (Manager::instance().hookPreference.getSipEnabled()) {
std::string header_value(fetchHeaderValue (rdata->msg_info.msg, Manager::instance().hookPreference.getUrlSipField()));
UrlHook::runAction (Manager::instance().hookPreference.getUrlCommand(), header_value);
}
SIPCall* call = new SIPCall(Manager::instance().getNewCallID(), Call::Incoming, _cp);
Manager::instance().associateCallToAccount(call->getCallId(), account_id);
// May use the published address as well
std::string addrToUse = SIPVoIPLink::instance()->getInterfaceAddrFromName(account->getLocalInterface());
std::string addrSdp = account->isStunEnabled()
? account->getPublishedAddress()
: addrToUse;
pjsip_tpselector *tp = SIPVoIPLink::instance()->initTransportSelector (account->transport, call->getMemoryPool());
if (addrToUse == "0.0.0.0") addrToUse = loadSIPLocalIP();
if (addrSdp == "0.0.0.0")
addrSdp = addrToUse;
char tmp[PJSIP_MAX_URL_SIZE];
int length = pjsip_uri_print(PJSIP_URI_IN_FROMTO_HDR, sip_from_uri, tmp, PJSIP_MAX_URL_SIZE);
std::string peerNumber(tmp, length);
stripSipUriPrefix(peerNumber);
call->setConnectionState (Call::Progressing);
call->setPeerNumber (peerNumber);
call->setDisplayName (displayName);
call->initRecFileName (peerNumber);
setCallMediaLocal (call, addrToUse);
call->getLocalSDP()->setLocalIP (addrSdp);
call->getAudioRtp()->initAudioRtpConfig ();
call->getAudioRtp()->initAudioSymmetricRtpSession ();
if (rdata->msg_info.msg->body) {
char sdpbuffer[1000];
int len = rdata->msg_info.msg->body->print_body (rdata->msg_info.msg->body, sdpbuffer, sizeof sdpbuffer);
if (len == -1) // error
len = 0;
std::string sdpoffer(sdpbuffer, len);
size_t start = sdpoffer.find ("a=crypto:");
// Found crypto header in SDP
if (start != std::string::npos) {
CryptoOffer crypto_offer;
crypto_offer.push_back (std::string(sdpoffer.substr (start, (sdpoffer.size() - start) - 1)));
std::vector<sfl::CryptoSuiteDefinition>localCapabilities;
for (int i = 0; i < 3; i++)
localCapabilities.push_back (sfl::CryptoSuites[i]);
sfl::SdesNegotiator sdesnego (localCapabilities, crypto_offer);
if (sdesnego.negotiate()) {
call->getAudioRtp()->setRemoteCryptoInfo (sdesnego);
call->getAudioRtp()->initLocalCryptoInfo ();
}
}
}
call->getLocalSDP()->receiveOffer (r_sdp, account->getActiveCodecs ());
sfl::Codec* audiocodec = Manager::instance().audioCodecFactory.instantiateCodec (PAYLOAD_CODEC_ULAW);
call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
pjsip_dialog* dialog;
if (pjsip_dlg_create_uas (pjsip_ua_instance(), rdata, NULL, &dialog) != PJ_SUCCESS)
goto fail;
pjsip_inv_create_uas (dialog, rdata, call->getLocalSDP()->getLocalSdpSession(), 0, &call->inv);
PJ_ASSERT_RETURN (pjsip_dlg_set_transport (dialog, tp) == PJ_SUCCESS, 1);
call->inv->mod_data[_mod_ua.id] = call;
// Check whether Replaces header is present in the request and process accordingly.
pjsip_dialog *replaced_dlg;
pjsip_tx_data *response;
if (pjsip_replaces_verify_request(rdata, &replaced_dlg, PJ_FALSE, &response) != PJ_SUCCESS) {
_error("Something wrong with Replaces request.");
pjsip_endpt_respond_stateless(_endpt, rdata, 500 /* internal server error */, NULL, NULL, NULL);
}
// Check if call has been transfered pjsip_tx_data *tdata;
if (replaced_dlg) { // If Replace header present
// Always answer the new INVITE with 200, regardless whether
// the replaced call is in early or confirmed state.
if (pjsip_inv_answer(call->inv, 200, NULL, NULL, &response) == PJ_SUCCESS)
pjsip_inv_send_msg(call->inv, response);
// Get the INVITE session associated with the replaced dialog.
pjsip_inv_session *replaced_inv = pjsip_dlg_get_inv_session(replaced_dlg);
// Disconnect the "replaced" INVITE session.
if (pjsip_inv_end_session(replaced_inv, PJSIP_SC_GONE, NULL, &tdata) == PJ_SUCCESS && tdata)
pjsip_inv_send_msg(replaced_inv, tdata);
} else { // Prooceed with normal call flow
PJ_ASSERT_RETURN (pjsip_inv_initial_answer (call->inv, rdata, PJSIP_SC_RINGING, NULL, NULL, &tdata) == PJ_SUCCESS, 1);
PJ_ASSERT_RETURN (pjsip_inv_send_msg (call->inv, tdata) == PJ_SUCCESS, 1);
call->setConnectionState (Call::Ringing);
if (!Manager::instance().incomingCall (call, account_id))
goto fail;
Manager::instance().getAccountLink (account_id)->addCall (call);
}
return true;
fail:
delete call;
pjsip_endpt_respond_stateless (_endpt, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL, NULL);
return false;
}
static pj_bool_t transaction_response_cb (pjsip_rx_data *rdata)
{
pjsip_dialog *dlg = pjsip_rdata_get_dlg (rdata);
if (!dlg)
return PJ_SUCCESS;
pjsip_transaction *tsx = pjsip_rdata_get_tsx (rdata);
if (!tsx || tsx->method.id != PJSIP_INVITE_METHOD)
return PJ_SUCCESS;
if (tsx->status_code / 100 == 2) {
/**
* Send an ACK message inside a transaction. PJSIP send automatically, non-2xx ACK response.
* ACK for a 2xx response must be send using this method.
*/
pjsip_tx_data *tdata;
pjsip_dlg_create_request(dlg, &pjsip_ack_method, rdata->msg_info.cseq->cseq, &tdata);
pjsip_dlg_send_request(dlg, tdata, -1, NULL);
}
return PJ_SUCCESS;
}
void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata)
{
SIPCall *currentCall = reinterpret_cast<SIPCall *>(inv->mod_data[_mod_ua.id]);
if (currentCall == NULL)
return;
static const pj_str_t str_refer_to = { (char*) "Refer-To", 8};
pjsip_generic_string_hdr *refer_to = static_cast<pjsip_generic_string_hdr*>
(pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_refer_to, NULL));
if (!refer_to) {
pjsip_dlg_respond (inv->dlg, rdata, 400, NULL, NULL, NULL);
return;
}
SIPVoIPLink::instance()->newOutgoingCall(Manager::instance().getNewCallID(), std::string(refer_to->hvalue.ptr, refer_to->hvalue.slen));
Manager::instance().hangupCall(currentCall->getCallId());
}
void transfer_client_cb (pjsip_evsub *sub, pjsip_event *event)
{
switch(pjsip_evsub_get_state(sub)) {
case PJSIP_EVSUB_STATE_ACCEPTED:
pj_assert(event->type == PJSIP_EVENT_TSX_STATE && event->body.tsx_state.type == PJSIP_EVENT_RX_MSG);
break;
case PJSIP_EVSUB_STATE_TERMINATED:
pjsip_evsub_set_mod_data (sub, _mod_ua.id, NULL);
break;
case PJSIP_EVSUB_STATE_ACTIVE: {
SIPVoIPLink *link = reinterpret_cast<SIPVoIPLink *>(pjsip_evsub_get_mod_data(sub, _mod_ua.id));
if (!link or !event)
return;
pjsip_rx_data* r_data = event->body.rx_msg.rdata;
if (!r_data)
return;
std::string request(pjsip_rx_data_get_info (r_data));
pjsip_status_line status_line = { 500, *pjsip_get_status_text (500) };
if (r_data->msg_info.msg->line.req.method.id == PJSIP_OTHER_METHOD and
request.find ("NOTIFY") != std::string::npos) {
pjsip_msg_body *body = r_data->msg_info.msg->body;
if (!body)
return;
if (pj_stricmp2 (&body->content_type.type, "message") or
pj_stricmp2 (&body->content_type.subtype, "sipfrag"))
return;
if (pjsip_parse_status_line ( (char*) body->data, body->len, &status_line) != PJ_SUCCESS)
return;
}
std::string transferID(r_data->msg_info.cid->id.ptr, r_data->msg_info.cid->id.slen);
SIPCall *call = dynamic_cast<SIPCall *> (link->getCall(transferCallID[transferID]));
if (!call)
return;
if (status_line.code/100 == 2) {
pjsip_tx_data *tdata;
if (pjsip_inv_end_session (call->inv, PJSIP_SC_GONE, NULL, &tdata) == PJ_SUCCESS)
pjsip_inv_send_msg (call->inv, tdata);
Manager::instance().hangupCall(call->getCallId());
pjsip_evsub_set_mod_data (sub, _mod_ua.id, NULL);
}
break;
}
default:
break;
}
}
void setCallMediaLocal (SIPCall* call, const std::string &localIP)
{
std::string account_id(Manager::instance().getAccountFromCall (call->getCallId ()));
SIPAccount *account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (account_id));
unsigned int callLocalAudioPort = ((rand() % 27250) + 5250) * 2;
unsigned int callLocalExternAudioPort = account->isStunEnabled() ? account->getStunPort()
: callLocalAudioPort;
call->setLocalIp (localIP);
call->setLocalAudioPort(callLocalAudioPort);
call->getLocalSDP()->setLocalPublishedAudioPort(callLocalExternAudioPort);
}
std::string fetchHeaderValue (pjsip_msg *msg, std::string field)
{
pj_str_t name = pj_str ( (char*) field.c_str());
pjsip_generic_string_hdr *hdr = (pjsip_generic_string_hdr*) pjsip_msg_find_hdr_by_name (msg, &name, NULL);
if (!hdr)
return "";
std::string value(std::string(hdr->hvalue.ptr, hdr->hvalue.slen));
size_t pos = value.find ("\n");
if (pos == std::string::npos)
return "";
return value.substr (0, pos);
}
} // end anonymous namespace
std::vector<std::string> SIPVoIPLink::getAllIpInterfaceByName (void)
{
static ifreq ifreqs[20];
ifconf ifconf;
std::vector<std::string> ifaceList;
ifaceList.push_back ("default");
ifconf.ifc_buf = (char*) (ifreqs);
ifconf.ifc_len = sizeof (ifreqs);
int sock = socket(AF_INET,SOCK_STREAM,0);
if (sock >= 0) {
if (ioctl (sock, SIOCGIFCONF, &ifconf) >= 0)
for (unsigned i = 0; i < ifconf.ifc_len/sizeof (struct ifreq); i++)
ifaceList.push_back (std::string (ifreqs[i].ifr_name));
close (sock);
}
return ifaceList;
}
std::string SIPVoIPLink::getInterfaceAddrFromName (const std::string &ifaceName)
{
int fd = socket (AF_INET, SOCK_DGRAM,0);
if (fd < 0) {
_error ("UserAgent: Error: could not open socket: %m");
return "";
}
ifreq ifr;
strcpy(ifr.ifr_name, ifaceName.c_str());
memset(&ifr.ifr_addr, 0, sizeof(ifr.ifr_addr));
ifr.ifr_addr.sa_family = AF_INET;
ioctl (fd, SIOCGIFADDR, &ifr);
close (fd);
sockaddr_in *saddr_in = (struct sockaddr_in *) &ifr.ifr_addr;
return inet_ntoa(saddr_in->sin_addr);
}
std::vector<std::string> SIPVoIPLink::getAllIpInterface (void){
pj_sockaddr addrList[16];
unsigned addrCnt = PJ_ARRAY_SIZE (addrList);
std::vector<std::string> ifaceList;
if (pj_enum_ip_interface (pj_AF_INET(), &addrCnt, addrList) == PJ_SUCCESS)
for (unsigned i = 0; i < addrCnt; i++) {
char addr[PJ_INET_ADDRSTRLEN];
pj_sockaddr_print (&addrList[i], addr, sizeof (addr), 0);
ifaceList.push_back (std::string (addr));
}
return ifaceList;
}