[asterisk-commits] oej: branch 1.6.2 r239703 - /branches/1.6.2/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: oej
Date: Wed Jan 13 05:25:55 2010
New Revision: 239703

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=239703
Log:
Show proper stats in "sip show channelstats"

SIP show channelstats show current RTCP statistics for calls - if we have it. Calls bridged
in RTP p2p bridge doesn't have any statistics.  In calls where the remote end doesn't send
RTCP or we can't receive it due to NAT, there's no reliable data as well.

Thanks, Klaus, for the patch. Sorry for the delay.
(closes issue #15819)
Reported by: klaus3000
Patches: 
      asterisk-sip-show-channelstats-1.6.2.txt uploaded by klaus3000 (license 65)
Tested by: klaus3000, oej


Modified:
    branches/1.6.2/channels/chan_sip.c

Modified: branches/1.6.2/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/channels/chan_sip.c?view=diff&rev=239703&r1=239702&r2=239703
==============================================================================
--- branches/1.6.2/channels/chan_sip.c (original)
+++ branches/1.6.2/channels/chan_sip.c Wed Jan 13 05:25:55 2010
@@ -15804,11 +15804,10 @@
 /*! \brief Callback for show_chanstats */
 static int show_chanstats_cb(void *__cur, void *__arg, int flags)
 {
-#define FORMAT2 "%-15.15s  %-11.11s  %-8.8s %-10.10s  %-10.10s (%-2.2s) %-6.6s %-10.10s  %-10.10s ( %%) %-6.6s\n"
-#define FORMAT  "%-15.15s  %-11.11s  %-8.8s %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u %-10.10u%-1.1s %-10.10u (%-2.2u%%) %-6.6u\n"
+#define FORMAT2 "%-15.15s  %-11.11s  %-8.8s %-10.10s  %-10.10s (     %%) %-6.6s %-10.10s  %-10.10s (     %%) %-6.6s\n"
+#define FORMAT  "%-15.15s  %-11.11s  %-8.8s %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.6u %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.6u\n"
 	struct sip_pvt *cur = __cur;
-	unsigned int rxcount;
-	unsigned int txcount;
+	unsigned int rxcount, txcount, rxploss, txploss;
 	char durbuf[10];
         int duration;
         int durh, durm, durs;
@@ -15827,6 +15826,8 @@
 	}
 	rxcount = ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXCOUNT);
 	txcount = ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXCOUNT);
+	rxploss = ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS);
+	txploss = ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS);
 
 	/* Find the duration of this channel */
 	if (c && c->cdr && !ast_tvzero(c->cdr->start)) {
@@ -15845,13 +15846,13 @@
 		durbuf,
 		rxcount > (unsigned int) 100000 ? (unsigned int) (rxcount)/(unsigned int) 1000 : rxcount,
 		rxcount > (unsigned int) 100000 ? "K":" ",
-		ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS),
-		rxcount > ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS) ? (unsigned int) ((double) ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS) / (ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXPLOSS) + rxcount) * 100) : 0,
+		rxploss,
+		(rxcount + rxploss) > 0 ? (double) rxploss / (rxcount + rxploss) * 100 : 0,
 		ast_rtp_get_qosvalue(cur->rtp, AST_RTP_RXJITTER),
 		txcount > (unsigned int) 100000 ? (unsigned int) (txcount)/(unsigned int) 1000 : txcount,
 		txcount > (unsigned int) 100000 ? "K":" ",
-		ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS),
-		txcount > ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS) ? (unsigned int) ((double) ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXPLOSS)/ txcount * 100) : 0,
+		txploss,
+		txcount > 0 ? (double) txploss / txcount * 100 : 0,
 		ast_rtp_get_qosvalue(cur->rtp, AST_RTP_TXJITTER)
 	);
 	arg->numchans++;
@@ -15879,7 +15880,7 @@
 	if (a->argc != 3)
 		return CLI_SHOWUSAGE;
 
-	ast_cli(a->fd, FORMAT2, "Peer", "Call ID", "Duration", "Recv: Pack", "Lost", "%", "Jitter", "Send: Pack", "Lost", "Jitter");
+	ast_cli(a->fd, FORMAT2, "Peer", "Call ID", "Duration", "Recv: Pack", "Lost", "Jitter", "Send: Pack", "Lost", "Jitter");
 	/* iterate on the container and invoke the callback on each item */
 	ao2_t_callback(dialogs, OBJ_NODATA, show_chanstats_cb, &arg, "callback to sip show chanstats");
 	ast_cli(a->fd, "%d active SIP channel%s\n", arg.numchans, (arg.numchans != 1) ? "s" : "");