[svn-commits] oej: trunk r47740 - /trunk/channels/chan_sip.c
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Thu Nov 16 09:02:41 MST 2006
Author: oej
Date: Thu Nov 16 10:02:41 2006
New Revision: 47740
URL: http://svn.digium.com/view/asterisk?view=rev&rev=47740
Log:
Merging implementation of invite states from my "invitestate" branch for 1.2. This is a bit more
clean platform for the handling of BYE/CANCEL than what we had. It might also need to changes
in other parts of the code, since we know the state of the INVITE transaction.
Please observe that this is is not dialog states at all, this is INVITE transaction states.
Hello Michael Proctor, and thank you! :-)
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=47740&r1=47739&r2=47740
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Nov 16 10:02:41 2006
@@ -238,6 +238,21 @@
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, /*!< Invite sent, no answer */
+ INV_PROCEEDING, /*!< We got 1xx message */
+ INV_EARLY_MEDIA, /*!< We got 18x message with to-tag back */
+ INV_COMPLETED, /*!< Got final response with error. Wait for ACK, then CONFIRMED */
+ INV_CONFIRMED, /*!< Confirmed response - we've got an ack (Incoming calls only) */
+ INV_TERMINATED, /*!< 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 /*!< 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
@@ -699,7 +714,7 @@
#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" */
@@ -870,6 +885,7 @@
/*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe */
struct sip_pvt {
ast_mutex_t pvt_lock; /*!< Dialog private lock */
+ enum invitestates invitestate; /*!< Track state of SIP_INVITEs */
int method; /*!< SIP method that opened this dialog */
AST_DECLARE_STRING_FIELDS(
AST_STRING_FIELD(callid); /*!< Global CallID */
@@ -2915,6 +2931,7 @@
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, SIP_TRANS_TIMEOUT, auto_congest, p);
@@ -3417,7 +3434,8 @@
__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 {
@@ -7467,6 +7485,9 @@
{
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);
@@ -11646,7 +11667,7 @@
{
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. */
@@ -11697,6 +11718,15 @@
if (resp > 100 && resp < 200 && 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 */
if (!ast_test_flag(req, SIP_PKT_IGNORE))
@@ -11714,13 +11744,13 @@
}
}
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;
@@ -11729,13 +11759,13 @@
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;
@@ -11833,8 +11863,8 @@
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;
@@ -13441,6 +13471,7 @@
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;
@@ -13450,6 +13481,7 @@
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
@@ -13457,6 +13489,7 @@
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
@@ -13493,6 +13526,7 @@
ast_setstate(c, AST_STATE_DOWN);
c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
}
+ p->invitestate = INV_COMPLETED;
ast_hangup(c);
sip_pvt_lock(p);
c = NULL;
@@ -13500,9 +13534,11 @@
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)
@@ -13588,6 +13624,7 @@
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);
@@ -13608,6 +13645,7 @@
transmit_response(p, msg, req);
else
transmit_response_reliable(p, msg, req);
+ p->invitestate = INV_COMPLETED;
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}
}
@@ -14063,6 +14101,7 @@
check_via(p, req);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+ p->invitestate = INV_CANCELLED;
if (p->owner && p->owner->_state == AST_STATE_UP) {
/* This call is up, cancel is ignored, we need a bye */
@@ -14095,8 +14134,10 @@
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);
if (sipdebug && option_debug)
@@ -14667,6 +14708,7 @@
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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