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merged changes from rel_0.10.0_rc1@106 made at sipit into trunk.

Browse Source 
git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@116 8eb893ce-cfd4-0310-b710-fb5ebe64c474
sayer/1.4-spce2.6
Raphael Coeffic 20 years ago
parent 3c8d91e96c
commit c75dd1bae7
16 changed files with 395 additions and 144 deletions
Show Stats Download Patch File Download Diff File

2
apps/Makefile
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@ -1,7 +1,7 @@
COREPATH ?= ../core
include $(COREPATH)/Makefile.defs
exclude_modules =
exclude_modules = mp3
modules = $(filter-out $(exclude_modules) \
$(wildcard Makefile*) CVS, \
$(wildcard *) )

10
apps/conf_auth/conf_auth.py
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@ -74,6 +74,7 @@ class IvrDialog(IvrDialogBase):
def onDtmf(self,key,duration):
if self.state == collect:
self.flush()
if key == 10:
c = xmlrpclib.ServerProxy(server_path )
@ -91,11 +92,12 @@ class IvrDialog(IvrDialogBase):
self.state = connect
self.removeTimer(HINT_TIMER)
else:
self.keys = ''
self.auth_fail_msg = IvrAudioFile()
self.auth_fail_msg.open(erg[1], ivr.AUDIO_READ)
self.flush()
#self.flush()
self.enqueue(self.auth_fail_msg, None)
self.keys = ''
self.setTimer(HINT_TIMER, HINT_TIMEOUT)
else:
self.keys += str(key)
@ -109,7 +111,8 @@ class IvrDialog(IvrDialogBase):
self.setRelayonly()
self.connectCallee(self.conf_to, self.conf_uri)
self.state = connected
self.setTimer(CONF_TIMER, self.conf_duration)
if (self.conf_duration > 0):
self.setTimer(CONF_TIMER, self.conf_duration)
elif self.state == collect:
self.setTimer(HINT_TIMER, HINT_TIMEOUT)
@ -118,6 +121,7 @@ class IvrDialog(IvrDialogBase):
if id == HINT_TIMER and self.state == collect:
self.enqueue(self.hint_msg, None)
elif id == CONF_TIMER and self.state == connected:
debug("conference timer timeout. disconnecting " + self.conf_to + " uri: " + self.conf_uri)
self.terminateOtherLeg()
self.bye()
self.stopSession()

59
apps/conference/AmRingTone.cpp
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@ -0,0 +1,59 @@
#include "AmRingTone.h"
#include <math.h>
#include "log.h"
#define PI 3.14159
AmRingTone::AmRingTone(int length, int on, int off, int f, int f2)
: AmAudio(),
length(length),
on_period(on),
off_period(off),
freq(f),freq2(f2)
{}
AmRingTone::~AmRingTone()
{}
int AmRingTone::read(unsigned int user_ts, unsigned int size)
{
int t = user_ts % ((on_period + off_period)<<3);
if(length < 0)
return -1;
if(t >= on_period<<3){
memset((unsigned char*)samples,0,size);
return size;
}
short* s = (short*)((unsigned char*)samples);
for(unsigned int i=0; i<PCM16_B2S(size); i++, s++, t++){
if(t < on_period<<3){
float fs = sin((float(t*freq)/8000.0)*2.0*PI)*15000.0;
if(freq2 != 0)
fs += sin((float(t*freq2)/8000.0)*2.0*PI)*15000.0;
*s = (short)(fs);
}
else
*s = 0;
}
if(length != 0){
length -= PCM16_B2S(size) / 8;
if(!length)
length--;
}
return size;
}
int AmRingTone::write(unsigned int user_ts, unsigned int size)
{
return -1;
}

23
apps/conference/AmRingTone.h
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@ -0,0 +1,23 @@
#ifndef _AmRingTone_h_
#define _AmRingTone_h_
#include "AmAudio.h"
class AmRingTone: public AmAudio
{
int on_period; // ms
int off_period;// ms
int freq; // Hz
int freq2; // Hz
int length;
public:
AmRingTone(int length, int on, int off, int f, int f2=0);
~AmRingTone();
int read(unsigned int user_ts, unsigned int size);
int write(unsigned int user_ts, unsigned int size);
};
#endif

228
apps/conference/Conference.cpp
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@ -70,11 +70,13 @@ int ConferenceFactory::onLoad()
JoinSound = cfg.getParameter("join_sound");
DropSound = cfg.getParameter("drop_sound");
DialoutSuffix = cfg.getParameter("dialout_suffix");
if(DialoutSuffix.empty()){
WARN("No dialout_suffix has been configured in the conference plug-in:\n");
WARN("\t -> dial out will not be available\n");
}
//RingTone = cfg.getParameter("ring_tone");
// DialoutSuffix = cfg.getParameter("dialout_suffix");
// if(DialoutSuffix.empty()){
// WARN("No dialout_suffix has been configured in the conference plug-in:\n");
// WARN("\t -> dial out will not be available\n");
// }
return 0;
}
@ -84,13 +86,28 @@ AmSession* ConferenceFactory::onInvite(const AmSipRequest& req)
return new ConferenceDialog(req.user);
}
AmSession* ConferenceFactory::onRefer(const AmSipRequest& req)
{
if(req.to_tag.empty())
throw AmSession::Exception(488,"Not accepted here");
AmSession* s = new ConferenceDialog(req.user);
s->dlg.local_tag = req.from_tag;
DBG("ConferenceFactory::onRefer: local_tag = %s\n",s->dlg.local_tag.c_str());
return s;
}
ConferenceDialog::ConferenceDialog(const string& conf_id,
AmConferenceChannel* dialout_channel)
: conf_id(conf_id),
channel(0),
play_list(this),
dialout_channel(dialout_channel),
state(CS_normal)
state(CS_normal),
allow_dialout(false)
{
dialedout = this->dialout_channel.get() != 0;
rtp_str.setAdaptivePlayout(true);
@ -107,14 +124,16 @@ void ConferenceDialog::onStart()
void ConferenceDialog::onSessionStart(const AmSipRequest& req)
{
setupAudio();
}
allow_dialout = (getHeader(req.hdrs,"P-Dialout") == "yes");
void ConferenceDialog::onSessionStart(const AmSipReply& reply)
{
setupAudio();
}
// void ConferenceDialog::onSessionStart(const AmSipReply& reply)
// {
// setupAudio();
// }
void ConferenceDialog::setupAudio()
{
if(!ConferenceFactory::JoinSound.empty()) {
@ -133,10 +152,19 @@ void ConferenceDialog::setupAudio()
DropSound.reset(0);
}
// if(!ConferenceFactory::RingTone.empty()) {
// RingTone.reset(new AmAudioFile());
// if(RingTone->open(ConferenceFactory::RingTone,
// AmAudioFile::Read))
// RingTone.reset(0);
// }
play_list.close();// !!!
if(dialout_channel.get()){
DBG("adding dialout_channel to the playlist (dialedout = %i)\n",dialedout);
play_list.addToPlaylist(new AmPlaylistItem(dialout_channel.get(),
dialout_channel.get()));
}
@ -238,13 +266,41 @@ void ConferenceDialog::process(AmEvent* ev)
case DoConfDisconnect:
DBG("****** Caller received DoConfDisconnect *******\n");
connectMainChannel();
state = CS_normal;
break;
case DoConfConnect:
state = CS_dialout_connected;
play_list.close(); // !!!
play_list.addToPlaylist(new AmPlaylistItem(dialout_channel.get(),
dialout_channel.get()));
break;
case DoConfRinging:
if(!RingTone.get())
RingTone.reset(new AmRingTone(0,2000,4000,440,480)); // US
DBG("adding ring tone to the playlist (dialedout = %i)\n",dialedout);
play_list.close();
play_list.addToPlaylist(new AmPlaylistItem(RingTone.get(),NULL));
break;
case DoConfError:
DBG("****** Caller received DoConfError *******\n");
if(!ErrorTone.get())
ErrorTone.reset(new AmRingTone(2000,250,250,440,480));
DBG("adding error tone to the playlist (dialedout = %i)\n",dialedout);
//play_list.close();
play_list.addToPlayListFront(new AmPlaylistItem(ErrorTone.get(),NULL));
break;
}
}
@ -269,15 +325,17 @@ string dtmf2str(int event)
}
}
void ConferenceDialog::onDtmf(int event, int duration)
{
DBG("ConferenceDialog::onDtmf, state = %d\n", state);
if(dialedout)
DBG("ConferenceDialog::onDtmf\n");
if(dialedout || !allow_dialout)
return;
switch(state){
case CS_normal:
DBG("CS_normal\n");
dtmf_seq += dtmf2str(event);
if(dtmf_seq.length() == 2){
@ -288,16 +346,17 @@ void ConferenceDialog::onDtmf(int event, int duration)
break;
case CS_dialing_out:{
DBG("CS_dialing_out\n");
string digit = dtmf2str(event);
if(digit == "*"){
if(!dtmf_seq.empty()){
createDialoutParticipant("sip:" + dtmf_seq +
ConferenceFactory::DialoutSuffix);
createDialoutParticipant(dtmf_seq);
state = CS_dialed_out;
}
else {
DBG("state = CS_normal; ????????\n");
state = CS_normal;
}
@ -310,6 +369,7 @@ void ConferenceDialog::onDtmf(int event, int duration)
case CS_dialout_connected:
DBG("CS_dialout_connected\n");
if(event == 10){ // '*'
AmSessionContainer::instance()
@ -319,17 +379,11 @@ void ConferenceDialog::onDtmf(int event, int duration)
connectMainChannel();
state = CS_normal;
} else if(event == 11){ // '#'
disconnectDialout();
state = CS_normal;
}
break;
case CS_dialed_out:
DBG("CS_dialed_out\n");
if(event == 11){ // '#'
disconnectDialout();
state = CS_normal;
}
@ -338,8 +392,10 @@ void ConferenceDialog::onDtmf(int event, int duration)
}
}
void ConferenceDialog::createDialoutParticipant(const string& uri)
void ConferenceDialog::createDialoutParticipant(const string& uri_user)
{
string uri = "sip:" + uri_user + "@" + dlg.domain;
dialout_channel.reset(AmConferenceStatus::getChannel(getLocalTag(),getLocalTag()));
dialout_id = AmSession::getNewId();
@ -363,11 +419,6 @@ void ConferenceDialog::createDialoutParticipant(const string& uri)
dialout_session->sdp.genRequest(AmConfig::LocalIP,local_port,body);
dialout_dlg.sendRequest("INVITE","application/sdp",body,"");
play_list.close(); // !!!
play_list.addToPlaylist(new AmPlaylistItem(dialout_channel.get(),
dialout_channel.get()));
dialout_session->start();
AmSessionContainer* sess_cont = AmSessionContainer::instance();
@ -392,6 +443,7 @@ void ConferenceDialog::disconnectDialout()
->postEvent(dialout_id,
new DialoutConfEvent(DoConfDisconnect,
getLocalTag()));
connectMainChannel();
}
}
@ -399,6 +451,7 @@ void ConferenceDialog::disconnectDialout()
void ConferenceDialog::connectMainChannel()
{
dialout_id = "";
dialedout = false;
dialout_channel.reset(NULL);
play_list.close();
@ -420,11 +473,46 @@ void ConferenceDialog::closeChannels()
dialout_channel.reset(NULL);
}
void ConferenceDialog::onSipRequest(const AmSipRequest& req)
{
AmSession::onSipRequest(req);
if((dlg.getStatus() >= AmSipDialog::Connected) ||
(req.method != "REFER"))
return;
std::swap(dlg.local_party,dlg.remote_party);
dlg.remote_tag = "";
dlg.setRoute(getHeader(req.hdrs,"P-Transfer-RR"));
dlg.next_hop = getHeader(req.hdrs,"P-Transfer-NH");
DBG("ConferenceDialog::onSipRequest: local_party = %s\n",dlg.local_party.c_str());
DBG("ConferenceDialog::onSipRequest: local_tag = %s\n",dlg.local_tag.c_str());
DBG("ConferenceDialog::onSipRequest: remote_party = %s\n",dlg.remote_party.c_str());
DBG("ConferenceDialog::onSipRequest: remote_tag = %s\n",dlg.remote_tag.c_str());
string body;
int local_port = rtp_str.getLocalPort();
sdp.genRequest(AmConfig::LocalIP,local_port,body);
dlg.sendRequest("INVITE","application/sdp",body,"");
transfer_req.reset(new AmSipRequest(req));
return;
}
void ConferenceDialog::onSipReply(const AmSipReply& reply)
{
int status = dlg.getStatus();
AmSession::onSipReply(reply);
DBG("ConferenceDialog::onSipReply: code = %i, reason = %s\n, status = %i",
reply.code,reply.reason.c_str(),dlg.getStatus());
if(!dialedout &&
!transfer_req.get())
return;
if(status < AmSipDialog::Connected){
switch(dlg.getStatus()){
@ -432,40 +520,90 @@ void ConferenceDialog::onSipReply(const AmSipReply& reply)
case AmSipDialog::Connected:
// connected!
if(acceptAudio(reply.body,reply.hdrs)){
ERROR("could not connect audio!!!\n");
dlg.bye();
setStopped();
}
else {
try {
acceptAudio(reply.body,reply.hdrs);
if(getDetached() && !getStopped()){
onSessionStart(reply);
setupAudio();
if(getInput() || getOutput())
AmMediaProcessor::instance()->addSession(this,
getCallgroup());
getCallgroup());
else {
ERROR("missing audio input and/or ouput.\n");
ERROR("missing audio input and/or ouput.\n");
return;
}
if(!transfer_req.get()){
// send connect event
AmSessionContainer::instance()
->postEvent(dialout_channel->getConfID(),
new DialoutConfEvent(DoConfConnect,
dialout_channel->getConfID()));
}
else {
dlg.reply(*(transfer_req.get()),202,"Accepted");
transfer_req.reset(0);
connectMainChannel();
}
// send connect event
AmSessionContainer::instance()
->postEvent(dialout_channel->getConfID(),
new DialoutConfEvent(DoConfConnect,
dialout_channel->getConfID()));
}
}
catch(const AmSession::Exception& e){
ERROR("%i %s\n",e.code,e.reason.c_str());
dlg.bye();
setStopped();
}
break;
case AmSipDialog::Pending:
switch(reply.code){
case 180: break;//TODO: local ring tone.
case 180:
// send ringing event
AmSessionContainer::instance()
->postEvent(dialout_channel->getConfID(),
new DialoutConfEvent(DoConfRinging,
dialout_channel->getConfID()));
break;
case 183: break;//TODO: remote ring tone.
default: break;// continue waiting.
}
break;
case AmSipDialog::Disconnected:
if(!transfer_req.get()){
disconnectDialout();
//switch(reply.code){
//default:
AmSessionContainer::instance()
->postEvent(dialout_channel->getConfID(),
new DialoutConfEvent(DoConfError,
dialout_channel->getConfID()));
//}
}
else {
dlg.reply(*(transfer_req.get()),reply.code,reply.reason);
transfer_req.reset(0);
setStopped();
}
break;
default: break;
}
}
}

20
apps/conference/Conference.h
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@ -33,6 +33,7 @@
#include "AmSession.h"
#include "AmConferenceChannel.h"
#include "AmPlaylist.h"
#include "AmRingTone.h"
#include <map>
#include <string>
@ -42,14 +43,17 @@ using std::string;
class ConferenceStatus;
class ConferenceStatusContainer;
enum { CS_normal=0,
CS_dialing_out,
CS_dialed_out,
CS_dialout_connected };
enum { DoConfConnect = 100,
DoConfDisconnect };
DoConfDisconnect,
DoConfRinging,
DoConfError
};
struct DialoutConfEvent : public AmEvent {
@ -61,7 +65,7 @@ struct DialoutConfEvent : public AmEvent {
conf_id(conf_id)
{}
};
class ConferenceFactory : public AmSessionFactory
{
public:
@ -69,9 +73,11 @@ public:
static string JoinSound;
static string DropSound;
static string DialoutSuffix;
//static string RingTone;
ConferenceFactory(const string& _app_name);
virtual AmSession* onInvite(const AmSipRequest&);
virtual AmSession* onRefer(const AmSipRequest& req);
virtual int onLoad();
};
@ -82,6 +88,8 @@ class ConferenceDialog : public AmSession
auto_ptr<AmAudioFile> LonelyUserFile;
auto_ptr<AmAudioFile> JoinSound;
auto_ptr<AmAudioFile> DropSound;
auto_ptr<AmRingTone> RingTone;
auto_ptr<AmRingTone> ErrorTone;
string conf_id;
@ -93,6 +101,10 @@ class ConferenceDialog : public AmSession
string dialout_id;
auto_ptr<AmConferenceChannel> dialout_channel;
bool allow_dialout;
auto_ptr<AmSipRequest> transfer_req;
void createDialoutParticipant(const string& uri);
void disconnectDialout();
@ -109,10 +121,10 @@ public:
void process(AmEvent* ev);
void onStart();
void onDtmf(int event, int duration);
void onSessionStart(const AmSipReply& reply);
void onSessionStart(const AmSipRequest& req);
void onBye(const AmSipRequest& req);
void onSipRequest(const AmSipRequest& req);
void onSipReply(const AmSipReply& reply);
};

27
apps/early_announce/EarlyAnnounce.cpp
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@ -73,20 +73,27 @@ int EarlyAnnounceFactory::onLoad()
void EarlyAnnounceDialog::onInvite(const AmSipRequest& req)
{
string sdp_reply;
if(acceptAudio(req.body,req.hdrs,&sdp_reply)!=0)
return;
try {
if(dlg.reply(req,183,"Session Progress",
"application/sdp",sdp_reply) != 0){
string sdp_reply;
acceptAudio(req.body,req.hdrs,&sdp_reply);
setStopped();
}
else {
if(dlg.reply(req,183,"Session Progress",
"application/sdp",sdp_reply) != 0){
localreq = req;
}
throw AmSession::Exception(500,"could not reply");
}
else {
localreq = req;
}
} catch(const AmSession::Exception& e) {
ERROR("%i %s\n",e.code,e.reason.c_str());
setStopped();
AmSipDialog::reply_error(req,e.code,e.reason);
}
}

2
apps/ivr/IvrAudio.h
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@ -3,7 +3,7 @@
// Python stuff
#include <Python.h>
#include "structmember.h"
#include <structmember.h>
#include "AmAudio.h"

11
apps/ivr/Makefile
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@ -1,22 +1,19 @@
plug_in_name = ivr
include Makefile.defs
PY_VER = $(PYTHON_VERSION)
PY_EXE = python$(PY_VER)
COREPATH ?=../../core
PYTHON_DIR = $(PYTHON_PREFIX)/include/python$(PY_VER)
PYTHON_LIBDIR = $(PYTHON_PREFIX)/lib/python$(PY_VER)
PYTHON_DIR = $(shell $(PY_EXE) ./python_inc.py)
PYTHON_LIBDIR = $(shell $(PY_EXE) ./python_lib.py)
# put used Python modules from lib-dynload here, e.g. time, mysql, _cvs.so etc.
PYTHON_DYNLOAD_MODULES = $(wildcard $(PYTHON_LIBDIR)/lib-dynload/*.so) \
$(wildcard $(PYTHON_LIBDIR)/site-packages/*.so)
PYTHON_module_cflags = -I$(PYTHON_DIR) -fno-strict-aliasing
PYTHON_module_ldflags = -Xlinker --export-dynamic \
-L$(PYTHON_LIBDIR)/config \
-lpython$(PY_VER) \
#$(PYTHON_DYNLOAD_MODULES)
-lpython$(PY_VER)
ifeq ($(TTS), y)
#

13
apps/ivr/Makefile.defs
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@ -1,19 +1,10 @@
# Python version: 2.5, 2.4 or 2.3
#
# PYTHON_VERSION might also be 2.2 -- except for the use of GIL
# do a ls /usr/include/python2.3/Python.h to see if it's there
PYTHON_VERSION = 2.4
# adjust to point to python include path
# can also be /usr/include/python$(PY_VER)
# look for Python.h in the specified path
# Python prefix is what you configured python with
# if you built from source (e.g. ./configure --with-prefix=/usr/local)
# on debian it's often /usr, on SuSE and FreeBSD /usr/local
PYTHON_PREFIX = /usr/local
PYTHON_VERSION ?= 2.4
# For flite text-to-speech set TTS = y
#
TTS = y
TTS ?= n
FLITE_DIR = /usr/src/flite-1.2-release
ALT_FLITE_DIR = /usr/include/flite

3
apps/ivr/python_inc.py
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@ -0,0 +1,3 @@
import distutils.sysconfig
print distutils.sysconfig.get_python_inc(True)

3
apps/ivr/python_lib.py
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@ -0,0 +1,3 @@
import distutils.sysconfig
print distutils.sysconfig.get_python_lib(True,True)

43
core/AmSdp.cpp
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@ -174,6 +174,7 @@ int AmSdp::parse()
}
telephone_event_pt = findPayload("telephone-event");
//DBG("telephone_event_pt = %i\n",telephone_event_pt);
return ret;
}
@ -205,7 +206,7 @@ int AmSdp::genResponse(const string& localip, int localport,
payloads += " " + int2str((*it)->payload_type);
if ((*it)->payload_type >= 96) // dynamic payload
//if ((*it)->payload_type >= 96) // dynamic payload
options += "a=rtpmap:" + int2str((*it)->payload_type) + " "
+ (*it)->encoding_name + "/" + int2str((*it)->clock_rate) + "\r\n";
@ -224,7 +225,7 @@ int AmSdp::genResponse(const string& localip, int localport,
+ options;
if (hasTelephoneEvent())
{
out_buf += "a=rtmap:" + int2str(telephone_event_pt->payload_type) + " " +
out_buf += "a=rtpmap:" + int2str(telephone_event_pt->payload_type) + " " +
telephone_event_pt->encoding_name + "/" +
int2str(telephone_event_pt->clock_rate) + "\r\n"
"a=fmtp:" + int2str(telephone_event_pt->payload_type) + " 0-15\r\n";
@ -271,12 +272,12 @@ int AmSdp::genRequest(const string& localip,int localport, string& out_buf)
for(it = payloads.begin();it != payloads.end();++it) {
if(it->first >= 96) {
//if(it->first >= 96) {
out_buf += "a=rtpmap:" + int2str(it->first)
+ " " + string(it->second->name)
+ "/" + int2str(it->second->sample_rate)
+ "\r\n";
}
//}
}
return 0;
@ -296,17 +297,26 @@ SdpPayload* AmSdp::getCompatiblePayload(int media_type, string& addr, int& port)
vector<SdpPayload>::iterator it = m_it->payloads.begin();
for(; it != m_it->payloads.end(); ++it ) {
if(it->payload_type < 96){ // static payload
if( pi->payload(it->payload_type) ) {
amci_payload_t* a_pl = NULL;
if(it->payload_type < 96){
// try static payloads
a_pl = pi->payload(it->payload_type);
}
if( a_pl ) {
payload = &(*it);
payload->int_pt = payload->payload_type;
goto end;
}
payload = &(*it);
payload->int_pt = a_pl->payload_id;
payload->encoding_name = a_pl->name;
payload->clock_rate = a_pl->sample_rate;
goto end;
}
else { // dynamic payload
else {
// Try dynamic payloads
// and give a chance to broken
// implementation using a static payload number
// for dynamic ones.
int int_pt = getDynPayload(it->encoding_name,
it->clock_rate);
@ -615,13 +625,14 @@ static bool parse_sdp_media(AmSdp* sdp_msg, char*& s, char*& next_line)
sdp_msg->media.push_back(m);
s = next_line;
DBG("next_line=<%s>\n",next_line);
ret = ret
// Media title
|| parse_sdp_line(sdp_msg,s,'i',true,NULL)
// connection information - optional if included at session-level
|| parse_sdp_line(sdp_msg,s,'c',true,parse_sdp_connection)
// bandwidth information
|| parse_sdp_line(sdp_msg,s,'b',true,NULL)
|| parse_sdp_line(sdp_msg,s,'b',true,NULL,false)
// encryption key
|| parse_sdp_line(sdp_msg,s,'k',true,NULL)
// zero or more media attribute lines
@ -633,12 +644,14 @@ static bool parse_sdp_media(AmSdp* sdp_msg, char*& s, char*& next_line)
}
next_line = get_next_line(s);
DBG("ret=%i; next_line=<%s>\n",ret,next_line);
return ret;
}
static bool parse_sdp_attribute(AmSdp* sdp_msg, char*& s, char*& next_line)
{
DBG("parse_sdp_attribute: s=%s\n",s);
if(sdp_msg->media.empty()){
ERROR("While parsing media options: no actual media !\n");
return true;
@ -672,6 +685,8 @@ static bool parse_sdp_attribute(AmSdp* sdp_msg, char*& s, char*& next_line)
}
parse_string_tok(s,params,'\0');
DBG("sdp attribute: pt=%u; enc=%s; cr=%u\n",
payload_type,encoding_name.c_str(),clock_rate);
vector<SdpPayload>::iterator pl_it;

67
core/AmSession.cpp
Unescape View File

@ -521,26 +521,29 @@ void AmSession::onSipReply(const AmSipReply& reply)
case AmSipDialog::Connected:
// connected!
if(acceptAudio(reply.body,reply.hdrs)){
ERROR("could not connect audio!!!\n");
dlg.bye();
setStopped();
}
else {
if(detached.get() && !getStopped()){
onSessionStart(reply);
try {
acceptAudio(reply.body,reply.hdrs);
if(detached.get() && !getStopped()){
if(input || output)
AmMediaProcessor::instance()->addSession(this,
callgroup);
else {
ERROR("missing audio input and/or ouput.\n");
}
}
}
break;
onSessionStart(reply);
if(input || output)
AmMediaProcessor::instance()->addSession(this,
callgroup);
else {
ERROR("missing audio input and/or ouput.\n");
}
}
}catch(const AmSession::Exception& e){
ERROR("could not connect audio!!!\n");
ERROR("%i %s\n",e.code,e.reason.c_str());
dlg.bye();
setStopped();
break;
}
break;
case AmSipDialog::Pending:
@ -556,15 +559,19 @@ void AmSession::onSipReply(const AmSipReply& reply)
void AmSession::onInvite(const AmSipRequest& req)
{
string sdp_reply;
if(acceptAudio(req.body,req.hdrs,&sdp_reply)!=0)
return;
try {
string sdp_reply;
if(dlg.reply(req,200,"OK",
"application/sdp",sdp_reply) != 0){
acceptAudio(req.body,req.hdrs,&sdp_reply);
if(dlg.reply(req,200,"OK",
"application/sdp",sdp_reply) != 0)
throw AmSession::Exception(500,"could not send response");
}catch(const AmSession::Exception& e){
//throw AmSession::Exception(500,"error while sending response");
ERROR("%i %s\n",e.code,e.reason.c_str());
setStopped();
AmSipDialog::reply_error(req,e.code,e.reason);
}
}
@ -612,10 +619,12 @@ int AmSession::acceptAudio(const string& body,
}
catch(const AmSession::Exception& e){
ERROR("%i %s\n",e.code,e.reason.c_str());
// if(dlg.reply(req,e.code,e.reason, "")){
// dlg.bye();
// }
setStopped();
throw;
// if(dlg.reply(req,e.code,e.reason, "")){
// dlg.bye();
// }
// setStopped();
}
return -1;

10
core/AmSession.h
Unescape View File

@ -149,16 +149,6 @@ public:
Exception(int c, string r) : code(c), reason(r) {}
};
// struct SessionTimerException : Exception {
// unsigned int minSE;
// SessionTimerException(unsigned int min_SE)
// : Exception(422, "Session Interval Too Small"),
// minSE(min_SE) { }
// string getErrorHeaders() const;
// };
/**
* Session constructor.
*/

18
core/AmSipDialog.cpp
Unescape View File

@ -31,6 +31,9 @@ void AmSipDialog::updateStatus(const AmSipRequest& req)
uas_trans[req.cseq] = AmSipTransaction(req.method,req.cseq);
remote_uri = req.from_uri;
sip_ip = req.dstip;
sip_port = req.port;
if(callid.empty()){
callid = req.callid;
remote_tag = req.from_tag;
@ -40,9 +43,6 @@ void AmSipDialog::updateStatus(const AmSipRequest& req)
remote_party = req.from;
local_party = req.to;
sip_ip = req.dstip;
sip_port = req.port;
setRoute(req.route);
next_hop = req.next_hop;
}
@ -200,14 +200,14 @@ string AmSipDialog::getContactHdr()
contact_uri = "Contact: <sip:";
if(user.empty() || !AmConfig::PrefixSep.empty())
contact_uri += CONTACT_USER_PREFIX;
contact_uri += CONTACT_USER_PREFIX;
if(!AmConfig::PrefixSep.empty())
contact_uri += AmConfig::PrefixSep;
contact_uri += AmConfig::PrefixSep;
if(!user.empty())
contact_uri += user;
contact_uri += user;
contact_uri += "@";
if(sip_ip.empty())

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