* #6905: simplify SIP code

try to make it more readable/understandable

daemon/src/account.h

@@ -70,7 +70,6 @@ static const char *const CONFIG_ACCOUNT_TYPE                 = "Account.type";
 static const char *const CONFIG_ACCOUNT_ALIAS                = "Account.alias";
 static const char *const CONFIG_ACCOUNT_MAILBOX	             = "Account.mailbox";
 static const char *const CONFIG_ACCOUNT_ENABLE               = "Account.enable";
-static const char *const CONFIG_ACCOUNT_RESOLVE_ONCE         = "Account.resolveOnce";
 static const char *const CONFIG_ACCOUNT_REGISTRATION_EXPIRE  = "Account.expire";
 static const char *const CONFIG_CREDENTIAL_NUMBER            = "Credential.count";
 static const char *const ACCOUNT_DTMF_TYPE                   = "Account.dtmfType";

daemon/src/audio/audiortp/AudioRtpFactory.cpp

@@ -248,5 +248,3 @@ void AudioRtpFactory::sendDtmfDigit (int digit)
 }
 
 }
-
-

daemon/src/audio/audiortp/AudioRtpRecordHandler.cpp

@@ -30,6 +30,7 @@
 #include "AudioRtpRecordHandler.h"
 #include <fstream>
 
+#include "sip/sipcall.h"
 #include "audio/audiolayer.h"
 #include "manager.h"
 

daemon/src/audio/audiortp/AudioRtpRecordHandler.h

@@ -36,7 +36,7 @@ using std::ptrdiff_t;
 #include <ccrtp/rtp.h>
 #include <list>
 
-#include "sip/sipcall.h"
+class SIPCall;
 #include "audio/codecs/audiocodec.h"
 #include "audio/samplerateconverter.h"
 #include "audio/noisesuppress.h"
@@ -105,7 +105,7 @@ class AudioRtpRecord
 class AudioRtpRecordHandler
 {
     public:
-        AudioRtpRecordHandler (SIPCall *ca);
+        AudioRtpRecordHandler(SIPCall *);
         virtual ~AudioRtpRecordHandler();
 
         /**

daemon/src/call.cpp

@@ -87,51 +87,26 @@ Call::getState()
 std::string
 Call::getStateStr ()
 {
-    CallState state = getState();
-    ConnectionState connection = getConnectionState ();
-    CallType type = _type;
-    std::string state_str;
-
-    switch (state) {
+    switch (getState()) {
     case Active:
-        switch (connection) {
-        case Ringing:
-        	(type == Incoming) ? state_str = "INCOMING":state_str = "RINGING";
-        	break;
+        switch (getConnectionState()) {
+        case Ringing: 	return isIncoming() ? "INCOMING" : "RINGING";
         case Connected:
-        default:
-            isRecording() ? state_str = "RECORD" : state_str = "CURRENT";
-            break;
+        default:		return isRecording() ? "RECORD" : "CURRENT";
         }
-    case Hold:
-    	state_str = "HOLD";
-    	break;
-    case Busy:
-    	state_str = "BUSY";
-    	break;
+    case Hold:			return "HOLD";
+    case Busy:			return "BUSY";
     case Inactive:
-    	switch (connection) {
-    	case Ringing:
-    		(type == Incoming) ? state_str = "INCOMING":state_str = "RINGING";
-    		break;
-    	case Connected:
-    		state_str = "CURRENT";
-    		break;
-    	default:
-    		state_str = "INACTIVE";
-    		break;
+    	switch (getConnectionState()) {
+    	case Ringing:	return isIncoming() ? "INCOMING" : "RINGING";
+    	case Connected:	return "CURRENT";
+    	default:		return "INACTIVE";
     	}
-    	break;
-    case Conferencing:
-    	state_str = "CONFERENCING";
-    	break;
+    case Conferencing:	return "CONFERENCING";
     case Refused:
     case Error:
-    default:
-    	state_str = "FAILURE";
-    	break;
+    default:			return "FAILURE";
     }
-    return state_str;
 }
 
 
@@ -152,36 +127,20 @@ Call::getLocalAudioPort()
 bool
 Call::setRecording()
 {
-    _debug ("Call: Set recording");
-
     bool recordStatus = Recordable::recAudio.isRecording();
 
     Recordable::recAudio.setRecording();
+    MainBuffer *mbuffer = Manager::instance().getMainBuffer();
+    std::string process_id = Recordable::recorder.getRecorderID();
 
-    // Start recording
     if (!recordStatus) {
-
-        _debug ("Call: Call not recording yet, set ringbuffers");
-
-        MainBuffer *mbuffer = Manager::instance().getMainBuffer();
-        std::string process_id = Recordable::recorder.getRecorderID();
-
         mbuffer->bindHalfDuplexOut (process_id, _id);
         mbuffer->bindHalfDuplexOut (process_id);
 
         Recordable::recorder.start();
-    }
-    // Stop recording
-    else {
-
-        _debug ("Call: Stop recording");
-
-        MainBuffer *mbuffer = Manager::instance().getMainBuffer();
-        std::string process_id = Recordable::recorder.getRecorderID();
-
+    } else {
         mbuffer->unBindHalfDuplexOut (process_id, _id);
         mbuffer->unBindHalfDuplexOut (process_id);
-
     }
 
     Manager::instance().getMainBuffer()->stateInfo();

daemon/src/call.h

@@ -170,7 +170,7 @@ class Call: public Recordable
          *	      false otherwise
          */
         bool isIncoming() {
-            return (_type == Incoming) ? true : false;
+            return _type == Incoming;
         }
 
         /**

daemon/src/dbus/callmanager.cpp

@@ -33,6 +33,7 @@
 #include "global.h"
 #include "callmanager.h"
 
+#include "sip/sipcall.h"
 #include "sip/sipvoiplink.h"
 #include "audio/audiortp/AudioRtpFactory.h"
 #include "audio/audiortp/AudioZrtpSession.h"
@@ -366,26 +367,15 @@ sfl::AudioZrtpSession * CallManager::getAudioZrtpSession (const std::string& cal
 
     SIPCall *call;
     try {
-        call = link->getSIPCall (callID);
+        call = link->getSIPCall(callID);
     }
     catch (const VoipLinkException &e) {
         throw CallManagerException("Call id " + callID + " is not valid");
     }
 
-    sfl::AudioRtpFactory * audioRtp = NULL;
-    audioRtp = call->getAudioRtp();
-
-    if (!audioRtp) {
-        throw CallManagerException("Failed to get AudioRtpFactory");
-    }
-
-    sfl::AudioZrtpSession * zSession = NULL;
-
-    zSession = audioRtp->getAudioZrtpSession();
-
-    if (!zSession) {
+    sfl::AudioZrtpSession * zSession = call->getAudioRtp()->getAudioZrtpSession();
+    if (!zSession)
         throw CallManagerException("Failed to get AudioZrtpSession");
-    }
 
     return zSession;
 }

daemon/src/sip/Makefile.am

@@ -6,7 +6,6 @@ libsiplink_la_SOURCES = \
 		Pattern.cpp \
 		SdesNegotiator.cpp \
 		sdp.cpp \
-		sdpmedia.cpp \
 		sipaccount.cpp \
 		sipcall.cpp \
 		sipvoiplink.cpp
@@ -15,7 +14,6 @@ noinst_HEADERS = \
 		Pattern.h \
 		SdesNegotiator.h \
 		sdp.h \
-		sdpmedia.h \
 		sipaccount.h \
 		sipcall.h \
 		sipvoiplink.h

daemon/src/sip/sdp.cpp

@@ -31,24 +31,9 @@
  */
 
 #include "sdp.h"
-#include "sdpmedia.h"
 #include "global.h"
 #include "manager.h"
-static const char* const ZRTP_VERSION = "1.10";
-
-static const pj_str_t STR_AUDIO = { (char*) "audio", 5};
-static const pj_str_t STR_VIDEO = { (char*) "video", 5};
-static const pj_str_t STR_IN = { (char*) "IN", 2 };
-static const pj_str_t STR_IP4 = { (char*) "IP4", 3};
-static const pj_str_t STR_IP6 = { (char*) "IP6", 3};
-static const pj_str_t STR_RTP_AVP = { (char*) "RTP/AVP", 7 };
-static const pj_str_t STR_RTP_SAVP = { (char*) "RTP/SAVP", 8 };
-static const pj_str_t STR_SDP_NAME = { (char*) "sflphone", 8 };
-static const pj_str_t STR_SENDRECV = { (char*) "sendrecv", 8 };
-static const pj_str_t STR_RTPMAP = { (char*) "rtpmap", 6 };
-static const pj_str_t STR_CRYPTO = { (char*) "crypto", 6 };
-static const pj_str_t STR_TELEPHONE_EVENT = { (char*) "telephone-event", 15};
-
+#include <cassert>
 
 Sdp::Sdp (pj_pool_t *pool)
     : memPool_(pool)
@@ -57,8 +42,6 @@ Sdp::Sdp (pj_pool_t *pool)
 	, remoteSession_(NULL)
     , activeLocalSession_(NULL)
     , activeRemoteSession_(NULL)
-    , localAudioMediaCap_()
-    , sessionAudioMedia_(0)
     , localIpAddr_("")
 	, remoteIpAddr_("")
     , localAudioPort_(0)
@@ -71,154 +54,104 @@ Sdp::Sdp (pj_pool_t *pool)
 
 void Sdp::setActiveLocalSdpSession (const pjmedia_sdp_session *sdp)
 {
-
-    int nb_media, nb_codecs;
-    int port;
-    pjmedia_sdp_media *current;
-    sdpMedia *media = NULL;
-    std::string dir;
-    pjmedia_sdp_attr *attribute = NULL;
-    pjmedia_sdp_rtpmap *rtpmap;
-
-    _debug ("SDP: Set active local SDP session");
-
     activeLocalSession_ = (pjmedia_sdp_session*) sdp;
 
-    // retrieve the media information
-    nb_media = activeLocalSession_->media_count;
-
-    for (int i = 0; i < nb_media ; i++) {
-        // Retrieve the media
-        current = activeLocalSession_->media[i];
-        std::string type (current->desc.media.ptr, current->desc.media.slen);
-        port = current->desc.port;
-        media = new sdpMedia (type, port);
-        // Retrieve the payload
-        nb_codecs = current->desc.fmt_count;  // Must be one
-
-        for (int j = 0; j < nb_codecs; j++) {
-            attribute = pjmedia_sdp_media_find_attr(current, &STR_RTPMAP, NULL);
-            // pj_strtoul(attribute->pt)
-
-            if (!attribute)
-            {
-                delete media;
-                return;
-            }
+    if (activeLocalSession_->media_count < 1)
+    	return;
 
-            pjmedia_sdp_attr_to_rtpmap (memPool_, attribute, &rtpmap);
+    pjmedia_sdp_media *current = activeLocalSession_->media[0];
 
-            sfl::Codec *codec = Manager::instance().audioCodecFactory.getCodec((int) pj_strtoul (&rtpmap->pt));
+	for (unsigned j = 0; j < current->desc.fmt_count; j++) {
+		static const pj_str_t STR_RTPMAP = { (char*) "rtpmap", 6 };
+	    pjmedia_sdp_attr *attribute = pjmedia_sdp_media_find_attr(current, &STR_RTPMAP, NULL);
+		if (!attribute) {
+			sessionAudioMedia_.clear();
+			return;
+		}
 
-            if (!codec)
-            {
-                delete media;
-                return;
-            }
+	    pjmedia_sdp_rtpmap *rtpmap;
+		pjmedia_sdp_attr_to_rtpmap (memPool_, attribute, &rtpmap);
 
-            media->add_codec(codec);
-        }
+		sfl::Codec *codec = Manager::instance().audioCodecFactory.getCodec((int) pj_strtoul (&rtpmap->pt));
+		if (!codec) {
+			sessionAudioMedia_.clear();
+			return;
+		}
 
-        sessionAudioMedia_.push_back (media);
-    }
+		sessionAudioMedia_.push_back(codec);
+	}
 }
 
+
+
 void Sdp::setActiveRemoteSdpSession (const pjmedia_sdp_session *sdp)
 {
-    _debug ("SDP: Set negotiated SDP");
-
     activeRemoteSession_ = (pjmedia_sdp_session*) sdp;
 
-    getRemoteSdpTelephoneEventFromOffer(sdp);
+    if(!sdp) {
+        _error("Sdp: Error: Remote sdp is NULL while parsing telephone event attribute");
+        return;
+    }
+
+    for (unsigned i = 0; i < sdp->media_count; i++)
+        if(pj_stricmp2(&sdp->media[i]->desc.media, "audio") == 0) {
+            pjmedia_sdp_media *r_media = sdp->media[i];
+            static const pj_str_t STR_TELEPHONE_EVENT = { (char*) "telephone-event", 15};
+            pjmedia_sdp_attr *attribute = pjmedia_sdp_attr_find(r_media->attr_count, r_media->attr, &STR_TELEPHONE_EVENT, NULL);
+            if (attribute != NULL) {
+                pjmedia_sdp_rtpmap *rtpmap;
+                pjmedia_sdp_attr_to_rtpmap (memPool_, attribute, &rtpmap);
+                telephoneEventPayload_ = pj_strtoul (&rtpmap->pt);
+            }
+            return;
+        }
+
+	_error("Sdp: Error: Could not found dtmf event from remote sdp");
 }
 
-bool Sdp::hasSessionMedia(void) const
+std::string Sdp::getCodecName(void) const
 {
-    return not sessionAudioMedia_.empty();
+	try {
+		sfl::AudioCodec *codec = getSessionMedia();
+		return codec ? codec->getMimeSubtype() : "";
+	} catch(...) {
+		return "";
+	}
 }
 
-sfl::AudioCodec* Sdp::getSessionMedia (void)
+sfl::AudioCodec* Sdp::getSessionMedia (void) const
 {
-
-    int nbMedia;
-    int nbCodec;
-    sfl::Codec *codec = NULL;
-    std::vector<sdpMedia *> mediaList;
-
-    _debug ("SDP: Get session media");
-
-    nbMedia = sessionAudioMedia_.size();
-
-    if (nbMedia <= 0) {
-        _error("SDP: Error: No media in session description");
-        throw SdpException("No media description for this SDP");
-    }
-
-    nbCodec = sessionAudioMedia_[0]->get_media_codec_list().size();
-
-    if (nbCodec <= 0) {
-        _error("SDP: Error: No codec description for this media");
+    if (sessionAudioMedia_.size() < 1)
         throw SdpException("No codec description for this media");
-    }
 
-    codec = sessionAudioMedia_[0]->get_media_codec_list()[0];
-
-    return static_cast<sfl::AudioCodec *>(codec);
+    return static_cast<sfl::AudioCodec *>(sessionAudioMedia_[0]);
 }
 
-namespace
-{
-std::string convertIntToString (int value)
-{
-    std::ostringstream result;
-    result << value;
-    return result.str();
-}
-} // end anonymous namespace
 
-
-void Sdp::setMediaDescriptorLine (sdpMedia *media, pjmedia_sdp_media** p_med)
+pjmedia_sdp_media *Sdp::setMediaDescriptorLine ()
 {
+    pjmedia_sdp_media *med = PJ_POOL_ZALLOC_T (memPool_, pjmedia_sdp_media);
 
-    pjmedia_sdp_media* med;
-    pjmedia_sdp_rtpmap rtpmap;
-    pjmedia_sdp_attr *attr;
-    sfl::Codec *codec;
-    int count, i;
-    std::string tmp;
-
-    med = PJ_POOL_ZALLOC_T (memPool_, pjmedia_sdp_media);
-
-    // Get the right media format
-    pj_strdup (memPool_, &med->desc.media,
-               (media->get_media_type() == MIME_TYPE_AUDIO) ? &STR_AUDIO : &STR_VIDEO);
+    med->desc.media = pj_str((char*)"audio");
     med->desc.port_count = 1;
-    med->desc.port = media->get_port();
-
+    med->desc.port = localAudioPort_;
     // in case of sdes, media are tagged as "RTP/SAVP", RTP/AVP elsewhere
-    if (srtpCrypto_.empty()) {
-        pj_strdup (memPool_, &med->desc.transport, &STR_RTP_AVP);
-    } else {
-
-        pj_strdup (memPool_, &med->desc.transport, &STR_RTP_SAVP);
-    }
+    med->desc.transport = pj_str(srtpCrypto_.empty() ? (char*)"RTP/AVP" : (char*)"RTP/SAVP");
 
-    // Media format ( RTP payload )
-    count = media->get_media_codec_list().size();
-    med->desc.fmt_count = count;
+    med->desc.fmt_count = codec_list_.size();
+    for (unsigned i=0; i<med->desc.fmt_count; i++) {
+        sfl::Codec *codec = codec_list_[i];
 
-    // add the payload list
+        std::ostringstream result;
+        result << (int)codec->getPayloadType ();
 
-    for (i=0; i<count; i++) {
-        codec = media->get_media_codec_list() [i];
-        tmp = convertIntToString (codec->getPayloadType ());
-        _debug ("%s", tmp.c_str());
-        pj_strdup2 (memPool_, &med->desc.fmt[i], tmp.c_str());
+        pj_strdup2(memPool_, &med->desc.fmt[i], result.str().c_str());
 
         // Add a rtpmap field for each codec
         // We could add one only for dynamic payloads because the codecs with static RTP payloads
         // are entirely defined in the RFC 3351, but if we want to add other attributes like an asymmetric
         // connection, the rtpmap attribute will be useful to specify for which codec it is applicable
+        pjmedia_sdp_rtpmap rtpmap;
         rtpmap.pt = med->desc.fmt[i];
         rtpmap.enc_name = pj_str ( (char*) codec->getMimeSubtype().c_str());
 
@@ -232,25 +165,20 @@ void Sdp::setMediaDescriptorLine (sdpMedia *media, pjmedia_sdp_media** p_med)
         rtpmap.param.ptr = ((char* const)"");
         rtpmap.param.slen = 0;
 
+        pjmedia_sdp_attr *attr;
         pjmedia_sdp_rtpmap_to_attr (memPool_, &rtpmap, &attr);
 
         med->attr[med->attr_count++] = attr;
     }
 
-    // Add the direction stream
-    attr = (pjmedia_sdp_attr*) pj_pool_zalloc (memPool_, sizeof (pjmedia_sdp_attr));
+    med->attr[ med->attr_count++] = pjmedia_sdp_attr_create(memPool_, "sendrecv", NULL);
 
-    pj_strdup2 (memPool_, &attr->name, media->get_stream_direction_str().c_str());
-
-    med->attr[ med->attr_count++] = attr;
-
-    if (!zrtpHelloHash_.empty()) {
+    if (!zrtpHelloHash_.empty())
         addZrtpAttribute (med, zrtpHelloHash_);
-    }
 
     setTelephoneEventRtpmap(med);
 
-    *p_med = med;
+    return med;
 }
 
 void Sdp::setTelephoneEventRtpmap(pjmedia_sdp_media *med)
@@ -273,299 +201,135 @@ void Sdp::setTelephoneEventRtpmap(pjmedia_sdp_media *med)
 
 void Sdp::setLocalMediaCapabilities (const CodecOrder &selectedCodecs)
 {
-    sdpMedia *audio;
-    // Clean it first
-    localAudioMediaCap_.clear();
-
-    _debug ("SDP: Fetch local media capabilities. Local extern audio port: %i" , getLocalPublishedAudioPort());
-
-    /* Only one audio media used right now */
-    audio = new sdpMedia (MIME_TYPE_AUDIO);
-    audio->set_port (getLocalPublishedAudioPort());
-
     if (selectedCodecs.size() == 0)
         _warn("No selected codec while building local SDP offer");
-    else {
-        for (CodecOrder::const_iterator iter = selectedCodecs.begin(); iter != selectedCodecs.end(); ++iter) {
-        	sfl::Codec *codec = Manager::instance().audioCodecFactory.getCodec(*iter);
-            if (codec)
-                audio->add_codec(codec);
-            else
-                _warn ("SDP: Couldn't find audio codec");
-        }
-    }
 
-    localAudioMediaCap_.push_back (audio);
+    codec_list_.clear();
+	for (CodecOrder::const_iterator iter = selectedCodecs.begin(); iter != selectedCodecs.end(); ++iter) {
+		sfl::Codec *codec = Manager::instance().audioCodecFactory.getCodec(*iter);
+		if (codec)
+			codec_list_.push_back(codec);
+		else
+			_warn ("SDP: Couldn't find audio codec");
+	}
 }
 
 int Sdp::createLocalSession (const CodecOrder &selectedCodecs)
 {
-    char buffer[1000];
+    setLocalMediaCapabilities (selectedCodecs);
 
-    _info ("SDP: Create local session");
+    localSession_ = PJ_POOL_ZALLOC_T(memPool_, pjmedia_sdp_session);
+    localSession_->conn = PJ_POOL_ZALLOC_T(memPool_, pjmedia_sdp_conn);
 
-    // Build local media capabilities
-    setLocalMediaCapabilities (selectedCodecs);
+    /* Initialize the fields of the struct */
+    localSession_->origin.version = 0;
+    pj_time_val tv;
+    pj_gettimeofday (&tv);
 
-    // Reference: RFC 4566 [5]
+    localSession_->origin.user = pj_str (pj_gethostname()->ptr);
+    // Use Network Time Protocol format timestamp to ensure uniqueness.
+    localSession_->origin.id = tv.sec + 2208988800UL;
+    localSession_->origin.net_type = pj_str((char*)"IN");
+    localSession_->origin.addr_type = pj_str((char*)"IP4");
+    localSession_->origin.addr = pj_str((char*)localIpAddr_.c_str());
 
-    /* Create and initialize basic SDP session */
-    localSession_ = PJ_POOL_ZALLOC_T (memPool_, pjmedia_sdp_session);
-    localSession_->conn = PJ_POOL_ZALLOC_T (memPool_, pjmedia_sdp_conn);
+    localSession_->name = pj_str((char*)"sflphone");
 
-    /* Initialize the fields of the struct */
-    addProtocol();
-    addOrigin();
-    addSessionName();
-    addConnectionInfo();
-    addTiming();
-    addAudioMediaDescription();
-
-    if (!srtpCrypto_.empty()) {
+    localSession_->conn->net_type = localSession_->origin.net_type;
+    localSession_->conn->addr_type = localSession_->origin.addr_type;
+    localSession_->conn->addr = localSession_->origin.addr;
+
+    // RFC 3264: An offer/answer model session description protocol
+    // As the session is created and destroyed through an external signaling mean (SIP), the line
+    // should have a value of "0 0".
+    localSession_->time.start = 0;
+    localSession_->time.stop = 0;
+
+    // For DTMF RTP events
+    localSession_->media_count = 1;
+    localSession_->media[0] = setMediaDescriptorLine();
+
+    if (!srtpCrypto_.empty())
         addSdesAttribute (srtpCrypto_);
-    }
 
-    memset(buffer, 0, 1000);
-    pjmedia_sdp_print(localSession_, buffer, 1000);
+    char buffer[1000];
+    pjmedia_sdp_print(localSession_, buffer, sizeof(buffer));
     _debug("SDP: Local SDP Session:\n%s", buffer);
 
-    // Validate the sdp session
     return pjmedia_sdp_validate (localSession_);
-
 }
 
-int Sdp::createOffer (const CodecOrder &selectedCodecs)
+void Sdp::createOffer (const CodecOrder &selectedCodecs)
 {
-    pj_status_t status;
-
-    // Build the SDP session descriptor
-    status = createLocalSession (selectedCodecs);
-    if (status != PJ_SUCCESS) {
+    if (createLocalSession (selectedCodecs) != PJ_SUCCESS)
         _error ("SDP: Error: Failed to create initial offer");
-        return status;
-    }
-