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Optimize chan_sip.c check_rtp_timeout() function. 

* Make check_rtp_timeout() remember the values returned by
ast_rtp_instance_get_timeout(), ast_rtp_instance_get_hold_timeout(), and
ast_rtp_instance_get_keepalive() instead of repeatedly calling them.

(closes issue ASTERISK-18319)
Reported by: Rob Gagnon
Patches:
      issue-18319-trunk-r333066.diff (License #6159) patch uploaded by Rob Gagnon

Review: https://reviewboard.asterisk.org/r/1377/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Richard Mudgett 2011-08-31 18:11:23 +00:00
parent dadc749dac
commit 89e79698fd
1 changed files with 14 additions and 6 deletions
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20
channels/chan_sip.c
View File

@ -25688,6 +25688,10 @@ static int sip_send_mwi_to_peer(struct sip_peer *peer, int cache_only)
/*! \brief helper function for the monitoring thread -- seems to be called with the assumption that the dialog is locked */
static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
{
int timeout;
int hold_timeout;
int keepalive;
/* If we have no active owner, no need to check timers */
if (!dialog->owner) {
dialog_unlink_rtpcheck(dialog);
@ -25710,15 +25714,19 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
return;
}
/* Store these values locally to avoid multiple function calls */
timeout = ast_rtp_instance_get_timeout(dialog->rtp);
hold_timeout = ast_rtp_instance_get_hold_timeout(dialog->rtp);
keepalive = ast_rtp_instance_get_keepalive(dialog->rtp);
/* If we have no timers set, return now */
if (!ast_rtp_instance_get_keepalive(dialog->rtp) && !ast_rtp_instance_get_timeout(dialog->rtp) && !ast_rtp_instance_get_hold_timeout(dialog->rtp)) {
if (!keepalive && !timeout && !hold_timeout) {
dialog_unlink_rtpcheck(dialog);
return;
}
/* Check AUDIO RTP keepalives */
if (dialog->lastrtptx && ast_rtp_instance_get_keepalive(dialog->rtp) &&
(t > dialog->lastrtptx + ast_rtp_instance_get_keepalive(dialog->rtp))) {
if (dialog->lastrtptx && keepalive && (t > dialog->lastrtptx + keepalive)) {
/* Need to send an empty RTP packet */
dialog->lastrtptx = time(NULL);
ast_rtp_instance_sendcng(dialog->rtp, 0);
@ -25731,10 +25739,10 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
*/
/* Check AUDIO RTP timers */
if (dialog->lastrtprx && (ast_rtp_instance_get_timeout(dialog->rtp) || ast_rtp_instance_get_hold_timeout(dialog->rtp)) && (t > dialog->lastrtprx + ast_rtp_instance_get_timeout(dialog->rtp))) {
if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD) || (ast_rtp_instance_get_hold_timeout(dialog->rtp) && (t > dialog->lastrtprx + ast_rtp_instance_get_hold_timeout(dialog->rtp)))) {
if (dialog->lastrtprx && (timeout || hold_timeout) && (t > dialog->lastrtprx + timeout)) {
if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD) || (hold_timeout && (t > dialog->lastrtprx + hold_timeout))) {
/* Needs a hangup */
if (ast_rtp_instance_get_timeout(dialog->rtp)) {
if (timeout) {
if (!dialog->owner || ast_channel_trylock(dialog->owner)) {
/*
* Don't block, just try again later.