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

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 46aaca2..613bea3 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -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.