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Qualify rtptimeout with a reinvite having taken place (bug #2286) 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer 2004-08-23 14:31:20 +00:00
parent 1afd98d916
commit 230d4b38a3
1 changed files with 21 additions and 7 deletions
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28
channels/chan_sip.c
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@ -7576,7 +7576,7 @@ restartsearch:
sip = iflist;
while(sip) {
ast_mutex_lock(&sip->lock);
if (sip->rtp && sip->lastrtprx && (sip->rtptimeout || sip->rtpholdtimeout)) {
if (sip->rtp && sip->lastrtprx && (sip->rtptimeout || sip->rtpholdtimeout) && !sip->redirip.sin_addr.s_addr) {
if (t > sip->lastrtprx + sip->rtptimeout) {
/* Might be a timeout now -- see if we're on hold */
struct sockaddr_in sin;
@ -8592,19 +8592,29 @@ static int reload_config(void)
static struct ast_rtp *sip_get_rtp_peer(struct ast_channel *chan)
{
struct sip_pvt *p;
struct ast_rtp *rtp = NULL;
p = chan->pvt->pvt;
if (p && p->rtp && p->canreinvite)
return p->rtp;
return NULL;
if (p) {
ast_mutex_lock(&p->lock);
if (p->rtp && p->canreinvite)
rtp = p->rtp;
ast_mutex_unlock(&p->lock);
}
return rtp;
}
static struct ast_rtp *sip_get_vrtp_peer(struct ast_channel *chan)
{
struct sip_pvt *p;
struct ast_rtp *rtp = NULL;
p = chan->pvt->pvt;
if (p && p->vrtp && p->canreinvite)
return p->vrtp;
return NULL;
if (p) {
ast_mutex_lock(&p->lock);
if (p->vrtp && p->canreinvite)
rtp = p->vrtp;
ast_mutex_unlock(&p->lock);
}
return rtp;
}
static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
@ -8612,6 +8622,7 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
struct sip_pvt *p;
p = chan->pvt->pvt;
if (p) {
ast_mutex_lock(&p->lock);
if (rtp)
ast_rtp_get_peer(rtp, &p->redirip);
else
@ -8629,6 +8640,9 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
p->needreinvite = 1;
}
}
/* Reset lastrtprx timer */
time(&p->lastrtprx);
ast_mutex_unlock(&p->lock);
return 0;
}
return -1;