[asterisk-commits] oej: branch oej/videocaps r50865 - in /team/oej/videocaps: ./ channels/

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Mon Jan 15 04:22:50 MST 2007


Author: oej
Date: Mon Jan 15 05:22:48 2007
New Revision: 50865

URL: http://svn.digium.com/view/asterisk?view=rev&rev=50865
Log:
Small changes for Tandberg compatibility

Modified:
    team/oej/videocaps/   (props changed)
    team/oej/videocaps/channels/chan_sip.c

Propchange: team/oej/videocaps/
------------------------------------------------------------------------------
    automerge = http://edvina.net/training/

Modified: team/oej/videocaps/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/videocaps/channels/chan_sip.c?view=diff&rev=50865&r1=50864&r2=50865
==============================================================================
--- team/oej/videocaps/channels/chan_sip.c (original)
+++ team/oej/videocaps/channels/chan_sip.c Mon Jan 15 05:22:48 2007
@@ -1,3 +1,5 @@
+/* TIAS support seems half done...
+*/
 /*
  * Asterisk -- An open source telephony toolkit.
  *
@@ -5219,15 +5221,17 @@
 		if (sscanf(b,"CT:%d", &ct) == 1) {
 			ct *= 1000;
 			p->peercaps.maxcallbitrate = ct;
-		}	else if (sscanf(b,"AS:%d", &as) == 1) {
+		} else if (sscanf(b,"AS:%d", &as) == 1) {
 			as *= 1000;
 			p->peercaps.maxvideobitrate = as;
-		} else if (sscanf(b,"TIAS:%d", &tias) == 1);
-		else
-			ast_verbose("Unable to parse b=%s in process_sdp\n", b);
-	}
-
-  if(sipdebug_caps)	
+		} else
+			/* OUCH CLEAR THIS UP */
+			if (sscanf(b,"TIAS:%d", &tias) == 1);
+			else
+				ast_verbose("Unable to parse b=%s in process_sdp\n", b);
+	}
+
+	if(sipdebug_caps)	
 		ast_verbose("SIP PROCESS SDP: CT=%d, AS=%d, TIAS=%d\n", ct, as, tias);
 	
 	/* XXX This could block for a long time, and block the main thread! XXX */
@@ -6529,7 +6533,7 @@
 	  ast_build_string(buf, size, "\r\n");
 	} else {
 		ast_verbose("WARNING Video cap is not valid but have been asked to add fmtp\n");
-		ast_build_string(buf, size, "a=fmtp:%d CIF=1 QCIF=1 maxbr=3840\r\n", rtp_code);
+		ast_build_string(buf, size, "a=fmtp:%d;CIF=1;QCIF=1;maxbr=3840\r\n", rtp_code);
 	}
 }
 



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