From 009905f451df2fc648fd7c26eefa41fbb22aad3f Mon Sep 17 00:00:00 2001
From: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
Date: Mon, 1 Nov 2010 13:04:58 -0400
Subject: [PATCH] [#4367] use symetric audio rtp session, not dual
---
sflphone-common/src/audio/audiortp/AudioRtpSession.cpp | 9 +--------
sflphone-common/src/audio/audiortp/AudioRtpSession.h | 3 ++-
2 files changed, 3 insertions(+), 9 deletions(-)
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
index fbe0079f6d..4f43da123e 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.cpp
@@ -49,7 +49,7 @@ timeval2microtimeout(const timeval& t)
AudioRtpSession::AudioRtpSession (ManagerImpl * manager, SIPCall * sipcall) :
// ost::SymmetricRTPSession (ost::InetHostAddress (sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort()),
AudioRtpRecordHandler(manager, sipcall),
- ost::TRTPSessionBase<ost::DualRTPUDPIPv4Channel,ost::DualRTPUDPIPv4Channel,ost::AVPQueue>(ost::InetHostAddress (sipcall->getLocalIp().c_str()),
+ ost::TRTPSessionBase<ost::SymmetricRTPChannel,ost::SymmetricRTPChannel,ost::AVPQueue>(ost::InetHostAddress (sipcall->getLocalIp().c_str()),
sipcall->getLocalAudioPort(),
0,
ost::MembershipBookkeeping::defaultMembersHashSize,
@@ -252,12 +252,8 @@ void AudioRtpSession::sendMicData()
_timestampCount = 0;
}
- // getCurrentTimestamp();
- // RTPDataQueue::getTimestampIncrement();
-
// Increment timestamp for outgoing packet
_timestamp += _timestampIncrement;
- _debug("sendMicData: %d, timestamp increment %d", _timestamp, _timestampIncrement);
// putData put the data on RTP queue, sendImmediate bypass this queue
putData (_timestamp, getMicDataEncoded(), compSize);
@@ -275,7 +271,6 @@ void AudioRtpSession::receiveSpeakerData ()
adu = getData (packetTimestamp);
if (!adu) {
- _debug("receiveSpeakerData: no data!");
return;
}
@@ -378,9 +373,7 @@ void AudioRtpSession::run ()
// make sure the scheduling timeout is
// <= the check interval for RTCP
// packets
- _debug("timeout before: %d, maxwait %d", timeout, maxWait);
timeout = (timeout > maxWait)? maxWait : timeout;
- _debug("timeout after: %d", timeout);
if ( timeout < 1000 ) { // !(timeout/1000)
setCancel(cancelDeferred);
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.h b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
index 3cc8477dbd..b948e5cf5a 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
@@ -64,7 +64,8 @@ static const int g722PayloadType = 9;
static const int g722RtpClockRate = 8000;
static const int g722RtpTimeincrement = 160;
-class AudioRtpSession : protected ost::Thread, public ost::TimerPort, public AudioRtpRecordHandler, public ost::TRTPSessionBase<ost::DualRTPUDPIPv4Channel,ost::DualRTPUDPIPv4Channel,ost::AVPQueue>
+// class AudioRtpSession : protected ost::Thread, public ost::TimerPort, public AudioRtpRecordHandler, public ost::TRTPSessionBase<ost::DualRTPUDPIPv4Channel,ost::DualRTPUDPIPv4Channel,ost::AVPQueue>
+class AudioRtpSession : protected ost::Thread, public ost::TimerPort, public AudioRtpRecordHandler, public ost::TRTPSessionBase<ost::SymmetricRTPChannel, ost::SymmetricRTPChannel, ost::AVPQueue>
{
public:
/**
--
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