Include all possible payload types into 200 OK and actually accept any of them

in the RTP stream. It is possible to disable this new behaviour using
single_codec_in_ok parameter in the sems.conf.

Developed by:	Sippy Software, Inc.
Sponsored by:	Digifonica Canada Limited


git-svn-id: http://svn.berlios.de/svnroot/repos/sems/trunk@279 8eb893ce-cfd4-0310-b710-fb5ebe64c474

core/AmAudio.cpp

@@ -28,6 +28,7 @@
 #include "AmAudio.h"
 #include "AmPlugIn.h"
 #include "AmUtils.h"
+#include "AmSdp.h"
 #include "amci/codecs.h"
 #include "log.h"
 
@@ -38,19 +39,89 @@
 
 #include <typeinfo>
 
-AmAudioRtpFormat::AmAudioRtpFormat(int payload, string format_parameters)
-  : AmAudioFormat(), payload(payload), amci_pl(0)
+struct CodecContainer
 {
-  sdp_format_parameters = format_parameters;
-  codec = getCodec();
- 
-  amci_payload_t* pl = getPayloadP();
-  if(pl && codec){
-    channels = pl->channels;
-    rate = pl->sample_rate;
-  } else {
-    ERROR("Could not find payload <%i>\n",payload);
-  } 
+  amci_codec_t *codec;
+  int frame_size;
+  int frame_length;
+  int frame_encoded_size;
+  long h_codec;
+};
+
+AmAudioRtpFormat::AmAudioRtpFormat(const vector<SdpPayload *>& payloads)
+  : AmAudioFormat(), m_payloads(payloads), m_currentPayload(-1)
+{
+  for (vector<SdpPayload *>::iterator it = m_payloads.begin();
+	  it != m_payloads.end(); ++it)
+  {
+    m_sdpPayloadByPayload[(*it)->payload_type] = *it;
+  }
+  setCurrentPayload(m_payloads[0]->payload_type);
+}
+
+void AmAudioRtpFormat::setCurrentPayload(int payload)
+{
+  if (m_currentPayload != payload)
+  {
+    map<int, SdpPayload *>::iterator p = m_sdpPayloadByPayload.find(payload);
+    if (p == m_sdpPayloadByPayload.end())
+    {
+      ERROR("Could not find payload <%i>\n", payload);
+      return;
+    }
+    map<int, amci_payload_t *>::iterator pp = m_payloadPByPayload.find(payload);
+    if (pp == m_payloadPByPayload.end())
+    {
+      m_currentPayloadP = AmPlugIn::instance()->payload(p->second->int_pt);
+      if (m_currentPayloadP == NULL)
+      {
+	ERROR("Could not find payload <%i>\n", payload);
+	return;
+      }
+      m_payloadPByPayload[payload] = m_currentPayloadP;
+    }
+    else
+      m_currentPayloadP = pp->second;
+    m_currentPayload = payload;
+    sdp_format_parameters = p->second->sdp_format_parameters;
+
+    map<int, CodecContainer *>::iterator c = m_codecContainerByPayload.find(payload);
+    if (c == m_codecContainerByPayload.end())
+    {
+      codec = NULL;
+      getCodec();
+      if (codec)
+      {
+	CodecContainer *cc = new CodecContainer();
+	cc->codec = codec;
+	cc->frame_size = frame_size;
+	cc->frame_length = frame_length;
+	cc->frame_encoded_size = frame_encoded_size;
+	cc->h_codec = h_codec;
+	m_codecContainerByPayload[payload] = cc;
+      }
+    }
+    else
+    {
+      codec = c->second->codec;
+      frame_size = c->second->frame_size;
+      frame_length = c->second->frame_length;
+      frame_encoded_size = c->second->frame_encoded_size;
+      h_codec = c->second->h_codec;
+    }
+    if (m_currentPayloadP && codec) {
+      channels = m_currentPayloadP->channels;
+      rate = m_currentPayloadP->sample_rate;
+    } else {
+      ERROR("Could not find payload <%i>\n", payload);
+    }
+  }
+}
+
+AmAudioRtpFormat::~AmAudioRtpFormat()
+{
+  for (map<int, CodecContainer *>::iterator it = m_codecContainerByPayload.begin(); it != m_codecContainerByPayload.end(); ++it)
+    delete it->second;
 }
 
 AmAudioFormat::AmAudioFormat()
@@ -677,22 +748,12 @@ int AmAudioFileFormat::getCodecId()
 }
 
 
-amci_payload_t* AmAudioRtpFormat::getPayloadP()
-{
-  if(!amci_pl)
-    amci_pl = AmPlugIn::instance()->payload(payload);
-
-  return amci_pl;
-}
-
 int AmAudioRtpFormat::getCodecId()
 {
-  amci_payload_t* pl = getPayloadP();
-  if(!pl){
-    ERROR("AmAudioRtpFormat::getCodecId: could not find payload %i\n",payload);
+  if(!m_currentPayloadP){
+    ERROR("AmAudioRtpFormat::getCodecId: could not find payload %i\n", m_currentPayload);
     return -1;
   }
   else 
-    return pl->codec_id;
+    return m_currentPayloadP->codec_id;
 }
-

core/AmAudio.h

@@ -38,10 +38,15 @@
 using std::auto_ptr;
 #include <string>
 using std::string;
+#include <map>
+using std::map;
 
 #define PCM16_B2S(b) ((b) >> 1)
 #define PCM16_S2B(s) ((s) << 1)
 
+class SdpPayload;
+class CodecContainer;
+
 /** \brief Audio Event */
 class AmAudioEvent: public AmEvent
 {
@@ -206,14 +211,15 @@ public:
 /** \brief RTP audio format */
 class AmAudioRtpFormat: public AmAudioFormat
 {
-  /** ==-1 if not yet initialized. */
-  int             payload;
-  /** == 0 if not yet initialized. */
-  amci_payload_t* amci_pl;
+  vector<SdpPayload *> m_payloads;
+  int m_currentPayload;
+  amci_payload_t *m_currentPayloadP;
+  map<int, SdpPayload *> m_sdpPayloadByPayload;
+  map<int, amci_payload_t *> m_payloadPByPayload;
+  map<int, CodecContainer *> m_codecContainerByPayload;
 
 protected:
   int getCodecId();
-  amci_payload_t* getPayloadP();
 
 public:
   /**
@@ -221,10 +227,10 @@ public:
    * All the information are taken from the 
    * payload description in the originating plug-in.
    */
-  AmAudioRtpFormat(int payload, string format_parameters);
+  AmAudioRtpFormat(const vector<SdpPayload *>& payloads);
+  ~AmAudioRtpFormat();
 
-  /** @return Payload ID. */
-  int           getPayload() { return payload; }
+  void setCurrentPayload(int payload);
 };
 
 /**

core/AmConfig.cpp

@@ -55,6 +55,7 @@ int          AmConfig::MediaProcessorThreads   = NUM_MEDIA_PROCESSORS;
 int          AmConfig::LocalSIPPort            = 5060;
 string       AmConfig::LocalSIPIP              = "";
 string       AmConfig::Signature               = "";
+bool	     AmConfig::SingleCodecInOK	       = false;
 
 AmSessionTimerConfig AmConfig::defaultSessionTimerConfig;
 
@@ -237,6 +238,10 @@ int AmConfig::readConfiguration()
       return -1;
     }
   }
+  // single codec in 200 OK
+  if(cfg.hasParameter("single_codec_in_ok")){
+    SingleCodecInOK = (cfg.getParameter("single_codec_in_ok") == "yes");
+  }
 
   return defaultSessionTimerConfig.readFromConfig(cfg);
 }	

core/AmConfig.h

@@ -85,6 +85,8 @@ struct AmConfig
   static int LocalSIPPort;
   /** Server/User-Agent header (optional) */
   static string Signature;
+  /** If 200 OK reply should be limited to preferred codec only */
+  static bool SingleCodecInOK;
 
   /** Init function. Resolves SMTP server address. */
   static int init();

core/AmRtpAudio.cpp

@@ -77,10 +77,14 @@ int AmRtpAudio::receive(unsigned int audio_buffer_ts)
   unsigned int rtp_ts;
 
   while(true) {
+    int payload;
     size = AmRtpStream::receive((unsigned char*)samples,
-				(unsigned int)AUDIO_BUFFER_SIZE, rtp_ts);
+				(unsigned int)AUDIO_BUFFER_SIZE, rtp_ts,
+				payload);
     if(size <= 0)
       break;
+
+    setCurrentPayload(payload);
 	
     if(send_only){
       playout_buffer->clearLastTs();
@@ -122,12 +126,16 @@ int AmRtpAudio::write(unsigned int user_ts, unsigned int size)
   return send(user_ts,(unsigned char*)samples,size);
 }
 
-void AmRtpAudio::init(const SdpPayload* sdp_payload)
+void AmRtpAudio::init(const vector<SdpPayload*>& sdp_payloads)
 {
   DBG("AmRtpAudio::init(...)\n");
-  AmRtpStream::init(sdp_payload);
-  fmt.reset(new AmAudioRtpFormat(int_payload, format_parameters));
+  AmRtpStream::init(sdp_payloads);
+  fmt.reset(new AmAudioRtpFormat(sdp_payloads));
+}
 
+void AmRtpAudio::setCurrentPayload(int payload)
+{
+  ((AmAudioRtpFormat *) fmt.get())->setCurrentPayload(payload);
   amci_codec_t* codec = fmt->getCodec();
   use_default_plc = !(codec && codec->plc);
 }

core/AmRtpAudio.h

@@ -70,6 +70,8 @@ public:
   bool checkInterval(unsigned int ts);
   bool sendIntReached();
 
+  void setCurrentPayload(int payload);
+
   int receive(unsigned int audio_buffer_ts);
 
   void setSendOnly(bool so){
@@ -84,7 +86,7 @@ public:
 	  unsigned int nb_samples);
 
   // AmRtpStream interface
-  void init(const SdpPayload* sdp_payload);
+  void init(const vector<SdpPayload*>& sdp_payloads);
 
   void setPlayoutType(PlayoutType type);
 

core/AmRtpStream.cpp

@@ -193,7 +193,7 @@ int AmRtpStream::send( unsigned int ts, unsigned char* buffer, unsigned int size
 // @param audio_buffer_ts [in]  current ts at the audio_buffer 
 
 int AmRtpStream::receive( unsigned char* buffer, unsigned int size,
-			  unsigned int& ts)
+			  unsigned int& ts, int &out_payload)
 {
   AmRtpPacket rp;
   int err = nextPacket(rp);
@@ -247,9 +247,11 @@ int AmRtpStream::receive( unsigned char* buffer, unsigned int size,
       return RTP_DTMF;
     }
 
+/*
   if (payload != rp.payload){
     return RTP_UNKNOWN_PL;
   }
+*/
 
   assert(rp.getData());
   if(rp.getDataSize() > size){
@@ -258,6 +260,7 @@ int AmRtpStream::receive( unsigned char* buffer, unsigned int size,
   }
   memcpy(buffer,rp.getData(),rp.getDataSize());
   ts = rp.timestamp;
+  out_payload = rp.payload;
 
   return rp.getDataSize();
 }
@@ -361,15 +364,12 @@ void AmRtpStream::setRAddr(const string& addr, unsigned short port)
 #endif
 }
 
-void AmRtpStream::init(const SdpPayload* sdp_payload)
+void AmRtpStream::init(const vector<SdpPayload*>& sdp_payloads)
 {
+  SdpPayload *sdp_payload = sdp_payloads[0];
   payload = sdp_payload->payload_type;
   int_payload = sdp_payload->int_pt;
   last_payload = payload;
-  if(!sdp_payload->sdp_format_parameters.empty())
-    format_parameters = sdp_payload->sdp_format_parameters;
-  else
-    format_parameters = "";
   resume();
 }
 

core/AmRtpStream.h

@@ -98,8 +98,6 @@ protected:
   */
   int         last_payload;
 
-  string      format_parameters;
-
   string             r_host;
   unsigned short     r_port;
 #ifdef SUPPORT_IPV6
@@ -153,7 +151,7 @@ public:
 
 
   int receive( unsigned char* buffer, unsigned int size,
-	       unsigned int& ts);
+	       unsigned int& ts, int& payload);
     
   /** Allocates resources for future use of RTP. */
   AmRtpStream(AmSession* _s=0);
@@ -211,7 +209,7 @@ public:
    * @warning It should be called only if the stream has been completly initialized,
    * @warning and only once per session. Use resume() then.
    */
-  virtual void init(const SdpPayload* sdp_payload);
+  virtual void init(const vector<SdpPayload*>& sdp_payloads);
 
   /**
    * Stops RTP stream.

core/AmSdp.cpp

@@ -185,7 +185,7 @@ int AmSdp::parse()
 
 
 int AmSdp::genResponse(const string& localip, int localport, 
-		       string& out_buf)
+		       string& out_buf, bool single_codec)
 {
   string l_ip = "IP4 " + localip;
 
@@ -218,6 +218,7 @@ int AmSdp::genResponse(const string& localip, int localport,
       options += "a=fmtp:" + int2str((*it)->payload_type) + " " 
 	+ (*it)->sdp_format_parameters + "\r\n";
     }
+    if (single_codec) break;
   }
 
   if (hasTelephoneEvent())
@@ -286,10 +287,11 @@ int AmSdp::genRequest(const string& localip,int localport, string& out_buf)
   return 0;
 }
 
-SdpPayload* AmSdp::getCompatiblePayload(int media_type, string& addr, int& port)
+const vector<SdpPayload *>& AmSdp::getCompatiblePayloads(int media_type, string& addr, int& port)
 {
   vector<SdpMedia>::iterator   m_it;
-  SdpPayload*                  payload=0;
+  SdpPayload *payload;
+  sup_pl.clear();
 
   AmPlugIn* pi = AmPlugIn::instance();
 
@@ -313,7 +315,7 @@ SdpPayload* AmSdp::getCompatiblePayload(int media_type, string& addr, int& port)
 	payload->int_pt = a_pl->payload_id;
 	payload->encoding_name = a_pl->name;
 	payload->clock_rate = a_pl->sample_rate;
-	goto end;
+	sup_pl.push_back(payload);
       }
       else { 
 	// Try dynamic payloads
@@ -331,37 +333,30 @@ SdpPayload* AmSdp::getCompatiblePayload(int media_type, string& addr, int& port)
 		    
 	  payload = &(*it);
 	  payload->int_pt = int_pt;
-	  goto end;
+	  sup_pl.push_back(payload);
 	}
       }
     }
-  }
-
- end:
-  sup_pl.clear();
-
-  if(payload){
-	
-    DBG("payload found: %i/%s/%i\n",
-	payload->int_pt,payload->encoding_name.c_str(),payload->clock_rate);
-
-    if(m_it->conn.address.empty()){
-      DBG("using global address: %s\n",conn.address.c_str());
-      addr = conn.address;
-    }
-    else {
-      DBG("using media specific address: %s\n",m_it->conn.address.c_str());
-      addr = m_it->conn.address;
+    if (sup_pl.size() > 0)
+    {
+      if (m_it->conn.address.empty())
+      {
+	DBG("using global address: %s\n",conn.address.c_str());
+	addr = conn.address;
+      }
+      else {
+	DBG("using media specific address: %s\n",m_it->conn.address.c_str());
+	addr = m_it->conn.address;
+      }
+      
+      if(m_it->dir == SdpMedia::DirActive)
+	remote_active = true;
+      
+      port = (int)m_it->port;
     }
-	
-    if(m_it->dir == SdpMedia::DirActive)
-      remote_active = true;
-	
-    sup_pl.push_back(payload);
-    port = (int)m_it->port;
+    break;
   }
-
-  return payload;
+  return sup_pl;
 }
 
 bool AmSdp::hasTelephoneEvent()

core/AmSdp.h

@@ -171,7 +171,7 @@ public:
    * @return !=0 if error encountered.
    */
   int genResponse(const string& localip, int localport, 
-		  string& out_buf);
+		  string& out_buf, bool single_codec = false);
 
   /** 
    * Generate an SDP offer. 
@@ -181,9 +181,9 @@ public:
 
   /** 
    * Get a compatible payload from SDP offer/response. 
-   * @return NULL if error encountered.
+   * @return empty vector if error encountered.
    */
-  SdpPayload* getCompatiblePayload(int media_type, string& addr, int& port);
+  const vector<SdpPayload*>& getCompatiblePayloads(int media_type, string& addr, int& port);
 
   /**
    * Test if remote UA supports 'telefone_event'.

core/AmSession.cpp

@@ -125,7 +125,7 @@ AmSession::AmSession()
     dlg(this),
     detached(true),
     sess_stopped(false),rtp_str(this),negotiate_onreply(false),
-    input(0), output(0), payload(0),
+    input(0), output(0),
     m_dtmfDetector(this), m_dtmfEventQueue(&m_dtmfDetector),
     m_dtmfDetectionEnabled(true)
 {
@@ -207,9 +207,9 @@ void AmSession::setLocalTag(const string& tag)
   dlg.local_tag = tag;
 }
 
-const SdpPayload* AmSession::getPayload()
+const vector<SdpPayload*>& AmSession::getPayloads()
 {
-  return &payload;
+  return m_payloads;
 }
 
 int AmSession::getRPort()
@@ -232,11 +232,12 @@ void AmSession::negotiate(const string& sdp_body,
   if(sdp.media.empty())
     throw AmSession::Exception(400,"no media line found in SDP message");
     
-  SdpPayload* tmp_pl = sdp.getCompatiblePayload(MT_AUDIO,r_host,r_port);
+  m_payloads = sdp.getCompatiblePayloads(MT_AUDIO, r_host, r_port);
 
-  if(!tmp_pl)
+  if (m_payloads.size() == 0)
     throw AmSession::Exception(606,"could not find compatible payload");
     
+/*
   if(payload.int_pt == -1){
 
     payload = *tmp_pl;
@@ -246,6 +247,7 @@ void AmSession::negotiate(const string& sdp_body,
     DBG("old payload: %i; new payload: %i\n",payload.int_pt,tmp_pl->int_pt);
     throw AmSession::Exception(400,"do not accept payload changes");
   }
+*/
 
   const SdpPayload *telephone_event_payload = sdp.telephoneEventPayload();
   if(telephone_event_payload)
@@ -277,7 +279,7 @@ void AmSession::negotiate(const string& sdp_body,
   unlockAudio();
 
   if(sdp_reply)
-    sdp.genResponse(AmConfig::LocalIP,rtp_str.getLocalPort(),*sdp_reply);
+    sdp.genResponse(AmConfig::LocalIP,rtp_str.getLocalPort(),*sdp_reply, AmConfig::SingleCodecInOK);
 }
 
 void AmSession::run()
@@ -620,7 +622,7 @@ int AmSession::acceptAudio(const string& body,
 	    
       // enable RTP stream
       lockAudio();
-      rtp_str.init(&payload);
+      rtp_str.init(m_payloads);
       unlockAudio();
 	    
       DBG("Sending Rtp data to %s/%i\n",

core/AmSession.h

@@ -108,7 +108,7 @@ class AmSession : public AmThread,
   AmAudio*     input;
   AmAudio*     output;
 
-  SdpPayload   payload;
+  vector<SdpPayload *>  m_payloads;
   bool         negotiate_onreply;
 
   AmDtmfDetector   m_dtmfDetector;
@@ -248,7 +248,7 @@ public:
   void setLocalTag(const string& tag);
 
   /** Gets the current RTP payload */
-  const SdpPayload* getPayload();
+  const vector<SdpPayload*>& getPayloads();
 
   /** Gets the port number of the remote part of the session */
   int getRPort();

core/sems.conf.sample

@@ -143,3 +143,11 @@ media_processor_threads=1
 #
 #
 # signature="SEMS media server 1.0"
+
+# optional parameter: single_codec_in_ok={yes|no}
+#
+# - use single codec in 200 OK response
+#
+#   default=no
+#
+# single_codec_in_ok=no