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Fixed DTMF to follow the protocol instead of the hacky ortp business.

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git-svn-id: http://wush.net/svn/range/software/public/openbts/trunk@4523 19bc5d8c-e614-43d4-8b26-e1612bc8e597
This commit is contained in:
kurtis.heimerl
2012-11-24 10:32:27 +00:00
parent adbe838ec8
commit 28d6c3c0fc
1 changed files with 56 additions and 7 deletions
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63
SIP/SIPEngine.cpp
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@@ -54,7 +54,34 @@ using namespace std;
using namespace SIP;
using namespace Control;
int get_rtp_tev_type(char dtmf){
switch (dtmf){
case '1': return TEV_DTMF_1;
case '2': return TEV_DTMF_2;
case '3': return TEV_DTMF_3;
case '4': return TEV_DTMF_4;
case '5': return TEV_DTMF_5;
case '6': return TEV_DTMF_6;
case '7': return TEV_DTMF_7;
case '8': return TEV_DTMF_8;
case '9': return TEV_DTMF_9;
case '0': return TEV_DTMF_0;
case '*': return TEV_DTMF_STAR;
case '#': return TEV_DTMF_POUND;
case 'a':
case 'A': return TEV_DTMF_A;
case 'B':
case 'b': return TEV_DTMF_B;
case 'C':
case 'c': return TEV_DTMF_C;
case 'D':
case 'd': return TEV_DTMF_D;
case '!': return TEV_FLASH;
default:
LOG(WARNING) << "Bad dtmf: " << dtmf;
return -1;
}
}
const char* SIP::SIPStateString(SIPState s)
@@ -1135,12 +1162,19 @@ bool SIPEngine::startDTMF(char key)
{
LOG (DEBUG) << key;
if (mState!=Active) return false;
if (get_rtp_tev_type(key) < 0){
return false;
}
mDTMF = key;
mDTMFDuration = 0;
mDTMFStartTime = mTxTime;
int code = rtp_session_send_dtmf2(mSession,mDTMF,mDTMFStartTime,mDTMFDuration);
//true means start
mblk_t *m = rtp_session_create_telephone_event_packet(mSession,true);
//volume 10 for some magic reason, false means not end
int code = rtp_session_add_telephone_event(mSession,m,get_rtp_tev_type(mDTMF),false,10,mDTMFDuration);
int bytes = rtp_session_sendm_with_ts(mSession,m,mDTMFStartTime);
mDTMFDuration += 160;
if (!code) return true;
if (!code && bytes > 0) return true;
// Error? Turn off DTMF sending.
LOG(WARNING) << "DTMF RFC-2833 failed on start.";
mDTMF = '\0';
@@ -1149,7 +1183,19 @@ bool SIPEngine::startDTMF(char key)
void SIPEngine::stopDTMF()
{
//false means not start
mblk_t *m = rtp_session_create_telephone_event_packet(mSession,false);
//volume 10 for some magic reason, end is true
int code = rtp_session_add_telephone_event(mSession,m,get_rtp_tev_type(mDTMF),true,10,mDTMFDuration);
int bytes = rtp_session_sendm_with_ts(mSession,m,mDTMFStartTime);
mDTMFDuration += 160;
LOG (DEBUG) << "DTMF RFC-2833 sending " << mDTMF << " " << mDTMFDuration;
// Turn it off if there's an error.
if (code || bytes <= 0) {
LOG(ERR) << "DTMF RFC-2833 failed at end";
}
mDTMF='\0';
}
@@ -1160,15 +1206,18 @@ void SIPEngine::txFrame(unsigned char* frame )
// HACK -- Hardcoded for GSM/8000.
// FIXME -- Make this work for multiple vocoder types.
rtp_session_send_with_ts(mSession, frame, 33, mTxTime);
mTxTime += 160;
mTxTime += 160;
if (mDTMF) {
// Any RFC-2833 action?
int code = rtp_session_send_dtmf2(mSession,mDTMF,mDTMFStartTime,mDTMFDuration);
//false means not start
mblk_t *m = rtp_session_create_telephone_event_packet(mSession,false);
//volume 10 for some magic reason, false means not end
int code = rtp_session_add_telephone_event(mSession,m,get_rtp_tev_type(mDTMF),false,10,mDTMFDuration);
int bytes = rtp_session_sendm_with_ts(mSession,m,mDTMFStartTime);
mDTMFDuration += 160;
LOG (DEBUG) << "DTMF RFC-2833 sending " << mDTMF << " " << mDTMFDuration;
// Turn it off if there's an error.
if (code) {
if (code || bytes <=0) {
LOG(ERR) << "DTMF RFC-2833 failed after start.";
mDTMF='\0';
}