[asterisk-commits] oej: trunk r42537 - /trunk/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Sat Sep 9 05:44:37 MST 2006
Author: oej
Date: Sat Sep 9 07:44:36 2006
New Revision: 42537
URL: http://svn.digium.com/view/asterisk?rev=42537&view=rev
Log:
Don't destruct sessions prematurely. Especially not when we want reliable retransmissions...
If this works properly, we might have to check 1.2 to implement a backport.
The theory is that if you get a final reply in a session, it is ok to destroy the session.
If you send a final reply, you need to keep the session open for potential retransmits
from the other side. If you send a HANGUP/CANCEL, wait to the other side confirms
or until you have a timeout. If you send HANGUP/CANCEL/ACK reliably, don't destroy
the session so that you cancel the needed retransmits.
I will have to change the timer to 64*T1, but that will be a separate patch. That will
mean that if we know the roundtrip time, we can destroy sessions quicker.
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=42537&r1=42536&r2=42537&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Sat Sep 9 07:44:36 2006
@@ -8951,14 +8951,14 @@
if (strcmp(content_type, "text/plain")) { /* No text/plain attachment */
transmit_response(p, "415 Unsupported Media Type", req); /* Good enough, or? */
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_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);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
return;
}
@@ -8977,7 +8977,7 @@
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? */
}
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
return;
}
@@ -10381,7 +10381,7 @@
if (!p->owner) { /* not a PBX call */
transmit_response(p, "481 Call leg/transaction does not exist", req);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
return;
}
@@ -11330,7 +11330,8 @@
} else if (p->t38.state == T38_DISABLED && bridgepeer && (bridgepvt->t38.state == T38_ENABLED)) {
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 */
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ /* XXXX Should we really destroy this session here, without any response at all??? */
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
}
} else {
if (option_debug > 1)
@@ -12338,7 +12339,7 @@
/* Here's room to implement incoming voicemail notifications :-) */
transmit_response(p, "489 Bad event", req);
if (!p->lastinvite)
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
return -1;
} else {
/* Save nesting depth for now, since there might be other events we will
@@ -12361,7 +12362,7 @@
if (strncasecmp(get_header(req, "Content-Type"), "message/sipfrag", strlen("message/sipfrag"))) {
/* We need a sipfrag */
transmit_response(p, "400 Bad request", req);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
return -1;
}
@@ -12369,7 +12370,7 @@
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);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
return -1;
}
@@ -12461,7 +12462,7 @@
/* Destroy if this OPTIONS was the opening request, but not if
it's in the middle of a normal call flow. */
if (!p->lastinvite)
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
return res;
}
@@ -12516,7 +12517,7 @@
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.");
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
ast_mutex_unlock(&p->refer->refer_call->lock);
return 1;
}
@@ -12655,7 +12656,7 @@
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)
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
return -1;
}
}
@@ -12705,7 +12706,7 @@
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.");
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
return -1;
}
@@ -12773,7 +12774,7 @@
if (error) { /* Give up this dialog */
append_history(p, "Xfer", "INVITE/Replace Failed.");
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
ast_mutex_unlock(&p->lock);
if (p->refer->refer_call) {
ast_mutex_unlock(&p->refer->refer_call->lock);
@@ -12806,7 +12807,7 @@
if (process_sdp(p, req)) {
transmit_response(p, "488 Not acceptable here", req);
if (!p->lastinvite)
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
return -1;
}
} else {
@@ -12834,7 +12835,7 @@
ast_log(LOG_NOTICE, "Failed to authenticate user %s\n", get_header(req, "From"));
transmit_response_reliable(p, "403 Forbidden", req);
}
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
ast_string_field_free(p, theirtag);
return 0;
}
@@ -12844,7 +12845,7 @@
if (process_sdp(p, req)) {
/* Unacceptable codecs */
transmit_response_reliable(p, "488 Not acceptable here", req);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
if (option_debug)
ast_log(LOG_DEBUG, "No compatible codecs for this SIP call.\n");
return -1;
@@ -12873,7 +12874,7 @@
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);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
}
return 0;
}
@@ -12892,7 +12893,7 @@
transmit_response_reliable(p, "404 Not Found", req);
update_call_counter(p, DEC_CALL_LIMIT);
}
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
} else {
/* If no extension was specified, use the s one */
/* Basically for calling to IP/Host name only */
@@ -13032,7 +13033,7 @@
transmit_response(p, "488 Not acceptable here", req);
else
transmit_response_reliable(p, "488 Not acceptable here", req);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
}
}
} else {
@@ -13044,7 +13045,7 @@
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>");
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
}
} else {
/* we are not bridged in a call */
@@ -13071,7 +13072,7 @@
transmit_response(p, "488 Not Acceptable Here (unsupported)", req);
else
transmit_response_reliable(p, "488 Not Acceptable Here (unsupported)", req);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
sendok = FALSE;
}
/* No bridged peer with T38 enabled*/
@@ -13101,7 +13102,7 @@
transmit_response(p, msg, req);
else
transmit_response_reliable(p, msg, req);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
}
}
return res;
@@ -13586,7 +13587,7 @@
if (p->owner)
ast_queue_hangup(p->owner);
else
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
if (p->initreq.len > 0) {
transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
transmit_response(p, "200 OK", req);
@@ -13668,7 +13669,7 @@
if (option_debug > 2)
ast_log(LOG_DEBUG, "Received bye, issuing owner hangup\n.");
} else {
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
if (option_debug > 2)
ast_log(LOG_DEBUG, "Received bye, no owner, selfdestruct soon.\n.");
}
@@ -14110,7 +14111,7 @@
/* Will cease to exist after ACK */
} else if (req->method != SIP_ACK) {
transmit_response(p, "481 Call/Transaction Does Not Exist", req);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
}
return res;
}
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