adds multihomed support for RTP streams

core/AmConfig.cpp

@@ -101,6 +101,38 @@ bool AmConfig::IgnoreSIGPIPE      = true;
 
 static int readInterfaces(AmConfigReader& cfg);
 
+AmConfig::IP_interface::IP_interface()
+  : LocalSIPIP(),
+    LocalSIPPort(5060),
+    LocalIP(),
+    PublicIP(),
+    RtpLowPort(RTP_LOWPORT),
+    RtpHighPort(RTP_HIGHPORT)
+{
+}
+
+int AmConfig::IP_interface::getNextRtpPort()
+{
+    
+  int port=0;
+
+  next_rtp_port_mut.lock();
+  if(next_rtp_port < 0){
+    next_rtp_port = RtpLowPort;
+  }
+    
+  port = next_rtp_port & 0xfffe;
+  next_rtp_port += 2;
+
+  if(next_rtp_port >= RtpHighPort){
+    next_rtp_port = RtpLowPort;
+  }
+  next_rtp_port_mut.unlock();
+    
+  return port;
+}
+
+
 int AmConfig::setLogLevel(const string& level, bool apply)
 {
   int n;

core/AmConfig.h

@@ -98,6 +98,14 @@ struct AmConfig
     string LocalSIPIP;
     /** the port SIP requests are sent from - optional (default 5060) */
     int LocalSIPPort;
+
+    IP_interface();
+
+    int getNextRtpPort();
+
+  private:
+    int next_rtp_port;
+    AmMutex next_rtp_port_mut;
   };
 
   static vector<IP_interface>            Ifs;

core/AmRtpAudio.cpp

@@ -31,8 +31,8 @@
 #include "AmSession.h"
 #include "AmPlayoutBuffer.h"
 
-AmRtpAudio::AmRtpAudio(AmSession* _s)
-  : AmRtpStream(_s), AmAudio(0), 
+AmRtpAudio::AmRtpAudio(AmSession* _s, int _if)
+  : AmRtpStream(_s,_if), AmAudio(0), 
     /*last_ts_i(false),*/ use_default_plc(true),
     send_only(false), playout_buffer(new AmPlayoutBuffer(this)),
     last_check(0),last_check_i(false),send_int(false)

core/AmRtpAudio.h

@@ -93,7 +93,7 @@ class AmRtpAudio: public AmRtpStream, public AmAudio, public AmPLCBuffer
 			   unsigned int   rate);
 
 public:
-  AmRtpAudio(AmSession* _s=0);
+  AmRtpAudio(AmSession* _s, int _if);
   ~AmRtpAudio();
 
   bool checkInterval(unsigned int ts, unsigned int frame_size);

core/AmRtpStream.cpp

@@ -75,35 +75,14 @@ void AmRtpStream::setLocalIP(const string& ip)
 #endif
 }
 
-int AmRtpStream::next_port = -1;
-AmMutex AmRtpStream::port_mut;
-
-int AmRtpStream::getNextPort()
-{
-    
-  int port=0;
-
-  port_mut.lock();
-  if(next_port < 0){
-    next_port = AmConfig::RtpLowPort();
-  }
-    
-  port = next_port & 0xfffe;
-  next_port += 2;
-
-  if(next_port >= AmConfig::RtpHighPort()){
-    next_port = AmConfig::RtpLowPort();
-  }
-  port_mut.unlock();
-    
-  return port;
-}
-
 void AmRtpStream::setLocalPort()
 {
   if(l_port)
     return;
   
+  assert(l_if >= 0);
+  assert(l_if < (int)AmConfig::Ifs.size());
+
   int sd=0;
 #ifdef SUPPORT_IPV6
   if((sd = socket(l_saddr.ss_family,SOCK_DGRAM,0)) == -1)
@@ -125,7 +104,7 @@ void AmRtpStream::setLocalPort()
   unsigned short port = 0;
   for(;retry; --retry){
 
-    port = getNextPort();
+    port = AmConfig::Ifs[l_if].getNextRtpPort();
 #ifdef SUPPORT_IPV6
     set_port_v6(&l_saddr,port);
     if(!bind(sd,(const struct sockaddr*)&l_saddr,
@@ -424,8 +403,9 @@ int AmRtpStream::receive( unsigned char* buffer, unsigned int size,
   return res;
 }
 
-AmRtpStream::AmRtpStream(AmSession* _s) 
+AmRtpStream::AmRtpStream(AmSession* _s, int _if) 
   : r_port(0),
+    l_if(_if),
     l_port(0),
     l_sd(0), 
     r_ssrc_i(false),

core/AmRtpStream.h

@@ -87,11 +87,6 @@ struct PacketMem {
 class AmRtpStream 
 {
 protected:
-  static int next_port;
-  static AmMutex port_mut;
-
-  static int getNextPort();
-
   /**
      Remote payload (only different from 
      int_payload if using dynamic payloads)
@@ -108,6 +103,10 @@ protected:
 
   string             r_host;
   unsigned short     r_port;
+
+  /* local interface */
+  int l_if;
+
 #ifdef SUPPORT_IPV6
   struct sockaddr_storage r_saddr;
   struct sockaddr_storage l_saddr;
@@ -191,7 +190,7 @@ public:
   int ping();
 
   /** Allocates resources for future use of RTP. */
-  AmRtpStream(AmSession* _s=0);
+  AmRtpStream(AmSession* _s, int _if);
   /** Stops the stream and frees all resources. */
   virtual ~AmRtpStream();
 

core/AmSession.h

@@ -620,7 +620,7 @@ inline AmRtpAudio* AmSession::RTPStream() {
   if (NULL == _rtp_str.get()) {
     DBG("creating RTP stream instance for session [%p]\n", 
 	this);
-    _rtp_str.reset(new AmRtpAudio(this));
+    _rtp_str.reset(new AmRtpAudio(this,rtp_interface));
   }
   return _rtp_str.get();
 }