[asterisk-commits] oej: branch oej/invitestate-1.4 r48131 - /team/oej/invitestate-1.4/channels/

asterisk-commits at lists.digium.com asterisk-commits at lists.digium.com
Thu Nov 30 03:30:27 MST 2006


Author: oej
Date: Thu Nov 30 04:30:25 2006
New Revision: 48131

URL: http://svn.digium.com/view/asterisk?view=rev&rev=48131
Log:
Adding the invitestate patch. This now exists embedded
in trunk, in the "invitestate" branch for 1.2 and in this
branch for 1.4. Please, please, test and report results.

There are a lot of bugs open for strange BYE/CANCEL requests
being sent. If you have one of those, please test this
branch a.s.a.p.

Modified:
    team/oej/invitestate-1.4/channels/chan_sip.c

Modified: team/oej/invitestate-1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/invitestate-1.4/channels/chan_sip.c?view=diff&rev=48131&r1=48130&r2=48131
==============================================================================
--- team/oej/invitestate-1.4/channels/chan_sip.c (original)
+++ team/oej/invitestate-1.4/channels/chan_sip.c Thu Nov 30 04:30:25 2006
@@ -244,6 +244,21 @@
 	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
@@ -702,7 +717,7 @@
 #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" */
@@ -876,6 +891,7 @@
 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 */
@@ -2801,6 +2817,7 @@
 			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, p->maxtime ? (p->maxtime * 4) : SIP_TRANS_TIMEOUT, auto_congest, p);
@@ -3302,7 +3319,8 @@
 				__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 {
@@ -7371,6 +7389,9 @@
 {
 	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);
@@ -11457,7 +11478,7 @@
 {
 	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. */
@@ -11508,6 +11529,15 @@
 	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 */
@@ -11526,13 +11556,13 @@
 			}
 		}
 		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;
 
@@ -11541,13 +11571,13 @@
 			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;
 
@@ -11645,8 +11675,8 @@
 				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;
 	case 407: /* Proxy authentication */
@@ -13246,6 +13276,7 @@
 			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;
@@ -13255,6 +13286,7 @@
 				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
@@ -13262,6 +13294,7 @@
 					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
@@ -13298,6 +13331,7 @@
 					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;
@@ -13305,9 +13339,11 @@
 			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)
@@ -13393,6 +13429,7 @@
 					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);
@@ -13413,6 +13450,7 @@
 				transmit_response(p, msg, req);
 			else
 				transmit_response_reliable(p, msg, req);
+			p->invitestate = INV_COMPLETED;
 			sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
 		}
 	}
@@ -13868,6 +13906,7 @@
 		
 	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 */
@@ -13900,8 +13939,10 @@
 	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);
 	check_via(p, req);
@@ -14471,6 +14512,7 @@
 	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)) {



More information about the asterisk-commits mailing list