* #6254: update destination IP and port for video

sflphone-common/src/call.h

@@ -29,8 +29,8 @@
  *  shall include the source code for the parts of OpenSSL used as well
  *  as that of the covered work.
  */
-#ifndef CALL_H
+#ifndef __CALL_H__
-#define CALL_H
+#define __CALL_H__
 
 #include <cc++/thread.h> // for mutex
 #include <sstream>
@@ -306,4 +306,4 @@ class Call: public Recordable
         std::string _filename;
 };
 
-#endif
+#endif // __CALL_H__

sflphone-common/src/sip/sdp.cpp

@@ -569,6 +569,21 @@ void Sdp::addAudioMediaDescription()
     }
 }
 
+
+void Sdp::addVideoMediaDescription()
+{
+#if 0
+    pjmedia_sdp_media* med;
+    int nb_media, i;
+
+    med = PJ_POOL_ZALLOC_T (memPool, pjmedia_sdp_media);
+    for (i=0; i<nb_media; i++) {
+        setMediaDescriptorLine (getLocalMediaCap() [i], &med);
+        localSession->media[i] = med;
+    }
+#endif
+}
+
 void Sdp::addSdesAttribute (const std::vector<std::string>& crypto)
 {
     // temporary buffer used to store crypto attribute

sflphone-common/src/sip/sdp.h

@@ -223,6 +223,13 @@ class Sdp
             localAudioPort_ = port;
         }
 
+        /**
+         * @param Set the published video port
+         */
+        void  setLocalPublishedVideoPort (int port) {
+            localVideoPort_ = port;
+        }
+
         /**
          * @return The published audio port
          */
@@ -230,6 +237,13 @@ class Sdp
             return localAudioPort_;
         }
 
+        /**
+         * @return The published video port
+         */
+        int  getLocalPublishedVideoPort (void) const {
+            return localVideoPort_;
+        }
+
         /**
          * Set remote's IP addr. [not protected]
          * @param ip  The remote IP address
@@ -254,6 +268,14 @@ class Sdp
             remoteAudioPort_ = port;
         }
 
+        /**
+         * Set remote's video port. [not protected]
+         * @param port  The remote video port
+         */
+        void setRemoteVideoPort (unsigned int port) {
+            remoteVideoPort_ = port;
+        }
+
         /**
          * Return audio port at destination [mutex protected]
          * @return unsigned int The remote audio port
@@ -263,6 +285,14 @@ class Sdp
         }
 
 
+        /**
+         * Return video port at destination [mutex protected]
+         * @return unsigned int The remote video port
+         */
+        unsigned int getRemoteVideoPort() const {
+            return remoteVideoPort_;
+        }
+
         /**
          * Get media list for this session
          */
@@ -368,11 +398,21 @@ class Sdp
          */
         int localAudioPort_;
 
+        /**
+         * Local video port
+         */
+        int localVideoPort_;
+
         /**
          * Remote audio port
          */
         unsigned int remoteAudioPort_;
 
+        /**
+         * Remote video port
+         */
+        unsigned int remoteVideoPort_;
+
         /**
          * Zrtp hello hash
          */
@@ -490,6 +530,11 @@ class Sdp
          */
         void addAudioMediaDescription();
 
+        /*
+         * Mandatory field: Media descriptions ("m=")
+         */
+        void addVideoMediaDescription();
+
         /*
          * Adds a sdes attribute to the given media section.
          *

sflphone-common/src/sip/sipcall.h

@@ -29,8 +29,8 @@
  *  shall include the source code for the parts of OpenSSL used as well
  *  as that of the covered work.
  */
-#ifndef SIPCALL_H
+#ifndef __SIPCALL_H__
-#define SIPCALL_H
+#define __SIPCALL_H__
 
 #include "call.h"
 
@@ -196,4 +196,4 @@ class SIPCall : public Call
 
 };
 
-#endif
+#endif // __SIPCALL_H__

sflphone-common/src/sip/sipvoiplink.cpp

@@ -724,6 +724,7 @@ Call *SIPVoIPLink::newOutgoingCall (const CallID& id, const std::string& toUrl)
 		_info ("UserAgent: Start audio rtp session");
 		call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
 		_info ("UserAgent: Start video rtp session");
+        call->getVideoRtp()->updateDestination(call->getLocalIp(), call->getLocalVideoPort());
 		call->getVideoRtp()->start();
 	} catch (...) {
 		throw VoipLinkException ("Could not start rtp session for early media");
@@ -1030,6 +1031,7 @@ SIPVoIPLink::offhold (const CallID& id) throw (VoipLinkException)
         call->getAudioRtp()->initAudioRtpConfig (call);
         call->getAudioRtp()->initAudioSymmetricRtpSession (call);
         call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
+        call->getVideoRtp()->updateDestination(call->getLocalIp(), call->getLocalVideoPort());
         call->getVideoRtp()->start();
 
     }
@@ -1811,6 +1813,7 @@ bool SIPVoIPLink::SIPNewIpToIpCall (const CallID& id, const std::string& to)
             call->getAudioRtp()->initAudioSymmetricRtpSession (call);
             call->getAudioRtp()->initLocalCryptoInfo (call);
             call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
+            call->getVideoRtp()->updateDestination(call->getLocalIp(), call->getLocalVideoPort());
             call->getVideoRtp()->start ();
         } catch (...) {
             _debug ("UserAgent: Unable to create RTP Session in new IP2IP call (%s:%d)", __FILE__, __LINE__);
@@ -3463,6 +3466,7 @@ void sdp_media_update_cb (pjsip_inv_session *inv, pj_status_t status)
 
     try {
         call->getAudioRtp()->updateDestinationIpAddress();
+        call->getVideoRtp()->updateDestination(call->getLocalIp(), call->getLocalVideoPort());
         call->getAudioRtp()->setDtmfPayloadType(sdpSession->getTelephoneEventType());
     } catch (...) {
 
@@ -4068,6 +4072,8 @@ transaction_request_cb (pjsip_rx_data *rdata)
     try {
         _debug ("UserAgent: Create RTP session for this call");
         call->getAudioRtp()->start (static_cast<sfl::AudioCodec *>(audiocodec));
+        call->getVideoRtp()->updateDestination(call->getLocalIp(), call->getLocalVideoPort());
+        call->getVideoRtp()->start();
     } catch (...) {
         _warn ("UserAgent: Error: Failed to create rtp thread from answer");
     }
@@ -4706,8 +4712,10 @@ bool setCallMediaLocal (SIPCall* call, const std::string &localIP)
 
         account = dynamic_cast<SIPAccount *> (Manager::instance().getAccount (account_id));
 
-        // Setting Audio
+        // Setting Audio and Video
         unsigned int callLocalAudioPort = RANDOM_LOCAL_PORT;
+        unsigned int callLocalVideoPort = RANDOM_LOCAL_PORT;
+        assert(callLocalAudioPort != callLocalVideoPort);
         unsigned int callLocalExternAudioPort = callLocalAudioPort;
 
         if (account->isStunEnabled ()) {
@@ -4718,11 +4726,13 @@ bool setCallMediaLocal (SIPCall* call, const std::string &localIP)
 
         _debug ("UserAgent: Setting local ip address: %s", localIP.c_str());
         _debug ("UserAgent: Setting local audio port to: %d", callLocalAudioPort);
+        _debug ("UserAgent: Setting local video port to: %d", callLocalVideoPort);
         _debug ("UserAgent: Setting local audio port (external) to: %d", callLocalExternAudioPort);
 
         // Set local audio port for SIPCall(id)
         call->setLocalIp (localIP);
         call->setLocalAudioPort (callLocalAudioPort);
+        call->setLocalVideoPort (callLocalVideoPort);
 
         call->getLocalSDP()->setPortToAllMedia (callLocalExternAudioPort);
 

sflphone-common/src/video/video_rtp_factory.cpp

@@ -61,4 +61,11 @@ void VideoRtpFactory::stop()
     session_->stop();
 }
 
+void VideoRtpFactory::updateDestination(const std::string &dest,
+                                        unsigned int port)
+{
+    session_->updateDestination(dest, port);
+}
+
+
 } // end namespace sfl

sflphone-common/src/video/video_rtp_factory.h

@@ -39,6 +39,8 @@ class VideoRtpSession;
 class VideoRtpFactory {
     public:
         VideoRtpFactory();
+        void updateDestination(const std::string &dest,
+                               unsigned int port);
 
         /**
          * Start the video RTP thread of the type specified in the configuration

sflphone-common/src/video/video_rtp_session.cpp

@@ -44,6 +44,15 @@ VideoRtpSession::VideoRtpSession(std::map<std::string,std::string> args) :
 {
 }
 
+void VideoRtpSession::updateDestination(const std::string &destination,
+        unsigned int port)
+{
+    std::stringstream tmp;
+    tmp << "rtp://" << destination << ":" << port;
+    args_["destination"] = tmp.str();
+    std::cerr << "updated dest to " << args_["destination"] << std::endl;
+}
+
 void VideoRtpSession::test()
 {
     assert(sendThread_.get() == 0);

sflphone-common/src/video/video_rtp_session.h

@@ -48,6 +48,8 @@ class VideoRtpSession {
         void test();
         void test_loopback();
         void stop();
+        void updateDestination(const std::string &destination,
+                               unsigned int port);
     private:
         std::tr1::shared_ptr<VideoSendThread> sendThread_;
         std::tr1::shared_ptr<VideoReceiveThread> receiveThread_;

sflphone-common/src/video/video_send_thread.cpp

@@ -65,12 +65,15 @@ void VideoSendThread::print_and_save_sdp()
     std::ofstream sdp_file("test.sdp");
     std::istringstream iss(sdp);
     std::string line;
+    sdp_ = "";
     while (std::getline(iss, line))
     {
         /* strip windows line ending */
-        sdp_file << line.substr(0, line.length() - 1) << std::endl;
+        line = line.substr(0, line.length() - 1);
-        std::cerr << line << std::endl;
+        sdp_file << line << std::endl;
+        sdp_ += line + "\n";
     }
+    std::cerr << sdp_ << std::endl;
     sdp_file << std::endl;
     sdp_file.close();
     free(sdp);
@@ -349,7 +352,8 @@ VideoSendThread::VideoSendThread(const std::map<std::string, std::string> &args)
     encoderCtx_(0),
     videoStream_(0),
     inputCtx_(0),
-    outputCtx_(0)
+    outputCtx_(0),
+    sdp_("")
 {}
 
 void VideoSendThread::run()

sflphone-common/src/video/video_send_thread.h

@@ -70,6 +70,7 @@ class VideoSendThread : public ost::Thread {
         AVStream *videoStream_;
         AVFormatContext *inputCtx_;
         AVFormatContext *outputCtx_;
+        std::string sdp_;
     public:
         explicit VideoSendThread(const std::map<std::string, std::string> &args);
         virtual ~VideoSendThread();

sflphone-common/test/Makefile.am

@@ -42,15 +42,15 @@ test_SOURCES = \
 	gaincontroltest.cpp
 
 LLIBS=$(CPPUNIT_LIBS) \
-	../src/sflphoned-logger.o \
+	../src/logger.o \
-	../src/sflphoned-managerimpl.o \
+	../src/managerimpl.o \
-	../src/sflphoned-account.o\
+	../src/account.o\
-	../src/sflphoned-accountcreator.o \
+	../src/accountcreator.o \
-	../src/sflphoned-call.o \
+	../src/call.o \
-	../src/sflphoned-conference.o \
+	../src/conference.o \
-	../src/sflphoned-eventthread.o \
+	../src/eventthread.o \
-	../src/sflphoned-managerimpl_registration.o \
+	../src/managerimpl_registration.o \
-	../src/sflphoned-numbercleaner.o \
+	../src/numbercleaner.o \
-	../src/sflphoned-voiplink.o \
+	../src/voiplink.o \
-	../src/sflphoned-preferences.o \
+	../src/preferences.o \
 	../src/libsflphone.la