[asterisk-commits] rizzo: branch rizzo/astobj2 r51228 - /team/rizzo/astobj2/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: rizzo
Date: Thu Jan 18 10:37:06 2007
New Revision: 51228

URL: http://svn.digium.com/view/asterisk?view=rev&rev=51228
Log:
merge from trunk 48965
Add an API call that initializes an RTP structure

add a comment on whether this api could as well bzero the data structure.


Modified:
    team/rizzo/astobj2/channels/chan_sip.c

Modified: team/rizzo/astobj2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/rizzo/astobj2/channels/chan_sip.c?view=diff&rev=51228&r1=51227&r2=51228
==============================================================================
--- team/rizzo/astobj2/channels/chan_sip.c (original)
+++ team/rizzo/astobj2/channels/chan_sip.c Thu Jan 18 10:37:06 2007
@@ -5199,10 +5199,12 @@
 	/* Initialize the temporary RTP structures we use to evaluate the offer from the peer */
 	newaudiortp = alloca(ast_rtp_alloc_size());
 	memset(newaudiortp, 0, ast_rtp_alloc_size());
+	ast_rtp_new_init(newaudiortp);	/* XXX could as well bzero ? */
 	ast_rtp_pt_clear(newaudiortp);
 
 	newvideortp = alloca(ast_rtp_alloc_size());
 	memset(newvideortp, 0, ast_rtp_alloc_size());
+	ast_rtp_new_init(newvideortp);
 	ast_rtp_pt_clear(newvideortp);
 
 	/* Update our last rtprx when we receive an SDP, too */