[Live-devel] [RFC PATCH] Add option to disable RTCP

[Julian Scheel]

Hi Ross,

we have the need to create a RTSP Session without a RTCP Instance because of a
Server that has a bug causing it not to terminate sessions after it received
an RTCP report from the client.
I came up with the attached patch, which feels really ugly and I'm pretty sure
could have unexpected side-effects. Would you mind taking a look and maybe
propose a better way to do this?

Thanks,
Julian

---

Allow connections without a RTCP port to be configured.
---
 liveMedia/MediaSession.cpp        |  6 +++++-
 liveMedia/RTSPClient.cpp          | 18 ++++++++++++++----
 liveMedia/include/MediaSession.hh |  3 +++
 3 files changed, 22 insertions(+), 5 deletions(-)

diff --git a/liveMedia/MediaSession.cpp b/liveMedia/MediaSession.cpp
index a5b13d0..54e9346 100644
--- a/liveMedia/MediaSession.cpp
+++ b/liveMedia/MediaSession.cpp
@@ -821,7 +821,7 @@ Boolean MediaSubsession::initiate(int useSpecialRTPoffset) {
     }
 
     // Finally, create our RTCP instance. (It starts running automatically)
-    if (fRTPSource != NULL && fRTCPSocket != NULL) {
+    if (fRTPSource != NULL && fRTCPSocket != NULL && !fNoRTCP) {
       // If bandwidth is specified, use it and add 5% for RTCP overhead.
       // Otherwise make a guess at 500 kbps.
       unsigned totSessionBandwidth
@@ -867,6 +867,10 @@ Boolean MediaSubsession::setClientPortNum(unsigned short portNum) {
   return True;
 }
 
+void MediaSubsession::setNoRTCP(Boolean enable) {
+  fNoRTCP = enable;
+}
+
 char const* MediaSubsession::attrVal_str(char const* attrName) const {
   SDPAttribute* attr = (SDPAttribute*)(fAttributeTable->Lookup(attrName));
   if (attr == NULL) return "";
diff --git a/liveMedia/RTSPClient.cpp b/liveMedia/RTSPClient.cpp
index fc68042..f659ff5 100644
--- a/liveMedia/RTSPClient.cpp
+++ b/liveMedia/RTSPClient.cpp
@@ -662,9 +662,15 @@ Boolean RTSPClient::setRequestFields(RequestRecord* request,
     char const* transportFmt;
     if (strcmp(subsession.protocolName(), "UDP") == 0) {
       suffix = "";
-      transportFmt = "Transport: RAW/RAW/UDP%s%s%s=%d-%d\r\n";
+      if (subsession.noRTCP())
+        transportFmt = "Transport: RAW/RAW/UDP%s%s%s=%d\r\n";
+      else
+        transportFmt = "Transport: RAW/RAW/UDP%s%s%s=%d-%d\r\n";
     } else {
-      transportFmt = "Transport: RTP/AVP%s%s%s=%d-%d\r\n";
+      if (subsession.noRTCP())
+        transportFmt = "Transport: RTP/AVP%s%s%s=%d\r\n";
+      else
+        transportFmt = "Transport: RTP/AVP%s%s%s=%d-%d\r\n";
     }
     
     cmdURL = new char[strlen(prefix) + strlen(separator) + strlen(suffix) + 1];
@@ -699,8 +705,12 @@ Boolean RTSPClient::setRequestFields(RequestRecord* request,
     unsigned transportSize = strlen(transportFmt)
       + strlen(transportTypeStr) + strlen(modeStr) + strlen(portTypeStr) + 2*5 /* max port len */;
     char* transportStr = new char[transportSize];
-    sprintf(transportStr, transportFmt,
-	    transportTypeStr, modeStr, portTypeStr, rtpNumber, rtcpNumber);
+    if (subsession.noRTCP())
+      sprintf(transportStr, transportFmt,
+	      transportTypeStr, modeStr, portTypeStr, rtpNumber);
+    else
+      sprintf(transportStr, transportFmt,
+	      transportTypeStr, modeStr, portTypeStr, rtpNumber, rtcpNumber);
     
     // When sending more than one "SETUP" request, include a "Session:" header in the 2nd and later commands:
     char* sessionStr = createSessionString(fLastSessionId);
diff --git a/liveMedia/include/MediaSession.hh b/liveMedia/include/MediaSession.hh
index f2ea909..107fcdc 100644
--- a/liveMedia/include/MediaSession.hh
+++ b/liveMedia/include/MediaSession.hh
@@ -168,6 +168,7 @@ public:
   char const* protocolName() const { return fProtocolName; }
   char const* controlPath() const { return fControlPath; }
   Boolean isSSM() const { return fSourceFilterAddr.s_addr != 0; }
+  Boolean noRTCP() const { return fNoRTCP; }
 
   unsigned short videoWidth() const { return fVideoWidth; }
   unsigned short videoHeight() const { return fVideoHeight; }
@@ -207,6 +208,7 @@ public:
       // description does not specfy a client port number - an ephemeral
       // (even) port number is chosen.)  This routine must *not* be
       // called after initiate().
+  void setNoRTCP(Boolean enable);
   void receiveRawMP3ADUs() { fReceiveRawMP3ADUs = True; } // optional hack for audio/MPA-ROBUST; must not be called after initiate()
   void receiveRawJPEGFrames() { fReceiveRawJPEGFrames = True; } // optional hack for video/JPEG; must not be called after initiate()
   char*& connectionEndpointName() { return fConnectionEndpointName; }
@@ -312,6 +314,7 @@ protected:
   char* fControlPath; // holds optional a=control: string
   struct in_addr fSourceFilterAddr; // used for SSM
   unsigned fBandwidth; // in kilobits-per-second, from b= line
+  Boolean fNoRTCP; // disable rtcp port announcement
 
   double fPlayStartTime;
   double fPlayEndTime;
-- 
2.7.0