Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@120885 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -1194,7 +1194,10 @@ 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);
+static void ts_ast_rtp_destroy(void *);
+
+AST_THREADSTORAGE_CUSTOM(ts_audio_rtp, ts_audio_rtp_init, ts_ast_rtp_destroy);
+AST_THREADSTORAGE_CUSTOM(ts_video_rtp, ts_video_rtp_init, ts_ast_rtp_destroy);
 
 /*! \todo Move the sip_auth list to AST_LIST */
 static struct sip_auth *authl = NULL;		/*!< Authentication list for realm authentication */
@@ -5063,7 +5066,7 @@ 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());
+	newaudiortp = ast_threadstorage_get(&ts_audio_rtp, ast_rtp_alloc_size());
 #else
 	newaudiortp = alloca(ast_rtp_alloc_size());
 #endif
@@ -5072,7 +5075,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 	ast_rtp_pt_clear(newaudiortp);
 
 #ifdef LOW_MEMORY
-	newvideortp = ast_threadstorage_get(&ast_rtp_buf, ast_rtp_alloc_size());
+	newvideortp = ast_threadstorage_get(&ts_video_rtp, ast_rtp_alloc_size());
 #else
 	newvideortp = alloca(ast_rtp_alloc_size());
 #endif
@@ -5607,6 +5610,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
 	return 0;
 }
 
+static void ts_ast_rtp_destroy(void *data)
+{
+    struct ast_rtp *tmp = data;
+    ast_rtp_destroy(tmp);
+}
 
 /*! \brief Add header to SIP message */
 static int add_header(struct sip_request *req, const char *var, const char *value)