This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struc...

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@120863 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 6a3619ae7de3377fb5b95686a34fefc6313343d3..50a7f43810bc045715b3e20a74013f893f52f64b 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1194,6 +1194,8 @@ static void temp_pvt_cleanup(void *);
 /*! \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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
        }
 
        /* 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);