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Merging the invitestate-1.4 branch after successful testing.

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Will check if I can solve this with less changes in 1.2.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
Olle Johansson 19 years ago
parent 7945d4ca35
commit 694205de93
1 changed files with 75 additions and 21 deletions
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96
channels/chan_sip.c
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@ -244,6 +244,21 @@ enum sip_result {
AST_FAILURE = -1,
};
/*! \brief States for the INVITE transaction, not the dialog
\note this is for the INVITE that sets up the dialog
*/
enum invitestates {
INV_NONE = 0, /*!< No state at all, maybe not an INVITE dialog */
INV_CALLING = 1, /*!< Invite sent, no answer */
INV_PROCEEDING = 2, /*!< We got/sent 1xx message */
INV_EARLY_MEDIA = 3, /*!< We got 18x message with to-tag back */
INV_COMPLETED = 4, /*!< Got final response with error. Wait for ACK, then CONFIRMED */
INV_CONFIRMED = 5, /*!< Confirmed response - we've got an ack (Incoming calls only) */
INV_TERMINATED = 6, /*!< Transaction done - either successful (AST_STATE_UP) or failed, but done
The only way out of this is a BYE from one side */
INV_CANCELLED = 7, /*!< Transaction cancelled by client or server in non-terminated state */
};
/* Do _NOT_ make any changes to this enum, or the array following it;
if you think you are doing the right thing, you are probably
not doing the right thing. If you think there are changes
@ -703,7 +718,7 @@ struct sip_auth {
#define SIP_REALTIME (1 << 11) /*!< Flag for realtime users */
#define SIP_USECLIENTCODE (1 << 12) /*!< Trust X-ClientCode info message */
#define SIP_OUTGOING (1 << 13) /*!< Direction of the last transaction in this dialog */
#define SIP_CAN_BYE (1 << 14) /*!< Can we send BYE on this dialog? */
#define SIP_FREE_BIT (1 << 14) /*!< ---- */
#define SIP_DEFER_BYE_ON_TRANSFER (1 << 15) /*!< Do not hangup at first ast_hangup */
#define SIP_DTMF (3 << 16) /*!< DTMF Support: four settings, uses two bits */
#define SIP_DTMF_RFC2833 (0 << 16) /*!< DTMF Support: RTP DTMF - "rfc2833" */
@ -877,6 +892,7 @@ struct sip_refer {
static struct sip_pvt {
ast_mutex_t lock; /*!< Dialog private lock */
int method; /*!< SIP method that opened this dialog */
enum invitestates invitestate; /*!< The state of the INVITE transaction only */
AST_DECLARE_STRING_FIELDS(
AST_STRING_FIELD(callid); /*!< Global CallID */
AST_STRING_FIELD(randdata); /*!< Random data */
@ -1593,6 +1609,13 @@ static void initialize_initreq(struct sip_pvt *p, struct sip_request *req)
ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
}
static void sip_alreadygone(struct sip_pvt *dialog)
{
if (option_debug > 2)
ast_log(LOG_DEBUG, "Setting SIP_ALREADYGONE on dialog %s\n", dialog->callid);
ast_set_flag(&dialog->flags[0], SIP_ALREADYGONE);
}
/*! \brief returns true if 'name' (with optional trailing whitespace)
* matches the sip method 'id'.
@ -1871,7 +1894,7 @@ static int retrans_pkt(void *data)
ast_mutex_lock(&pkt->owner->lock);
}
if (pkt->owner->owner) {
ast_set_flag(&pkt->owner->flags[0], SIP_ALREADYGONE);
sip_alreadygone(pkt->owner);
ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet.\n", pkt->owner->callid);
ast_queue_hangup(pkt->owner->owner);
ast_channel_unlock(pkt->owner->owner);
@ -2802,6 +2825,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
if (option_debug > 1)
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);
p->invitestate = INV_CALLING;
/* Initialize auto-congest time */
p->initid = ast_sched_add(sched, p->maxtime ? (p->maxtime * 4) : SIP_TRANS_TIMEOUT, auto_congest, p);
@ -3269,7 +3293,7 @@ static int sip_hangup(struct ast_channel *ast)
return 0;
}
/* If the call is not UP, we need to send CANCEL instead of BYE */
if (ast->_state == AST_STATE_RING || ast->_state == AST_STATE_RINGING) {
if (ast->_state == AST_STATE_RING || ast->_state == AST_STATE_RINGING || p->invitestate < INV_COMPLETED) {
needcancel = TRUE;
if (option_debug > 3)
ast_log(LOG_DEBUG, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast->_state));
@ -3293,7 +3317,7 @@ static int sip_hangup(struct ast_channel *ast)
*/
if (ast_test_flag(&p->flags[0], SIP_ALREADYGONE))
needdestroy = 1; /* Set destroy flag at end of this function */
else
else if (p->invitestate != INV_CALLING)
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
/* Start the process if it's not already started */
@ -3304,7 +3328,8 @@ static int sip_hangup(struct ast_channel *ast)
__sip_pretend_ack(p);
/* if we can't send right now, mark it pending */
if (!ast_test_flag(&p->flags[0], SIP_CAN_BYE)) {
if (p->invitestate == INV_CALLING) {
/* We can't send anything in CALLING state */
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
/* Do we need a timer here if we don't hear from them at all? */
} else {
@ -3354,6 +3379,7 @@ static int sip_hangup(struct ast_channel *ast)
but we can't send one while we have "INVITE" outstanding. */
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);
sip_cancel_destroy(p);
}
}
}
@ -3608,6 +3634,7 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
switch(condition) {
case AST_CONTROL_RINGING:
if (ast->_state == AST_STATE_RING) {
p->invitestate = INV_EARLY_MEDIA;
if (!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) ||
(ast_test_flag(&p->flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER)) {
/* Send 180 ringing if out-of-band seems reasonable */
@ -3624,7 +3651,8 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
case AST_CONTROL_BUSY:
if (ast->_state != AST_STATE_UP) {
transmit_response(p, "486 Busy Here", &p->initreq);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
p->invitestate = INV_TERMINATED;
sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
break;
}
@ -3633,7 +3661,8 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
case AST_CONTROL_CONGESTION:
if (ast->_state != AST_STATE_UP) {
transmit_response(p, "503 Service Unavailable", &p->initreq);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
p->invitestate = INV_TERMINATED;
sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
break;
}
@ -3644,6 +3673,7 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
transmit_response(p, "100 Trying", &p->initreq);
p->invitestate = INV_PROCEEDING;
break;
}
res = -1;
@ -3652,6 +3682,7 @@ static int sip_indicate(struct ast_channel *ast, int condition, const void *data
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
p->invitestate = INV_EARLY_MEDIA;
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
break;
@ -7374,6 +7405,9 @@ static int transmit_request(struct sip_pvt *p, int sipmethod, int seqno, enum xm
{
struct sip_request resp;
if (sipmethod == SIP_ACK)
p->invitestate = INV_CONFIRMED;
reqprep(&resp, p, sipmethod, seqno, newbranch);
add_header_contentLength(&resp, 0);
return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);
@ -11462,7 +11496,7 @@ static void check_pendings(struct sip_pvt *p)
{
if (ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
/* if we can't BYE, then this is really a pending CANCEL */
if (!ast_test_flag(&p->flags[0], SIP_CAN_BYE))
if (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA)
transmit_request(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, FALSE);
/* 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. */
@ -11513,6 +11547,15 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 183)
resp = 183;
/* Any response between 100 and 199 is PROCEEDING */
if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING)
p->invitestate = INV_PROCEEDING;
/* Final response, not 200 ? */
if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA ))
p->invitestate = INV_COMPLETED;
switch (resp) {
case 100: /* Trying */
case 101: /* Dialog establishment */
@ -11531,13 +11574,13 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
}
}
if (find_sdp(req)) {
p->invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req);
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
/* Queue a progress frame only if we have SDP in 180 */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
}
ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
@ -11546,13 +11589,13 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
sip_cancel_destroy(p);
/* Ignore 183 Session progress without SDP */
if (find_sdp(req)) {
p->invitestate = INV_EARLY_MEDIA;
res = process_sdp(p, req);
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
/* Queue a progress frame */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
}
ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
@ -11653,8 +11696,8 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
}
/* If I understand this right, the branch is different for a non-200 ACK only */
p->invitestate = INV_TERMINATED;
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
case 407: /* Proxy authentication */
@ -11672,7 +11715,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, authenticate, authorization, SIP_INVITE, 1)) {
ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", get_header(&p->initreq, "From"));
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
sip_alreadygone(p);
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
}
@ -11686,14 +11729,14 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
sip_alreadygone(p);
break;
case 404: /* Not found */
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
sip_alreadygone(p);
break;
case 481: /* Call leg does not exist */
@ -12150,7 +12193,6 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
/* Fatal response */
if ((option_verbose > 2) && (resp != 487))
ast_verbose(VERBOSE_PREFIX_3 "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_inet_ntoa(p->sa.sin_addr));
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
@ -12209,7 +12251,7 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
/* ACK on invite */
if (sipmethod == SIP_INVITE)
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
sip_alreadygone(p);
if (!p->owner)
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
} else if ((resp >= 100) && (resp < 200)) {
@ -13254,6 +13296,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
if (option_debug > 1)
ast_log(LOG_DEBUG, "%s: New call is still down.... Trying... \n", c->name);
transmit_response(p, "100 Trying", req);
p->invitestate = INV_PROCEEDING;
ast_setstate(c, AST_STATE_RING);
if (strcmp(p->exten, ast_pickup_ext())) { /* Call to extension -start pbx on this call */
enum ast_pbx_result res;
@ -13263,6 +13306,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
switch(res) {
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
p->invitestate = INV_COMPLETED;
if (ast_test_flag(req, SIP_PKT_IGNORE))
transmit_response(p, "503 Unavailable", req);
else
@ -13270,6 +13314,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
break;
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
p->invitestate = INV_COMPLETED;
if (ast_test_flag(req, SIP_PKT_IGNORE))
transmit_response(p, "480 Temporarily Unavailable", req);
else
@ -13297,7 +13342,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
transmit_response(p, "503 Unavailable", req); /* OEJ - Right answer? */
else
transmit_response_reliable(p, "503 Unavailable", req);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
sip_alreadygone(p);
/* Unlock locks so ast_hangup can do its magic */
ast_mutex_unlock(&p->lock);
c->hangupcause = AST_CAUSE_CALL_REJECTED;
@ -13306,6 +13351,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
ast_setstate(c, AST_STATE_DOWN);
c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
}
p->invitestate = INV_COMPLETED;
ast_hangup(c);
ast_mutex_lock(&p->lock);
c = NULL;
@ -13313,9 +13359,11 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
break;
case AST_STATE_RING:
transmit_response(p, "100 Trying", req);
p->invitestate = INV_PROCEEDING;
break;
case AST_STATE_RINGING:
transmit_response(p, "180 Ringing", req);
p->invitestate = INV_PROCEEDING;
break;
case AST_STATE_UP:
if (option_debug > 1)
@ -13401,6 +13449,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
transmit_response_with_sdp(p, "200 OK", req, XMIT_CRITICAL);
}
p->invitestate = INV_TERMINATED;
break;
default:
ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", c->_state);
@ -13421,6 +13470,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
transmit_response(p, msg, req);
else
transmit_response_reliable(p, msg, req);
p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
}
@ -13616,7 +13666,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
transmit_response(p, "603 Declined (No dialog)", req);
if (!ast_test_flag(req, SIP_PKT_IGNORE)) {
append_history(p, "Xfer", "Refer failed. Outside of dialog.");
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
sip_alreadygone(p);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
}
return 0;
@ -13875,7 +13925,8 @@ static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req)
{
check_via(p, req);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
sip_alreadygone(p);
p->invitestate = INV_CANCELLED;
if (p->owner && p->owner->_state == AST_STATE_UP) {
/* This call is up, cancel is ignored, we need a bye */
@ -13908,12 +13959,14 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
struct ast_channel *bridged_to;
/* If we have an INCOMING invite that we haven't answered, terminate that transaction */
if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !ast_test_flag(req, SIP_PKT_IGNORE) && !p->owner)
if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !ast_test_flag(req, SIP_PKT_IGNORE) && !p->owner)
transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
p->invitestate = INV_TERMINATED;
copy_request(&p->initreq, req);
check_via(p, req);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
sip_alreadygone(p);
/* Get RTCP quality before end of call */
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY) || p->owner) {
@ -14479,6 +14532,7 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
case SIP_ACK:
/* Make sure we don't ignore this */
if (seqno == p->pendinginvite) {
p->invitestate = INV_CONFIRMED;
p->pendinginvite = 0;
__sip_ack(p, seqno, FLAG_RESPONSE, 0);
if (find_sdp(req)) {

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