[asterisk-commits] tzafrir: branch tzafrir/monitor-rtp-14 r213893 - /team/tzafrir/monitor-rtp-14...

[SVN commits to the Asterisk project]

Author: tzafrir
Date: Mon Aug 24 18:34:01 2009
New Revision: 213893

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=213893
Log:
send dummy SIP packet with metadata on startup/end

On stream startup send a SIP INVITE message and on stream end send a BYE
message.

Each of those include practically the same data and are intended to pass
information about the stream to the remote party. The remote recording
server is not expected to respond to those messages.

This method is used to pass CDR information to the remote side. Currrently
only src, dst, and clid are passed along.

Also note that unlike any standard SIP call, a separate INVITE/BYE pair
will be used for any separate direction (tx, rx) of the channel.

Modified:
    team/tzafrir/monitor-rtp-14/res/res_monitor.c

Modified: team/tzafrir/monitor-rtp-14/res/res_monitor.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/tzafrir/monitor-rtp-14/res/res_monitor.c?view=diff&rev=213893&r1=213892&r2=213893
==============================================================================
--- team/tzafrir/monitor-rtp-14/res/res_monitor.c (original)
+++ team/tzafrir/monitor-rtp-14/res/res_monitor.c Mon Aug 24 18:34:01 2009
@@ -212,6 +212,126 @@
 	return fs;
 }
 
+#ifdef RTP_MONITOR_DEBUG
+static	const char *cdr_readonly_vars[] = { "clid", "src", "dst", "dcontext", "channel", "dstchannel",
+				    "lastapp", "lastdata", "start", "answer", "end", "duration",
+				    "billsec", "disposition", "amaflags", "accountcode", "uniqueid",
+				    "userfield", NULL };
+#endif /* ifdef RTP_MONITOR_DEBUG */
+
+static int send_cdr(struct ast_channel *chan, int begin, int is_rx, struct monitor_rtp_state *rtp_state)
+{
+	char		*src;
+	char		*dst;
+	char		*clid;
+	char		srcbuf[BUFSIZ];
+	char		dstbuf[BUFSIZ];
+	char		clidbuf[BUFSIZ];
+	char		header[BUFSIZ];
+	char		payload[BUFSIZ];
+	char		packet[BUFSIZ];
+	struct sockaddr_in	my_addr;
+	char		my_ip[INET_ADDRSTRLEN + 1];
+	in_port_t	rtp_port = rtp_state->dest.sin_port;
+	struct ast_cdr *cdr = chan ? chan->cdr : NULL;
+	int		content_length;
+	int		i;
+
+	if (!cdr)
+		return -1;
+	ast_log(LOG_NOTICE, "%s-%s(%d): %s\n",
+		(begin) ? "BEGIN" : "END",
+		(is_rx) ? "RX" : "TX",
+		ntohs(rtp_port),
+		chan->uniqueid);
+#if 0
+	/* Debugging code */
+	{
+		char		buf[BUFSIZ];
+
+		buf[0] = '\0';
+		ast_cdr_serialize_variables(cdr, buf, sizeof(buf), '\t', '\n', 1);
+		ast_log(LOG_NOTICE, "%s\n", buf);
+	}
+#endif
+#ifdef RTP_MONITOR_DEBUG
+	{
+		char		buf[BUFSIZ];
+		int		len = sizeof(buf);
+
+		for(i = 0; cdr_readonly_vars[i]; i++) {
+			buf[0] = '\0';
+			ast_cdr_getvar(cdr, cdr_readonly_vars[i], &ret, buf, len, 0, 0);
+			if(ret) {
+				ast_log(LOG_NOTICE, "%-20s - '%s'\n",
+						cdr_readonly_vars[i], ret);
+			}
+		}
+	}
+#endif /* ifdef RTP_MONITOR_DEBUG */
+	if(!get_ipaddr(reporting_host, &(my_addr.sin_addr))) {
+		ast_log(LOG_WARNING, "Bad reporting_host: '%s'\n", reporting_host);
+		return -1;
+	}
+	if(inet_ntop(AF_INET, &my_addr.sin_addr, my_ip, sizeof(my_ip)) == NULL) {
+		ast_log(LOG_WARNING, "Cannot convert ip address: %s\n", strerror(errno));
+		return -1;
+	}
+	ast_cdr_getvar(cdr, "src", &src, srcbuf, sizeof(srcbuf), 0, 0);
+	ast_cdr_getvar(cdr, "dst", &dst, dstbuf, sizeof(dstbuf), 0, 0);
+	ast_cdr_getvar(cdr, "clid", &clid, clidbuf, sizeof(clidbuf), 0, 0);
+	if(begin) {
+		snprintf(payload, sizeof(payload),
+				"v=0\r\n"
+				"o=asterisk %s 1 IN IP4 %s\r\n"
+				"c=IN IP4 %s\r\n"
+				"s=Phone recording (%s)\r\n"
+				"i=Asterisk Monitor (pid=%d)\r\n"
+				"m=audio %d RTP/AVP 0 \r\n"
+				"a=rtpmap:0 PCMU/8000\r\n",
+				chan->uniqueid, my_ip,
+				my_ip,
+				(is_rx) ? "RX" : "TX",
+				getpid(),
+				ntohs(rtp_port));
+	} else
+		payload[0] = '\0';
+	content_length = strlen(payload);
+	i = 0;
+	i += snprintf(header + i, sizeof(header) - i,
+		"%s sip:%s@%s:5060 SIP/2.0\r\n",
+		(begin) ? "INVITE" : "BYE",
+		srcbuf, my_ip);
+	i += snprintf(header + i, sizeof(header) - i,
+		"Via: SIP/2.0/UDP %s:5061;branch=z9hG4bK-%s\r\n",
+		my_ip, chan->uniqueid);
+	i += snprintf(header + i, sizeof(header) - i, "From: <sip:%s@%s:5061>;tag=1\r\n", src, my_ip);
+	i += snprintf(header + i, sizeof(header) - i, "To: <sip:%s@%s:5060>\r\n", dst, my_ip);
+	i += snprintf(header + i, sizeof(header) - i, "Call-ID: %s\r\n", chan->uniqueid);
+	i += snprintf(header + i, sizeof(header) - i, "CSeq: 1 %s\r\n", (begin) ? "INVITE" : "BYE");
+	i += snprintf(header + i, sizeof(header) - i,
+		"Contact: sip:asterisk@%s:5061\r\n"
+		"Max-Forwards: 70\r\n"
+		"Subject: %s %s recording of %s\r\n",
+		my_ip,
+		(begin) ? "BEGIN" : "END",
+		(is_rx) ? "RX" : "TX",
+		clid);
+	if(begin) {
+		i += snprintf(header + i, sizeof(header) - i, "Content-Type: application/sdp\r\n");
+	}
+	i += snprintf(header + i, sizeof(header) - i, "Content-Length: %d\r\n", content_length);
+	i = snprintf(packet, sizeof(packet), "%s\r\n%s", header, payload);
+	ast_log(LOG_WARNING, "SIP header: %s\n", header);
+	ast_log(LOG_WARNING, "SIP payload: %s\n", payload);
+	ast_log(LOG_WARNING, "SIP packet: %s\n", packet);
+	if(sendto(sip_socket, packet, i, 0, &sip_server_addr, sizeof(sip_server_addr)) < 0) {
+		ast_log(LOG_WARNING, "Bad SIP sendto (%d bytes): %s\n", i, strerror(errno));
+		return -1;
+	}
+	return 0;
+}
+
 /* Start monitoring a channel */
 int ast_monitor_start(	struct ast_channel *chan, const char *format_spec,
 		const char *fname_base, int need_lock)