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-beta1~401
X-Git-Url: http://git.asterisk.org/gitweb/?p=asterisk%2Fasterisk.git;a=commitdiff_plain;h=e693a515cc80412e2c328f43513859e6ff72645b

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 6e5a3b1..fc9c49b 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);