[asterisk-commits] oej: branch oej/sipchanstats r123825 - in /team/oej/sipchanstats: channels/ m...

[SVN commits to the Asterisk project]

Author: oej
Date: Thu Jun 19 05:04:31 2008
New Revision: 123825

URL: http://svn.digium.com/view/asterisk?view=rev&rev=123825
Log:
Add doc and explanation of why you don't see the RTCP stats for some calls.

Modified:
    team/oej/sipchanstats/channels/chan_sip.c
    team/oej/sipchanstats/main/rtp.c

Modified: team/oej/sipchanstats/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/sipchanstats/channels/chan_sip.c?view=diff&rev=123825&r1=123824&r2=123825
==============================================================================
--- team/oej/sipchanstats/channels/chan_sip.c (original)
+++ team/oej/sipchanstats/channels/chan_sip.c Thu Jun 19 05:04:31 2008
@@ -11821,7 +11821,8 @@
 
 static char show_channelstats_usage[] = 
 "Usage: sip show channelstats\n"
-"       Lists all currently active SIP channel's RTCP statistics.\n";
+"       Lists all currently active SIP channel's RTCP statistics.\n"
+"       Note that calls in the much optimized P2P bridge mode will not show any packets here.";
 
 static char show_channel_usage[] = 
 "Usage: sip show channel <channel>\n"

Modified: team/oej/sipchanstats/main/rtp.c
URL: http://svn.digium.com/view/asterisk/team/oej/sipchanstats/main/rtp.c?view=diff&rev=123825&r1=123824&r2=123825
==============================================================================
--- team/oej/sipchanstats/main/rtp.c (original)
+++ team/oej/sipchanstats/main/rtp.c Thu Jun 19 05:04:31 2008
@@ -2083,6 +2083,9 @@
 		if (option_debug > 1)
 			ast_log(LOG_DEBUG, "NO RTP Structure? Kidding me? \n");
 		return 0;
+	}
+	if (option_debug > 1 && rtp->rtcp == NULL) {
+		ast_log(LOG_DEBUG, "NO RTCP structure. Maybe in RTP p2p bridging mode? \n");
 	}
 
 	switch (value) {