[asterisk-commits] oej: trunk r46650 - /trunk/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Oct 31 06:56:33 MST 2006
Author: oej
Date: Tue Oct 31 07:56:33 2006
New Revision: 46650
URL: http://svn.digium.com/view/asterisk?rev=46650&view=rev
Log:
Set #define for TIMER T1 value
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=46650&r1=46649&r2=46650&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Oct 31 07:56:33 2006
@@ -195,6 +195,7 @@
#define DEFAULT_RETRANS 1000 /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
#define MAX_RETRANS 6 /*!< Try only 6 times for retransmissions, a total of 7 transmissions */
+#define SIP_TIMER_T1 500 /* SIP timer T1 (according to RFC 3261) */
#define SIP_TRANS_TIMEOUT 32000 /*!< SIP request timeout (rfc 3261) 64*T1
\todo Use known T1 for timeout (peerpoke)
*/
@@ -2003,7 +2004,7 @@
{
if (ms < 0) {
if (p->timer_t1 == 0)
- p->timer_t1 = 500; /* Set timer T1 if not set (RFC 3261) */
+ p->timer_t1 = SIP_TIMER_T1; /* Set timer T1 if not set (RFC 3261) */
ms = p->timer_t1 * 64;
}
if (sip_debug_test_pvt(p))
@@ -2751,7 +2752,7 @@
if (port)
*port++ = '\0';
dialog->sa.sin_family = AF_INET;
- dialog->timer_t1 = 500; /* Default SIP retransmission timer T1 (RFC 3261) */
+ dialog->timer_t1 = SIP_TIMER_T1; /* Default SIP retransmission timer T1 (RFC 3261) */
peer = find_peer(peername, NULL, 1);
if (peer) {
@@ -4199,7 +4200,7 @@
p->prefs = default_prefs; /* Set default codecs for this call */
if (intended_method != SIP_OPTIONS) /* Peerpoke has it's own system */
- p->timer_t1 = 500; /* Default SIP retransmission timer T1 (RFC 3261) */
+ p->timer_t1 = SIP_TIMER_T1; /* Default SIP retransmission timer T1 (RFC 3261) */
if (sin) {
p->sa = *sin;
More information about the asterisk-commits
mailing list