bool AmSession::isAudioSet()
{
lockAudio();
bool set = input || output;
unlockAudio();
return set;
}
void AmSession::setInOut(AmAudio* in,AmAudio* out)
{
lockAudio();
input = in;
output = out;
unlockAudio();
}
void AmSession::setOnHold(bool hold)
{
lockAudio();
bool old_hold = RTPStream()->getOnHold();
RTPStream()->setOnHold(hold);
if (hold != old_hold)
sendReinvite();
unlockAudio();
}
int AmSession::onSdpCompleted(const AmSdp& local_sdp, const AmSdp& remote_sdp)
{
DBG("AmSession::onSdpCompleted(...) ...\n");
if(local_sdp.media.empty() || remote_sdp.media.empty()) {
ERROR("Invalid SDP");
string debug_str;
local_sdp.print(debug_str);
ERROR("Local SDP:\n%s",
debug_str.empty() ? "<empty>"
: debug_str.c_str());
remote_sdp.print(debug_str);
ERROR("Remote SDP:\n%s",
debug_str.empty() ? "<empty>"
: debug_str.c_str());
return -1;
}
bool set_on_hold = false;
if (!remote_sdp.media.empty()) {
vector<SdpAttribute>::const_iterator pos =
std::find(remote_sdp.media[0].attributes.begin(), remote_sdp.media[0].attributes.end(), SdpAttribute("sendonly"));
set_on_hold = pos != remote_sdp.media[0].attributes.end();
}
lockAudio();
// TODO:
// - get the right media ID
// - check if the stream coresponding to the media ID
// should be created or updated
//
int ret = 0;
try {
ret = RTPStream()->init(local_sdp, remote_sdp, AmConfig::ForceSymmetricRtp);
} catch (const string& s) {
ERROR("Error while initializing RTP stream: '%s'\n", s.c_str());
ret = -1;
} catch (...) {
ERROR("Error while initializing RTP stream (unknown exception in AmRTPStream::init)\n");
ret = -1;
}
unlockAudio();
if (!isProcessingMedia()) {
setInbandDetector(AmConfig::DefaultDTMFDetector);
}
return ret;
}
int AmSession::acceptAudio(const string& body,
const string& hdrs,
string* sdp_reply)
{
try {
try {
// handle codec and send reply
string str_msg_flags = getHeader(hdrs,"P-MsgFlags");
unsigned int msg_flags = 0;
if(reverse_hex2int(str_msg_flags,msg_flags)){
ERROR("while parsing 'P-MsgFlags' header\n");
msg_flags = 0;
}
negotiate( body,
msg_flags & FL_FORCE_ACTIVE,
sdp_reply);
// enable RTP stream
lockAudio();
rtp_str.init(&payload);
unlockAudio();
DBG("Sending Rtp data to %s/%i\n",
rtp_str.getRHost().c_str(),rtp_str.getRPort());
return 0;
}
catch(const AmSession::Exception& e){ throw e; }
catch(const string& str){
ERROR("%s\n",str.c_str());
throw AmSession::Exception(500,str);
}
catch(...){
throw AmSession::Exception(500,"unexpected exception.");
}
}
catch(const AmSession::Exception& e){
ERROR("%i %s\n",e.code,e.reason.c_str());
throw;
// if(dlg.reply(req,e.code,e.reason, "")){
// dlg.bye();
// }
// setStopped();
}
return -1;
}
void AmSession::clearAudio()
{
lockAudio();
if(input){
input->close();
input = 0;
}
if(output){
output->close();
output = 0;
}
unlockAudio();
DBG("Audio cleared !!!\n");
postEvent(new AmAudioEvent(AmAudioEvent::cleared));
}
int AmSession::writeStreams(unsigned long long ts, unsigned char *buffer)
{
int res = 0;
lockAudio();
AmRtpAudio *stream = RTPStream();
if (stream->sendIntReached()) { // FIXME: shouldn't depend on checkInterval call before!
unsigned int f_size = stream->getFrameSize();
int got = 0;
if (output) got = output->get(ts, buffer, stream->getSampleRate(), f_size);
if (got < 0) res = -1;
if (got > 0) res = stream->put(ts, buffer, stream->getSampleRate(), got);
}
unlockAudio();
return res;
}
int AmSession::readStreams(unsigned long long ts, unsigned char *buffer)
{
int res = 0;
lockAudio();
AmRtpAudio *stream = RTPStream();
unsigned int f_size = stream->getFrameSize();
if (stream->checkInterval(ts)) {
int got = stream->get(ts, buffer, stream->getSampleRate(), f_size);
if (got < 0) res = -1;
if (got > 0) {
if (isDtmfDetectionEnabled())
putDtmfAudio(buffer, got, ts);
if (input) res = input->put(ts, buffer, stream->getSampleRate(), got);
}
}
unlockAudio();
return res;
}
void AmSession::setOutput(AmAudio* out)
{
lockAudio();
output = out;
unlockAudio();
}
void AmSession::setInput(AmAudio* in)
{
lockAudio();
input = in;
unlockAudio();
}
void AmSession::negotiate(const string& sdp_body,
bool force_symmetric_rtp,
string* sdp_reply)
{
string r_host = "";
int r_port = 0;
sdp.setBody(sdp_body.c_str());
if(sdp.parse())
throw AmSession::Exception(400,"session description parsing failed");
if(sdp.media.empty())
throw AmSession::Exception(400,"no media line found in SDP message");
SdpPayload* tmp_pl = sdp.getCompatiblePayload(MT_AUDIO,r_host,r_port);
if(!tmp_pl)
throw AmSession::Exception(606,"could not find compatible payload");
if(payload.int_pt == -1){
payload = *tmp_pl;
DBG("new payload: %i\n",payload.int_pt);
}
else if(payload.int_pt != tmp_pl->int_pt){
DBG("old payload: %i; new payload: %i\n",payload.int_pt,tmp_pl->int_pt);
throw AmSession::Exception(400,"do not accept payload changes");
}
const SdpPayload *telephone_event_payload = sdp.telephoneEventPayload();
if(telephone_event_payload)
{
DBG("remote party supports telephone events (pt=%i)\n",
telephone_event_payload->payload_type);
lockAudio();
rtp_str.setTelephoneEventPT(telephone_event_payload);
unlockAudio();
}
else {
DBG("remote party doesn't support telephone events\n");
}
bool passive_mode = false;
if( sdp.remote_active || force_symmetric_rtp) {
DBG("The other UA is NATed: switched to passive mode.\n");
DBG("remote_active = %i; force_symmetric_rtp = %i\n",
sdp.remote_active,force_symmetric_rtp);
passive_mode = true;
}
lockAudio();
rtp_str.setLocalIP(AmConfig::LocalIP);
rtp_str.setPassiveMode(passive_mode);
rtp_str.setRAddr(r_host, r_port);
unlockAudio();
if(sdp_reply)
sdp.genResponse(AmConfig::LocalIP,rtp_str.getLocalPort(),*sdp_reply);
}
