[asterisk-commits] oej: branch oej/pinefrog-1.4 r239568 - /team/oej/pinefrog-1.4/

[SVN commits to the Asterisk project]

Author: oej
Date: Tue Jan 12 13:30:10 2010
New Revision: 239568

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=239568
Log:
Sorry, review at 
http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-1.4/README.pinefrog-rtcp

(The old URI pointed to the old version of the document... Life is hard)

Modified:
    team/oej/pinefrog-1.4/README.pinefrog-rtcp

Modified: team/oej/pinefrog-1.4/README.pinefrog-rtcp
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-1.4/README.pinefrog-rtcp?view=diff&rev=239568&r1=239567&r2=239568
==============================================================================
--- team/oej/pinefrog-1.4/README.pinefrog-rtcp (original)
+++ team/oej/pinefrog-1.4/README.pinefrog-rtcp Tue Jan 12 13:30:10 2010
@@ -77,6 +77,7 @@
 - Can we have some sort of ring buffer for the latest RTCP reports for a device (peer) 
   and use that to determine the status of the connection to the peer?
 - Can we use the APP packet for relaying events in joined bridges, like meetme?
+- What should we use as CNAME? Currently SIP call ID.
 
 
 Scenarios to test