fixes sip CANCEL race condition

If Asterisk sends a 4xx error and the other side sends a CANCEl
before receiving the 4xx and responding with the ACK, Asterisk
will process the CANCEL and send a 487 Request Terminated as
a new final response to the INVITE.  Since we are issuing a new
final response to the INVITE, the old one must be pretend_acked
else it will keep retransmitting.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
David Vossel
2010-07-20 21:41:21 +00:00
parent c149b851b3
commit c26791d5f8

View File

@@ -22131,20 +22131,10 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
/*! \brief Handle incoming CANCEL request */
static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
{
check_via(p, req);
sip_alreadygone(p);
/* At this point, we could have cancelled the invite at the same time
as the other side sends a CANCEL. Our final reply with error code
might not have been received by the other side before the CANCEL
was sent, so let's just give up retransmissions and waiting for
ACK on our error code. The call is hanging up any way. */
if (p->invitestate == INV_TERMINATED)
__sip_pretend_ack(p);
else
p->invitestate = INV_CANCELLED;
if (p->owner && p->owner->_state == AST_STATE_UP) {
/* This call is up, cancel is ignored, we need a bye */
transmit_response(p, "200 OK", req);
@@ -22152,6 +22142,17 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
return 0;
}
/* At this point, we could have cancelled the invite at the same time
as the other side sends a CANCEL. Our final reply with error code
might not have been received by the other side before the CANCEL
was sent, so let's just give up retransmissions and waiting for
ACK on our error code. The call is hanging up any way. */
if (p->invitestate == INV_TERMINATED || p->invitestate == INV_COMPLETED) {
__sip_pretend_ack(p);
}
if (p->invitestate != INV_TERMINATED)
p->invitestate = INV_CANCELLED;
if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD))
update_call_counter(p, DEC_CALL_LIMIT);