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Timout or error on INFO or MESSAGE transaction causes call to be lost.

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When exchanging INFO messages within a call, 4xx error causes the call to
be disconnected although RFC 2976 explicitly states that such transactions
do not modify the state of the dialog.

When exchanging MESSAGE messages within a call, 4xx error causes the call
to be disconnected.  To provide least surprise, we should not disconnect
the call since a MESSAGE is like INFO in this case.  (Implied by RFC 3428
Section 2)

(closes issue ASTERISK-17901)
Reported by: neutrino88

Review: https://reviewboard.asterisk.org/r/1257/
Review: https://reviewboard.asterisk.org/r/1258/

JIRA SWP-3486


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Richard Mudgett
2011-06-22 18:41:20 +00:00
parent f5e0f04c19
commit 9de3aa9c60
1 changed files with 86 additions and 12 deletions
Download Patch File Download Diff File Expand all files Collapse all files

98
channels/chan_sip.c
View File

@@ -20100,6 +20100,83 @@ static void handle_response_peerpoke(struct sip_pvt *p, int resp, struct sip_req
ref_peer(peer, "adding poke peer ref"));
}
/*!
* \internal
* \brief Handle responses to INFO messages
*
* \note The INFO method MUST NOT change the state of calls or
* related sessions (RFC 2976).
*/
static void handle_response_info(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno)
{
int sipmethod = SIP_INFO;
switch (resp) {
case 401: /* Not www-authorized on SIP method */
case 407: /* Proxy auth required */
ast_log(LOG_WARNING, "Host '%s' requests authentication (%d) for '%s'\n",
ast_sockaddr_stringify(&p->sa), resp, sip_methods[sipmethod].text);
break;
case 405: /* Method not allowed */
case 501: /* Not Implemented */
mark_method_unallowed(&p->allowed_methods, sipmethod);
if (p->relatedpeer) {
mark_method_allowed(&p->relatedpeer->disallowed_methods, sipmethod);
}
ast_log(LOG_WARNING, "Host '%s' does not implement '%s'\n",
ast_sockaddr_stringify(&p->sa), sip_methods[sipmethod].text);
break;
default:
if (300 <= resp && resp < 700) {
ast_verb(3, "Got SIP %s response %d \"%s\" back from host '%s'\n",
sip_methods[sipmethod].text, resp, rest, ast_sockaddr_stringify(&p->sa));
}
break;
}
}
/*!
* \internal
* \brief Handle responses to MESSAGE messages
*
* \note The MESSAGE method should not change the state of calls
* or related sessions if associated with a dialog. (Implied by
* RFC 3428 Section 2).
*/
static void handle_response_message(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, int seqno)
{
int sipmethod = SIP_MESSAGE;
/* Out-of-dialog MESSAGE currently not supported. */
//int in_dialog = ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
switch (resp) {
case 401: /* Not www-authorized on SIP method */
case 407: /* Proxy auth required */
ast_log(LOG_WARNING, "Host '%s' requests authentication (%d) for '%s'\n",
ast_sockaddr_stringify(&p->sa), resp, sip_methods[sipmethod].text);
break;
case 405: /* Method not allowed */
case 501: /* Not Implemented */
mark_method_unallowed(&p->allowed_methods, sipmethod);
if (p->relatedpeer) {
mark_method_allowed(&p->relatedpeer->disallowed_methods, sipmethod);
}
ast_log(LOG_WARNING, "Host '%s' does not implement '%s'\n",
ast_sockaddr_stringify(&p->sa), sip_methods[sipmethod].text);
break;
default:
if (100 <= resp && resp < 200) {
/* Must allow provisional responses for out-of-dialog requests. */
} else if (200 <= resp && resp < 300) {
p->authtries = 0; /* Reset authentication counter */
} else if (300 <= resp && resp < 700) {
ast_verb(3, "Got SIP %s response %d \"%s\" back from host '%s'\n",
sip_methods[sipmethod].text, resp, rest, ast_sockaddr_stringify(&p->sa));
}
break;
}
}
/*! \brief Immediately stop RTP, VRTP and UDPTL as applicable */
static void stop_media_flows(struct sip_pvt *p)
{
@@ -20220,6 +20297,12 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
* we just always call the response handler. Good gravy!
*/
handle_response_publish(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_INFO) {
/* More good gravy! */
handle_response_info(p, resp, rest, req, seqno);
} else if (sipmethod == SIP_MESSAGE) {
/* More good gravy! */
handle_response_message(p, resp, rest, req, seqno);
} else if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
switch(resp) {
case 100: /* 100 Trying */
@@ -20233,11 +20316,7 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
break;
case 200: /* 200 OK */
p->authtries = 0; /* Reset authentication counter */
if (sipmethod == SIP_MESSAGE || sipmethod == SIP_INFO) {
/* We successfully transmitted a message
or a video update request in INFO */
/* Nothing happens here - the message is inside a dialog */
} else if (sipmethod == SIP_INVITE) {
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);
@@ -20429,15 +20508,14 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
break;
default:
/* Send hangup */
if (owner && sipmethod != SIP_MESSAGE && sipmethod != SIP_INFO && sipmethod != SIP_BYE)
if (owner && sipmethod != SIP_BYE)
ast_queue_hangup_with_cause(p->owner, AST_CAUSE_PROTOCOL_ERROR);
break;
}
/* ACK on invite */
if (sipmethod == SIP_INVITE)
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
if (sipmethod != SIP_MESSAGE && sipmethod != SIP_INFO)
sip_alreadygone(p);
sip_alreadygone(p);
if (!p->owner) {
pvt_set_needdestroy(p, "transaction completed");
}
@@ -20498,10 +20576,6 @@ static void handle_response(struct sip_pvt *p, int resp, const char *rest, struc
}
} else if (sipmethod == SIP_BYE) {
pvt_set_needdestroy(p, "transaction completed");
} else if (sipmethod == SIP_MESSAGE || sipmethod == SIP_INFO) {
/* We successfully transmitted a message or
a video update request in INFO */
;
}
break;
case 401: /* www-auth */