[svn-commits] branch group/sip-threading r26803 - in
/team/group/sip-threading: ./ channels/
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Thu May 11 02:36:21 MST 2006
Author: file
Date: Thu May 11 04:36:21 2006
New Revision: 26803
URL: http://svn.digium.com/view/asterisk?rev=26803&view=rev
Log:
Merged revisions 26755-26759,26764,26781,26799-26800 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r26755 | oej | 2006-05-11 03:08:54 -0500 (Thu, 11 May 2006) | 3 lines
Issue #OEJ12399.3 - Set default maxcallbitrate for users too.
(Well, could have opened a branch for this, but felt community pressure not to... ;-)
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r26756 | oej | 2006-05-11 03:20:59 -0500 (Thu, 11 May 2006) | 2 lines
Small formatting fixes
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r26757 | oej | 2006-05-11 03:25:20 -0500 (Thu, 11 May 2006) | 2 lines
Ignore CANCEL on call that is already answered
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r26758 | markster | 2006-05-11 03:28:16 -0500 (Thu, 11 May 2006) | 2 lines
Make sure we handle long information elements (bug #6654)
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r26759 | markster | 2006-05-11 03:47:37 -0500 (Thu, 11 May 2006) | 2 lines
Set "MARK" whenever SSRC changes (bug #6934)
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r26764 | kpfleming | 2006-05-11 04:01:53 -0500 (Thu, 11 May 2006) | 10 lines
Merged revisions 26760 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
........
r26760 | kpfleming | 2006-05-11 03:52:46 -0500 (Thu, 11 May 2006) | 2 lines
ensure that we send a response to REGISTER requests that are successfully authenticated but contain invalid Contact URIs
........
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r26781 | kpfleming | 2006-05-11 04:07:45 -0500 (Thu, 11 May 2006) | 2 lines
block revision for fix that was committed here before 1.2
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r26799 | rizzo | 2006-05-11 04:24:35 -0500 (Thu, 11 May 2006) | 4 lines
simplify sip_pretend_ack also removing a bug in the existing
code which called ast_skip_blanks erroneously.
................
r26800 | rizzo | 2006-05-11 04:29:37 -0500 (Thu, 11 May 2006) | 5 lines
replace list unlinking with UNLINK macro
(when the list becomes an astobj this will be revisited;
however at the moment the change would be too intrusive).
................
Modified:
team/group/sip-threading/ (props changed)
team/group/sip-threading/channels/chan_sip.c
team/group/sip-threading/enum.c
team/group/sip-threading/rtp.c
Propchange: team/group/sip-threading/
------------------------------------------------------------------------------
automerge = AstriDevConOMG
Propchange: team/group/sip-threading/
------------------------------------------------------------------------------
Binary property 'branch-1.2-blocked' - no diff available.
Propchange: team/group/sip-threading/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.
Propchange: team/group/sip-threading/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Thu May 11 04:36:21 2006
@@ -1,1 +1,1 @@
-/trunk:1-26723
+/trunk:1-26800
Modified: team/group/sip-threading/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/group/sip-threading/channels/chan_sip.c?rev=26803&r1=26802&r2=26803&view=diff
==============================================================================
--- team/group/sip-threading/channels/chan_sip.c (original)
+++ team/group/sip-threading/channels/chan_sip.c Thu May 11 04:36:21 2006
@@ -1277,6 +1277,15 @@
.send_text = sip_sendtext,
};
+/**--- some list management macros. **/
+
+#define UNLINK(element, head, prev) do { \
+ if (prev) \
+ (prev)->next = (element)->next; \
+ else \
+ (head) = (element)->next; \
+ } while (0)
+
/*! \brief Interface structure with callbacks used to connect to RTP module */
static struct ast_rtp_protocol sip_rtp = {
type: "SIP",
@@ -1744,27 +1753,20 @@
}
/*! \brief Pretend to ack all packets */
+/* maybe the lock on p is not strictly necessary but there might be a race */
static int __sip_pretend_ack(struct sip_pvt *p)
{
struct sip_pkt *cur = NULL;
-
+
while (!AST_LIST_EMPTY(&p->packets)) {
+ int method;
if (cur == AST_LIST_FIRST(&p->packets)) {
ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text);
return -1;
}
cur = AST_LIST_FIRST(&p->packets);
- if (cur->method)
- __sip_ack(p, cur->seqno, (ast_test_flag(cur, FLAG_RESPONSE)), cur->method, FALSE);
- else { /* Unknown packet type */
- char *c;
- char method[128];
-
- ast_copy_string(method, cur->data, sizeof(method));
- c = ast_skip_blanks(method); /* XXX what ? */
- *c = '\0';
- __sip_ack(p, cur->seqno, (ast_test_flag(cur, FLAG_RESPONSE)), find_sip_method(method), FALSE);
- }
+ method = (cur->method) ? cur->method : find_sip_method(cur->data);
+ __sip_ack(p, cur->seqno, ast_test_flag(cur, FLAG_RESPONSE), method, FALSE);
}
return 0;
}
@@ -2837,7 +2839,7 @@
__sip_pretend_ack(p);
/* Send a new request: CANCEL */
- transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, 0);
+ transmit_request_with_auth(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. */
ast_clear_flag(&locflags, SIP_NEEDDESTROY);
@@ -6812,6 +6814,9 @@
switch (parse_register_contact(p, peer, req)) {
case PARSE_REGISTER_FAILED:
ast_log(LOG_WARNING, "Failed to parse contact info\n");
+ transmit_response_with_date(p, "400 Bad Request", req);
+ peer->lastmsgssent = -1;
+ res = 0;
break;
case PARSE_REGISTER_QUERY:
transmit_response_with_date(p, "200 OK", req);
@@ -6835,11 +6840,13 @@
peer = temp_peer(name);
if (peer) {
ASTOBJ_CONTAINER_LINK(&peerl, peer);
- peer->lastmsgssent = -1;
sip_cancel_destroy(p);
switch (parse_register_contact(p, peer, req)) {
case PARSE_REGISTER_FAILED:
ast_log(LOG_WARNING, "Failed to parse contact info\n");
+ transmit_response_with_date(p, "400 Bad Request", req);
+ peer->lastmsgssent = -1;
+ res = 0;
break;
case PARSE_REGISTER_QUERY:
transmit_response_with_date(p, "200 OK", req);
@@ -9936,7 +9943,7 @@
{
/* Go ahead and send bye at this point */
if (ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
- transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
+ transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);
} else if (ast_test_flag(&p->flags[0], SIP_NEEDREINVITE)) {
@@ -10048,13 +10055,13 @@
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
}
/* If I understand this right, the branch is different for a non-200 ACK only */
- transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 1);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
check_pendings(p);
break;
case 407: /* Proxy authentication */
case 401: /* Www auth */
/* First we ACK */
- transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
if (p->options)
p->options->auth_type = (resp == 401 ? WWW_AUTH : PROXY_AUTH);
@@ -10074,7 +10081,7 @@
break;
case 403: /* Forbidden */
/* First we ACK */
- transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
ast_log(LOG_WARNING, "Received response: \"Forbidden\" from '%s'\n", get_header(&p->initreq, "From"));
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
@@ -10082,7 +10089,7 @@
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
break;
case 404: /* Not found */
- transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
+ 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);
@@ -10090,7 +10097,7 @@
case 481: /* Call leg does not exist */
/* Could be REFER or INVITE */
ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
- transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
break;
case 491: /* Pending */
/* we have to wait a while, then retransmit */
@@ -10458,7 +10465,7 @@
case 481: /* Call leg does not exist */
if (sipmethod == SIP_INVITE) {
/* First we ACK */
- transmit_request(p, SIP_ACK, seqno, 0, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
ast_log(LOG_WARNING, "INVITE with REPLACEs failed to '%s'\n", get_header(&p->initreq, "From"));
if (owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
@@ -10571,7 +10578,7 @@
}
/* ACK on invite */
if (sipmethod == SIP_INVITE)
- transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, 0);
+ transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
if (!p->owner)
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
@@ -11427,7 +11434,7 @@
transmit_notify_with_sipfrag(p, seqno, "200 OK", 1);
/* Always increment on a BYE */
if (!nobye) {
- transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
+ transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
}
}
@@ -11440,6 +11447,14 @@
check_via(p, req);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+
+ if (p->owner && p->owner->_state == AST_STATE_UP) {
+ /* This call is up, cancel is ignored, we need a bye */
+ transmit_response(p, "200 OK", req);
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Got CANCEL on an answered call. Ignoring... \n");
+ return 0;
+ }
if (p->rtp) {
/* Immediately stop RTP */
ast_rtp_stop(p->rtp);
@@ -12881,6 +12896,7 @@
ast_copy_flags(&user->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);
user->capability = global_capability;
user->allowtransfer = global_allowtransfer;
+ user->maxcallbitrate = default_maxcallbitrate;
user->prefs = default_prefs;
/* set default context */
strcpy(user->context, default_context);
Modified: team/group/sip-threading/enum.c
URL: http://svn.digium.com/view/asterisk/team/group/sip-threading/enum.c?rev=26803&r1=26802&r2=26803&view=diff
==============================================================================
--- team/group/sip-threading/enum.c (original)
+++ team/group/sip-threading/enum.c Thu May 11 04:36:21 2006
@@ -95,7 +95,7 @@
} __attribute__ ((__packed__));
/*! \brief Parse NAPTR record information elements */
-static int parse_ie(char *data, int maxdatalen, char *src, int srclen)
+static int parse_ie(unsigned char *data, int maxdatalen, unsigned char *src, int srclen)
{
int len, olen;
@@ -113,7 +113,7 @@
}
/*! \brief Parse DNS NAPTR record used in ENUM ---*/
-static int parse_naptr(char *dst, int dstsize, char *tech, int techsize, char *answer, int len, char *naptrinput)
+static int parse_naptr(unsigned char *dst, int dstsize, char *tech, int techsize, unsigned char *answer, int len, unsigned char *naptrinput)
{
char tech_return[80];
Modified: team/group/sip-threading/rtp.c
URL: http://svn.digium.com/view/asterisk/team/group/sip-threading/rtp.c?rev=26803&r1=26802&r2=26803&view=diff
==============================================================================
--- team/group/sip-threading/rtp.c (original)
+++ team/group/sip-threading/rtp.c Thu May 11 04:36:21 2006
@@ -93,6 +93,7 @@
struct ast_frame f;
unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
unsigned int ssrc; /*!< Synchronization source, RFC 3550, page 10. */
+ unsigned int rxssrc;
unsigned int lastts;
unsigned int lastdigitts;
unsigned int lastrxts;
@@ -430,6 +431,7 @@
int ext;
int x;
char iabuf[INET_ADDRSTRLEN];
+ unsigned int ssrc;
unsigned int timestamp;
unsigned int *rtpheader;
struct rtpPayloadType rtpPT;
@@ -485,6 +487,14 @@
ext = seqno & (1 << 28);
seqno &= 0xffff;
timestamp = ntohl(rtpheader[1]);
+ ssrc = ntohl(rtpheader[2]);
+
+ if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
+ ast_log(LOG_WARNING, "Forcing Marker bit, because SSRC has changed\n");
+ mark = 1;
+ }
+
+ rtp->rxssrc = ssrc;
if (padding) {
/* Remove padding bytes */
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