#4702: Send RTP packets immediately, no need of outgoing queue

sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h

@@ -47,8 +47,8 @@ namespace sfl
 // Factor use to increase volume in fade in
 #define FADEIN_STEP_SIZE 4;
 
-static const int schedulingTimeout = 1000000;
-static const int expireTimeout = 10000000;
+static const int schedulingTimeout = 8000;
+static const int expireTimeout = 1000000;
 
 // G.722 VoIP is typically carried in RTP payload type 9.[2] Note that IANA records the clock rate for type 9 G.722 as 8 kHz
 // (instead of 16 kHz), RFC3551[3]  clarifies that this is due to a historical error and is retained in order to maintain backward

sflphone-common/src/audio/audiortp/AudioRtpSession.cpp

@@ -198,7 +198,8 @@ void AudioRtpSession::sendDtmfEvent (sfl::DtmfEvent *dtmf)
     if (dtmf->newevent)
         setMark (true);
 
-    putData (_timestamp, (const unsigned char*) (& (dtmf->payload)), sizeof (ost::RTPPacket::RFC2833Payload));
+    // putData (_timestamp, (const unsigned char*) (& (dtmf->payload)), sizeof (ost::RTPPacket::RFC2833Payload));
+    sendImmediate (_timestamp, (const unsigned char*) (& (dtmf->payload)), sizeof (ost::RTPPacket::RFC2833Payload));
 
     // This is no more a new event
     if (dtmf->newevent) {
@@ -253,7 +254,8 @@ void AudioRtpSession::sendMicData()
     _timestamp += _timestampIncrement;
 
     // putData put the data on RTP queue, sendImmediate bypass this queue
-    putData (_timestamp, getMicDataEncoded(), compSize);
+    // putData (_timestamp, getMicDataEncoded(), compSize);
+    sendImmediate (_timestamp, getMicDataEncoded(), compSize);
 }
 
 
@@ -375,9 +377,10 @@ void AudioRtpSession::run ()
         // packets
         timeout = (timeout > maxWait) ? maxWait : timeout;
 
+
         if (timeout < 1000) {   // !(timeout/1000)
             setCancel (cancelDeferred);
-            dispatchDataPacket();
+            // dispatchDataPacket();
             setCancel (cancelImmediate);
             timerTick();
         } else {