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Fix RTP reference leak.

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If a blind transfer were initiated using a REFER without a prior
reINVITE to place the call on hold, AND if Asterisk were sending
RTCP reports, then there was a reference for the RTP instance
of the transferer.

This fixes the issue by merging two similar but slightly conflicting
sections of code into a single area. It also adds a stop_media_flows()
call in the case that the transferer's UA never sends a BYE to us
like it is supposed to.

(issue ASTERISK-19192)

Review: https://reviewboard.asterisk.org/r/1681/



git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352014 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Michelson
2012-01-21 00:04:13 +00:00
parent 2cefb60505
commit 27d894d624
1 changed files with 20 additions and 32 deletions
Download Patch File Download Diff File Expand all files Collapse all files

52
channels/chan_sip.c
View File

@@ -3860,6 +3860,7 @@ static int __sip_autodestruct(const void *data)
ast_channel_unref(owner);
} else if (p->refer && !p->alreadygone) {
ast_debug(3, "Finally hanging up channel after transfer: %s\n", p->callid);
stop_media_flows(p);
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
append_history(p, "ReferBYE", "Sending BYE on transferer call leg %s", p->callid);
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
@@ -20262,15 +20263,22 @@ static void handle_response_notify(struct sip_pvt *p, int resp, const char *rest
case 200: /* Notify accepted */
/* They got the notify, this is the end */
if (p->owner) {
if (!p->refer) {
ast_log(LOG_WARNING, "Notify answer on an owned channel? - %s\n", p->owner->name);
ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_UNSPECIFIED);
if (p->refer) {
ast_log(LOG_NOTICE, "Got OK on REFER Notify message\n");
} else {
ast_debug(4, "Got OK on REFER Notify message\n");
ast_log(LOG_WARNING, "Notify answer on an owned channel? - %s\n", p->owner->name);
/*
* XXX There is discrepancy on whether a hangup should be queued
* or not. This code used to be duplicated in two places, and the more
* frequently hit area had this disabled, making it the de facto
* "correct" way to go.
*
* ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_UNSPECIFIED);
*/
}
} else {
if (p->subscribed == NONE) {
ast_debug(4, "Got 200 accepted on NOTIFY\n");
if (p->subscribed == NONE && !p->refer) {
ast_debug(4, "Got 200 accepted on NOTIFY %s\n", p->callid);
pvt_set_needdestroy(p, "received 200 response");
}
if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
@@ -20295,6 +20303,9 @@ static void handle_response_notify(struct sip_pvt *p, int resp, const char *rest
pvt_set_needdestroy(p, "failed to authenticate NOTIFY");
}
break;
case 481: /* Call leg does not exist */
pvt_set_needdestroy(p, "Received 481 response for NOTIFY");
break;
}
}
@@ -20889,6 +20900,9 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
} else if (sipmethod == SIP_MESSAGE) {
/* More good gravy! */
handle_response_message(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_NOTIFY) {
/* The gravy train continues to roll */
handle_response_notify(p, resp, rest, req, seqno);
} else if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
switch(resp) {
case 100: /* 100 Trying */
@@ -20904,8 +20918,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
p->authtries = 0; /* Reset authentication counter */
if (sipmethod == SIP_INVITE) {
handle_response_invite(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_NOTIFY) {
handle_response_notify(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_REGISTER) {
handle_response_register(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_SUBSCRIBE) {
@@ -20920,8 +20932,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
case 407: /* Proxy auth required */
if (sipmethod == SIP_INVITE)
handle_response_invite(p, resp, rest, req, seqno);
else if (sipmethod == SIP_NOTIFY)
handle_response_notify(p, resp, rest, req, seqno);
else if (sipmethod == SIP_SUBSCRIBE)
handle_response_subscribe(p, resp, rest, req, seqno);
else if (p->registry && sipmethod == SIP_REGISTER)
@@ -20997,8 +21007,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
handle_response_invite(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_SUBSCRIBE) {
handle_response_subscribe(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_NOTIFY) {
pvt_set_needdestroy(p, "received 481 response");
} else if (sipmethod == SIP_BYE) {
/* The other side has no transaction to bye,
just assume it's all right then */
@@ -21155,24 +21163,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
ast_debug(1, "Got 200 OK on CANCEL\n");
/* Wait for 487, then destroy */
} else if (sipmethod == SIP_NOTIFY) {
/* They got the notify, this is the end */
if (p->owner) {
if (p->refer) {
ast_debug(1, "Got 200 OK on NOTIFY for transfer\n");
} else
ast_log(LOG_WARNING, "Notify answer on an owned channel?\n");
/* ast_queue_hangup(p->owner); Disabled */
} else {
if (!p->subscribed && !p->refer) {
pvt_set_needdestroy(p, "transaction completed");
}
if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
/* Ready to send the next state we have on queue */
ast_clear_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
cb_extensionstate((char *)p->context, (char *)p->exten, p->laststate, (void *) p);
}
}
} else if (sipmethod == SIP_BYE) {
pvt_set_needdestroy(p, "transaction completed");
}
@@ -21194,8 +21184,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
handle_response_invite(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_BYE) {
pvt_set_needdestroy(p, "received 481 response");
} else if (sipmethod == SIP_NOTIFY) {
pvt_set_needdestroy(p, "received 481 response");
} else if (sipdebug) {
ast_debug(1, "Remote host can't match request %s to call '%s'. Giving up\n", sip_methods[sipmethod].text, p->callid);
}