[svn-commits] trunk r8926 - in /trunk: channels/chan_sip.c
configs/sip.conf.sample
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue Jan 31 14:20:57 MST 2006
Author: oej
Date: Mon Jan 30 13:50:39 2006
New Revision: 8926
URL: http://svn.digium.com/view/asterisk?rev=8926&view=rev
Log:
Issue 5892: Set a minimum T1 timer for calls. Reporter: twisted
Modified:
trunk/channels/chan_sip.c
trunk/configs/sip.conf.sample
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=8926&r1=8925&r2=8926&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Jan 30 13:50:39 2006
@@ -363,6 +363,7 @@
#define DEFAULT_PEDANTIC FALSE
#define DEFAULT_AUTOCREATEPEER FALSE
#define DEFAULT_QUALIFY FALSE
+#define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
#ifndef DEFAULT_USERAGENT
#define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
#endif
@@ -403,6 +404,7 @@
static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
static int allow_external_domains; /*!< Accept calls to external SIP domains? */
static int global_callevents; /*!< Whether we send manager events or not */
+static int global_t1min; /*!< T1 roundtrip time minimum */
/*! \brief Codecs that we support by default: */
static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
@@ -1915,8 +1917,9 @@
r->callgroup = peer->callgroup;
r->pickupgroup = peer->pickupgroup;
/* Set timer T1 to RTT for this peer (if known by qualify=) */
+ /* Minimum is settable or default to 100 ms */
if (peer->maxms && peer->lastms)
- r->timer_t1 = peer->lastms;
+ r->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
if ((ast_test_flag(r, SIP_DTMF) == SIP_DTMF_RFC2833) || (ast_test_flag(r, SIP_DTMF) == SIP_DTMF_AUTO))
r->noncodeccapability |= AST_RTP_DTMF;
else
@@ -8289,6 +8292,7 @@
ast_cli(fd, " Codecs: ");
print_codec_to_cli(fd, &prefs);
ast_cli(fd, "\n");
+ ast_cli(fd, " T1 minimum: %d\n", global_t1min);
ast_cli(fd, " Relax DTMF: %s\n", global_relaxdtmf ? "Yes" : "No");
ast_cli(fd, " Compact SIP headers: %s\n", compactheaders ? "Yes" : "No");
ast_cli(fd, " RTP Timeout: %d %s\n", global_rtptimeout, global_rtptimeout ? "" : "(Disabled)" );
@@ -12403,6 +12407,7 @@
/* Misc settings for the channel */
global_relaxdtmf = FALSE;
global_callevents = FALSE;
+ global_t1min = DEFAULT_T1MIN;
/* Read the [general] config section of sip.conf (or from realtime config) */
for (v = ast_variable_browse(cfg, "general"); v; v = v->next) {
@@ -12423,6 +12428,8 @@
ast_set2_flag((&global_flags_page2), ast_true(v->value), SIP_PAGE2_RTUPDATE);
} else if (!strcasecmp(v->name, "ignoreregexpire")) {
ast_set2_flag((&global_flags_page2), ast_true(v->value), SIP_PAGE2_IGNOREREGEXPIRE);
+ } else if (!strcasecmp(v->name, "t1min")) {
+ global_t1min = atoi(v->value);
} else if (!strcasecmp(v->name, "rtautoclear")) {
int i = atoi(v->value);
if (i > 0)
Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?rev=8926&r1=8925&r2=8926&view=diff
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Mon Jan 30 13:50:39 2006
@@ -61,6 +61,8 @@
;maxexpiry=3600 ; Max length of incoming registrations/subscriptions we allow (seconds)
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outoing registration
+;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
+ ; Defaults to 100 ms
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
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