From b49ffaf7917cfb900e3513c67fc1b94f29a27510 Mon Sep 17 00:00:00 2001
From: yanmorin <yanmorin>
Date: Mon, 17 Oct 2005 19:38:13 +0000
Subject: [PATCH] Debugging on Now the wav play correctly, and more often
---
src/audio/audiocodec.h | 2 -
src/audio/tonegenerator.cpp | 4 +-
src/managerimpl.cpp | 36 +++++++----
src/sipvoiplink.cpp | 117 ++++++++++++++++++------------------
src/sipvoiplink.h | 2 +-
5 files changed, 84 insertions(+), 77 deletions(-)
diff --git a/src/audio/audiocodec.h b/src/audio/audiocodec.h
index cf682f89bf..a2cfdb14ed 100644
--- a/src/audio/audiocodec.h
+++ b/src/audio/audiocodec.h
@@ -21,9 +21,7 @@
#ifndef __CODEC_AUDIO_H__
#define __CODEC_AUDIO_H__
-
#include <string>
-#include "../global.h"
class AudioCodec {
public:
diff --git a/src/audio/tonegenerator.cpp b/src/audio/tonegenerator.cpp
index 31f83bd0bd..615efc6fe8 100644
--- a/src/audio/tonegenerator.cpp
+++ b/src/audio/tonegenerator.cpp
@@ -385,6 +385,7 @@ ToneGenerator::playRingtone (const char *fileName) {
file.close();
// Decode file.ul
+ // expandedsize is the number of bytes, not the number of int
expandedsize = _ulaw->codecDecode (_dst, (unsigned char *)_src, length);
_debug("length (pre-ulaw) : %d\n", length);
@@ -392,7 +393,8 @@ ToneGenerator::playRingtone (const char *fileName) {
if (tonethread == NULL) {
_debug("Thread: start tonethread\n");
- tonethread = new ToneThread ((int16*)_dst, expandedsize);
+ // send the number of int16, so device by two
+ tonethread = new ToneThread ((int16*)_dst, expandedsize>>1);
tonethread->start();
}
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 9494ce5d83..9f0b0e64bd 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -449,7 +449,7 @@ ManagerImpl::registerVoIPLink (void)
if (_voIPLinkVector.at(DFT_VOIP_LINK)->setRegister() >= 0) {
returnValue = true;
} else {
- _debug("ManagerImpl::registerVoIPLink: Registration Failed\n");
+ _debug("ManagerImpl::registerVoIPLink: Registration failed\n");
}
}
return returnValue;
@@ -741,10 +741,10 @@ ManagerImpl::displayErrorText (CALLID id, const std::string& message)
* for outgoing call, send by SipEvent
*/
void
-ManagerImpl::displayError (const std::string& error)
+ManagerImpl::displayError (const std::string& voIPError)
{
if(_gui) {
- _gui->displayError(error);
+ _gui->displayError(voIPError);
}
}
@@ -1218,9 +1218,14 @@ ManagerImpl::getCallStatus(const std::string& sequenceId)
while(iter!=_callVector.end()){
call = (*iter);
switch( call->getState() ) {
- case Call::Busy:
- code="113";
- status = "Busy";
+ case Call::Progressing:
+ code="110";
+ status="Trying";
+ break;
+
+ case Call::Ringing:
+ code="111";
+ status = "Ringing";
break;
case Call::Answered:
@@ -1228,18 +1233,23 @@ ManagerImpl::getCallStatus(const std::string& sequenceId)
status = "Established";
break;
- case Call::Ringing:
- code="111";
- status = "Ringing";
+ case Call::Busy:
+ code="113";
+ status = "Busy";
break;
- case Call::Progressing:
- code="110";
- status="Trying";
+ case Call::OnHold:
+ code="114";
+ status = "Holded";
break;
- default:
+ case Call::OffHold:
code="115";
+ status = "Unholded";
+ break;
+
+ default:
+ code="125";
status="Other";
}
// No Congestion
diff --git a/src/sipvoiplink.cpp b/src/sipvoiplink.cpp
index f0873dd4b9..49b7f35426 100644
--- a/src/sipvoiplink.cpp
+++ b/src/sipvoiplink.cpp
@@ -338,14 +338,13 @@ SipVoIPLink::answer (CALLID id)
bzero (tmpbuf, 64);
// Get port
snprintf (tmpbuf, 63, "%d", getSipCall(id)->getLocalAudioPort());
-
- _debug("Answer call [id = %d, cid = %d, did = %d]\n",
- id, getSipCall(id)->getCid(), getSipCall(id)->getDid());
+
+ _debug("Answer call [id = %d, cid = %d, did = %d]\n", id, getSipCall(id)->getCid(), getSipCall(id)->getDid());
port = getSipCall(id)->getLocalAudioPort();
_debug("Local audio port: %d\n", port);
-
-
- osip_message_t *answer = NULL;
+
+
+ osip_message_t *answerMessage = NULL;
SipCall* ca = getSipCall(id);
// Send 180 RINGING
@@ -355,18 +354,18 @@ SipVoIPLink::answer (CALLID id)
// Send 200 OK
eXosip_lock();
- i = eXosip_call_build_answer (ca->getTid(), OK, &answer);
+ i = eXosip_call_build_answer (ca->getTid(), OK, &answerMessage);
if (i != 0) {
// Send 400 BAD_REQUEST
eXosip_call_send_answer (ca->getTid(), BAD_REQ, NULL);
} else {
- i = sdp_complete_200ok (ca->getDid(), answer, port);
+ i = sdp_complete_200ok (ca->getDid(), answerMessage, port);
if (i != 0) {
- osip_message_free (answer);
+ osip_message_free (answerMessage);
// Send 415 UNSUPPORTED_MEDIA_TYPE
eXosip_call_send_answer (ca->getTid(), UNSUP_MEDIA_TYPE, NULL);
} else {
- eXosip_call_send_answer (ca->getTid(), OK, answer);
+ eXosip_call_send_answer (ca->getTid(), OK, answerMessage);
}
}
eXosip_unlock();
@@ -587,12 +586,12 @@ SipVoIPLink::refuse (CALLID id)
// Get local port
snprintf (tmpbuf, 63, "%d", getSipCall(id)->getLocalAudioPort());
- osip_message_t *answer = NULL;
+ osip_message_t *answerMessage = NULL;
eXosip_lock();
// not BUSY.. where decline the invitation!
- i = eXosip_call_build_answer (getSipCall(id)->getTid(), SIP_DECLINE, &answer);
+ i = eXosip_call_build_answer (getSipCall(id)->getTid(), SIP_DECLINE, &answerMessage);
if (i == 0) {
- i = eXosip_call_send_answer (getSipCall(id)->getTid(), SIP_DECLINE, answer);
+ i = eXosip_call_send_answer (getSipCall(id)->getTid(), SIP_DECLINE, answerMessage);
}
eXosip_unlock();
return i;
@@ -753,7 +752,7 @@ SipVoIPLink::getEvent (void)
_debug("Call is closed [id = %d, cid = %d, did = %d]\n", id, event->cid, event->did);
if (id != 0) {
if (Manager::instance().callCanBeClosed(id)) {
- SipCall* sipcall = getSipCall(id);
+ sipcall = getSipCall(id);
if ( sipcall != NULL ) { sipcall->enable_audio = false; }
_audiortp.closeRtpSession();
}
@@ -1139,7 +1138,7 @@ SipVoIPLink::sdp_off_hold_call (sdp_message_t * sdp)
}
int
-SipVoIPLink::sdp_complete_200ok (int did, osip_message_t * answer, int port)
+SipVoIPLink::sdp_complete_200ok (int did, osip_message_t * answerMessage, int port)
{
sdp_message_t *remote_sdp;
sdp_media_t *remote_med;
@@ -1161,10 +1160,10 @@ SipVoIPLink::sdp_complete_200ok (int did, osip_message_t * answer, int port)
eXosip_guess_localip (AF_INET, localip, 128);
snprintf (buf, 4096,
- "v=0\r\n"
- "o=user 0 0 IN IP4 %s\r\n"
- "s=session\r\n" "c=IN IP4 %s\r\n" "t=0 0\r\n", localip,localip);
-
+ "v=0\r\n"
+ "o=user 0 0 IN IP4 %s\r\n"
+ "s=session\r\n" "c=IN IP4 %s\r\n" "t=0 0\r\n", localip,localip);
+
pos = 0;
while (!osip_list_eol (remote_sdp->m_medias, pos)) {
char payloads[128];
@@ -1176,40 +1175,40 @@ SipVoIPLink::sdp_complete_200ok (int did, osip_message_t * answer, int port)
if (0 == osip_strcasecmp (remote_med->m_media, "audio")) {
pos2 = 0;
while (!osip_list_eol (remote_med->m_payloads, pos2)) {
- tmp = (char *) osip_list_get (remote_med->m_payloads, pos2);
- if (tmp != NULL && (0 == osip_strcasecmp (tmp, "0")
- || 0 == osip_strcasecmp (tmp, "8")
- || 0 == osip_strcasecmp (tmp, "3"))) {
- strcat (payloads, tmp);
- strcat (payloads, " ");
- }
- pos2++;
+ tmp = (char *) osip_list_get (remote_med->m_payloads, pos2);
+ if (tmp != NULL && (0 == osip_strcasecmp (tmp, "0")
+ || 0 == osip_strcasecmp (tmp, "8")
+ || 0 == osip_strcasecmp (tmp, "3"))) {
+ strcat (payloads, tmp);
+ strcat (payloads, " ");
+ }
+ pos2++;
}
strcat (buf, "m=");
strcat (buf, remote_med->m_media);
if (pos2 == 0 || payloads[0] == '\0') {
- strcat (buf, " 0 RTP/AVP \r\n");
- sdp_message_free (remote_sdp);
- return -1; /* refuse anyway */
+ strcat (buf, " 0 RTP/AVP \r\n");
+ sdp_message_free (remote_sdp);
+ return -1; /* refuse anyway */
} else {
- strcat (buf, " ");
- strcat (buf, port_tmp);
- strcat (buf, " RTP/AVP ");
- strcat (buf, payloads);
- strcat (buf, "\r\n");
-
- if (NULL != strstr (payloads, " 0 ")
- || (payloads[0] == '0' && payloads[1] == ' ')) {
- strcat (buf, "a=rtpmap:0 PCMU/8000\r\n");
- }
- if (NULL != strstr (payloads, " 8 ")
- || (payloads[0] == '8' && payloads[1] == ' ')) {
- strcat (buf, "a=rtpmap:8 PCMA/8000\r\n");
- }
- if (NULL != strstr (payloads, " 3")
- || (payloads[0] == '3' && payloads[1] == ' ')) {
- strcat (buf, "a=rtpmap:3 GSM/8000\r\n");
- }
+ strcat (buf, " ");
+ strcat (buf, port_tmp);
+ strcat (buf, " RTP/AVP ");
+ strcat (buf, payloads);
+ strcat (buf, "\r\n");
+
+ if (NULL != strstr (payloads, " 0 ")
+ || (payloads[0] == '0' && payloads[1] == ' ')) {
+ strcat (buf, "a=rtpmap:0 PCMU/8000\r\n");
+ }
+ if (NULL != strstr (payloads, " 8 ")
+ || (payloads[0] == '8' && payloads[1] == ' ')) {
+ strcat (buf, "a=rtpmap:8 PCMA/8000\r\n");
+ }
+ if (NULL != strstr (payloads, " 3")
+ || (payloads[0] == '3' && payloads[1] == ' ')) {
+ strcat (buf, "a=rtpmap:3 GSM/8000\r\n");
+ }
}
} else {
strcat (buf, "m=");
@@ -1221,8 +1220,8 @@ SipVoIPLink::sdp_complete_200ok (int did, osip_message_t * answer, int port)
pos++;
}
- osip_message_set_body (answer, buf, strlen (buf));
- osip_message_set_content_type (answer, "application/sdp");
+ osip_message_set_body (answerMessage, buf, strlen (buf));
+ osip_message_set_content_type (answerMessage, "application/sdp");
sdp_message_free (remote_sdp);
return 0;
}
@@ -1335,15 +1334,13 @@ SipVoIPLink::toHeader(const std::string& to)
}
int
-SipVoIPLink::startCall (CALLID id, const std::string& from, const std::string& to,
- const std::string& subject, const std::string& route)
+SipVoIPLink::startCall (CALLID id, const std::string& from, const std::string& to, const std::string& subject, const std::string& route)
{
SipCall* sipcall = getSipCall(id);
if ( sipcall == NULL) {
return -1; // error, we can't find the sipcall
}
osip_message_t *invite;
- int i;
if (checkUrl(from) != 0) {
Manager::instance().displayConfigError("Error for 'From' header");
@@ -1372,7 +1369,7 @@ SipVoIPLink::startCall (CALLID id, const std::string& from, const std::string& t
sipcall->setLocalAudioPort(_localPort);
sipcall->setLocalIp(getLocalIpAddress());
- i = eXosip_call_build_initial_invite (&invite, (char*)to.data(),
+ int i = eXosip_call_build_initial_invite (&invite, (char*)to.data(),
(char*)from.data(),
(char*)route.data(),
(char*)subject.data());
@@ -1390,20 +1387,20 @@ SipVoIPLink::startCall (CALLID id, const std::string& from, const std::string& t
bzero(rtpmap_attr, 2048);
bzero(media, 64);
bzero(media_audio, 64);
-
+
// Set rtpmap according to the supported codec order
CodecDescriptorVector* cdv = Manager::instance().getCodecDescVector();
unsigned int nb = cdv->size();
- for (unsigned int i = 0; i < nb; i++) {
- payload = cdv->at(i)->getPayload();
+ for (unsigned int iCodec = 0; iCodec < nb; iCodec++) {
+ payload = cdv->at(iCodec)->getPayload();
// Add payload to rtpmap if it is not already added
- if (!isInRtpmap(i, payload, cdv)) {
+ if (!isInRtpmap(iCodec, payload, cdv)) {
snprintf(media, 63, "%d ", payload);
strcat (media_audio, media);
-
+
snprintf(rtpmap, 127, "a=rtpmap: %d %s/%d\r\n", payload,
- cdv->at(i)->rtpmapPayload(payload).data(), SAMPLING_RATE);
+ cdv->at(iCodec)->rtpmapPayload(payload).data(), SAMPLING_RATE);
strcat(rtpmap_attr, rtpmap);
}
}
diff --git a/src/sipvoiplink.h b/src/sipvoiplink.h
index 1e79dbebe7..aae96dafa7 100644
--- a/src/sipvoiplink.h
+++ b/src/sipvoiplink.h
@@ -187,7 +187,7 @@ private:
/*
* To build sdp with 200 OK when answered call
*/
- int sdp_complete_200ok(int did, osip_message_t * answer, int port);
+ int sdp_complete_200ok(int did, osip_message_t * answerMessage, int port);
/*
* To build sdp when call is on-hold
--
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