[asterisk-commits] oej: branch oej/no-premature-183 r215882 - in /team/oej/no-premature-183: ./ ...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Sep 3 01:58:51 CDT 2009


Author: oej
Date: Thu Sep  3 01:58:47 2009
New Revision: 215882

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=215882
Log:
Reset automerge on this branch

Modified:
    team/oej/no-premature-183/   (props changed)
    team/oej/no-premature-183/channels/chan_sip.c
    team/oej/no-premature-183/utils/Makefile

Propchange: team/oej/no-premature-183/
------------------------------------------------------------------------------
    automerge = http://www.codename-pineapple.org/

Propchange: team/oej/no-premature-183/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Thu Sep  3 01:58:47 2009
@@ -1,1 +1,1 @@
-/branches/1.4:1-215283
+/branches/1.4:1-215879

Modified: team/oej/no-premature-183/channels/chan_sip.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/oej/no-premature-183/channels/chan_sip.c?view=diff&rev=215882&r1=215881&r2=215882
==============================================================================
--- team/oej/no-premature-183/channels/chan_sip.c (original)
+++ team/oej/no-premature-183/channels/chan_sip.c Thu Sep  3 01:58:47 2009
@@ -211,6 +211,7 @@
                                                       \todo Use known T1 for timeout (peerpoke)
                                                       */
 #define DEFAULT_TRANS_TIMEOUT        -1               /* Use default SIP transaction timeout */
+#define PROVIS_KEEPALIVE_TIMEOUT     60000            /*!< How long to wait before retransmitting a provisional response (rfc 3261 13.3.1.1) */
 #define MAX_AUTHTRIES                3                /*!< Try authentication three times, then fail */
 
 #define SIP_MAX_HEADERS              64               /*!< Max amount of SIP headers to read */
@@ -1035,6 +1036,8 @@
 	struct ast_variable *chanvars;		/*!< Channel variables to set for inbound call */
 	AST_LIST_HEAD_NOLOCK(request_queue, sip_request) request_queue; /*!< Requests that arrived but could not be processed immediately */
 	int request_queue_sched_id;		/*!< Scheduler ID of any scheduled action to process queued requests */
+	int provisional_keepalive_sched_id; /*!< Scheduler ID for provisional responses that need to be sent out to avoid cancellation */
+	const char *last_provisional;   /*!< The last successfully transmitted provisonal response message */
 	struct sip_pvt *next;			/*!< Next dialog in chain */
 	struct sip_invite_param *options;	/*!< Options for INVITE */
 	int autoframing;
@@ -1291,6 +1294,7 @@
 static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
 static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported);
 static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *rand, enum xmittype reliable, const char *header, int stale);
+static int transmit_provisional_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, int with_sdp);
 static int transmit_response_with_allow(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
 static void transmit_fake_auth_response(struct sip_pvt *p, int sipmethod, struct sip_request *req, enum xmittype reliable);
 static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch);
@@ -2322,6 +2326,46 @@
 	}
 }
 
+static int send_provisional_keepalive_full(struct sip_pvt *pvt, int with_sdp)
+{
+	const char *msg = NULL;
+
+	if (!pvt->last_provisional || !strncasecmp(pvt->last_provisional, "100", 3)) {
+		msg = "183 Session Progress";
+	}
+
+	if (pvt->invitestate < INV_COMPLETED) {
+		if (with_sdp) {
+			transmit_response_with_sdp(pvt, S_OR(msg, pvt->last_provisional), &pvt->initreq, XMIT_UNRELIABLE);
+		} else {
+			transmit_response(pvt, S_OR(msg, pvt->last_provisional), &pvt->initreq);
+		}
+		return PROVIS_KEEPALIVE_TIMEOUT;
+	}
+
+	return 0;
+}
+
+static int send_provisional_keepalive(const void *data) {
+	struct sip_pvt *pvt = (struct sip_pvt *) data;
+
+	return send_provisional_keepalive_full(pvt, 0);
+}
+
+static int send_provisional_keepalive_with_sdp(const void *data) {
+	struct sip_pvt *pvt = (void *)data;
+
+	return send_provisional_keepalive_full(pvt, 1);
+}
+
+static void update_provisional_keepalive(struct sip_pvt *pvt, int with_sdp)
+{
+	AST_SCHED_DEL(sched, pvt->provisional_keepalive_sched_id);
+
+	pvt->provisional_keepalive_sched_id = ast_sched_add(sched, PROVIS_KEEPALIVE_TIMEOUT,
+		with_sdp ? send_provisional_keepalive_with_sdp : send_provisional_keepalive, pvt);
+}
+
 /*! \brief Transmit response on SIP request*/
 static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
 {
@@ -2342,6 +2386,12 @@
 		append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), 
 			(tmp.method == SIP_RESPONSE || tmp.method == SIP_UNKNOWN) ? tmp.rlPart2 : sip_methods[tmp.method].text);
 	}
+
+	/* If we are sending a final response to an INVITE, stop retransmitting provisional responses */
+	if (p->initreq.method == SIP_INVITE && reliable == XMIT_CRITICAL) {
+		AST_SCHED_DEL(sched, p->provisional_keepalive_sched_id);
+	}
+
 	res = (reliable) ?
 		 __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
 		__sip_xmit(p, req->data, req->len);
@@ -3229,6 +3279,7 @@
 	AST_SCHED_DEL(sched, p->waitid);
 	AST_SCHED_DEL(sched, p->autokillid);
 	AST_SCHED_DEL(sched, p->request_queue_sched_id);
+	AST_SCHED_DEL(sched, p->provisional_keepalive_sched_id);
 
 	if (p->rtp) {
 		ast_rtp_destroy(p->rtp);
@@ -3846,7 +3897,7 @@
 					ast_rtp_new_source(p->rtp);
 					if (!global_prematuremediafilter) {
 						p->invitestate = INV_EARLY_MEDIA;
-						transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE);
+						transmit_provisional_response(p, "183 Session Progress", &p->initreq, 1);
 						ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
 					}
 				} else if (p->t38.state == T38_ENABLED && !p->t38.direct) {
@@ -3869,7 +3920,7 @@
 				    !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);
+					transmit_provisional_response(p, "183 Session Progress", &p->initreq, 1);
 					ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);	
 				}
 				p->lastrtptx = time(NULL);
@@ -4040,7 +4091,7 @@
 			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 */
-				transmit_response(p, "180 Ringing", &p->initreq);
+				transmit_provisional_response(p, "180 Ringing", &p->initreq, 0);
 				ast_set_flag(&p->flags[0], SIP_RINGING);
 				if (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) != SIP_PROG_INBAND_YES)
 					break;
@@ -4085,7 +4136,7 @@
 		    !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);
+			transmit_provisional_response(p, "183 Session Progress", &p->initreq, 1);
 			ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);	
 			break;
 		}
@@ -4597,6 +4648,7 @@
 	p->waitid = -1;
 	p->autokillid = -1;
 	p->request_queue_sched_id = -1;
+	p->provisional_keepalive_sched_id = -1;
 	p->subscribed = NONE;
 	p->stateid = -1;
 	p->prefs = default_prefs;		/* Set default codecs for this call */
@@ -6520,6 +6572,19 @@
 	add_header_contentLength(&resp, 0);
 	append_history(p, "AuthChal", "Auth challenge sent for %s - nc %d", p->username, p->noncecount);
 	return send_response(p, &resp, reliable, seqno);
+}
+
+/* Only use a static string for the msg, here! */
+static int transmit_provisional_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, int with_sdp)
+{
+	int res;
+
+	if (!(res = with_sdp ? transmit_response_with_sdp(p, msg, req, XMIT_UNRELIABLE) : transmit_response(p, msg, req))) {
+		p->last_provisional = msg;
+		update_provisional_keepalive(p, with_sdp);
+	}
+
+	return res;
 }
 
 /*! \brief Add text body to SIP message */
@@ -14979,7 +15044,7 @@
 		case AST_STATE_DOWN:
 			if (option_debug > 1)
 				ast_log(LOG_DEBUG, "%s: New call is still down.... Trying... \n", c->name);
-			transmit_response(p, "100 Trying", req);
+			transmit_provisional_response(p, "100 Trying", req, 0);
 			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 */
@@ -15043,11 +15108,11 @@
 			}
 			break;
 		case AST_STATE_RING:
-			transmit_response(p, "100 Trying", req);
+			transmit_provisional_response(p, "100 Trying", req, 0);
 			p->invitestate = INV_PROCEEDING;
 			break;
 		case AST_STATE_RINGING:
-			transmit_response(p, "180 Ringing", req);
+			transmit_provisional_response(p, "180 Ringing", req, 0);
 			p->invitestate = INV_PROCEEDING;
 			break;
 		case AST_STATE_UP:

Modified: team/oej/no-premature-183/utils/Makefile
URL: http://svn.asterisk.org/svn-view/asterisk/team/oej/no-premature-183/utils/Makefile?view=diff&rev=215882&r1=215881&r2=215882
==============================================================================
--- team/oej/no-premature-183/utils/Makefile (original)
+++ team/oej/no-premature-183/utils/Makefile Thu Sep  3 01:58:47 2009
@@ -26,7 +26,7 @@
 #     changes are made to ast_expr2.y or ast_expr2.fl (or the corresponding .c files),
 #     as a regression test. Others (mere mortals?) need not bother, but they are
 #     more than welcome to play! The regression test itself is in expr2.testinput.
-ALL_UTILS:=astman smsq stereorize streamplayer aelparse muted
+ALL_UTILS:=astman smsq stereorize streamplayer aelparse
 UTILS:=$(ALL_UTILS)
 
 include $(ASTTOPDIR)/Makefile.rules




More information about the asterisk-commits mailing list