[asterisk-commits] jpeeler: branch 1.4 r120863 - /branches/1.4/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: jpeeler
Date: Fri Jun  6 10:33:15 2008
New Revision: 120863

URL: http://svn.digium.com/view/asterisk?view=rev&rev=120863
Log:
This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead.


Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=120863&r1=120862&r2=120863
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Fri Jun  6 10:33:15 2008
@@ -1194,6 +1194,8 @@
 /*! \brief A per-thread temporary pvt structure */
 AST_THREADSTORAGE_CUSTOM(ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup);
 
+AST_THREADSTORAGE(ast_rtp_buf, ast_rtp_buf_init);
+
 /*! \todo Move the sip_auth list to AST_LIST */
 static struct sip_auth *authl = NULL;		/*!< Authentication list for realm authentication */
 
@@ -5060,12 +5062,20 @@
 	}
 
 	/* Initialize the temporary RTP structures we use to evaluate the offer from the peer */
+#ifdef LOW_MEMORY
+	newaudiortp = ast_threadstorage_get(&ast_rtp_buf, ast_rtp_alloc_size());
+#else
 	newaudiortp = alloca(ast_rtp_alloc_size());
+#endif
 	memset(newaudiortp, 0, ast_rtp_alloc_size());
 	ast_rtp_new_init(newaudiortp);
 	ast_rtp_pt_clear(newaudiortp);
 
+#ifdef LOW_MEMORY
+	newvideortp = ast_threadstorage_get(&ast_rtp_buf, ast_rtp_alloc_size());
+#else
 	newvideortp = alloca(ast_rtp_alloc_size());
+#endif
 	memset(newvideortp, 0, ast_rtp_alloc_size());
 	ast_rtp_new_init(newvideortp);
 	ast_rtp_pt_clear(newvideortp);