From: Joshua Colp <jcolp@digium.com>
Date: Mon, 17 Dec 2007 15:18:58 +0000 (+0000)
Subject: Fix usage of rtptimeout. It can be used without rtpkeepalive, and the value can not... 
X-Git-Tag: 1.6.0-beta1~3^2~396
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=e693a515cc80412e2c328f43513859e6ff72645b;p=thirdparty%2Fasterisk.git

Fix usage of rtptimeout. It can be used without rtpkeepalive, and the value can not be accessed directly in the SIP pvt structure. All RTP related timeouts have to be retrieved using the ast_rtp_* function calls.
(closes issue #11562)
Reported by: ibc


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 6e5a3b17a0..fc9c49b8fd 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -16921,7 +16921,7 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
 		return;
 
 	/* If we have no timers set, return now */
-	if (ast_rtp_get_rtpkeepalive(dialog->rtp) == 0 || (ast_rtp_get_rtptimeout(dialog->rtp) == 0 && ast_rtp_get_rtpholdtimeout(dialog->rtp) == 0))
+	if ((ast_rtp_get_rtpkeepalive(dialog->rtp) == 0) && (ast_rtp_get_rtptimeout(dialog->rtp) == 0) && (ast_rtp_get_rtpholdtimeout(dialog->rtp) == 0))
 		return;
 
 	/* Check AUDIO RTP keepalives */
@@ -16948,7 +16948,7 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
 		if (sin.sin_addr.s_addr || (ast_rtp_get_rtpholdtimeout(dialog->rtp) &&
 		     (t > dialog->lastrtprx + ast_rtp_get_rtpholdtimeout(dialog->rtp)))) {
 			/* Needs a hangup */
-			if (dialog->rtptimeout) {
+			if (ast_rtp_get_rtptimeout(dialog->rtp)) {
 				while (dialog->owner && ast_channel_trylock(dialog->owner)) {
 					sip_pvt_unlock(dialog);
 					usleep(1);