diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index 7a1664ec90a38cb9d78c41d2e96ee9a22a2ef787..3f754c8df169c1bbb9fe49e3b79bab666e71c4fc 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -357,12 +357,17 @@ PulseLayer::stopStream (void)
{
_info("Audio: Stop audio stream");
+
+ pa_threaded_mainloop_lock (m);
+
if(playback)
pa_stream_flush (playback->pulseStream(), NULL, NULL);
if(record)
pa_stream_flush (record->pulseStream(), NULL, NULL);
+ pa_threaded_mainloop_unlock (m);
+
disconnectAudioStream();
}
diff --git a/sflphone-common/src/audio/sound/Makefile.am b/sflphone-common/src/audio/sound/Makefile.am
index 37d7ddc61eb82225f25c86da9c270da8da0dc208..10e9960b7a14c29d2c66d7fb1447f2e18e2bdc1f 100644
--- a/sflphone-common/src/audio/sound/Makefile.am
+++ b/sflphone-common/src/audio/sound/Makefile.am
@@ -5,14 +5,12 @@ noinst_LTLIBRARIES = libsound.la
libsound_la_SOURCES = \
audiofile.cpp \
tone.cpp \
- tonegenerator.cpp \
tonelist.cpp \
dtmf.cpp \
dtmfgenerator.cpp
noinst_HEADERS = \
audiofile.h \
- tonegenerator.h \
tone.h \
tonelist.h \
dtmfgenerator.h \
diff --git a/sflphone-common/src/audio/sound/tone.cpp b/sflphone-common/src/audio/sound/tone.cpp
index d8849cbf3235ffe1db8b1c0b9daa64f41238b55e..7f60f1ecca749103c8e42fe94f92231f9675677d 100644
--- a/sflphone-common/src/audio/sound/tone.cpp
+++ b/sflphone-common/src/audio/sound/tone.cpp
@@ -28,8 +28,12 @@
#include <cstdlib>
#include <strings.h>
-Tone::Tone (const std::string& definition, unsigned int sampleRate) : AudioLoop(), _sampleRate (sampleRate)
+#define TABLE_LENGTH 4096
+double TWOPI = 2 * M_PI;
+
+Tone::Tone (const std::string& definition, unsigned int sampleRate) : AudioLoop(), _sampleRate (sampleRate), _xhigher(0.0), _xlower(0.0)
{
+ fillWavetable();
genBuffer (definition); // allocate memory with definition parameter
}
@@ -101,6 +105,7 @@ Tone::genBuffer (const std::string& definition)
}
// Generate SAMPLING_RATE samples of sinus, buffer is the result
+ _debug("genSin(%d, %d)", freq1, freq2);
genSin (bufferPos, freq1, freq2, count);
// To concatenate the different buffers for each section.
@@ -124,20 +129,70 @@ Tone::genBuffer (const std::string& definition)
bufferPos=0;
}
+void
+Tone::fillWavetable()
+{
+ double tableSize = (double)TABLE_LENGTH;
+
+ for(int i = 0; i < TABLE_LENGTH; i++) {
+ _wavetable[i] = sin( ((double)i / (tableSize - 1.0)) * TWOPI );
+ }
+}
+
+double
+Tone::interpolate(double x)
+{
+ int xi_0, xi_1;
+ double yi_0, yi_1, A, B;
+
+ xi_0 = (int)x;
+ xi_1 = xi_0+1;
+
+ yi_0 =_wavetable[xi_0];
+ yi_1 = _wavetable[xi_1];
+
+ A = (x - xi_0);
+ B = 1.0 - A;
+
+ return A*yi_0 + B*yi_1;
+}
+
void
Tone::genSin (SFLDataFormat* buffer, int frequency1, int frequency2, int nb)
{
+ _xhigher = 0.0;
+ _xlower = 0.0;
- double pi2 = 6.28318520;
- double var1 = pi2 * (double) frequency1 / (double) _sampleRate;
- double var2 = pi2 * (double) frequency2 / (double) _sampleRate;
+ double sr = (double)_sampleRate;
+ double tableSize = (double)TABLE_LENGTH;
- // softer
- double amp = (double) SFLDataAmplitude;
+ double N_h = sr / (double) (frequency1);
+ double N_l = sr / (double) (frequency2);
- for (int t = 0; t < nb; t++) {
- buffer[t] = (SFLDataFormat) (amp * ( (sin (var1 * t) + sin (var2 * t))));
- }
+ double dx_h = tableSize / N_h;
+ double dx_l = tableSize / N_l;
+
+ double x_h = _xhigher;
+ double x_l = _xlower;
+
+ double amp = (double)SFLDataAmplitude;
+
+ for (int t = 0; t < nb; t ++) {
+ buffer[t] = (int16)(amp*(interpolate(x_h) + interpolate(x_l)));
+ x_h += dx_h;
+ x_l += dx_l;
+
+ if(x_h > tableSize) {
+ x_h -= tableSize;
+ }
+
+ if(x_l > tableSize) {
+ x_l -= tableSize;
+ }
+ }
+
+ _xhigher = x_h;
+ _xlower = x_l;
}
diff --git a/sflphone-common/src/audio/sound/tone.h b/sflphone-common/src/audio/sound/tone.h
index a725059dcf20a6d5ca509ffa8d0bc07c4187de28..94be60a0b8ecbe4a9b23465dfeec310231d2eedc 100644
--- a/sflphone-common/src/audio/sound/tone.h
+++ b/sflphone-common/src/audio/sound/tone.h
@@ -28,6 +28,8 @@
#define TONE_NBTONE 4
#define TONE_NBCOUNTRY 7
+#define TABLE_LENGTH 4096
+
/**
* @file tone.h
* @brief Tone sample (dial, busy, ring, congestion)
@@ -65,6 +67,17 @@ public:
*/
void genSin(SFLDataFormat* buffer, int frequency1, int frequency2, int nb);
+ /**
+ *
+ */
+ void fillWavetable(void);
+
+ /**
+ *
+ */
+ double interpolate(double x);
+
+
private:
/**
@@ -75,6 +88,11 @@ private:
/** Sample rate */
unsigned int _sampleRate;
+
+ double _wavetable[TABLE_LENGTH];
+
+ double _xhigher;
+ double _xlower;
};
#endif // __TONE_H__
diff --git a/sflphone-common/src/audio/sound/tonegenerator.cpp b/sflphone-common/src/audio/sound/tonegenerator.cpp
deleted file mode 100644
index dcdbb6be078724e8792321fb8cebfb829476c016..0000000000000000000000000000000000000000
--- a/sflphone-common/src/audio/sound/tonegenerator.cpp
+++ /dev/null
@@ -1,60 +0,0 @@
-/**
- * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-
-#include <fstream>
-#include <math.h>
-#include <stdlib.h>
-
-#include "tonegenerator.h"
-#include "global.h"
-
-int AMPLITUDE = 32767;
-
-///////////////////////////////////////////////////////////////////////////////
-// ToneGenerator implementation
-///////////////////////////////////////////////////////////////////////////////
-
-ToneGenerator::ToneGenerator (unsigned int sampleRate) : sample (NULL), freq1(), freq2(), time(), totalbytes(), _sampleRate (sampleRate)
-{
-}
-
-ToneGenerator::~ToneGenerator (void)
-{
-}
-
-/**
- * Calculate superposition of 2 sinus
- *
- */
-void
-ToneGenerator::generateSin (int lowerfreq, int higherfreq, int16* ptr, int len) const
-{
- double var1, var2;
-
- var1 = (double) 2 * (double) M_PI * (double) higherfreq / (double) _sampleRate;
- var2 = (double) 2 * (double) M_PI * (double) lowerfreq / (double) _sampleRate;
-
- double amp = (double) (AMPLITUDE >> 2);
-
- for (int t = 0; t < len; t++) {
- ptr[t] = (int16) (amp * ( (sin (var1 * t) + sin (var2 * t))));
- }
-}
-
diff --git a/sflphone-common/src/audio/sound/tonegenerator.h b/sflphone-common/src/audio/sound/tonegenerator.h
deleted file mode 100644
index ce1ad56e618ebce339448607027f7b694e686ba1..0000000000000000000000000000000000000000
--- a/sflphone-common/src/audio/sound/tonegenerator.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * Copyright (C) 2004-2006 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-
-#ifndef __TONE_GENERATOR_H__
-#define __TONE_GENERATOR_H__
-
-#include <string>
-#include <cc++/thread.h>
-
-#include "global.h"
-
-/**
- * @file tonegenerator.h
- * @brief Sine generator to create tone with string definition
- */
-
-class ToneGenerator {
- public:
- /**
- * Constructor
- * @param sampleRate The sample rate of the generated samples
- */
- ToneGenerator (unsigned int sampleRate);
-
- /**
- * Destructor
- */
- ~ToneGenerator (void);
-
- /**
- * Calculate sinus with superposition of 2 frequencies
- * @param lowerfreq Lower frequency
- * @param higherfreq Higher frequency
- * @param ptr For result buffer
- * @param len The length of the data to be generated
- */
- void generateSin (int, int, int16 *, int len) const;
-
-
- ///////////////////////////
- // Public members variable
- //////////////////////////
- int16 *sample;
- int freq1, freq2;
- int time;
- int totalbytes;
-
- private:
- // Copy Constructor
- ToneGenerator(const ToneGenerator& rh);
-
- // Assignment Operator
- ToneGenerator& operator=( const ToneGenerator& rh);
-
- /*
- * Initialisation of the supported tones according to the countries.
- */
- void initTone (void);
-
- int16 _buf[SIZEBUF];
- int _sampleRate;
-};
-
-#endif // __TONE_GENRATOR_H__
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index a7058bbb7bed2735fae67677c1a973fa6ff74cc6..022cdcbe0c2dd16cc5b97e6115570756b9cb7bd6 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -921,6 +921,8 @@ SIPVoIPLink::peerHungup (const CallID& id)
pjsip_tx_data *tdata = NULL;
SIPCall* call;
+ _info("SIP: Peer hungup");
+
call = getSIPCall (id);
if (call==0) {
@@ -3476,7 +3478,7 @@ mod_on_rx_request (pjsip_rx_data *rdata)
// No need to go any further on incoming ACK
if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
_info("UserAgent: received an ACK");
- return true;
+ return true;
}
// Handle the incoming call invite in this function
diff --git a/sippxml/account_uac_send_hangup.xml b/sippxml/account_uac_send_hangup.xml
index b07251cb2193abf14a4d70dd5351b790976c5dbf..8971769c591c52bbfd1c89cc1e96e46c71fe3860 100644
--- a/sippxml/account_uac_send_hangup.xml
+++ b/sippxml/account_uac_send_hangup.xml
@@ -19,47 +19,6 @@
<scenario name="accountcall_client">
- <send retrans="500">
- <![CDATA[
-
- REGISTER sip:[remote_ip] SIP/2.0
- Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
- Max-Forward: 70
- From: <sip:27182@[remote_ip]:[remote_port]>;tag=[call_number]
- To: <sip:27182@[remote_ip]:[remote_port]>
- Call-ID: REG///[call_id]
- CSeq: 1 REGISTER
- Contact: <sip:27182@[remote_ip]:[remote_port]>
- Content-Length: 0
- Expires: 300
-
- ]]>
- </send>
-
- <recv response="401" auth="true">
- </recv>
-
- <send retrans="500">
- <![CDATA[
-
- REGISTER sip:[remote_ip] SIP/2.0
- Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
- Max-Forwards: 70
- From: <sip:27182@[remote_ip]:[remote_port]>;tag=[call_number]
- To: <sip:27182@[remote_ip]:[remote_port]>
- Call-ID: REG///[call_id]
- CSeq: 2 REGISTER
- Contact: <sip:27182@[remote_ip]:[remote_port]>
- Content-Length: 0
- Expires: 300
- [authentication username=27182 password=1234]
-
- ]]>
- </send>
-
- <recv response="200">
- </recv>
-
<pause milliseconds="200"/>
<send retrans="500">
diff --git a/sippxml/account_uac_send_peer_hungup.xml b/sippxml/account_uac_send_peer_hungup.xml
new file mode 100644
index 0000000000000000000000000000000000000000..716da9dd6de55534e79d29e2efa54ad6248ccc5c
--- /dev/null
+++ b/sippxml/account_uac_send_peer_hungup.xml
@@ -0,0 +1,155 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- This program is free software; you can redistribute it and/or -->
+<!-- modify it under the terms of the GNU General Public License as -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version. -->
+<!-- -->
+<!-- This program is distributed in the hope that it will be useful, -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
+<!-- GNU General Public License for more details. -->
+<!-- -->
+<!-- You should have received a copy of the GNU General Public License -->
+<!-- along with this program; if not, write to the -->
+<!-- Free Software Foundation, Inc., -->
+<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
+
+
+<scenario name="accountcall_client">
+
+ <pause milliseconds="200"/>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:2000@[remote_ip] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
+ From: <sip:27182@[remote_ip]>;tag=[call_number]
+ To: <sip:2000@[remote_ip]>
+ Call-ID: [call_id]
+ CSeq: 3 INVITE
+ Contact: sip:27182@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Functional Test
+ 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>
+
+ <recv response="401" auth="true">
+ </recv>
+
+ <pause milliseconds="200"/>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:27182@[remote_ip] SIP/2.0
+ Max-Forwards: 70
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:27182@[remote_ip]>;tag=[call_number]
+ To: <sip:2000@[remote_ip]>
+ Call-ID: [call_id]
+ CSeq: 4 ACK
+ Subject: Functional Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <send retrans="500">
+ <![CDATA[
+
+ INVITE sip:2000@[remote_ip] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
+ From: <sip:27182@:[remote_ip]>;tag=[call_number]
+ To: <sip:2000@[remote_ip]>
+ Call-ID: [call_id]
+ CSeq: 5 INVITE
+ Contact: sip:27182@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Functional Test
+ Content-Type: application/sdp
+ Content-Length: [len]
+ [authentication username=27182 password=1234]
+
+ 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>
+
+ <recv response="100">
+ </recv>
+
+ <recv response="180">
+ </recv>
+
+
+ <recv response="200">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ ACK sip:2000@192.168.50.79 SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:27182@[remote_ip]>;tag=[call_number]
+ To: <sip:2000@[remote_ip]>
+ Call-ID: [call_id]
+ CSeq: 5 ACK
+ Contact: sip:27182@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Functional Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv request="BYE">
+ </recv>
+
+
+ <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-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>
+
+ <pause milliseconds="1000"/>
+
+</scenario>
diff --git a/sippxml/account_uas_receive_transfer.xml b/sippxml/account_uas_receive_transfer.xml
new file mode 100644
index 0000000000000000000000000000000000000000..80a875fbf707be6c6f3c03c2484be68c7c255ce4
--- /dev/null
+++ b/sippxml/account_uas_receive_transfer.xml
@@ -0,0 +1,101 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- This program is free software; you can redistribute it and/or -->
+<!-- modify it under the terms of the GNU General Public License as -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version. -->
+<!-- -->
+<!-- This program is distributed in the hope that it will be useful, -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
+<!-- GNU General Public License for more details. -->
+<!-- -->
+<!-- You should have received a copy of the GNU General Public License -->
+<!-- along with this program; if not, write to the -->
+<!-- Free Software Foundation, Inc., -->
+<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
+
+
+<scenario name="accountcall_client">
+
+ <recv request="INVITE">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 180 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 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-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>
+
+ <recv request="ACK">
+ </recv>
+
+ <recv request="INVITE">
+ </recv>
+
+ <pause milliseconds="200">
+ </recv>
+
+ <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-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>
+
+ <recv request="ACK">
+ </recv>
+
+
+</scenario>
diff --git a/sippxml/account_uas_recv_hangup.xml b/sippxml/account_uas_recv_hangup.xml
new file mode 100644
index 0000000000000000000000000000000000000000..101e6f4c97442b091dc86fedc913078353126c63
--- /dev/null
+++ b/sippxml/account_uas_recv_hangup.xml
@@ -0,0 +1,121 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- This program is free software; you can redistribute it and/or -->
+<!-- modify it under the terms of the GNU General Public License as -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version. -->
+<!-- -->
+<!-- This program is distributed in the hope that it will be useful, -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
+<!-- GNU General Public License for more details. -->
+<!-- -->
+<!-- You should have received a copy of the GNU General Public License -->
+<!-- along with this program; if not, write to the -->
+<!-- Free Software Foundation, Inc., -->
+<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
+
+
+<scenario name="accountcall_client">
+
+ <recv request="INVITE">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 180 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>
+
+ <pause milliseconds="200"/>
+
+ <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-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>
+
+ <recv request="ACK">
+ </recv>
+
+ <recv request="INVITE">
+ </recv>
+
+ <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-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>
+
+ <recv request="ACK">
+ </recv>
+
+ <pause milliseconds="500"/>
+
+ <send retrans="500">
+ <![CDATA[
+
+ BYE sip:2000@192.168.50.79 SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+ From: <sip:27182@[local_ip]:[local_port]>;tag=[call_number]
+ To: <sip:2000@192.168.50.79:[remote_port]>
+ Call-ID: [call_id]
+ CSeq: 6 BYE
+ Contact: sip:27182@[local_ip]:[local_port]
+ Max-Forwards: 70
+ Subject: Functional Test
+ Content-Length: 0
+
+ ]]>
+ </send>
+
+ <recv response="200">
+ </recv>
+
+</scenario>
diff --git a/sippxml/account_uas_recv_peer_hungup.xml b/sippxml/account_uas_recv_peer_hungup.xml
index cfa9acb151e23f8234f3daf4a35ae21eca05c8dd..c91230c7fc32e2c853d934ec5dbc3f499060d958 100644
--- a/sippxml/account_uas_recv_peer_hungup.xml
+++ b/sippxml/account_uas_recv_peer_hungup.xml
@@ -19,54 +19,9 @@
<scenario name="accountcall_client">
- <send retrans="500">
- <![CDATA[
-
- REGISTER sip:[remote_ip] SIP/2.0
- Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
- Max-Forward: 70
- From: <sip:27182@[remote_ip]:[remote_port]>;tag=[call_number]
- To: <sip:27182@[remote_ip]:[remote_port]>
- Call-ID: REG///[call_id]
- CSeq: 1 REGISTER
- Contact: <sip:27182@[remote_ip]:[remote_port]>
- Content-Length: 0
- Expires: 300
-
- ]]>
- </send>
-
- <recv response="401" auth="true">
- </recv>
-
- <send retrans="500">
- <![CDATA[
-
- REGISTER sip:[remote_ip] SIP/2.0
- Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
- Max-Forwards: 70
- From: <sip:27182@[remote_ip]:[remote_port]>;tag=[call_number]
- To: <sip:27182@[remote_ip]:[remote_port]>
- Call-ID: REG///[call_id]
- CSeq: 2 REGISTER
- Contact: <sip:27182@[remote_ip]:[remote_port]>
- Content-Length: 0
- Expires: 300
- [authentication username=27182 password=1234]
-
- ]]>
- </send>
-
- <recv response="200">
- </recv>
-
-
<recv request="INVITE">
</recv>
- <recv response="401" auth="true">
- </recv>
-
<send>
<![CDATA[
@@ -82,7 +37,38 @@
]]>
</send>
- <pause milliseconds="200"/>
+ <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-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>
+
+ <recv request="ACK">
+ </recv>
+
+ <recv request="INVITE">
+ </recv>
+
+ <pause milliseconds="200">
+ </recv>
<send retrans="500">
<![CDATA[
diff --git a/sippxml/account_uas_recv_transfered.xml b/sippxml/account_uas_recv_transfered.xml
new file mode 100644
index 0000000000000000000000000000000000000000..210f5621b4b5a002adcbbd0a1880756c49b33088
--- /dev/null
+++ b/sippxml/account_uas_recv_transfered.xml
@@ -0,0 +1,132 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- This program is free software; you can redistribute it and/or -->
+<!-- modify it under the terms of the GNU General Public License as -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version. -->
+<!-- -->
+<!-- This program is distributed in the hope that it will be useful, -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
+<!-- GNU General Public License for more details. -->
+<!-- -->
+<!-- You should have received a copy of the GNU General Public License -->
+<!-- along with this program; if not, write to the -->
+<!-- Free Software Foundation, Inc., -->
+<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
+
+
+<scenario name="accountcall_client">
+
+ <recv request="INVITE">
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 180 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 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-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>
+
+ <recv request="ACK">
+ </recv>
+
+ <recv request="INVITE">
+ </recv>
+
+ <pause milliseconds="200">
+ </recv>
+
+ <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-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>
+
+ <recv request="ACK">
+ </recv>
+
+
+ <!--
+
+ <recv request="BYE">
+ </recv>
+
+ <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-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>
+
+ -->
+
+</scenario>
diff --git a/sippxml/account_uas_register.xml b/sippxml/account_uas_register.xml
new file mode 100644
index 0000000000000000000000000000000000000000..e4ebb812276270bd5a3334ecd61a4b4c2825900c
--- /dev/null
+++ b/sippxml/account_uas_register.xml
@@ -0,0 +1,63 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- This program is free software; you can redistribute it and/or -->
+<!-- modify it under the terms of the GNU General Public License as -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version. -->
+<!-- -->
+<!-- This program is distributed in the hope that it will be useful, -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
+<!-- GNU General Public License for more details. -->
+<!-- -->
+<!-- You should have received a copy of the GNU General Public License -->
+<!-- along with this program; if not, write to the -->
+<!-- Free Software Foundation, Inc., -->
+<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
+
+
+<scenario name="accountcall_client">
+
+ <send retrans="500">
+ <![CDATA[
+
+ REGISTER sip:[remote_ip] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
+ Max-Forwards: 70
+ From: <sip:27182@[remote_ip]>;tag=[call_number]
+ To: <sip:27182@[remote_ip]>
+ Call-ID: REG///[call_id]
+ CSeq: 1 REGISTER
+ Contact: <sip:27182@[local_ip]:[local_port]>
+ Content-Length: 0
+ Expires: 300
+
+ ]]>
+ </send>
+
+ <recv response="401" auth="true">
+ </recv>
+
+ <send retrans="500">
+ <![CDATA[
+
+ REGISTER sip:[remote_ip] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
+ Max-Forwards: 70
+ From: <sip:27182@[remote_ip]>;tag=[call_number]
+ To: <sip:27182@[remote_ip]>
+ Call-ID: REG///[call_id]
+ CSeq: 2 REGISTER
+ Contact: <sip:27182@[local_ip]:[local_port]>
+ Content-Length: 0
+ Expires: 300
+ [authentication username=27182 password=1234]
+
+ ]]>
+ </send>
+
+ <recv response="200">
+ </recv>
+
+</scenario>
diff --git a/sippxml/ip2ip_uac_send_hangup.xml b/sippxml/ip2ip_uac_send_hangup.xml
index b38cf17361a568261f5441bd8b651639e759c4e2..49cc63fca06e2cd1f2a40f1f65929de436da7d6b 100644
--- a/sippxml/ip2ip_uac_send_hangup.xml
+++ b/sippxml/ip2ip_uac_send_hangup.xml
@@ -53,15 +53,18 @@
<recv response="200">
+ <action>
+ <ereg regexp="tag=.*" search_in="hdr" header="To:" check_it="true" assign_to="1" />
+ </action>
</recv>
<send>
<![CDATA[
- ACK sip:192.168.50.79 SIP/2.0
+ ACK sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[local_ip]:[local_port]>;tag=[call_number]
- To: <sip:192.168.50.79:[remote_port]>
+ To: <sip:[remote_ip]:[remote_port]>;tag=[$1]
Call-ID: [call_id]
CSeq: 2 ACK
Contact: sip:[local_ip]:[local_port]
@@ -78,7 +81,7 @@
BYE sip:192.168.50.79 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[local_ip]:[local_port]>;tag=[call_number]
- To: <sip:192.168.50.79:[remote_port]>
+ To: <sip:[remote_ip]:[remote_port]>;tag=[$1]
Call-ID: [call_id]
CSeq: 3 BYE
Contact: sip:[local_ip]:[local_port]
diff --git a/sippxml/ip2ip_uac_send_peer_hungup.xml b/sippxml/ip2ip_uac_send_peer_hungup.xml
index 3ffd02a3723285a72ce25cd4b026585b96e4dcef..4e811e38262f10680417f9415e960e33585b26ee 100644
--- a/sippxml/ip2ip_uac_send_peer_hungup.xml
+++ b/sippxml/ip2ip_uac_send_peer_hungup.xml
@@ -60,7 +60,7 @@
ACK sip:192.168.50.79 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[local_ip]:[local_port]>;tag=[call_number]
- To: <sip:192.168.50.79:[remote_port]>
+ To: <sip:[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 2 ACK
Contact: sip:[local_ip]:[local_port]
diff --git a/sippxml/ip2ip_uas_recv_hangup.xml b/sippxml/ip2ip_uas_recv_hangup.xml
index 571dda97ccc73d9200591f795d2c3ed08458c91e..4e86b47480bd761e3d4ccdf071dcbe5738f5c5f9 100644
--- a/sippxml/ip2ip_uas_recv_hangup.xml
+++ b/sippxml/ip2ip_uas_recv_hangup.xml
@@ -16,7 +16,7 @@
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
- Contact: <sip:127.0.1.1:5062;transport=[transport]>
+ Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
@@ -46,22 +46,27 @@
]]>
</send>
+ <!-- tag from From: header is required to build the To: header in
+ -- Bye request. -->
- <recv request="ACK">
+ <recv request="ACK">
+ <action>
+ <ereg regexp="tag=.*" search_in="hdr" header="From:" check_it="true" assign_to="1" />
+ </action>
</recv>
<pause milliseconds="500"/>
-
+
<send retrans="500">
<![CDATA[
- BYE sip:127.0.0.1:5060;transport=[transport] SIP/2.0
- [last_Via:]
- [last_From:]
- [last_To:]
+ BYE sip:[service] SIP/2.0
+ Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch]
+ From: <sip:[local_ip]:[local_port]>;tag=[call_number]
+ To: <sip:[remote_ip]:[remote_port]>;[$1]
[last_Call-ID:]
CSeq: [cseq] BYE
- Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+ Contact: <sip:test@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Functional Test
Content-Length: 0
diff --git a/sippxml/ip2ip_uas_recv_hold_offhold.xml b/sippxml/ip2ip_uas_recv_hold_offhold.xml
new file mode 100644
index 0000000000000000000000000000000000000000..568a97a9e62ddf5bd4a323b920f425f300104273
--- /dev/null
+++ b/sippxml/ip2ip_uas_recv_hold_offhold.xml
@@ -0,0 +1,187 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+
+<scenario name="UAS HOLD/OFFHOLD">
+
+ <!-- Receive a new call -->
+
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp="sendrecv" search_in="body" check_it="true" assign_to="1"/>
+ <log message="Media is [$1]"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 180 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 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-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>
+
+ <recv request="ACK" optional="true" rtd="true" crlf="true">
+ </recv>
+
+ <!-- This call is now on HOLD: sendonly tell to PBX to send music on hold-->
+
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp="sendonly" search_in="body" check_it="true" assign_to="2"/>
+ <log message="Media is [$2]"/>
+ </action>
+ </recv>
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 180 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 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-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>
+
+ <recv request="ACK" optional="true" rtd="true" crlf="true">
+ </recv>
+
+ <!-- OFFHOLD this call -->
+
+ <recv request="INVITE" crlf="true">
+ <action>
+ <ereg regexp="sendrecv" search_in="body" check_it="true" assign_to="3"/>
+ <log message="Media is [$3]"/>
+ </action>
+ </recv>
+
+
+ <send>
+ <![CDATA[
+
+ SIP/2.0 180 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 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-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>
+
+ <recv request="ACK" optional="true" rtd="true" crlf="true">
+ </recv>
+
+ <!-- Hangup this call -->
+
+ <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>
diff --git a/sippxml/ip2ip_uas_recv_peer_hungup.xml b/sippxml/ip2ip_uas_recv_peer_hungup.xml
index 4b615f216cf96daf1ecfb35cfcf78f2d2f1daade..9f2721f5dc71742819d4d39003fc1279b0cd18db 100644
--- a/sippxml/ip2ip_uas_recv_peer_hungup.xml
+++ b/sippxml/ip2ip_uas_recv_peer_hungup.xml
@@ -46,10 +46,7 @@
]]>
</send>
- <recv request="ACK"
- optional="true"
- rtd="true"
- crlf="true">
+ <recv request="ACK" optional="true" rtd="true" crlf="true">
</recv>
<recv request="BYE">
diff --git a/sippxml/tempscript.sh b/sippxml/tempscript.sh
new file mode 100644
index 0000000000000000000000000000000000000000..090a058ed8491511f8ef33551e260d56237be826
--- /dev/null
+++ b/sippxml/tempscript.sh
@@ -0,0 +1,5 @@
+SERVERPORT=5064
+
+sipp -sf account_uas_register_bis.xml 192.168.50.79 -i 192.168.50.182 -p ${SERVERPORT} -l 1 -m 1
+
+sipp -sf account_uas_receive_transfer.xml 192.168.50.79 -i 192.168.50.182 -p ${SERVERPORT} -l 1
\ No newline at end of file
diff --git a/sippxml/testsuiteuac.sh b/sippxml/testsuiteuac.sh
index c63983c1e0319464254c924542d06b7472860126..be9455244341e1074b25b4fbee1f8cbbf9c28391 100644
--- a/sippxml/testsuiteuac.sh
+++ b/sippxml/testsuiteuac.sh
@@ -3,10 +3,11 @@
SERVERPORT=5062
+# SCENARIO 1 Test 1
function test_ip2ip_send_hangup {
# start sipp server to receive calls from sflphone
- sipp -sf ip2ip_uas_recv_peer_hungup.xml -p ${SERVERPORT}
+ sipp -sf ip2ip_uas_recv_peer_hungup.xml 127.0.0.1:5060 -i 127.0.0.1 -p ${SERVERPORT}
# start sflphoned
# /usr/lib/sflphone/sflphoned&
@@ -21,10 +22,11 @@ function test_ip2ip_send_hangup {
# bashtrap
}
+# SCENARIO 1 Test 2
function test_ip2ip_send_peer_hungup {
# start sipp server to receive calls from sflphone and then hangup
- sipp -sf ip2ip_uas_recv_hangup.xml -p ${SERVERPORT}
+ sipp -sf ip2ip_uas_recv_hangup.xml 127.0.0.1:5060 -s 127.0.0.1:5060 -i 127.0.0.1 -p ${SERVERPORT}
# start sflphoned
# /usr/lib/sflphone/sflphoned&
@@ -40,6 +42,7 @@ function test_ip2ip_send_peer_hungup {
}
+# SCENARIO 1 Test 3
function test_ip2ip_recv_hangup {
# start sflphoned
@@ -60,6 +63,8 @@ function test_ip2ip_recv_hangup {
# bashtrap
}
+
+# SCENARIO 1 Test 4
function test_ip2ip_recv_peer_hungup {
# start sflphoned
@@ -80,6 +85,7 @@ function test_ip2ip_recv_peer_hungup {
# bashtrap
}
+# SCENARIO 2 Test 1
function test_account_send_hangup {
# start sflphoned
@@ -93,6 +99,9 @@ function test_account_send_hangup {
# wait some time to make sure client is bound
# sleep 1;
+ # process only one registration
+ sipp -sf account_uas_register.xml 192.168.50.79 -i 192.168.50.182 -p ${SERVERPORT} -l 1 -m 1
+
# start sipp client and send calls
sipp -sf account_uas_recv_peer_hungup.xml 192.168.50.79 -i 192.168.50.182 -p ${SERVERPORT} -l 1
@@ -100,6 +109,55 @@ function test_account_send_hangup {
# bashtrap
}
+# SCENARIO 2 Test 2
+function test_account_send_peer_hungup {
+
+ # start sflphoned
+ # /usr/lib/sflphone/sflphoned&
+
+ # wait some time to make sure sflphoned is started
+ # sleep 1;
+
+ # python ../tools/pysflphone/pysflphone_testdbus.py &
+
+ # wait some time to make sure client is bound
+ # sleep 1;
+
+ # process only one registration
+ sipp -sf account_uas_register.xml 192.168.50.79 -i 192.168.50.182 -p ${SERVERPORT} -l 1 -m 1
+
+ # start sipp client and send calls
+ sipp -sf account_uas_recv_hangup.xml 192.168.50.79 -i 192.168.50.182 -p ${SERVERPORT} -l 1
+
+ # kill every one
+ # bashtrap
+}
+
+# SCENARIO 2 Test 3
+function test_account_recv_hangup {
+
+ # start sflphoned
+ # /usr/lib/sflphone/sflphoned&
+
+ # wait some time to make sure sflphoned is started
+ # sleep 1;
+
+ # python ../tools/pysflphone/pysflphone_testdbus.py &
+
+ # wait some time to make sure client is bound
+ # sleep 1;
+
+ # process only one registration
+ sipp -sf account_uas_register.xml 192.168.50.79 -i 192.168.50.182 -p ${SERVERPORT} -l 1 -m 1
+
+ # start sipp client and send calls
+ sipp -sf account_uac_send_peer_hungup.xml 192.168.50.79 -i 192.168.50.182 -p ${SERVERPORT} -l 1
+
+ # kill every one
+ # bashtrap
+}
+
+# SCENARIO 2 Test 4
function test_account_recv_peer_hungup {
# start sflphoned
@@ -113,6 +171,9 @@ function test_account_recv_peer_hungup {
# wait some time to make sure client is bound
# sleep 1;
+ # process only one registration
+ sipp -sf account_uas_register.xml 192.168.50.79 -i 192.168.50.182 -p ${SERVERPORT} -l 1 -m 1
+
# start sipp client and send calls
sipp -sf account_uac_send_hangup.xml 192.168.50.79 -i 192.168.50.182 -p ${SERVERPORT} -l 1
@@ -120,6 +181,51 @@ function test_account_recv_peer_hungup {
# bashtrap
}
+# SCENARIO 3 Test 1
+function test_ip2ip_send_hold_offhold {
+
+ # start sflphoned
+ # /usr/lib/sflphone/sflphoned&
+
+ # wait some time to make sure sflphoned is started
+ # sleep 1;
+
+ # python ../tools/pysflphone/pysflphone_testdbus.py &
+
+ # wait some time to make sure client is bound
+ # sleep 1;
+
+ # start sipp client and send calls
+ sipp -sf ip2ip_uas_recv_hold_offhold.xml 127.0.0.1:5060 -i 127.0.0.1 -p ${SERVERPORT}
+ # kill every one
+ # bashtrap
+}
+
+# SCENARIO 4 Test 1
+function test_account_send_transfer {
+
+ # start sflphoned
+ # /usr/lib/sflphone/sflphoned&
+
+ # wait some time to make sure sflphoned is started
+ # sleep 1;
+
+ # python ../tools/pysflphone/pysflphone_testdbus.py &
+
+ # wait some time to make sure client is bound
+ # sleep 1;
+
+ # process only one registration
+ sipp -sf account_uas_register.xml 192.168.50.79 -i 192.168.50.182 -p ${SERVERPORT} -l 1 -m 1
+
+ # start sipp client and send calls
+ sipp -sf account_uas_recv_transfered.xml 192.168.50.79 -i 192.168.50.182 -p ${SERVERPORT} -l 1
+
+ # kill every one
+ # bashtrap
+}
+
+
# function called if CTRL-C detected
bashtrap()
{
@@ -136,6 +242,14 @@ bashtrap()
# test_ip2ip_recv_hangup
# test_ip2ip_recv_peer_hungup
-# SCENARIO X: Normal flow calls (Account)
-test_account_send_hangup
-# test_account_recv_peer_hungup
\ No newline at end of file
+# SCENARIO 2: Normal flow calls (Account)
+# test_account_send_hangup
+# test_account_send_peer_hungup
+# test_account_recv_hangup
+# test_account_recv_peer_hungup
+
+# SCENARIO 3: Hold/offHold calls (Account)
+# test_ip2ip_send_hold_offhold
+
+# SCENARIO 4: Transfer calls (Account)
+test_account_send_transfer
\ No newline at end of file
diff --git a/tools/pysflphone/pysflphone_testdbus.py b/tools/pysflphone/pysflphone_testdbus.py
index 563b683456ec16685896a661109eba094da578ca..3e7c97f0b03dc2df97dc4f643275e11224d75ce2 100644
--- a/tools/pysflphone/pysflphone_testdbus.py
+++ b/tools/pysflphone/pysflphone_testdbus.py
@@ -113,6 +113,7 @@ class SflPhoneTests():
# Start Glib mainloop
self.sflphone.start()
+
def test_ip2ip_recv_peer_hungup(self):
"""Wait for calls, answer, peer hangup"""
# Add callback for this test
@@ -121,24 +122,49 @@ class SflPhoneTests():
# Start Glib mainloop
self.sflphone.start()
+
def test_account_send_hangup(self):
"""Send new account call, hangup once peer answered"""
- print "test account send hangup"
+
i = 0
- while(i < 10):
+ while(i < 1):
callid = self.sflphone.Call("27182")
- time.sleep(0.5)
+ time.sleep(1.0)
self.sflphone.HangUp(callid)
- time.sleep(0.5)
+ time.sleep(1.0)
+
+ i = i+1
+
+ # del self.sflphone
+
+
+ def test_account_send_peer_hungup(self):
+ """Send new account call, hangup once peer answered"""
+
+ i = 0
+ while(i < 10):
+
+ callid = self.sflphone.Call("27182")
+ time.sleep(1.0)
i = i+1
- print "account hangup done"
del self.sflphone
- def test_account_recv_call_peer_hungup(self):
+
+ def test_account_recv_hangup(self):
+ """Register an account and wait for incoming calls"""
+
+ # Add callback for this test
+ self.sflphone.onIncomingCall_cb = acceptOnIncomingCallHangup
+
+ # Start Glib mainloop
+ self.sflphone.start()
+
+
+ def test_account_recv_peer_hungup(self):
"""Register an account and wait for incoming calls"""
# Add callback for this test
@@ -147,24 +173,43 @@ class SflPhoneTests():
# Start Glib mainloop
self.sflphone.start()
-
-# def test_account_send_call_peer_hungup(self):
-# """Register an account on a remote server and make several calls"""
-#
-# self.setAccount("Account:1258495784");
-# time.sleep(3)
-#
-# i = 0
-# while(i < 50):
-#
-# callid = self.Call("5000")
-# time.sleep(0.4)
-#
-# self.HangUp(callid)
-# time.sleep(0.4)
-#
-# i = i+1
+ def test_ip2ip_send_hold_offhold(self):
+ """Send new call, hold this call, offhold, hangup"""
+ i = 0
+ while(i < 10):
+
+ callid = self.sflphone.Call("sip:test@127.0.0.1:5062")
+ time.sleep(0.5)
+
+ self.sflphone.Hold(callid)
+ time.sleep(0.5)
+
+ self.sflphone.UnHold(callid)
+ time.sleep(0.5)
+
+ self.sflphone.HangUp(callid)
+ time.sleep(0.5)
+
+ i = i+1
+
+ del self.sflphone
+
+
+ def test_account_send_transfer(self):
+ """Send new calls, transfer it to a new instance"""
+
+ i = 0
+ while(i < 1):
+
+ callid = self.sflphone.Call("27182")
+ time.sleep(1.0)
+
+ self.sflphone.Transfer(callid,"14142")
+ # self.sflphone.HangUp(callid)
+ # time.sleep(1.0)
+
+ i = i+1
@@ -183,18 +228,52 @@ sflphone.setFirstRegisteredAccount();
# Test 1: - Send an IP2IP call
# - Hangup
# testsuite.test_ip2ip_send_hangup()
+
+# Test 2: - Send an IP2IP call
+# - Peer Hangup
# testsuite.test_ip2ip_send_peer_hungup()
+
+# Test 3: - Receive an IP2IP call
+# - Hangup
# testsuite.test_ip2ip_recv_hangup()
+
+# Test 4: - Receive an IP2IP call
+# - Peer Hangup
# testsuite.test_ip2ip_recv_peer_hungup()
-# SCENARIO 2: IP2IP Normal flow calls
+# SCENARIO 2: ACCOUNT Normal flow calls
+
+# Test 1: - Send an ACCOUNT call
+# - Hangup
+# testsuite.test_account_send_hangup()
+
+# Test 2: - Send an ACCOUNT call
+# - Peer Hangup
+# testsuite.test_account_send_peer_hungup()
+
+# Test 3: - Receive an ACCOUNT call
+# - Hangup
+# testsuite.test_account_recv_hangup()
-# Test 1: - Create an account on Asterisk
-# - Wait for incoming calls
-# - Answer
-# - Call is hanged up by calle
-testsuite.test_account_send_hangup()
-# testsuite.test_account_recv_call_peer_hungup()
+# Test 4: - Receive an ACCOUNT call
+# - Peer Hangup
+# testsuite.test_account_recv_peer_hungup()
+
+# SCENARIO 3: IP2IP Call, HOLD/OFFHOLD
+
+# Test 1: - Send an IP2IP call
+# - Put this call on HOLD
+# - Off HOLD this call
+# - Hangup
+# testsuite.test_ip2ip_send_hold_offhold()
+
+
+# SCENARIO 4: IP2IP Call, HOLD/OFFHOLD
+
+# Test 1: - Send an IP2IP call
+# - Transfer this call to another sipp instance
+# - Hangup
+testsuite.test_account_send_transfer()
diff --git a/tools/pysflphone/sflphonectrlsimple.py b/tools/pysflphone/sflphonectrlsimple.py
index a073e6ca04372fe09f1b74203a2ce1251ddb5624..a19d4ab0901f4b28fb2081d3f68fc58a5419b2e1 100755
--- a/tools/pysflphone/sflphonectrlsimple.py
+++ b/tools/pysflphone/sflphonectrlsimple.py
@@ -518,13 +518,13 @@ class SflPhoneCtrlSimple(Thread):
self.callmanager.hangUp(callid)
- def Transfert(self, callid, to):
+ def Transfer(self, callid, to):
"""Transfert a call identified by a CallID"""
- if not self.account:
- self.setFirstRegisteredAccount()
+ # if not self.account:
+ # self.setFirstRegisteredAccount()
- if not self.isAccountRegistered():
- raise SflPhoneError("Can't transfert a call without a registered account")
+ # if not self.isAccountRegistered():
+ # raise SflPhoneError("Can't transfert a call without a registered account")
if callid is None or callid == "":
raise SflPhoneError("Invalid callID")
@@ -562,11 +562,11 @@ class SflPhoneCtrlSimple(Thread):
def Hold(self, callid):
"""Hold a call identified by a CallID"""
- if not self.account:
- self.setFirstRegisteredAccount()
+ # if not self.account:
+ # self.setFirstRegisteredAccount()
- if not self.isAccountRegistered():
- raise SflPhoneError("Can't hold a call without a registered account")
+ # if not self.isAccountRegistered():
+ # raise SflPhoneError("Can't hold a call without a registered account")
if callid is None or callid == "":
raise SflPhoneError("Invalid callID")
@@ -576,11 +576,11 @@ class SflPhoneCtrlSimple(Thread):
def UnHold(self, callid):
"""Unhold an incoming call identified by a CallID"""
- if not self.account:
- self.setFirstRegisteredAccount()
+ # if not self.account:
+ # self.setFirstRegisteredAccount()
- if not self.isAccountRegistered():
- raise SflPhoneError("Can't unhold a call without a registered account")
+ # if not self.isAccountRegistered():
+ # raise SflPhoneError("Can't unhold a call without a registered account")
if callid is None or callid == "":
raise SflPhoneError("Invalid callID")