From 8c5e5dd5b68bc25ccd881e25f3a7280425fb5821 Mon Sep 17 00:00:00 2001
From: Alexandre Savard <alexandre.savard@savoirfairelinux.com>
Date: Fri, 16 Jul 2010 17:19:32 -0400
Subject: [PATCH] [#3366] Handle early media received from RTP
---
sflphone-common/src/audio/alsa/alsalayer.cpp | 8 +-
.../src/audio/audiortp/AudioRtpSession.h | 13 ++-
.../audio/audiortp/AudioSymmetricRtpSession.h | 2 +-
.../src/audio/pulseaudio/pulselayer.cpp | 11 +-
sflphone-common/src/managerimpl.cpp | 46 ++++----
sippxml/testEarlyMedia.xml | 107 ++++++++++++++++++
6 files changed, 150 insertions(+), 37 deletions(-)
create mode 100644 sippxml/testEarlyMedia.xml
diff --git a/sflphone-common/src/audio/alsa/alsalayer.cpp b/sflphone-common/src/audio/alsa/alsalayer.cpp
index 89e6f41513..5c19f3da8c 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.cpp
+++ b/sflphone-common/src/audio/alsa/alsalayer.cpp
@@ -947,7 +947,9 @@ void AlsaLayer::audioCallback(void)
} else {
- if (tone) {
+ normalAvailBytes = getMainBuffer()->availForGet();
+
+ if (tone && (normalAvailBytes <= 0)) {
out = (SFLDataFormat *) malloc (playbackAvailBytes);
tone->getNext (out, playbackAvailSmpl, spkrVolume);
@@ -957,7 +959,7 @@ void AlsaLayer::audioCallback(void)
out = 0;
}
- else if (file_tone && !_RingtoneHandle) {
+ else if (file_tone && !_RingtoneHandle && (normalAvailBytes <= 0)) {
out = (SFLDataFormat *) malloc (playbackAvailBytes);
file_tone->getNext (out, playbackAvailSmpl, spkrVolume);
@@ -985,8 +987,6 @@ void AlsaLayer::audioCallback(void)
}
-
- normalAvailBytes = getMainBuffer()->availForGet();
toGet = (normalAvailBytes < (int) maxNbBytesToGet) ? normalAvailBytes : maxNbBytesToGet;
out = (SFLDataFormat*) malloc (maxNbBytesToGet);
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.h b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
index 5144d37679..cd4f7edc3c 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
@@ -574,8 +574,10 @@ namespace sfl {
// set available byte to maxByteToGet
int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
- if (bytesAvail == 0)
- return 0;
+ if (bytesAvail == 0) {
+ memset(_micDataEncoded, 0, sizeof(SFLDataFormat));
+ return _audiocodec->getFrameSize();
+ }
// Get bytes from micRingBuffer to data_from_mic
int nbSample = _manager->getAudioDriver()->getMainBuffer()->getData(_micData , bytesAvail, 100, _ca->getCallId()) / sizeof (SFLDataFormat);
@@ -693,6 +695,7 @@ namespace sfl {
template <typename D>
void AudioRtpSession<D>::receiveSpeakerData ()
{
+
if (!_audiolayer) {
_debug ("No audiolayer available for speaker");
return;
@@ -707,8 +710,9 @@ namespace sfl {
int packetTimestamp = static_cast<D*>(this)->getFirstTimestamp();
adu = static_cast<D*>(this)->getData(packetTimestamp);
- if(!adu)
+ if(!adu) {
return;
+ }
unsigned char* spkrDataIn = NULL;
unsigned int size = 0;
@@ -723,8 +727,6 @@ namespace sfl {
spkrDataIn = (unsigned char*) adu->getData(); // data in char
size = adu->getSize(); // size in char
-
-
result = jb_put(_jbuffer, spkrDataIn, JB_TYPE_VOICE, _packetLength, _ts+=20, _currentTime);
}
@@ -758,6 +760,7 @@ namespace sfl {
template <typename D>
void AudioRtpSession<D>::run ()
{
+
// Timestamp must be initialized randomly
_timestamp = static_cast<D*>(this)->getCurrentTimestamp();
diff --git a/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.h b/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.h
index 0713f3db0c..c1260a6fc3 100644
--- a/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioSymmetricRtpSession.h
@@ -40,7 +40,7 @@ namespace sfl {
{
public:
AudioSymmetricRtpSession(ManagerImpl * manager, SIPCall * sipcall) :
- ost::SymmetricRTPSession(ost::InetHostAddress(sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort()),
+ ost::SymmetricRTPSession(ost::InetHostAddress(sipcall->getLocalIp().c_str()), sipcall->getLocalAudioPort()),
AudioRtpSession<AudioSymmetricRtpSession>(manager, sipcall)
{
_debug("AudioSymmetricRtpSession initialized\n");
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index 403ec11ad0..667f0cf272 100755
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -798,7 +798,6 @@ void PulseLayer::writeToSpeaker (void)
SFLDataFormat* out;// = (SFLDataFormat*)pa_xmalloc(framesPerBuffer);
urgentAvailBytes = _urgentRingBuffer.AvailForGet();
-
// available bytes to be written in pulseaudio internal buffer
int writeableSize = pa_stream_writable_size (playback->pulseStream());
@@ -821,12 +820,16 @@ void PulseLayer::writeToSpeaker (void)
} else {
+ // Get ringtone
AudioLoop* tone = _manager->getTelephoneTone();
+ // We must test if data have been received from network in case of early media
+ normalAvailBytes = getMainBuffer()->availForGet();
+
// flush remaining samples in _urgentRingBuffer
flushUrgent();
- if (tone != 0) {
+ if ((tone != 0) && (normalAvailBytes <= 0)) {
if (playback->getStreamState() == PA_STREAM_READY) {
@@ -845,9 +848,7 @@ void PulseLayer::writeToSpeaker (void)
int maxNbBytesToGet = 0;
-
// test if audio resampling is needed
-
if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
// upsamplefactor is used to compute the number of bytes to get in the ring buffer
@@ -862,8 +863,6 @@ void PulseLayer::writeToSpeaker (void)
}
- normalAvailBytes = getMainBuffer()->availForGet();
-
byteToGet = (normalAvailBytes < (int) (maxNbBytesToGet)) ? normalAvailBytes : maxNbBytesToGet;
if (byteToGet) {
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index 8eafb6212d..662c7f1646 100755
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -1355,38 +1355,42 @@ void ManagerImpl::addStream (const CallID& call_id) {
if (participToConference(call_id)) {
- // bind to conference participant
- ConferenceMap::iterator iter = _conferencemap.find(call->getConfId());
+ _debug("Manager: Add stream to conference");
- if (iter != _conferencemap.end()) {
- Conference* conf = iter->second;
+ // bind to conference participant
+ ConferenceMap::iterator iter = _conferencemap.find(call->getConfId());
- conf->bindParticipant(call_id);
+ if (iter != _conferencemap.end()) {
+ Conference* conf = iter->second;
- ParticipantSet participants = conf->getParticipantList();
- // reset ring buffer for all conference participant
- ParticipantSet::iterator iter_p = participants.begin();
+ conf->bindParticipant(call_id);
- while (iter_p != participants.end()) {
+ ParticipantSet participants = conf->getParticipantList();
+ // reset ring buffer for all conference participant
+ ParticipantSet::iterator iter_p = participants.begin();
- // to avoid puting onhold the call
- // switchCall("");
- _audiodriver->getMainBuffer()->flush(*iter_p);
+ while (iter_p != participants.end()) {
- iter_p++;
- }
+ // to avoid puting onhold the call
+ // switchCall("");
+ _audiodriver->getMainBuffer()->flush(*iter_p);
+
+ iter_p++;
+ }
- _audiodriver->getMainBuffer()->flush(default_id);
- }
+ _audiodriver->getMainBuffer()->flush(default_id);
+ }
} else {
- // bind to main
- getAudioDriver()->getMainBuffer()->bindCallID(call_id);
+ _debug("Manager: Add stream to call");
- // _audiodriver->getMainBuffer()->flush(default_id);
- _audiodriver->flushUrgent();
- _audiodriver->flushMain();
+ // bind to main
+ getAudioDriver()->getMainBuffer()->bindCallID(call_id);
+
+ // _audiodriver->getMainBuffer()->flush(default_id);
+ _audiodriver->flushUrgent();
+ _audiodriver->flushMain();
}
diff --git a/sippxml/testEarlyMedia.xml b/sippxml/testEarlyMedia.xml
new file mode 100644
index 0000000000..28f34ff4f0
--- /dev/null
+++ b/sippxml/testEarlyMedia.xml
@@ -0,0 +1,107 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- sudo sipp -sf testEarlyMedia.xml 127.0.0.1 -i 127.0.0.1 -p 5062 -l 1 -m 1 -mp 6000 -->
+
+<scenario name="Basic UAS responder">
+
+ <recv request="INVITE" crlf="true">
+ </recv>
+
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 100 Ringing
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 183 Trying
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Type: application/sdp
+ Content-Length: [len]
+
+ v=0
+ o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+ s=-
+ c=IN IP[media_ip_type] [media_ip]
+ t=0 0
+ m=audio [media_port] RTP/AVP 0
+ a=rtpmap:0 PCMU/8000
+
+ ]]>
+ </send>
+
+ <!-- Play a pre-recorded PCAP file (RTP stream) -->
+ <nop>
+ <action>
+ <exec play_pcap_audio="g711a.pcap"/>
+ </action>
+ </nop>
+
+ <!-- Pause 8 seconds, which is approximately the duration of the -->
+ <!-- PCAP file -->
+ <pause milliseconds="8000"/>
+
+ <send retrans="500">
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:];tag=[call_number]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: [len]
+
+ ]]>
+ </send>
+
+ <recv request="ACK"
+ optional="true"
+ rtd="true"
+ crlf="true">
+ </recv>
+
+ <recv request="BYE">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 200 OK
+ [last_Via:]
+ [last_From:]
+ [last_To:]
+ [last_Call-ID:]
+ [last_CSeq:]
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <pause milliseconds="4000"/>
+
+
+ <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+ <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
--
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