[asterisk-commits] elguero: trunk r388602 - in /trunk: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: elguero
Date: Mon May 13 16:07:02 2013
New Revision: 388602

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=388602
Log:
Fix Crash Caused By One-way Audio With auto_* NAT Settings Fix

The prior code committed, r385473, failed to take into consideration that not
all outgoing calls will be to a peer.  My fault.

This patch does the following:

* Check if there is a related peer involved.  If there is, check and set NAT 
  settings according to the peer's settings.

* Fix a problem with realtime peers.  If the global setting has auto_force_rport
  set and we issued a "sip reload" while a peer is still registered, the peer's
  flags for NAT are reset to off.  When this happens, we were always setting the
  contact address of the peer to that of the full contact info that we had.

(closes issue ASTERISK-21374)
Reported by: jmls
Tested by: Michael L. Young
Patches:
   asterisk-21374-fix-crash-and-rt-peers.diff by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2524/
........

Merged revisions 388601 from http://svn.asterisk.org/svn/asterisk/branches/11

Modified:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-11-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=388602&r1=388601&r2=388602
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon May 13 16:07:02 2013
@@ -30108,18 +30108,25 @@
 		ast_string_field_set(p, peername, ext);
 	/* Recalculate our side, and recalculate Call ID */
 	ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
-	/* When chan_sip is first loaded, we may have a peer entry but it hasn't re-registered yet.
-	   If the peer hasn't re-registered, we have not checked for NAT yet.  With the new
-	   auto_* settings, we need to check for NAT so we do not have one-way audio. */
-	check_for_nat(&p->ourip, p);
-	set_peer_nat(p, p->relatedpeer);
-
-	if (p->natdetected && ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
-		ast_copy_flags(&p->flags[0], &p->relatedpeer->flags[0], SIP_NAT_FORCE_RPORT);
-	}
-
-	if (p->natdetected && ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
-		ast_copy_flags(&p->flags[1], &p->relatedpeer->flags[1], SIP_PAGE2_SYMMETRICRTP);
+	/* When chan_sip is first loaded or reloaded, we need to check for NAT and set the appropiate flags
+	   now that we have the auto_* settings. */
+	check_for_nat(&p->sa, p);
+	/* If there is a peer related to this outgoing call and it hasn't re-registered after
+	   a reload, we need to set the peer's NAT flags accordingly. */
+	if (p->relatedpeer) {
+
+		if (!ast_strlen_zero(p->relatedpeer->fullcontact) && !p->natdetected &&
+			(ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) && !ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT))) {
+			/* We need to make an attempt to determine if a peer is behind NAT
+			   if the peer has the auto_force_rport flag set. */
+			struct ast_sockaddr tmpaddr;
+
+			__set_address_from_contact(p->relatedpeer->fullcontact, &tmpaddr, 0);
+
+			check_for_nat(&tmpaddr, p);
+		}
+
+		set_peer_nat(p, p->relatedpeer);
 	}
 
 	do_setnat(p);
@@ -31340,7 +31347,8 @@
 		 * specified, use that address instead. */
 		/* XXX May need to revisit the final argument; does the realtime DB store whether
 		 * the original contact was over TLS or not? XXX */
-		if (!ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) || ast_sockaddr_isnull(&peer->addr)) {
+		if ((!ast_test_flag(&peer->flags[2],  SIP_PAGE3_NAT_AUTO_RPORT) && !ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT))
+		    || ast_sockaddr_isnull(&peer->addr)) {
 			__set_address_from_contact(ast_str_buffer(fullcontact), &peer->addr, 0);
 		}
 	}