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OpenBTS-UMTS/SIP/SIPEngine.cpp

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/**@file SIP Call Control -- SIP IETF RFC-3261, RTP IETF RFC-3550. */
/*
* OpenBTS provides an open source alternative to legacy telco protocols and
* traditionally complex, proprietary hardware systems.
*
* Copyright 2008, 2009, 2010 Free Software Foundation, Inc.
* Copyright 2011, 2014 Range Networks, Inc.
*
* This software is distributed under the terms of the GNU Affero General
* Public License version 3. See the COPYING and NOTICE files in the main
* directory for licensing information.
*
* This use of this software may be subject to additional restrictions.
* See the LEGAL file in the main directory for details.
*/
#include <stdio.h>
#include <stdlib.h>
#include <iostream>
#include <sys/types.h>
#include <semaphore.h>
#include <ortp/telephonyevents.h>
#include <Timeval.h>
#include <UMTSConfig.h>
#include <ControlCommon.h>
#include <UMTSCommon.h>
#include "SIPInterface.h"
#include "SIPUtility.h"
#include "SIPMessage.h"
#include "SIPEngine.h"
#undef WARNING
using namespace std;
using namespace SIP;
using namespace Control;
const char* SIP::SIPStateString(SIPState s)
{
switch(s)
{
case NullState: return "Null";
case Timeout: return "Timeout";
case Starting: return "Starting";
case Proceeding: return "Proceeding";
case Ringing: return "Ringing";
case Connecting: return "Connecting";
case Active: return "Active";
case Fail: return "Fail";
case Busy: return "Busy";
case MODClearing: return "MODClearing";
case MTDClearing: return "MTDClearing";
case Cleared: return "Cleared";
case MessageSubmit: return "SMS-Submit";
default: return NULL;
}
}
ostream& SIP::operator<<(ostream& os, SIPState s)
{
const char* str = SIPStateString(s);
if (str) os << str;
else os << "?" << s << "?";
return os;
}
SIPEngine::SIPEngine(const char* proxy, const char* IMSI)
:mCSeq(random()%1000),
mMyToFromHeader(NULL), mRemoteToFromHeader(NULL),
mCallIDHeader(NULL),
mSIPPort(gConfig.getNum("SIP.Local.Port")),
mSIPIP(gConfig.getStr("SIP.Local.IP")),
mINVITE(NULL), mLastResponse(NULL), mBYE(NULL),
mSession(NULL),mTxTime(0), mRxTime(0),
mState(NullState),
mDTMF('\0'),mDTMFDuration(0)
{
if (IMSI) user(IMSI);
resolveAddress(&mProxyAddr,proxy);
char host[256];
const char* ret = inet_ntop(AF_INET,&(mProxyAddr.sin_addr),host,255);
if (!ret) {
LOG(ALERT) << "cannot translate proxy IP address";
return;
}
mProxyIP = string(host);
mProxyPort = ntohs(mProxyAddr.sin_port);
// generate a tag now
char tmp[50];
make_tag(tmp);
mMyTag=tmp;
// set our CSeq in case we need one
mCSeq = random()%600;
}
SIPEngine::~SIPEngine()
{
if (mINVITE!=NULL) osip_message_free(mINVITE);
if (mLastResponse!=NULL) osip_message_free(mLastResponse);
if (mBYE!=NULL) osip_message_free(mBYE);
// FIXME -- Do we need to dispose of the RtpSesion *mSesison?
}
void SIPEngine::saveINVITE(const osip_message_t *INVITE, bool mine)
{
// Instead of cloning, why not just keep the old one?
// Because that doesn't work in all calling contexts.
// This simplifies the call-handling logic.
if (mINVITE!=NULL) osip_message_free(mINVITE);
osip_message_clone(INVITE,&mINVITE);
mCallIDHeader = mINVITE->call_id;
// If this our own INVITE? Did we initiate the transaciton?
if (mine) {
mMyToFromHeader = mINVITE->from;
mRemoteToFromHeader = mINVITE->to;
return;
}
// It's not our own. The From: is the remote party.
mMyToFromHeader = mINVITE->to;
mRemoteToFromHeader = mINVITE->from;
// We need to set our tag, too.
osip_from_set_tag(mMyToFromHeader, strdup(mMyTag.c_str()));
}
void SIPEngine::saveResponse(osip_message_t *response)
{
if (mLastResponse!=NULL) osip_message_free(mLastResponse);
osip_message_clone(response,&mLastResponse);
// The To: is the remote party and might have an new tag.
mRemoteToFromHeader = mLastResponse->to;
}
void SIPEngine::saveBYE(const osip_message_t *BYE, bool mine)
{
// Instead of cloning, why not just keep the old one?
// Because that doesn't work in all calling contexts.
// This simplifies the call-handling logic.
if (mBYE!=NULL) osip_message_free(mBYE);
osip_message_clone(BYE,&mBYE);
}
void SIPEngine::user( const char * IMSI )
{
LOG(DEBUG) << "IMSI=" << IMSI;
unsigned id = random();
char tmp[20];
sprintf(tmp, "%u", id);
mCallID = tmp;
// IMSI gets prefixed with "IMSI" to form a SIP username
mSIPUsername = string("IMSI") + IMSI;
}
void SIPEngine::user( const char * wCallID, const char * IMSI, const char *origID, const char *origHost)
{
LOG(DEBUG) << "IMSI=" << IMSI << " " << wCallID << " " << origID << "@" << origHost;
mSIPUsername = string("IMSI") + IMSI;
mCallID = string(wCallID);
mRemoteUsername = string(origID);
mRemoteDomain = string(origHost);
}
string randy401(osip_message_t *msg)
{
if (msg->status_code != 401) return "";
osip_www_authenticate_t *auth = (osip_www_authenticate_t*)osip_list_get(&msg->www_authenticates, 0);
if (auth == NULL) return "";
char *rand = osip_www_authenticate_get_nonce(auth);
string rands = rand ? string(rand) : "";
if (rands.length()!=32) {
LOG(WARNING) << "SIP RAND wrong length: " << rands;
return "";
}
return rands;
}
bool SIPEngine::Register( Method wMethod , string *RAND, const char *IMSI, const char *SRES)
{
LOG(INFO) << "user " << mSIPUsername << " state " << mState << " " << wMethod << " callID " << mCallID;
// Before start, need to add mCallID
gSIPInterface.addCall(mCallID);
// Initial configuration for sip message.
// Make a new from tag and new branch.
// make new mCSeq.
// Generate SIP Message
// Either a register or unregister. Only difference
// is expiration period.
osip_message_t * reg;
if (wMethod == SIPRegister ){
reg = sip_register( mSIPUsername.c_str(),
60*gConfig.getNum("SIP.RegistrationPeriod"),
mSIPPort, mSIPIP.c_str(),
mProxyIP.c_str(), mMyTag.c_str(),
mViaBranch.c_str(), mCallID.c_str(), mCSeq,
RAND, IMSI, SRES
);
} else if (wMethod == SIPUnregister ) {
reg = sip_register( mSIPUsername.c_str(),
0,
mSIPPort, mSIPIP.c_str(),
mProxyIP.c_str(), mMyTag.c_str(),
mViaBranch.c_str(), mCallID.c_str(), mCSeq,
NULL, NULL, NULL
);
} else { assert(0); }
LOG(DEBUG) << "writing " << reg;
gSIPInterface.write(&mProxyAddr,reg);
bool success = false;
osip_message_t *msg = NULL;
Timeval timeout(gConfig.getNum("SIP.Timer.F"));
while (!timeout.passed()) {
try {
// SIPInterface::read will throw SIPTIimeout if it times out.
// It should not return NULL.
msg = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.E"));
} catch (SIPTimeout) {
// send again
gSIPInterface.write(&mProxyAddr,reg);
continue;
}
assert(msg);
int status = msg->status_code;
LOG(INFO) << "received status " << msg->status_code << " " << msg->reason_phrase;
// specific status
if (status==200) {
LOG(INFO) << "REGISTER success";
success = true;
break;
}
if (status==401) {
string wRAND = randy401(msg);
// if rand is included on 401 unauthorized, then the challenge-response game is afoot
if (wRAND.length() != 0 && RAND != NULL) {
LOG(INFO) << "REGISTER challenge RAND=" << wRAND;
*RAND = wRAND;
osip_message_free(msg);
osip_message_free(reg);
return false;
} else {
LOG(INFO) << "REGISTER fail -- unauthorized";
break;
}
}
if (status==404) {
LOG(INFO) << "REGISTER fail -- not found";
break;
}
if (status>=200) {
LOG(NOTICE) << "REGISTER unexpected response " << status;
break;
}
}
if (!msg) {
LOG(ALERT) << "SIP REGISTER timed out; is the registration server " << mProxyIP << ":" << mProxyPort << " OK?";
throw SIPTimeout();
}
osip_message_free(reg);
osip_message_free(msg);
gSIPInterface.removeCall(mCallID);
return success;
}
const char* geoprivTemplate =
"<?xml version=\"1.0\" encoding=\"UTF-8\"?>\n"
"<presence xmlns=\"urn:ietf:params:xml:ns:pidf\"\n"
"xmlns:gp=\"urn:ietf:params:xml:ns:pidf:geopriv10\"\n"
"xmlns:gml=\"urn:opengis:specification:gml:schema-xsd:feature:v3.0\"\n"
"entity=\"pres:%s@%s\">\n"
"<tuple id=\"1\">\n"
"<status>\n"
"<gp:geopriv>\n"
"<gp:location-info>\n"
"<gml:location>\n"
"<gml:Point gml:id=\"point1\" srsName=\"epsg:4326\">\n"
"<gml:coordinates>%s</gml:coordinates>\n"
"</gml:Point>\n"
"</gml:location>\n"
"</gp:location-info>\n"
"<gp:usage-rules>\n"
"<gp:retransmission-allowed>no</gp:retransmission-allowed>\n"
"</gp:usage-rules>\n"
"</gp:geopriv>\n"
"</status>\n"
"</tuple>\n"
"</presence>\n";
SIPState SIPEngine::MOCSendINVITE( const char * wCalledUsername,
const char * wCalledDomain , short wRtp_port, unsigned wCodec)
{
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
// Before start, need to add mCallID
gSIPInterface.addCall(mCallID);
// Set Invite params.
// new CSEQ and codec
char tmp[50];
make_branch(tmp);
mViaBranch = tmp;
mCodec = wCodec;
mCSeq++;
mRemoteUsername = wCalledUsername;
mRemoteDomain = wCalledDomain;
mRTPPort= wRtp_port;
LOG(DEBUG) << "mRemoteUsername=" << mRemoteUsername;
LOG(DEBUG) << "mSIPUsername=" << mSIPUsername;
osip_message_t * invite = sip_invite(
mRemoteUsername.c_str(), mRTPPort, mSIPUsername.c_str(),
mSIPPort, mSIPIP.c_str(), mProxyIP.c_str(),
mMyTag.c_str(), mViaBranch.c_str(), mCallID.c_str(), mCSeq, mCodec);
// P-Access-Network-Info
// See 3GPP 24.229 7.2.
char cgi_3gpp[50];
sprintf(cgi_3gpp,"3GPP-GERAN; cgi-3gpp=%s%s%04x%04x",
gConfig.getStr("UMTS.Identity.MCC").c_str(),gConfig.getStr("UMTS.Identity.MNC").c_str(),
(unsigned)gConfig.getNum("UMTS.Identity.LAC"),(unsigned)gConfig.getNum("UMTS.Identity.CI"));
osip_message_set_header(invite,"P-Access-Network-Info",cgi_3gpp);
// Send Invite.
gSIPInterface.write(&mProxyAddr,invite);
saveINVITE(invite,true);
osip_message_free(invite);
mState = Starting;
return mState;
};
SIPState SIPEngine::MOCResendINVITE()
{
assert(mINVITE);
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
gSIPInterface.write(&mProxyAddr,mINVITE);
return mState;
}
SIPState SIPEngine::MOCWaitForOK()
{
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
osip_message_t * msg;
// Read off the fifo. if time out will
// clean up and return false.
try {
msg = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.A"));
}
catch (SIPTimeout& e) {
LOG(DEBUG) << "timeout";
mState = Timeout;
return mState;
}
int status = msg->status_code;
LOG(DEBUG) << "received status " << status;
saveResponse(msg);
switch (status) {
case 100: // Trying
case 183: // Progress
mState = Proceeding;
break;
case 180: // Ringing
mState = Ringing;
break;
case 200: // OK
// Save the response and update the state,
// but the ACK doesn't happen until the call connects.
mState = Active;
break;
// Anything 400 or above terminates the call, so we ACK.
// FIXME -- It would be nice to save more information about the
// specific failure cause.
case 486:
case 503:
mState = Busy;
MOCSendACK();
break;
default:
LOG(NOTICE) << "unhandled status code " << status;
mState = Fail;
MOCSendACK();
}
osip_message_free(msg);
LOG(DEBUG) << " new state: " << mState;
return mState;
}
SIPState SIPEngine::MOCSendACK()
{
assert(mLastResponse);
// new branch
char tmp[50];
make_branch(tmp);
mViaBranch = tmp;
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
osip_message_t* ack = sip_ack( mRemoteDomain.c_str(),
mRemoteUsername.c_str(),
mSIPUsername.c_str(),
mSIPPort, mSIPIP.c_str(), mProxyIP.c_str(),
mMyToFromHeader, mRemoteToFromHeader,
mViaBranch.c_str(), mCallIDHeader, mCSeq
);
gSIPInterface.write(&mProxyAddr,ack);
osip_message_free(ack);
return mState;
}
SIPState SIPEngine::MODSendBYE()
{
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
assert(mINVITE);
char tmp[50];
make_branch(tmp);
mViaBranch = tmp;
mCSeq++;
osip_message_t * bye = sip_bye(mRemoteDomain.c_str(), mRemoteUsername.c_str(),
mSIPUsername.c_str(),
mSIPPort, mSIPIP.c_str(), mProxyIP.c_str(), mProxyPort,
mMyToFromHeader, mRemoteToFromHeader,
mViaBranch.c_str(), mCallIDHeader, mCSeq );
gSIPInterface.write(&mProxyAddr,bye);
saveBYE(bye,true);
osip_message_free(bye);
mState = MODClearing;
return mState;
}
SIPState SIPEngine::MODResendBYE()
{
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
assert(mState==MODClearing);
assert(mBYE);
gSIPInterface.write(&mProxyAddr,mBYE);
return mState;
}
SIPState SIPEngine::MODWaitForOK()
{
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
bool responded = false;
Timeval byeTimeout(gConfig.getNum("SIP.Timer.F"));
while (!byeTimeout.passed()) {
try {
osip_message_t * ok = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.E"));
responded = true;
unsigned code = ok->status_code;
saveResponse(ok);
osip_message_free(ok);
if (code!=200) {
LOG(WARNING) << "unexpected " << code << " response to BYE, from proxy " << mProxyIP << ":" << mProxyPort;
}
break;
}
catch (SIPTimeout& e) {
LOG(NOTICE) << "response timeout, resending BYE";
MODResendBYE();
}
}
if (!responded) { LOG(ALERT) << "lost contact with proxy " << mProxyIP << ":" << mProxyPort; }
// However we got here, the SIP side of the call is cleared now.
mState = Cleared;
return mState;
}
SIPState SIPEngine::MTDCheckBYE()
{
//LOG(DEBUG) << "user " << mSIPUsername << " state " << mState;
// If the call is not active, there should be nothing to check.
if (mState!=Active) return mState;
// Need to check size of osip_message_t* fifo,
// so need to get fifo pointer and get size.
// HACK -- reach deep inside to get damn thing
int fifoSize = gSIPInterface.fifoSize(mCallID);
// Size of -1 means the FIFO does not exist.
// Treat the call as cleared.
if (fifoSize==-1) {
LOG(NOTICE) << "MTDCheckBYE attempt to check BYE on non-existant SIP FIFO";
mState=Cleared;
return mState;
}
// If no messages, there is no change in state.
if (fifoSize==0) return mState;
osip_message_t * msg = gSIPInterface.read(mCallID);
if ((msg->sip_method!=NULL) && (strcmp(msg->sip_method,"BYE")==0)) {
LOG(DEBUG) << "found msg="<<msg->sip_method;
saveBYE(msg,false);
mState = MTDClearing;
}
// FIXME -- Check for repeated ACK and send OK if needed.
// FIXME -- Check for repeated OK and send ACK if needed.
osip_message_free(msg);
return mState;
}
SIPState SIPEngine::MTDSendOK()
{
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
assert(mBYE);
osip_message_t * okay = sip_b_okay(mBYE);
gSIPInterface.write(&mProxyAddr,okay);
osip_message_free(okay);
mState = Cleared;
return mState;
}
SIPState SIPEngine::MTCSendTrying()
{
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
if (mINVITE==NULL) mState=Fail;
if (mState==Fail) return mState;
osip_message_t * trying = sip_trying(mINVITE, mSIPUsername.c_str(), mProxyIP.c_str());
gSIPInterface.write(&mProxyAddr,trying);
osip_message_free(trying);
mState=Proceeding;
return mState;
}
SIPState SIPEngine::MTCSendRinging()
{
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
assert(mINVITE);
LOG(DEBUG) << "send ringing";
osip_message_t * ringing = sip_ringing(mINVITE,
mSIPUsername.c_str(), mProxyIP.c_str());
gSIPInterface.write(&mProxyAddr,ringing);
osip_message_free(ringing);
mState = Proceeding;
return mState;
}
SIPState SIPEngine::MTCSendOK( short wRTPPort, unsigned wCodec )
{
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
assert(mINVITE);
mRTPPort = wRTPPort;
mCodec = wCodec;
LOG(DEBUG) << "port=" << wRTPPort << " codec=" << mCodec;
// Form ack from invite and new parameters.
osip_message_t * okay = sip_okay(mINVITE, mSIPUsername.c_str(),
mSIPIP.c_str(), mSIPPort, mRTPPort, mCodec);
gSIPInterface.write(&mProxyAddr,okay);
osip_message_free(okay);
mState=Connecting;
return mState;
}
SIPState SIPEngine::MTCWaitForACK()
{
// wait for ack,set this to timeout of
// of call channel. If want a longer timeout
// period, need to split into 2 handle situation
// like MOC where this fxn if called multiple times.
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
osip_message_t * ack;
// FIXME -- This is supposed to retransmit BYE on timer I.
try {
ack = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.H"));
}
catch (SIPTimeout& e) {
LOG(NOTICE) << "timeout";
mState = Timeout;
return mState;
}
catch (SIPError& e) {
LOG(NOTICE) << "read error";
mState = Fail;
return mState;
}
if (ack->sip_method==NULL) {
LOG(NOTICE) << "SIP message with no method, status " << ack->status_code;
mState = Fail;
osip_message_free(ack);
return mState;
}
LOG(INFO) << "received sip_method="<<ack->sip_method;
// check for duplicated INVITE
if( strcmp(ack->sip_method,"INVITE") == 0){
LOG(NOTICE) << "received duplicate INVITE";
}
// check for the ACK
else if( strcmp(ack->sip_method,"ACK") == 0){
LOG(INFO) << "received ACK";
mState=Active;
}
// check for the CANCEL
else if( strcmp(ack->sip_method,"CANCEL") == 0){
LOG(INFO) << "received CANCEL";
mState=Fail;
// FIXME -- Send 200 OK, see ticket #173.
}
// check for strays
else {
LOG(NOTICE) << "unexpected Message "<<ack->sip_method;
mState = Fail;
}
osip_message_free(ack);
return mState;
}
SIPState SIPEngine::MTCCheckForCancel()
{
// wait for ack,set this to timeout of
// of call channel. If want a longer timeout
// period, need to split into 2 handle situation
// like MOC where this fxn if called multiple times.
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
osip_message_t * msg;
try {
// 1 ms timeout, effectively non-blocking
msg = gSIPInterface.read(mCallID,1);
}
catch (SIPTimeout& e) {
return mState;
}
catch (SIPError& e) {
LOG(NOTICE) << "read error";
mState = Fail;
return mState;
}
if (msg->sip_method==NULL) {
LOG(NOTICE) << "SIP message with no method, status " << msg->status_code;
mState = Fail;
osip_message_free(msg);
return mState;
}
LOG(INFO) << "received sip_method=" << msg->sip_method;
// check for duplicated INVITE
if (strcmp(msg->sip_method,"INVITE")==0) {
LOG(NOTICE) << "received duplicate INVITE";
}
// check for the CANCEL
else if (strcmp(msg->sip_method,"CANCEL")==0) {
LOG(INFO) << "received CANCEL";
mState=Fail;
}
// check for strays
else {
LOG(NOTICE) << "unexpected Message " << msg->sip_method;
mState = Fail;
}
osip_message_free(msg);
return mState;
}
void SIPEngine::InitRTP(const osip_message_t * msg )
{
if(mSession == NULL)
mSession = rtp_session_new(RTP_SESSION_SENDRECV);
bool rfc2833 = gConfig.defines("SIP.DTMF.RFC2833");
if (rfc2833) {
RtpProfile* profile = rtp_session_get_send_profile(mSession);
int index = gConfig.getNum("SIP.DTMF.RFC2833.PayloadType");
rtp_profile_set_payload(profile,index,&payload_type_telephone_event);
// Do we really need this next line?
rtp_session_set_send_profile(mSession,profile);
}
rtp_session_set_blocking_mode(mSession, TRUE);
rtp_session_set_scheduling_mode(mSession, TRUE);
rtp_session_set_connected_mode(mSession, TRUE);
rtp_session_set_symmetric_rtp(mSession, TRUE);
// Hardcode RTP session type to GSM full rate (GSM 06.10).
// FIXME -- Make this work for multiple vocoder types.
rtp_session_set_payload_type(mSession, 3);
char d_ip_addr[20];
char d_port[10];
get_rtp_params(msg, d_port, d_ip_addr);
LOG(DEBUG) << "IP="<<d_ip_addr<<" "<<d_port<<" "<<mRTPPort;
#ifdef ORTP_NEW_API
rtp_session_set_local_addr(mSession, "0.0.0.0", mRTPPort, -1);
#else
rtp_session_set_local_addr(mSession, "0.0.0.0", mRTPPort, mRTPPort+1);
#endif
rtp_session_set_remote_addr(mSession, d_ip_addr, atoi(d_port));
// Check for event support.
int code = rtp_session_telephone_events_supported(mSession);
if (code == -1) {
if (rfc2833) { LOG(ALERT) << "RTP session does not support selected DTMF method RFC-2833"; }
else { LOG(WARNING) << "RTP session does not support telephone events"; }
}
}
void SIPEngine::MTCInitRTP()
{
assert(mINVITE);
InitRTP(mINVITE);
}
void SIPEngine::MOCInitRTP()
{
assert(mLastResponse);
InitRTP(mLastResponse);
}
bool SIPEngine::startDTMF(char key)
{
LOG (DEBUG) << key;
if (mState!=Active) return false;
mDTMF = key;
mDTMFDuration = 0;
mDTMFStartTime = mTxTime;
int code = rtp_session_send_dtmf2(mSession,mDTMF,mDTMFStartTime,mDTMFDuration);
mDTMFDuration += 160;
if (!code) return true;
// Error? Turn off DTMF sending.
LOG(WARNING) << "DTMF RFC-2833 failed on start.";
mDTMF = '\0';
return false;
}
void SIPEngine::stopDTMF()
{
mDTMF='\0';
}
void SIPEngine::txFrame(unsigned char* frame )
{
if(mState!=Active) return;
// HACK -- Hardcoded for GSM/8000.
// FIXME -- Make this work for multiple vocoder types.
rtp_session_send_with_ts(mSession, frame, 33, mTxTime);
mTxTime += 160;
if (mDTMF) {
// Any RFC-2833 action?
int code = rtp_session_send_dtmf2(mSession,mDTMF,mDTMFStartTime,mDTMFDuration);
mDTMFDuration += 160;
LOG (DEBUG) << "DTMF RFC-2833 sending " << mDTMF << " " << mDTMFDuration;
// Turn it off if there's an error.
if (code) {
LOG(ERR) << "DTMF RFC-2833 failed after start.";
mDTMF='\0';
}
}
}
int SIPEngine::rxFrame(unsigned char* frame)
{
if(mState!=Active) return 0;
int more;
int ret=0;
// HACK -- Hardcoded for GSM/8000.
// FIXME -- Make this work for multiple vocoder types.
ret = rtp_session_recv_with_ts(mSession, frame, 33, mRxTime, &more);
mRxTime += 160;
return ret;
}
SIPState SIPEngine::MOSMSSendMESSAGE(const char * wCalledUsername,
const char * wCalledDomain , const char *messageText, const char *contentType)
{
LOG(DEBUG) << "mState=" << mState;
LOG(INFO) << "SIP send to " << wCalledUsername << "@" << wCalledDomain << " MESSAGE " << messageText;
// Before start, need to add mCallID
gSIPInterface.addCall(mCallID);
// Set MESSAGE params.
char tmp[50];
make_branch(tmp);
mViaBranch = tmp;
mCSeq++;
mRemoteUsername = wCalledUsername;
mRemoteDomain = wCalledDomain;
osip_message_t * message = sip_message(
mRemoteUsername.c_str(), mSIPUsername.c_str(),
mSIPPort, mSIPIP.c_str(), mProxyIP.c_str(),
mMyTag.c_str(), mViaBranch.c_str(), mCallID.c_str(), mCSeq,
messageText, contentType);
// Send Invite to the SIP proxy.
gSIPInterface.write(&mProxyAddr,message);
saveINVITE(message,true);
osip_message_free(message);
mState = MessageSubmit;
return mState;
};
SIPState SIPEngine::MOSMSWaitForSubmit()
{
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
try {
osip_message_t * ok = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.A"));
// That should never return NULL.
assert(ok);
if((ok->status_code==200) || (ok->status_code==202) ) {
mState = Cleared;
LOG(INFO) << "successful";
}
osip_message_free(ok);
}
catch (SIPTimeout& e) {
LOG(ALERT) << "SIP MESSAGE timed out; is the SMS server " << mProxyIP << ":" << mProxyPort << " OK?";
mState = Fail;
}
return mState;
}
SIPState SIPEngine::MTSMSSendOK()
{
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
// If this operation was initiated from the CLI, there was no INVITE.
if (!mINVITE) {
LOG(INFO) << "clearing CLI-generated transaction";
mState=Cleared;
return mState;
}
// Form ack from invite and new parameters.
osip_message_t * okay = sip_okay_SMS(mINVITE, mSIPUsername.c_str(),
mSIPIP.c_str(), mSIPPort);
gSIPInterface.write(&mProxyAddr,okay);
osip_message_free(okay);
mState=Cleared;
return mState;
}
bool SIPEngine::sendINFOAndWaitForOK(unsigned wInfo)
{
LOG(INFO) << "user " << mSIPUsername << " state " << mState;
char tmp[50];
make_branch(tmp);
mViaBranch = tmp;
mCSeq++;
osip_message_t * info = sip_info( wInfo,
mRemoteUsername.c_str(), mRTPPort, mSIPUsername.c_str(),
mSIPPort, mSIPIP.c_str(), mProxyIP.c_str(),
mMyTag.c_str(), mViaBranch.c_str(), mCallIDHeader, mCSeq);
gSIPInterface.write(&mProxyAddr,info);
osip_message_free(info);
try {
// This will timeout on failure. It will not return NULL.
osip_message_t *msg = gSIPInterface.read(mCallID, gConfig.getNum("SIP.Timer.A"));
LOG(DEBUG) << "received status " << msg->status_code << " " << msg->reason_phrase;
bool retVal = (msg->status_code==200);
osip_message_free(msg);
if (!retVal) LOG(CRIT) << "DTMF RFC-2967 failed.";
return retVal;
}
catch (SIPTimeout& e) {
LOG(NOTICE) << "timeout";
return false;
}
};
// vim: ts=4 sw=4
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