[asterisk-commits] oej: branch oej/pinefrog-1.4 r239709 - in /team/oej/pinefrog-1.4: ./ include/...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Jan 13 07:28:42 CST 2010


Author: oej
Date: Wed Jan 13 07:28:40 2010
New Revision: 239709

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=239709
Log:
Adding a few more items to the report structure

Modified:
    team/oej/pinefrog-1.4/README.pinefrog-rtcp
    team/oej/pinefrog-1.4/include/asterisk/rtp.h
    team/oej/pinefrog-1.4/main/rtp.c

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=239709&r1=239708&r2=239709
==============================================================================
--- team/oej/pinefrog-1.4/README.pinefrog-rtcp (original)
+++ team/oej/pinefrog-1.4/README.pinefrog-rtcp Wed Jan 13 07:28:40 2010
@@ -78,7 +78,8 @@
   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.
-
+- Separate on the IPs of different media servers. IE we can have one SIP peer with
+  multiple media IPs with different properties
 
 Scenarios to test
 ------------------

Modified: team/oej/pinefrog-1.4/include/asterisk/rtp.h
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-1.4/include/asterisk/rtp.h?view=diff&rev=239709&r1=239708&r2=239709
==============================================================================
--- team/oej/pinefrog-1.4/include/asterisk/rtp.h (original)
+++ team/oej/pinefrog-1.4/include/asterisk/rtp.h Wed Jan 13 07:28:40 2010
@@ -77,6 +77,7 @@
 	AST_LIST_ENTRY(ast_rtp_protocol) list;
 };
 
+/*! \brief Data structure only used for RTCP reports */
 struct ast_rtp_quality {
 	unsigned int local_ssrc;          /*!< Our SSRC */
 	unsigned int local_lostpackets;   /*!< Our lost packets */
@@ -87,12 +88,15 @@
 	unsigned int remote_ssrc;         /*!< Their SSRC */
 	unsigned int remote_lostpackets;  /*!< Their lost packets */
 	double       remote_jitter;       /*!< Their reported jitter */
+	double       remote_jitter_max;   /*!< Their reported jitter */
+	double       remote_jitter_min;   /*!< Their reported jitter */
 	unsigned int remote_count;        /*!< Number of transmitted packets */
 	double       rtt;                 /*!< Round trip time */
 	double       rttmax;              /*!< Max observed round trip time */
 	double       rttmin;              /*!< Max observed round trip time */
 	int lasttxformat;		  /*!< Last used codec on transmitted stream */
 	int lastrxformat;		  /*!< Last used codec on received stream */
+	struct sockaddr_in them;	  /*!< The Ip address used for media by remote end */
 };
 
 

Modified: team/oej/pinefrog-1.4/main/rtp.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-1.4/main/rtp.c?view=diff&rev=239709&r1=239708&r2=239709
==============================================================================
--- team/oej/pinefrog-1.4/main/rtp.c (original)
+++ team/oej/pinefrog-1.4/main/rtp.c Wed Jan 13 07:28:40 2010
@@ -255,23 +255,23 @@
 	double accumulated_transit;	/*!< accumulated a-dlsr-lsr */
 	double rtt;			/*!< Last reported rtt */
 	unsigned int reported_jitter;	/*!< The contents of their last jitter entry in the RR */
-	double reported_maxjitter;	/*!< The contents of their last jitter entry in the RR */
-	double reported_minjitter;	/*!< The contents of their last jitter entry in the RR */
-	unsigned int reported_jitter_count;
+	double reported_maxjitter;	/*!< The contents of their max jitter entry received by us */
+	double reported_minjitter;	/*!< The contents of their min jitter entry received by us */
+	unsigned int reported_jitter_count;	/*! Number of reports received */
 	unsigned int reported_lost;	/*!< Reported lost packets in their RR */
 	double reported_maxlost;
 	double reported_minlost;
 	double rxlost;
 	double maxrxlost;
 	double minrxlost;
-	unsigned int rxlost_count;
+	unsigned int rxlost_count;	/*! Number of reports received */
 	char quality[AST_MAX_USER_FIELD];
 	double maxrxjitter;
 	double minrxjitter;
-	unsigned int rxjitter_count;
+	unsigned int rxjitter_count;	/*! Number of reports received */
 	double maxrtt;
 	double minrtt;
-	unsigned int rtt_count;
+	unsigned int rtt_count;		/*! Number of reports received */
 	int sendfur;
 };
 
@@ -2408,6 +2408,7 @@
 	*txcount       transmitted packets
 	*rlp           remote lost packets
 	*rtt           round trip time
+
 	*/
 
 	if (qual && rtp) {
@@ -2418,11 +2419,18 @@
 		qual->local_count = rtp->rxcount;
 		qual->remote_ssrc = rtp->themssrc;
 		qual->remote_count = rtp->txcount;
+		qual->them = rtp->them;	/* IP address and port */
 		if (rtp->rtcp) {
+			qual->local_jitter_max = rtp->rtcp->maxrxjitter;
+			qual->local_jitter_min = rtp->rtcp->minrxjitter;
 			qual->local_lostpackets = rtp->rtcp->expected_prior - rtp->rtcp->received_prior;
 			qual->remote_lostpackets = rtp->rtcp->reported_lost;
 			qual->remote_jitter = rtp->rtcp->reported_jitter / 65536.0;
+			qual->remote_jitter_max = rtp->rtcp->reported_maxjitter;
+			qual->remote_jitter_min = rtp->rtcp->reported_minjitter;
 			qual->rtt = rtp->rtcp->rtt;
+			qual->rttmax = rtp->rtcp->maxrtt;
+			qual->rttmin = rtp->rtcp->minrtt;
 		}
 	}
 	if (rtp->rtcp) {




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