diff --git a/sflphone-client-gnome/src/codeclist.c b/sflphone-client-gnome/src/codeclist.c
index 8887452247081200ed34155ca701900b38e6e205..88136547d4430a69c324425b91b2968826e1a768 100644
--- a/sflphone-client-gnome/src/codeclist.c
+++ b/sflphone-client-gnome/src/codeclist.c
@@ -290,8 +290,6 @@ void codec_list_update_to_daemon (account_t *acc) {
int c = 0;
unsigned int i = 0;
- g_print ("List of active codecs :\n");
-
for(i = 0; i < length; i++)
{
codec_t* currentCodec = codec_list_get_nth (i, acc->codecs);
@@ -301,7 +299,6 @@ void codec_list_update_to_daemon (account_t *acc) {
// Save only if active
if(currentCodec->is_active)
{
- g_print ("Codec %s\n", currentCodec->name);
// Reallocate memory each time more than one active codec is found
if(c!=0)
codecList = (void*)realloc(codecList, (c+1)*sizeof(void*));
@@ -311,7 +308,6 @@ void codec_list_update_to_daemon (account_t *acc) {
// Put payload string in char array
sprintf(payload, "%d", currentCodec->_payload);
strcpy((char*)*(codecList+c), payload);
- g_print(" %s", *(codecList+c));
c++;
}
}
diff --git a/sflphone-client-gnome/src/uimanager.c b/sflphone-client-gnome/src/uimanager.c
index 1042bf0c13682a5aef2eee6a5d1a4bfc8c174bab..26b35661b5ae22367fe6d355df1360cbd3d79cf4 100644
--- a/sflphone-client-gnome/src/uimanager.c
+++ b/sflphone-client-gnome/src/uimanager.c
@@ -115,6 +115,7 @@ update_actions()
gtk_widget_set_sensitive(GTK_WIDGET (holdToolbar), FALSE);
gtk_widget_set_sensitive(GTK_WIDGET (offHoldToolbar), FALSE);
gtk_action_set_sensitive(GTK_ACTION (recordAction), FALSE);
+ gtk_widget_set_sensitive(GTK_WIDGET (recordWidget), FALSE);
gtk_action_set_sensitive(GTK_ACTION (copyAction), FALSE);
gtk_widget_set_sensitive(GTK_WIDGET(contactButton), FALSE);
gtk_widget_set_tooltip_text(GTK_WIDGET (contactButton),
@@ -145,9 +146,9 @@ update_actions()
}
}
- // g_signal_handler_block (GTK_OBJECT (transferToolbar), transfertButtonConnId);
- // gtk_toggle_tool_button_set_active (GTK_TOGGLE_TOOL_BUTTON (transferToolbar), FALSE);
- // g_signal_handler_unblock ( GTK_OBJECT (transferToolbar), transfertButtonConnId);
+ // g_signal_handler_block (GTK_OBJECT (recordWidget), recordButtonConnId);
+ // gtk_toggle_tool_button_set_active (GTK_TOGGLE_TOOL_BUTTON (recordWidget), FALSE);
+ // g_signal_handler_unblock ( GTK_OBJECT (recordWidget), recordButtonConnId);
callable_obj_t * selectedCall = calltab_get_selected_call(active_calltree);
conference_obj_t * selectedConf = calltab_get_selected_conf(active_calltree);
@@ -211,9 +212,7 @@ update_actions()
1);
gtk_widget_set_sensitive(GTK_WIDGET (holdMenu), TRUE);
gtk_widget_set_sensitive(GTK_WIDGET (holdToolbar), TRUE);
- //gtk_image_menu_item_set_image (GTK_IMAGE_MENU_ITEM (holdMenu), gtk_image_new_from_stock (GTK_STOCK_ONHOLD, GTK_ICON_SIZE_MENU));
gtk_widget_set_sensitive(GTK_WIDGET (transferToolbar), TRUE);
- //gtk_action_set_sensitive( GTK_ACTION(newCallMenu),TRUE);
gtk_action_set_sensitive(GTK_ACTION (recordAction), TRUE);
gtk_toolbar_insert(GTK_TOOLBAR (toolbar), GTK_TOOL_ITEM (holdToolbar),
2);
@@ -225,6 +224,7 @@ update_actions()
gtk_toggle_tool_button_set_active(
GTK_TOGGLE_TOOL_BUTTON (transferToolbar), FALSE);
gtk_signal_handler_unblock (transferToolbar, transfertButtonConnId);
+
break;
case CALL_STATE_BUSY:
case CALL_STATE_FAILURE:
@@ -546,7 +546,14 @@ call_pick_up(void * foo UNUSED)
static void
call_hang_up(void)
{
- sflphone_hang_up();
+ /*
+ * [#3020] Restore the record toggle button
+ * We set it to FALSE, as when we hang up a call, the recording is stopped.
+ */
+ gtk_toggle_tool_button_set_active (GTK_TOGGLE_TOOL_BUTTON (recordWidget), FALSE);
+
+ sflphone_hang_up();
+
}
static void
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/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index af885c9964af77e6a9000d2fc603fe00e906cb21..34b6ab9700cd2cf026aa629b08974b9382fb7bcc 100644
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -650,13 +650,15 @@ bool ManagerImpl::transferCall (const CallID& call_id, const std::string& to) {
AccountID accountid;
bool returnValue;
+ _info("Manager: Transfer Call\n");
+
stopTone();
CallID current_call_id = getCurrentCallId();
if (participToConference(call_id)) {
- _debug("Particip to a conference\n");
+ _info("Manager: Particip to a conference\n");
Conference *conf = getConferenceFromCallID(call_id);
@@ -668,8 +670,6 @@ bool ManagerImpl::transferCall (const CallID& call_id, const std::string& to) {
}
} else {
- _debug("Do not Particip to a conference\n");
-
// we are not participating to a conference, current call switched to ""
if (!isConference(current_call_id))
switchCall("");
@@ -685,7 +685,7 @@ bool ManagerImpl::transferCall (const CallID& call_id, const std::string& to) {
accountid = getAccountFromCall(call_id);
if (accountid == AccountNULL) {
- _debug ("! Manager Transfer Call: Call doesn't exists");
+ _warn ("Manager: Call doesn't exists");
return false;
}
@@ -696,18 +696,21 @@ bool ManagerImpl::transferCall (const CallID& call_id, const std::string& to) {
removeWaitingCall(call_id);
- if (_dbus)
- _dbus->getCallManager()->callStateChanged(call_id, "HUNGUP");
-
return returnValue;
}
void ManagerImpl::transferFailed () {
+
+ _debug("UserAgent: Transfer failed");
+
if (_dbus)
_dbus->getCallManager()->transferFailed();
}
void ManagerImpl::transferSucceded () {
+
+ _debug("UserAgent: Transfer succeded");
+
if (_dbus)
_dbus->getCallManager()->transferSucceded();
@@ -1713,7 +1716,7 @@ void ManagerImpl::peerHungupCall (const CallID& call_id) {
AccountID account_id;
bool returnValue;
- _debug ("ManagerImpl::peerHungupCall(%s)", call_id.c_str());
+ _debug ("Manager: Peer hungup call %s", call_id.c_str());
// store the current call id
CallID current_call_id = getCurrentCallId();
@@ -1745,11 +1748,6 @@ void ManagerImpl::peerHungupCall (const CallID& call_id) {
account_id = getAccountFromCall(call_id);
- if (account_id == AccountNULL) {
- _debug ("peerHungupCall: Call doesn't exists");
- return;
- }
-
returnValue = getAccountLink(account_id)->peerHungup(call_id);
}
diff --git a/sflphone-common/src/sip/sipcall.h b/sflphone-common/src/sip/sipcall.h
index f2d93259da6872000a51a3491da79e2d37eef3b4..7d14936729055c65cbd348263fcac35b6fe620f9 100644
--- a/sflphone-common/src/sip/sipcall.h
+++ b/sflphone-common/src/sip/sipcall.h
@@ -97,9 +97,11 @@ class SIPCall : public Call
void setTid(int tid) { _tid = tid; }
void setXferSub(pjsip_evsub* sub) {_xferSub = sub;}
+
pjsip_evsub *getXferSub() {return _xferSub;}
void setInvSession(pjsip_inv_session* inv) {_invSession = inv;}
+
pjsip_inv_session *getInvSession() {return _invSession;}
Sdp* getLocalSDP (void) { return _local_sdp; }
diff --git a/sflphone-common/src/sip/sipvoiplink.cpp b/sflphone-common/src/sip/sipvoiplink.cpp
index a7058bbb7bed2735fae67677c1a973fa6ff74cc6..2309c86923288825ba59aed8ce1b1ab8a04c50ff 100644
--- a/sflphone-common/src/sip/sipvoiplink.cpp
+++ b/sflphone-common/src/sip/sipvoiplink.cpp
@@ -50,6 +50,8 @@
#include <sys/ioctl.h>
#include <linux/if.h>
+#include <map>
+
#define CAN_REINVITE 1
static char * invitationStateMap[] = {
@@ -79,9 +81,17 @@ struct result {
pjsip_server_addresses servers;
};
+/** The default transport (5060) */
pjsip_transport *_localUDPTransport = NULL;
+
+/** The local tls listener */
pjsip_tpfactory *_localTlsListener = NULL;
+/** A map to retreive SFLphone internal call id
+ * Given a SIP call ID (usefull for transaction sucha as transfer)*/
+std::map<std::string, CallID> transferCallID;
+
+
const pj_str_t STR_USER_AGENT = { (char*) "User-Agent", 10 };
/**************** EXTERN VARIABLES AND FUNCTIONS (callbacks) **************************/
@@ -921,10 +931,12 @@ SIPVoIPLink::peerHungup (const CallID& id)
pjsip_tx_data *tdata = NULL;
SIPCall* call;
+ _info("UserAgent: Peer hungup");
+
call = getSIPCall (id);
if (call==0) {
- _debug ("! SIP Error: Call doesn't exist");
+ _warn ("UserAgent: Call doesn't exist");
return false;
}
@@ -947,7 +959,7 @@ SIPVoIPLink::peerHungup (const CallID& id)
// Release RTP thread
if (Manager::instance().isCurrentCall (id)) {
- _debug ("* SIP Info: Stopping AudioRTP for hangup");
+ _debug ("UserAgent: Stopping AudioRTP for hangup");
call->getAudioRtp()->stop();
}
@@ -1107,13 +1119,13 @@ SIPVoIPLink::transfer (const CallID& id, const std::string& to)
account_id = Manager::instance().getAccountFromCall (id);
account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (account_id));
- if (account == NULL) {
- _debug ("SIPVoIPLink::transfer account is null. Returning.");
+ if (!account) {
+ _error("UserAgent: Error: Transfer account is null. Returning.");
return false;
}
- if (call==0) {
- _debug ("! SIP Failure: Call doesn't exist");
+ if (!call) {
+ _error ("UserAgent: Error: Call doesn't exist");
return false;
}
@@ -1126,7 +1138,7 @@ SIPVoIPLink::transfer (const CallID& id, const std::string& to)
pj_cstr (&pjDest, dest.c_str());
}
- _debug ("Transfering to %s", dest.c_str());
+ _info ("UserAgent: Transfering to %s", dest.c_str());
/* Create xfer client subscription. */
pj_bzero (&xfer_cb, sizeof (xfer_cb));
@@ -1135,13 +1147,13 @@ SIPVoIPLink::transfer (const CallID& id, const std::string& to)
status = pjsip_xfer_create_uac (call->getInvSession()->dlg, &xfer_cb, &sub);
if (status != PJ_SUCCESS) {
- _debug ("UserAgent: Unable to create xfer -- %d", status);
+ _warn ("UserAgent: Unable to create xfer -- %d", status);
return false;
}
/* Associate this voiplink of call with the client subscription
* We can not just associate call with the client subscription
- * because after this function, we can not find the cooresponding
+ * because after this function, we can no find the cooresponding
* voiplink from the call any more. But the voiplink is useful!
*/
pjsip_evsub_set_mod_data (sub, getModId(), this);
@@ -1152,15 +1164,21 @@ SIPVoIPLink::transfer (const CallID& id, const std::string& to)
status = pjsip_xfer_initiate (sub, &pjDest, &tdata);
if (status != PJ_SUCCESS) {
- _debug ("UserAgent: Unable to create REFER request -- %d", status);
+ _error ("UserAgent: Unable to create REFER request -- %d", status);
return false;
}
+ // Put SIP call id in map in order to retrieve call during transfer callback
+ std::string callidtransfer(call->getInvSession()->dlg->call_id->id.ptr, call->getInvSession()->dlg->call_id->id.slen);
+ _debug("%s", callidtransfer.c_str());
+ transferCallID.insert(std::pair<std::string, CallID>(callidtransfer, call->getCallId()));
+
+
/* Send. */
status = pjsip_xfer_send_request (sub, tdata);
if (status != PJ_SUCCESS) {
- _debug ("UserAgent: Unable to send REFER request -- %d", status);
+ _error ("UserAgent: Unable to send REFER request -- %d", status);
return false;
}
@@ -1169,7 +1187,12 @@ SIPVoIPLink::transfer (const CallID& id, const std::string& to)
bool SIPVoIPLink::transferStep2 (SIPCall* call)
{
- call->getAudioRtp()->stop();
+
+ // TODO is this the best way to proceed?
+ Manager::instance().peerHungupCall(call->getCallId());
+
+
+
return true;
}
@@ -1490,7 +1513,10 @@ SIPVoIPLink::SIPCallServerFailure (SIPCall *call)
void
SIPVoIPLink::SIPCallClosed (SIPCall *call)
{
+ _info("UserAgent: Closing call");
+
if (!call) {
+ _warn("UserAgent: Error: CAll pointer is NULL\n");
return;
}
@@ -1498,16 +1524,14 @@ SIPVoIPLink::SIPCallClosed (SIPCall *call)
if (Manager::instance().isCurrentCall (id)) {
call->setAudioStart (false);
- _debug ("* SIP Info: Stopping AudioRTP when closing");
+ _debug ("UserAgent: Stopping AudioRTP when closing");
call->getAudioRtp()->stop();
}
- _debug ("After close RTP");
-
Manager::instance().peerHungupCall (id);
terminateOneCall (id);
removeCall (id);
- _debug ("After remove call ID");
+
}
void
@@ -1518,7 +1542,7 @@ SIPVoIPLink::SIPCallReleased (SIPCall *call)
}
// if we are here.. something when wrong before...
- _debug ("SIP call release");
+ _debug ("UserAgent: SIP call release");
CallID id = call->getCallId();
@@ -3313,9 +3337,8 @@ void call_on_forked (pjsip_inv_session *inv, pjsip_event *e)
void call_on_tsx_changed (pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e)
{
- _debug("call_on_tsx_changed to state %s", transactionStateMap[tsx->state]);
+ _debug("UserAgent: Transaction changed to state %s", transactionStateMap[tsx->state]);
-
if (tsx->role==PJSIP_ROLE_UAS && tsx->state==PJSIP_TSX_STATE_TRYING &&
pjsip_method_cmp (&tsx->method, &pjsip_refer_method) ==0) {
@@ -3330,12 +3353,18 @@ void call_on_tsx_changed (pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_
if (r_data->msg_info.msg->line.req.method.id == PJSIP_OTHER_METHOD) {
- std::string method_name = "INFO";
- std::string request = r_data->msg_info.msg->line.req.method.name.ptr;
+ std::string method_info = "INFO";
+ std::string method_notify = "NOTIFY";
+
+ std::string request = pjsip_rx_data_get_info (r_data);
+
+ _debug("UserAgent: %s", request.c_str());
- if (request.find (method_name) != (size_t)-1) {
+ if(request.find (method_notify) != (size_t)-1) {
- _debug ("%s", pjsip_rx_data_get_info (r_data));
+ }
+ // Must reply 200 OK on SIP INFO request
+ else if (request.find (method_info) != (size_t)-1) {
pjsip_dlg_create_response (inv->dlg, r_data, PJSIP_SC_OK, NULL, &t_data);
@@ -3476,7 +3505,7 @@ 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
@@ -3815,8 +3844,7 @@ void onCallTransfered (pjsip_inv_session *inv, pjsip_rx_data *rdata)
* request.
*/
ref_by_hdr = (pjsip_hdr*)
- pjsip_msg_find_hdr_by_name (rdata->msg_info.msg, &str_ref_by,
- NULL);
+ pjsip_msg_find_hdr_by_name (rdata->msg_info.msg, &str_ref_by, NULL);
/* Notify callback */
code = PJSIP_SC_ACCEPTED;
@@ -4002,10 +4030,9 @@ void xfer_func_cb (pjsip_evsub *sub, pjsip_event *event)
* When subscription is accepted (got 200/OK to REFER), check if
* subscription suppressed.
*/
-
if (pjsip_evsub_get_state (sub) == PJSIP_EVSUB_STATE_ACCEPTED) {
- _debug ("Transfer accepted! Waiting for notifications. ");
+ _debug ("UserAgent: Transfer received, waiting for notifications. ");
}
@@ -4014,6 +4041,7 @@ void xfer_func_cb (pjsip_evsub *sub, pjsip_event *event)
*/
else if (pjsip_evsub_get_state (sub) == PJSIP_EVSUB_STATE_ACTIVE ||
pjsip_evsub_get_state (sub) == PJSIP_EVSUB_STATE_TERMINATED) {
+
pjsip_msg *msg;
pjsip_msg_body *body;
pjsip_status_line status_line;
@@ -4021,13 +4049,6 @@ void xfer_func_cb (pjsip_evsub *sub, pjsip_event *event)
pj_bool_t cont;
pj_status_t status;
- std::string noresource;
- std::string ringing;
- std::string request;
-
- noresource = "noresource";
- ringing = "Ringing";
-
SIPVoIPLink *link = reinterpret_cast<SIPVoIPLink *> (pjsip_evsub_get_mod_data (sub, _mod_ua.id));
@@ -4041,101 +4062,73 @@ void xfer_func_cb (pjsip_evsub *sub, pjsip_event *event)
}
+ /* Application is not interested with call progress status */
if (!link || !event) {
- /* Application is not interested with call progress status */
- _debug ("UserAgent: Either link or event is empty!");
+ _warn ("UserAgent: Either link or event is empty in transfer callback");
return;
}
+ pjsip_rx_data* r_data = event->body.rx_msg.rdata;
- /* This better be a NOTIFY request */
- if (event->type == PJSIP_EVENT_TSX_STATE &&
- event->body.tsx_state.type == PJSIP_EVENT_RX_MSG) {
-
- pjsip_rx_data *rdata;
-
- rdata = event->body.tsx_state.src.rdata;
+ std::string method_notify = "NOTIFY";
+ std::string request = pjsip_rx_data_get_info (r_data);
+ /* This better be a NOTIFY request */
+ if (r_data->msg_info.msg->line.req.method.id == PJSIP_OTHER_METHOD &&
+ request.find(method_notify) != (size_t)-1) {
/* Check if there's body */
- msg = rdata->msg_info.msg;
+ msg = r_data->msg_info.msg;
body = msg->body;
if (!body) {
- // if (call->getCallConfiguration () == Call::IPtoIP) {
- // _debug("UserAgent: IptoIp NOTIFY without message body");
- // }
- // else{
- _debug ("UserAgent: Warning! Received NOTIFY without message body");
+ _warn ("UserAgent: Warning! Received NOTIFY without message body");
return;
- // }
}
-
-
/* Check for appropriate content */
if (pj_stricmp2 (&body->content_type.type, "message") != 0 ||
pj_stricmp2 (&body->content_type.subtype, "sipfrag") != 0) {
- _debug ("UserAgent: Warning! Received NOTIFY with non message/sipfrag content");
+ _warn ("UserAgent: Warning! Received NOTIFY without message/sipfrag content");
return;
}
/* Try to parse the content */
- status = pjsip_parse_status_line ( (char*) body->data, body->len,
- &status_line);
+ status = pjsip_parse_status_line ( (char*) body->data, body->len, &status_line);
if (status != PJ_SUCCESS) {
- _debug ("UserAgent: Warning! Received NOTIFY with invalid message/sipfrag content");
+ _warn ("UserAgent: Warning! Received NOTIFY with invalid message/sipfrag content");
return;
}
} else {
- _debug ("UserAgent: Set code to 500!");
+ _error ("UserAgent: Error: Set code to 500 during transfer");
status_line.code = 500;
status_line.reason = *pjsip_get_status_text (500);
}
- // Get current call
- SIPCall *call = dynamic_cast<SIPCall *> (link->getCall (Manager::instance().getCurrentCallId()));
+ // Get call coresponding to this transaction
+ std::string transferID(r_data->msg_info.cid->id.ptr, r_data->msg_info.cid->id.slen);
+ std::map<std::string, CallID>::iterator it = transferCallID.find(transferID);
+ CallID cid = it->second;
+ SIPCall *call = dynamic_cast<SIPCall *> (link->getCall (cid));
if (!call) {
- _debug ("UserAgent: Call doesn't exit!");
+ _warn ("UserAgent: Call with id %s doesn't exit!", cid.c_str());
return;
}
- if (event->body.rx_msg.rdata->msg_info.msg_buf != NULL) {
- request = event->body.rx_msg.rdata->msg_info.msg_buf;
-
- if ( (int) request.find (noresource) != -1) {
- _debug ("UserAgent: NORESOURCE for transfer!");
- link->transferStep2 (call);
- pjsip_evsub_terminate (sub, PJ_TRUE);
-
- Manager::instance().transferFailed();
- return;
- }
-
- if ( (int) request.find (ringing) != -1) {
- _debug ("UserAgent: transfered call RINGING!");
- link->transferStep2 (call);
- pjsip_evsub_terminate (sub, PJ_TRUE);
-
- Manager::instance().transferSucceded();
- return;
- }
- }
-
-
/* Notify application */
is_last = (pjsip_evsub_get_state (sub) ==PJSIP_EVSUB_STATE_TERMINATED);
cont = !is_last;
+ _debug("UserAgent: Notification status line: %d", status_line.code);
if (status_line.code/100 == 2) {
- _debug ("UserAgent: Try to stop rtp!");
+ _debug ("UserAgent: Received 200 OK on call transfered, stop call!");
pjsip_tx_data *tdata;
status = pjsip_inv_end_session (call->getInvSession(), PJSIP_SC_GONE, NULL, &tdata);
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..94bea4704232386a5518de7995c3138bc3007850
--- /dev/null
+++ b/sippxml/account_uas_recv_hangup.xml
@@ -0,0 +1,119 @@
+<?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>
+
+ <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..1ed0db416504d9abc8288a3a56e8b848d0e46263 100644
--- a/sippxml/testsuiteuac.sh
+++ b/sippxml/testsuiteuac.sh
@@ -1,16 +1,22 @@
#!/bin/bash
-# sleep 5;
-SERVERPORT=5062
-function test_ip2ip_send_hangup {
+LOCALPORT=5062
+LOCALIP_lo=127.0.0.1
+LOCALIP_eth0=192.168.50.182
- # start sipp server to receive calls from sflphone
- sipp -sf ip2ip_uas_recv_peer_hungup.xml -p ${SERVERPORT}
+REMOTEADDR_lo=127.0.0.1:5060
+REMOTEADDR_ast=192.168.50.79
+
+# SCENARIO 1 Test 1
+function test_ip2ip_send_hangup {
# start sflphoned
# /usr/lib/sflphone/sflphoned&
+ # start sipp server to receive calls from sflphone
+ sipp -sf ip2ip_uas_recv_peer_hungup.xml ${REMOTEADDR_lo} -i ${LOCALIP_lo} -p ${LOCALPORT}
+
# wait some time to make sure sflphoned is started
# sleep 1;
@@ -21,14 +27,15 @@ 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}
-
# start sflphoned
# /usr/lib/sflphone/sflphoned&
+ # start sipp server to receive calls from sflphone and then hangup
+ sipp -sf ip2ip_uas_recv_hangup.xml ${REMOTEADDR_lo} -s ${REMOTEADDR_lo} -i ${LOCALIP_lo} -p ${LOCALPORT}
+
# wait some time to make sure sflphoned is started
# sleep 1;
@@ -40,6 +47,7 @@ function test_ip2ip_send_peer_hungup {
}
+# SCENARIO 1 Test 3
function test_ip2ip_recv_hangup {
# start sflphoned
@@ -54,12 +62,14 @@ function test_ip2ip_recv_hangup {
# sleep 1;
# start sipp client and send calls
- sipp -sf ip2ip_uac_send_peer_hungup.xml 127.0.0.1:5060 -i 127.0.0.1 -p ${SERVERPORT} -l 1
+ sipp -sf ip2ip_uac_send_peer_hungup.xml ${REMOTEADDR_lo} -i ${LOCALIP_lo} -p ${LOCALPORT} -l 1 -m 10
# kill every one
# bashtrap
}
+
+# SCENARIO 1 Test 4
function test_ip2ip_recv_peer_hungup {
# start sflphoned
@@ -74,12 +84,13 @@ function test_ip2ip_recv_peer_hungup {
# sleep 1;
# start sipp client and send calls
- sipp -sf ip2ip_uac_send_hangup.xml 127.0.0.1:5060 -i 127.0.0.1 -p ${SERVERPORT} -l 1
+ sipp -sf ip2ip_uac_send_hangup.xml ${REMOTEADDR_lo} -i ${LOCALIP_lo} -p ${LOCALPORT} -l 1 -m 10
# kill every one
# bashtrap
}
+# SCENARIO 2 Test 1
function test_account_send_hangup {
# start sflphoned
@@ -93,13 +104,64 @@ 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 ${REMOTEADDR_ast} -i ${LOCALIP_eth0} -p ${LOCALPORT} -l 1 -m 1
+
+ # start sipp client and send calls
+ sipp -sf account_uas_recv_peer_hungup.xml ${REMOTEADDR_ast} -i ${LOCALIP_eth0} -p ${LOCALPORT} -l 1
+
+ # kill every one
+ # 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 ${REMOTEADDR_ast} -i ${LOCALIP_eth0} -p ${LOCALPORT} -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
+ sipp -sf account_uas_recv_hangup.xml ${REMOTEADDR_ast} -i ${LOCALIP_eth0} -p ${LOCALPORT} -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 ${REMOTEADDR_ast} -i ${LOCALIP_eth0} -p ${LOCALPORT} -l 1 -m 1
+
+ # start sipp client and send calls
+ sipp -sf account_uac_send_peer_hungup.xml ${REMOTEADDR_ast} -i ${LOCALIP_eth0} -p ${LOCALPORT} -l 1 -m 10
+ # kill every one
+ # bashtrap
+}
+
+# SCENARIO 2 Test 4
function test_account_recv_peer_hungup {
# start sflphoned
@@ -113,13 +175,61 @@ 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 ${REMOTEADDR_ast} -i ${LOCALIP_eth0} -p ${LOCALPORT} -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
+ sipp -sf account_uac_send_hangup.xml ${REMOTEADDR_ast} -i ${LOCALIP_eth0} -p ${LOCALPORT} -l 1 -m 10
# kill every one
# 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 ${REMOTEADDR_lo} -i ${LOCALIP_lo} -p ${LOCALPORT}
+ # 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 ${REMOTEADDR_ast} -i ${LOCALIP_eth0} -p ${LOCALPORT} -l 1 -m 1
+
+ # start sipp client and send calls
+ sipp -sf account_uas_recv_transfered.xml ${REMOTEADDR_ast} -i ${LOCALIP_eth0} -p ${LOCALPORT} -l 1
+
+ # kill every one
+ # bashtrap
+}
+
+
# function called if CTRL-C detected
bashtrap()
{
@@ -130,12 +240,61 @@ bashtrap()
# ============================ Test Suite ============================
-# SCENARIO 1: Normal flow calls (IP2IP)
+
+
+# SCENARIO 1: (IP2IP) Normal flow calls
+
+# Test 1: - Send an IP2IP call
+# - Hangup
# test_ip2ip_send_hangup
+
+# Test 2: - Send an IP2IP call
+# - Peer Hangup
# test_ip2ip_send_peer_hungup
+
+# Test 3: - Receive an IP2IP call
+# - Hangup
# test_ip2ip_recv_hangup
+
+# Test 4: - Receive an IP2IP call
+# - Peer 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: (ACCOUNT) Normal flow calls
+
+# Test 1: - Send an ACCOUNT call
+# - Hangup
+# test_account_send_hangup
+
+# Test 2: - Send an ACCOUNT call
+# - Peer Hangup
+# test_account_send_peer_hungup
+
+# Test 3: - Receive an ACCOUNT call
+# - Hangup
+# test_account_recv_hangup
+
+# Test 4: - Receive an ACCOUNT call
+# - Peer Hangup
+# test_account_recv_peer_hungup
+
+
+
+# SCENARIO 3: Hold/offHold calls (Account)
+
+# Test 1: - Send an IP2IP call
+# - Put this call on HOLD
+# - Off HOLD this call
+# - Hangup
+test_ip2ip_send_hold_offhold
+
+
+
+# SCENARIO 4: Transfer calls (Account)
+
+# Test 1: - Send an IP2IP call
+# - Transfer this call to another sipp instance
+# - Hangup
+# 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..a1c909c531fb3a2b73306c59466039e26bbcefb1 100644
--- a/tools/pysflphone/pysflphone_testdbus.py
+++ b/tools/pysflphone/pysflphone_testdbus.py
@@ -13,21 +13,29 @@ from threading import Event
print "Import SFLphone"
from sflphonectrlsimple import SflPhoneCtrlSimple
-# def killhandler(signum, frame):
-# raise IOError("Couldn't open device!")
+# Define remote IP address constant
+REMOTEADDR_lo="127.0.0.1:5062"
-# signal.signal(signal.SIGKILL, killhandler)
+# Defines phone numbers
+PHONE1="27182"
+PHONE2="31416"
+PHONE3="14142"
+# Define function callback to emulate UA behavior on
+# recieving a call
def acceptOnIncomingCall(sflphone):
time.sleep(0.2)
sflphone.Accept(sflphone.currentCallId)
+# Define function callback to emulate UA behavior on
+# receiving a call and hanging up
def acceptOnIncomingCallHangup(sflphone):
time.sleep(0.2)
sflphone.Accept(sflphone.currentCallId)
time.sleep(0.5)
sflphone.HangUp(sflphone.currentCallId)
+
class SflPhoneTests():
def __init__(self, sfl):
@@ -75,12 +83,13 @@ class SflPhoneTests():
print "Account with ID " + accountID + " removed"
+ # SCENARIO 1 Test 1
def test_ip2ip_send_hangup(self):
"""Make a call to a server (sipp) on port 5062"""
i = 0
while(i < 10):
- callid = self.sflphone.Call("sip:test@127.0.0.1:5062")
+ callid = self.sflphone.Call("sip:test@" + REMOTEADDR_lo)
time.sleep(0.5)
self.sflphone.HangUp(callid)
@@ -91,19 +100,21 @@ class SflPhoneTests():
del self.sflphone
+ # SCENARIO 1 Test 2
def test_ip2ip_send_peer_hungup(self):
"""Make a call to a server (sipp) on port 5062"""
i = 0
- while(i < 1):
+ while(i < 10):
- callid = self.sflphone.Call("sip:test@127.0.0.1:5062")
+ callid = self.sflphone.Call("sip:test@" + REMOTEADDR_lo)
time.sleep(1.0)
i = i+1
del self.sflphone
-
+
+ # SCENARIO 1 Test 3
def test_ip2ip_recv_hangup(self):
"""Wait for calls, answer then hangup"""
@@ -113,6 +124,8 @@ class SflPhoneTests():
# Start Glib mainloop
self.sflphone.start()
+
+ # SCENARIO 1 Test 4
def test_ip2ip_recv_peer_hungup(self):
"""Wait for calls, answer, peer hangup"""
# Add callback for this test
@@ -121,24 +134,53 @@ class SflPhoneTests():
# Start Glib mainloop
self.sflphone.start()
+
+ # SCENARIO 2 Test 1
def test_account_send_hangup(self):
"""Send new account call, hangup once peer answered"""
- print "test account send hangup"
+
i = 0
while(i < 10):
- callid = self.sflphone.Call("27182")
- time.sleep(0.5)
+ callid = self.sflphone.Call(PHONE1)
+ time.sleep(0.2)
self.sflphone.HangUp(callid)
- time.sleep(0.5)
+ time.sleep(0.2)
+
+ i = i+1
+
+ # del self.sflphone
+
+
+ # SCENARIO 2 Test 2
+ def test_account_send_peer_hungup(self):
+ """Send new account call, hangup once peer answered"""
+
+ i = 0
+ while(i < 10):
+
+ callid = self.sflphone.Call(PHONE1)
+ time.sleep(1.0)
i = i+1
- print "account hangup done"
del self.sflphone
- def test_account_recv_call_peer_hungup(self):
+
+ # SCENARIO 2 Test 3
+ 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()
+
+
+ # SCENARIO 2 Test 4
+ def test_account_recv_peer_hungup(self):
"""Register an account and wait for incoming calls"""
# Add callback for this test
@@ -147,24 +189,45 @@ 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
+ # SCENARIO 3 Test 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@" + REMOTEADDR_lo)
+ 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
+
+
+ # SCENARIO 4 Test 1
+ def test_account_send_transfer(self):
+ """Send new calls, transfer it to a new instance"""
+
+ i = 0
+ while(i < 1):
+
+ callid = self.sflphone.Call(PHONE1)
+ time.sleep(1.0)
+
+ self.sflphone.Transfer(callid,PHONE3)
+ # self.sflphone.HangUp(callid)
+ # time.sleep(1.0)
+
+ i = i+1
@@ -178,23 +241,63 @@ testsuite = SflPhoneTests(sflphone)
sflphone.setFirstRegisteredAccount();
+# ============================ Test Suite ============================
+
+
+
# SCENARIO 1: IP2IP Normal flow calls
# 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 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()
+
+
-# 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()
+# 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")