/** Hook called when an SDP offer is required */
bool AmSession::getSdpOffer(AmSdp& offer)
{
DBG("AmSession::getSdpOffer(...) ...\n");
offer.version = 0;
offer.origin.user = "******";
//offer.origin.sessId = 1;
//offer.origin.sessV = 1;
offer.sessionName = "sems";
offer.conn.network = NT_IN;
offer.conn.addrType = AT_V4;
offer.conn.address = advertisedIP();
// TODO: support mutiple media types (needs multiples RTP streams)
// TODO: support update instead of clearing everything
if(RTPStream()->getSdpMediaIndex() < 0)
offer.media.clear();
unsigned int media_idx = 0;
if(!offer.media.size()) {
offer.media.push_back(SdpMedia());
offer.media.back().type=MT_AUDIO;
}
else {
media_idx = RTPStream()->getSdpMediaIndex();
}
RTPStream()->setLocalIP(localMediaIP());
RTPStream()->getSdpOffer(media_idx,offer.media.back());
return true;
}
void AmSession::setOnHold(bool hold)
{
lockAudio();
bool old_hold = RTPStream()->getOnHold();
RTPStream()->setOnHold(hold);
if (hold != old_hold)
sendReinvite();
unlockAudio();
}
string AmSession::sid4dbg()
{
string dbg;
dbg = dlg->getCallid() + "/" + dlg->getLocalTag() + "/"
+ dlg->getRemoteTag() + "/"
+ int2str(RTPStream()->getLocalPort()) + "/"
+ RTPStream()->getRHost() + ":" + int2str(RTPStream()->getRPort());
return dbg;
}
CallBackDialog::CallBackDialog(AmPromptCollection& prompts,
UACAuthCred* cred)
: play_list(this), prompts(prompts), cred(cred),
state(CBNone)
{
// set configured playout type
RTPStream()->setPlayoutType(CallBackFactory::m_PlayoutType);
}
CallBackCalleeDialog::CallBackCalleeDialog(const string& other_tag,
AmSessionAudioConnector* connector,
UACAuthCred* cred)
: AmB2ABCalleeSession(other_tag, connector), cred(cred)
{
// set configured playout type
RTPStream()->setPlayoutType(CallBackFactory::m_PlayoutType);
setDtmfDetectionEnabled(false);
}
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;
}
WebConferenceDialog::WebConferenceDialog(AmPromptCollection& prompts,
WebConferenceFactory* my_f,
UACAuthCred* cred)
: play_list(this), separator(this, 0), prompts(prompts), state(None),
factory(my_f), cred(cred), muted(false),
connect_ts(-1), disconnect_ts(-1)
{
is_dialout = (cred != NULL);
accept_early_session = is_dialout;
// set configured playout type
RTPStream()->setPlayoutType(WebConferenceFactory::m_PlayoutType);
}
WebConferenceDialog::WebConferenceDialog(AmPromptCollection& prompts,
WebConferenceFactory* my_f,
const string& room)
: play_list(this), separator(this, 0), prompts(prompts), state(None),
factory(my_f), muted(false),
connect_ts(-1), disconnect_ts(-1)
{
conf_id = room;
DBG("set conf_id to %s\n", conf_id.c_str());
is_dialout = false;
// set configured playout type
RTPStream()->setPlayoutType(WebConferenceFactory::m_PlayoutType);
}
/* 循环产生一个能唯一标识该RTPSession的RTPStream数组中新的RTPStream的随机数(作为SSRC),搭建一个新的RTPStream,放入RTPStream的数组 */
QTSS_Error RTPSession::AddStream(RTSPRequestInterface* request, RTPStream** outStream, QTSS_AddStreamFlags inFlags)
{
Assert(outStream != NULL);
// Create a new SSRC for this stream. This should just be a random number unique
// to all the streams in the session
/* 循环产生一个能唯一标识该RTPSession的RTPStream数组中新的RTPStream的随机数,作为SSRC(它和该数组中现存的每个RTPStream的SSRC都不同) */
UInt32 theSSRC = 0;
while (theSSRC == 0)
{
/* 产生随机数,它在该RTPSession的RTPStream数组中唯一标识一个RTPSream */
theSSRC = (SInt32)::rand();
RTPStream** theStream = NULL;
UInt32 theLen = 0;
/* 遍历RTPSession中的每个RTPStream,若它的SSRC恰和theSSRC相同,就将theSSRC置0,使得该循环继续 */
for (int x = 0; this->GetValuePtr(qtssCliSesStreamObjects, x, (void**)&theStream, &theLen) == QTSS_NoErr; x++)
{
Assert(theStream != NULL);
Assert(theLen == sizeof(RTPStream*));
if (*theStream != NULL)
if ((*theStream)->GetSSRC() == theSSRC)
theSSRC = 0;
}
}
/***************** 新建一个RTPStream对象,它有唯一标识它的SSRC(随机数) *************/
*outStream = NEW RTPStream(theSSRC, this);
/***************** 新建一个RTPStream对象 *************/
/* 以指定加入方式搭建该RTSP Request的RTPStream */
QTSS_Error theErr = (*outStream)->Setup(request, inFlags);
/* 假如搭建不成功,就删去新建RTPStream对象 */
if (theErr != QTSS_NoErr)
// If we couldn't setup the stream, make sure not to leak memory!
delete *outStream;
/* 假如搭建成功,将其放入RTPStream Array中,注意下面qtssCliSesStreamObjects是个多值索引的属性 */
else
{
// If the stream init succeeded, then put it into the array of setup streams
theErr = this->SetValue(qtssCliSesStreamObjects, this->GetNumValues(qtssCliSesStreamObjects),
outStream, sizeof(RTPStream*), QTSSDictionary::kDontObeyReadOnly);
Assert(theErr == QTSS_NoErr);
fHasAnRTPStream = true;/* 现在有一个RTPStream */
}
return theErr;
}
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;
}
QTSS_Error RTPSession::AddStream(RTSPRequestInterface* request, RTPStream** outStream,
QTSS_AddStreamFlags inFlags)
{
Assert(outStream != NULL);
// Create a new SSRC for this stream. This should just be a random number unique
// to all the streams in the session
UInt32 theSSRC = 0;
while (theSSRC == 0)
{
theSSRC = (SInt32)::rand();
RTPStream** theStream = NULL;
UInt32 theLen = 0;
for (int x = 0; this->GetValuePtr(qtssCliSesStreamObjects, x, (void**)&theStream, &theLen) == QTSS_NoErr; x++)
{
Assert(theStream != NULL);
Assert(theLen == sizeof(RTPStream*));
if (*theStream != NULL)
if ((*theStream)->GetSSRC() == theSSRC)
theSSRC = 0;
}
}
*outStream = NEW RTPStream(theSSRC, this);
QTSS_Error theErr = (*outStream)->Setup(request, inFlags);
if (theErr != QTSS_NoErr)
// If we couldn't setup the stream, make sure not to leak memory!
delete *outStream;
else
{
// If the stream init succeeded, then put it into the array of setup streams
theErr = this->SetValue(qtssCliSesStreamObjects, this->GetNumValues(qtssCliSesStreamObjects),
outStream, sizeof(RTPStream*), QTSSDictionary::kDontObeyReadOnly);
Assert(theErr == QTSS_NoErr);
fHasAnRTPStream = true;
}
return theErr;
}
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;
}
int AmB2BSession::onSdpCompleted(const AmSdp& local_sdp, const AmSdp& remote_sdp)
{
if (rtp_relay_mode != RTP_Direct) {
if (!media_session) {
// report missing media session (here we get for rtp_relay_mode == RTP_Relay)
ERROR("BUG: media session is missing, can't update SDP\n");
}
else {
media_session->updateStreams(a_leg, local_sdp, remote_sdp, this);
}
}
if(hasRtpStream() && RTPStream()->getSdpMediaIndex() >= 0) {
if(!sip_relay_only){
return AmSession::onSdpCompleted(local_sdp,remote_sdp);
}
DBG("sip_relay_only = true: doing nothing!\n");
}
return 0;
}
/** Hook called when an SDP answer is required */
bool AmSession::getSdpAnswer(const AmSdp& offer, AmSdp& answer)
{
DBG("AmSession::getSdpAnswer(...) ...\n");
answer.version = 0;
answer.origin.user = "******";
//answer.origin.sessId = 1;
//answer.origin.sessV = 1;
answer.sessionName = "sems";
answer.conn.network = NT_IN;
if (offer.conn.address.empty()) answer.conn.addrType = AT_V4; // or use first stream connection?
else answer.conn.addrType = offer.conn.addrType;
answer.conn.address = advertisedIP(answer.conn.addrType);
answer.media.clear();
bool audio_1st_stream = true;
unsigned int media_index = 0;
for(vector<SdpMedia>::const_iterator m_it = offer.media.begin();
m_it != offer.media.end(); ++m_it) {
answer.media.push_back(SdpMedia());
SdpMedia& answer_media = answer.media.back();
if( m_it->type == MT_AUDIO
&& m_it->transport == TP_RTPAVP
&& audio_1st_stream
&& (m_it->port != 0) ) {
RTPStream()->setLocalIP(localMediaIP(answer.conn.addrType));
RTPStream()->getSdpAnswer(media_index,*m_it,answer_media);
if(answer_media.payloads.empty() ||
((answer_media.payloads.size() == 1) &&
(answer_media.payloads[0].encoding_name == "telephone-event"))
){
// no compatible media found
throw Exception(488,"no compatible payload");
}
audio_1st_stream = false;
}
else {
answer_media.type = m_it->type;
answer_media.port = 0;
answer_media.nports = 0;
answer_media.transport = m_it->transport;
answer_media.send = false;
answer_media.recv = false;
answer_media.payloads.clear();
if(!m_it->payloads.empty()) {
SdpPayload dummy_pl = m_it->payloads.front();
dummy_pl.encoding_name.clear();
dummy_pl.sdp_format_parameters.clear();
answer_media.payloads.push_back(dummy_pl);
}
answer_media.attributes.clear();
}
// sort payload type in the answer according to the priority given in the codec_order configuration key
std::stable_sort(answer_media.payloads.begin(),answer_media.payloads.end(),codec_priority_cmp());
media_index++;
}
return true;
}
Click2TransSession::Click2TransSession(Click2TransDialog* parentDialog) :
dialog(parentDialog),ringTone(new AmRingTone(0,2000,4000,440,480))
{
RTPStream()->setPlayoutType(ADAPTIVE_PLAYOUT);
setCallgroup(dialog->getID());
}
void AmSession::sendDtmf(int event, unsigned int duration_ms) {
RTPStream()->sendDtmf(event, duration_ms);
}
void AmSession::setInbandDetector(Dtmf::InbandDetectorType t)
{
m_dtmfDetector.setInbandDetector(t, RTPStream()->getSampleRate());
}
int AmSession::getRPort()
{
return RTPStream()->getRPort();
}
PinAuthConferenceDialog::PinAuthConferenceDialog(AmPromptCollection& prompts)
: play_list(this), separator(this, 0), prompts(prompts), state(None)
{
// set configured playout type
RTPStream()->setPlayoutType(PinAuthConferenceFactory::m_PlayoutType);
}
