[asterisk-commits] oej: trunk r42751 - /trunk/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Mon Sep 11 11:06:45 MST 2006
Author: oej
Date: Mon Sep 11 13:06:44 2006
New Revision: 42751
URL: http://svn.digium.com/view/asterisk?rev=42751&view=rev
Log:
Use Timer T1 for dialog timeouts/destruction
(If you have qualify=yes, we will use the actual roundtrip time)
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=42751&r1=42750&r2=42751&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Sep 11 13:06:44 2006
@@ -201,6 +201,7 @@
#define SIP_TRANS_TIMEOUT 32000 /*!< SIP request timeout (rfc 3261) 64*T1
\todo Use known T1 for timeout (peerpoke)
*/
+#define DEFAULT_TRANS_TIMEOUT -1 /* Use default SIP transaction timeout */
#define MAX_AUTHTRIES 3 /*!< Try authentication three times, then fail */
#define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
@@ -1954,6 +1955,11 @@
/*! \brief Schedule destruction of SIP call */
static void sip_scheddestroy(struct sip_pvt *p, int ms)
{
+ if (ms < 0) {
+ if (p->timer_t1 == 0)
+ p->timer_t1 = 500; /* Set timer T1 if not set (RFC 3261) */
+ ms = p->timer_t1 * 64;
+ }
if (sip_debug_test_pvt(p))
ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text);
if (recordhistory)
@@ -2709,9 +2715,8 @@
if (sipdebug && option_debug > 2)
ast_log(LOG_DEBUG, "Call for %s transfered by %s\n", p->username, referer);
snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer);
- } else {
+ } else
snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name);
- }
ast_string_field_set(p, cid_name, buf);
}
if (option_debug)
@@ -2725,12 +2730,11 @@
if (option_debug)
ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
transmit_invite(p, SIP_INVITE, 1, 2);
- if (p->maxtime) {
+ if (p->maxtime)
/* Initialize auto-congest time */
p->initid = ast_sched_add(sched, p->maxtime * 4, auto_congest, p);
- } else {
+ else
p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, p);
- }
}
return res;
}
@@ -3166,7 +3170,7 @@
ast_log(LOG_DEBUG, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid);
if (p->autokillid > -1)
sip_cancel_destroy(p);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Really hang up next time */
ast_clear_flag(&p->flags[0], SIP_NEEDDESTROY);
p->owner->tech_pvt = NULL;
@@ -3220,7 +3224,7 @@
if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE))
needdestroy = 1; /* Set destroy flag at end of this function */
else
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
/* Start the process if it's not already started */
if (!ast_test_flag(&p->flags[0], SIP_ALREADYGONE) && !ast_strlen_zero(p->initreq.data)) {
@@ -3239,7 +3243,7 @@
/* Actually don't destroy us yet, wait for the 487 on our original
INVITE, but do set an autodestruct just in case we never get it. */
needdestroy = 0;
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
if ( p->initid != -1 ) {
/* channel still up - reverse dec of inUse counter
@@ -7716,7 +7720,7 @@
retransmission should get it */
transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, 0);
/* Schedule auto destroy in 32 seconds (according to RFC 3261) */
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
return AUTH_CHALLENGE_SENT;
} else if (ast_strlen_zero(p->randdata) || ast_strlen_zero(authtoken)) {
@@ -7724,7 +7728,7 @@
ast_string_field_build(p, randdata, "%08lx", ast_random()); /* Create nonce for challenge */
transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, 0);
/* Schedule auto destroy in 32 seconds */
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return AUTH_CHALLENGE_SENT;
}
@@ -7810,7 +7814,7 @@
}
/* Schedule auto destroy in 32 seconds */
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return AUTH_CHALLENGE_SENT;
}
if (good_response)
@@ -7819,7 +7823,7 @@
/* Ok, we have a bad username/secret pair */
/* Challenge again, and again, and again */
transmit_response_with_auth(p, response, req, p->randdata, reliable, respheader, 0);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return AUTH_CHALLENGE_SENT;
}
@@ -7856,7 +7860,7 @@
case AST_EXTENSION_REMOVED: /* Extension is gone */
if (p->autokillid > -1)
sip_cancel_destroy(p); /* Remove subscription expiry for renewals */
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT); /* Delete subscription in 32 secs */
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); /* Delete subscription in 32 secs */
ast_verbose(VERBOSE_PREFIX_2 "Extension state: Watcher for hint %s %s. Notify User %s\n", exten, state == AST_EXTENSION_DEACTIVATED ? "deactivated" : "removed", p->username);
p->stateid = -1;
p->subscribed = NONE;
@@ -8951,14 +8955,14 @@
if (strcmp(content_type, "text/plain")) { /* No text/plain attachment */
transmit_response(p, "415 Unsupported Media Type", req); /* Good enough, or? */
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return;
}
if (get_msg_text(buf, sizeof(buf), req)) {
ast_log(LOG_WARNING, "Unable to retrieve text from %s\n", p->callid);
transmit_response(p, "202 Accepted", req);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return;
}
@@ -8977,7 +8981,7 @@
ast_log(LOG_WARNING,"Received message to %s from %s, dropped it...\n Content-Type:%s\n Message: %s\n", get_header(req,"To"), get_header(req,"From"), content_type, buf);
transmit_response(p, "405 Method Not Allowed", req); /* Good enough, or? */
}
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return;
}
@@ -10381,7 +10385,7 @@
if (!p->owner) { /* not a PBX call */
transmit_response(p, "481 Call leg/transaction does not exist", req);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return;
}
@@ -10572,7 +10576,7 @@
build_callid_pvt(p);
ast_cli(fd, "Sending NOTIFY of type '%s' to '%s'\n", argv[2], argv[i]);
transmit_sip_request(p, &req);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
return RESULT_SUCCESS;
@@ -11196,7 +11200,7 @@
else
transmit_request_with_auth(p, SIP_BYE, 0, 1, 1);
ast_clear_flag(&p->flags[0], SIP_PENDINGBYE);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
} else if (ast_test_flag(&p->flags[0], SIP_NEEDREINVITE)) {
if (option_debug)
ast_log(LOG_DEBUG, "Sending pending reinvite on '%s'\n", p->callid);
@@ -11331,7 +11335,7 @@
ast_log(LOG_WARNING, "RTP re-inivte after T38 session not handled yet !\n");
/* Insted of this we should somehow re-invite the other side of the bridge to RTP */
/* XXXX Should we really destroy this session here, without any response at all??? */
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
} else {
if (option_debug > 1)
@@ -11549,7 +11553,7 @@
r->call = NULL;
p->registry = NULL;
/* Let this one hang around until we have all the responses */
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
/* ast_set_flag(&p->flags[0], SIP_NEEDDESTROY); */
/* set us up for re-registering */
@@ -11795,7 +11799,7 @@
ast_log(LOG_WARNING, "INVITE with REPLACEs failed to '%s'\n", get_header(&p->initreq, "From"));
if (owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
} else if (sipmethod == SIP_REFER) {
/* A transfer with Replaces did not work */
/* OEJ: We should Set flag, cancel the REFER, go back
@@ -12339,7 +12343,7 @@
/* Here's room to implement incoming voicemail notifications :-) */
transmit_response(p, "489 Bad event", req);
if (!p->lastinvite)
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
} else {
/* Save nesting depth for now, since there might be other events we will
@@ -12362,7 +12366,7 @@
if (strncasecmp(get_header(req, "Content-Type"), "message/sipfrag", strlen("message/sipfrag"))) {
/* We need a sipfrag */
transmit_response(p, "400 Bad request", req);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
}
@@ -12370,7 +12374,7 @@
if (get_msg_text(buf, sizeof(buf), req)) {
ast_log(LOG_WARNING, "Unable to retrieve attachment from NOTIFY %s\n", p->callid);
transmit_response(p, "400 Bad request", req);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
}
@@ -12462,7 +12466,7 @@
/* Destroy if this OPTIONS was the opening request, but not if
it's in the middle of a normal call flow. */
if (!p->lastinvite)
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return res;
}
@@ -12517,7 +12521,7 @@
ast_log(LOG_ERROR, "Unable to create new channel. Invite/replace failed.\n");
transmit_response_with_sdp(p, "503 Service Unavailable", req, 1);
append_history(p, "Xfer", "INVITE/Replace Failed. No new channel.");
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_mutex_unlock(&p->refer->refer_call->lock);
return 1;
}
@@ -12656,7 +12660,7 @@
transmit_response_with_unsupported(p, "420 Bad extension (unsupported)", req, required);
ast_log(LOG_WARNING,"Received SIP INVITE with unsupported required extension: %s\n", required);
if (!p->lastinvite)
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
}
}
@@ -12706,7 +12710,7 @@
if (!p->refer && !sip_refer_allocate(p)) {
transmit_response(p, "500 Server Internal Error", req);
append_history(p, "Xfer", "INVITE/Replace Failed. Out of memory.");
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
}
@@ -12774,7 +12778,7 @@
if (error) { /* Give up this dialog */
append_history(p, "Xfer", "INVITE/Replace Failed.");
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_mutex_unlock(&p->lock);
if (p->refer->refer_call) {
ast_mutex_unlock(&p->refer->refer_call->lock);
@@ -12807,7 +12811,7 @@
if (process_sdp(p, req)) {
transmit_response(p, "488 Not acceptable here", req);
if (!p->lastinvite)
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return -1;
}
} else {
@@ -12835,7 +12839,7 @@
ast_log(LOG_NOTICE, "Failed to authenticate user %s\n", get_header(req, "From"));
transmit_response_reliable(p, "403 Forbidden", req);
}
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
ast_string_field_free(p, theirtag);
return 0;
}
@@ -12845,7 +12849,7 @@
if (process_sdp(p, req)) {
/* Unacceptable codecs */
transmit_response_reliable(p, "488 Not acceptable here", req);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
if (option_debug)
ast_log(LOG_DEBUG, "No compatible codecs for this SIP call.\n");
return -1;
@@ -12874,7 +12878,7 @@
if (res < 0) {
ast_log(LOG_NOTICE, "Failed to place call for user %s, too many calls\n", p->username);
transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
return 0;
}
@@ -12893,7 +12897,7 @@
transmit_response_reliable(p, "404 Not Found", req);
update_call_counter(p, DEC_CALL_LIMIT);
}
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
} else {
/* If no extension was specified, use the s one */
/* Basically for calling to IP/Host name only */
@@ -13033,7 +13037,7 @@
transmit_response(p, "488 Not acceptable here", req);
else
transmit_response_reliable(p, "488 Not acceptable here", req);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
}
} else {
@@ -13045,7 +13049,7 @@
p->t38.state = T38_DISABLED;
if (option_debug > 1)
ast_log(LOG_DEBUG,"T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
} else {
/* we are not bridged in a call */
@@ -13072,7 +13076,7 @@
transmit_response(p, "488 Not Acceptable Here (unsupported)", req);
else
transmit_response_reliable(p, "488 Not Acceptable Here (unsupported)", req);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
sendok = FALSE;
}
/* No bridged peer with T38 enabled*/
@@ -13102,7 +13106,7 @@
transmit_response(p, msg, req);
else
transmit_response_reliable(p, msg, req);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
}
return res;
@@ -13587,7 +13591,7 @@
if (p->owner)
ast_queue_hangup(p->owner);
else
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
if (p->initreq.len > 0) {
transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
transmit_response(p, "200 OK", req);
@@ -13669,7 +13673,7 @@
if (option_debug > 2)
ast_log(LOG_DEBUG, "Received bye, issuing owner hangup\n.");
} else {
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
if (option_debug > 2)
ast_log(LOG_DEBUG, "Received bye, no owner, selfdestruct soon.\n.");
}
@@ -13966,7 +13970,7 @@
if (res < 1) {
/* Destroy the session, but keep us around for just a bit in case they don't
get our 200 OK */
- sip_scheddestroy(p, 15000);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
append_history(p, "RegRequest", "%s : Account %s", res ? "Failed": "Succeeded", get_header(req, "To"));
return res;
@@ -14111,7 +14115,7 @@
/* Will cease to exist after ACK */
} else if (req->method != SIP_ACK) {
transmit_response(p, "481 Call/Transaction Does Not Exist", req);
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
return res;
}
@@ -14328,7 +14332,7 @@
build_via(p);
build_callid_pvt(p);
/* Destroy this session after 32 secs */
- sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
+ sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
/* Send MWI */
ast_set_flag(&p->flags[0], SIP_OUTGOING);
@@ -14689,8 +14693,8 @@
return res;
}
-/*! \brief PBX interface function -build SIP pvt structure */
-/* SIP calls initiated by the PBX arrive here */
+/*! \brief PBX interface function -build SIP pvt structure
+ SIP calls initiated by the PBX arrive here */
static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause)
{
int oldformat;
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