[asterisk-commits] mnicholson: branch 1.4 r315891 - /branches/1.4/channels/chan_sip.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Apr 27 13:58:02 CDT 2011


Author: mnicholson
Date: Wed Apr 27 13:57:56 2011
New Revision: 315891

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=315891
Log:
Fix our compliance with RFC 3261 section 18.2.2.

This change optimizes the free_via() function and removes some redundant null
checking. It also fixes compliance with RFC 3261 section 18.2.2 by always using
the port specified in the Via header for routing responses (even when maddr is
not set). Also the htons() function is now used when setting the port.
Additional documentation comments have been added in various places to make the
logic in the code clearer.

(closes issue #18951)
Reported by: jmls
Patches:
      issue18951_set_proper_port_from_via.patch uploaded by wdoekes (license 717) (modified)

Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=315891&r1=315890&r2=315891
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Wed Apr 27 13:57:56 2011
@@ -4973,10 +4973,7 @@
 		return;
 	}
 
-	if (v->via) {
-		ast_free(v->via);
-	}
-
+	ast_free(v->via);
 	ast_free(v);
 }
 
@@ -5099,6 +5096,21 @@
 	return ((ntohl(addr->s_addr) & 0xf0000000) == 0xe0000000);
 }
 
+/*!
+ * \brief Process the Via header according to RFC 3261 section 18.2.2.
+ * \param p a sip_pvt structure that will be modified according to the received
+ * header
+ * \param req a sip request with a Via header to process
+ *
+ * This function will update the destination of the response according to the
+ * Via header in the request and RFC 3261 section 18.2.2. We do not have a
+ * transport layer so we ignore certain values like the 'received' param (we
+ * set the destination address to the addres the request came from in the
+ * respprep() function).
+ *
+ * \retval -1 error
+ * \retval 0 success
+ */
 static int process_via(struct sip_pvt *p, const struct sip_request *req)
 {
 	struct sip_via *via = parse_via(get_header(req, "Via"));
@@ -5123,16 +5135,12 @@
 		p->sa.sin_family = AF_INET;
 		memcpy(&p->sa.sin_addr, hp->h_addr, sizeof(p->sa.sin_addr));
 
-		if (via->port) {
-			p->sa.sin_port = via->port;
-		} else {
-			p->sa.sin_port = STANDARD_SIP_PORT;
-		}
-
 		if (addr_is_multicast(&p->sa.sin_addr)) {
 			setsockopt(sipsock, IPPROTO_IP, IP_MULTICAST_TTL, &via->ttl, sizeof(via->ttl));
 		}
 	}
+
+	p->sa.sin_port = htons(via->port ? via->port : STANDARD_SIP_PORT);
 
 	free_via(via);
 	return 0;
@@ -6785,6 +6793,9 @@
 
 	/* default to routing the response to the address where the request
 	 * came from.  Since we don't have a transport layer, we do this here.
+	 * The process_via() function will update the port to either the port
+	 * specified in the via header or the default port later on (per RFC
+	 * 3261 section 18.2.2).
 	 */
 	p->sa = p->recv;
 




More information about the asterisk-commits mailing list