[svn-commits] oej: branch oej/invitestate-1.4 r48131 -
/team/oej/invitestate-1.4/channels/
svn-commits at lists.digium.com
svn-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)) {
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