Update of /usr/cvsroot/asterisk/channels
In directory localhost.localdomain:/tmp/cvs-serv24024/channels
Modified Files:
chan_sip.c
Log Message:
attempt to not allocate RTP ports for SIP private structures unless they are
needed (bug #3986)
Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.715
retrieving revision 1.716
diff -u -d -r1.715 -r1.716
--- chan_sip.c 1 May 2005 23:17:12 -0000 1.715
+++ chan_sip.c 2 May 2005 01:15:23 -0000 1.716
@@ -133,23 +133,24 @@
const struct cfsip_methods {
int id;
+ int need_rtp; /* when this is the 'primary' use for a pvt
structure, does it need RTP? */
char *text;
} sip_methods[] = {
- { 0, "-UNKNOWN-" },
- { SIP_REGISTER, "REGISTER" },
- { SIP_OPTIONS, "OPTIONS" },
- { SIP_NOTIFY, "NOTIFY" },
- { SIP_INVITE, "INVITE" },
- { SIP_ACK, "ACK" },
- { SIP_PRACK, "PRACK" },
- { SIP_BYE, "BYE" },
- { SIP_REFER, "REFER" },
- { SIP_SUBSCRIBE,"SUBSCRIBE" },
- { SIP_MESSAGE, "MESSAGE" },
- { SIP_UPDATE, "UPDATE" },
- { SIP_INFO, "INFO" },
- { SIP_CANCEL, "CANCEL" },
- { SIP_PUBLISH, "PUBLISH" }
+ { 0, 1, "-UNKNOWN-" },
+ { SIP_REGISTER, 0, "REGISTER" },
+ { SIP_OPTIONS, 0, "OPTIONS" },
+ { SIP_NOTIFY, 0, "NOTIFY" },
+ { SIP_INVITE, 1, "INVITE" },
+ { SIP_ACK, 0, "ACK" },
+ { SIP_PRACK, 0, "PRACK" },
+ { SIP_BYE, 0, "BYE" },
+ { SIP_REFER, 0, "REFER" },
+ { SIP_SUBSCRIBE, 0, "SUBSCRIBE" },
+ { SIP_MESSAGE, 0, "MESSAGE" },
+ { SIP_UPDATE, 0, "UPDATE" },
+ { SIP_INFO, 0, "INFO" },
+ { SIP_CANCEL, 0, "CANCEL" },
+ { SIP_PUBLISH, 0, "PUBLISH" }
};
@@ -2494,7 +2495,7 @@
}
/*--- sip_alloc: Allocate SIP_PVT structure and set defaults ---*/
-static struct sip_pvt *sip_alloc(char *callid, struct sockaddr_in *sin, int
useglobal_nat)
+static struct sip_pvt *sip_alloc(char *callid, struct sockaddr_in *sin, int
useglobal_nat, const int intended_method)
{
struct sip_pvt *p;
@@ -2519,32 +2520,40 @@
} else {
memcpy(&p->ourip, &__ourip, sizeof(p->ourip));
}
- p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
- if (videosupport)
- p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0,
bindaddr.sin_addr);
+
p->branch = rand();
p->tag = rand();
-
/* Start with 101 instead of 1 */
p->ocseq = 101;
- if (!p->rtp) {
- ast_log(LOG_WARNING, "Unable to create RTP session: %s\n",
strerror(errno));
- ast_mutex_destroy(&p->lock);
- if(p->chanvars) {
- ast_variables_destroy(p->chanvars);
- p->chanvars = NULL;
+
+ if (sip_methods[intended_method].need_rtp) {
+ p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0,
bindaddr.sin_addr);
+ if (videosupport)
+ p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0,
bindaddr.sin_addr);
+ if (!p->rtp) {
+ ast_log(LOG_WARNING, "Unable to create RTP session:
%s\n", strerror(errno));
+ ast_mutex_destroy(&p->lock);
+ if(p->chanvars) {
+ ast_variables_destroy(p->chanvars);
+ p->chanvars = NULL;
+ }
+ free(p);
+ return NULL;
}
- free(p);
- return NULL;
+ ast_rtp_settos(p->rtp, tos);
+ if (p->vrtp)
+ ast_rtp_settos(p->vrtp, tos);
+ p->rtptimeout = global_rtptimeout;
+ p->rtpholdtimeout = global_rtpholdtimeout;
+ p->rtpkeepalive = global_rtpkeepalive;
}
- ast_rtp_settos(p->rtp, tos);
- if (p->vrtp)
- ast_rtp_settos(p->vrtp, tos);
+
if (useglobal_nat && sin) {
/* Setup NAT structure according to global settings if we have
an address */
ast_copy_flags(p, &global_flags, SIP_NAT);
memcpy(&p->recv, sin, sizeof(p->recv));
- ast_rtp_setnat(p->rtp, (ast_test_flag(p, SIP_NAT) &
SIP_NAT_ROUTE));
+ if (p->rtp)
+ ast_rtp_setnat(p->rtp, (ast_test_flag(p, SIP_NAT) &
SIP_NAT_ROUTE));
if (p->vrtp)
ast_rtp_setnat(p->vrtp, (ast_test_flag(p, SIP_NAT) &
SIP_NAT_ROUTE));
}
@@ -2558,9 +2567,6 @@
ast_copy_flags(p, (&global_flags), SIP_PROMISCREDIR | SIP_TRUSTRPID |
SIP_DTMF | SIP_REINVITE | SIP_PROG_INBAND | SIP_OSPAUTH);
/* Assign default music on hold class */
strcpy(p->musicclass, global_musicclass);
- p->rtptimeout = global_rtptimeout;
- p->rtpholdtimeout = global_rtpholdtimeout;
- p->rtpkeepalive = global_rtpkeepalive;
p->capability = global_capability;
if (ast_test_flag(p, SIP_DTMF) == SIP_DTMF_RFC2833)
p->noncodeccapability |= AST_RTP_DTMF;
@@ -2577,7 +2583,7 @@
/*--- find_call: Connect incoming SIP message to current call or create new
call structure */
/* Called by handle_request ,sipsock_read */
-static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in
*sin)
+static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in
*sin, const int intended_method)
{
struct sip_pvt *p;
char *callid;
@@ -2633,7 +2639,7 @@
p = p->next;
}
ast_mutex_unlock(&iflock);
- p = sip_alloc(callid, sin, 1);
+ p = sip_alloc(callid, sin, 1, intended_method);
if (p)
ast_mutex_lock(&p->lock);
return p;
@@ -4479,7 +4485,7 @@
r->callid_valid = 1;
}
/* Allocate SIP packet for registration */
- p=sip_alloc( r->callid, NULL, 0);
+ p=sip_alloc( r->callid, NULL, 0, SIP_REGISTER);
if (!p) {
ast_log(LOG_WARNING, "Unable to allocate registration
call\n");
return 0;
@@ -7423,7 +7429,7 @@
struct sip_request req;
struct ast_variable *var;
- p = sip_alloc(NULL, NULL, 0);
+ p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY);
if (!p) {
ast_log(LOG_WARNING, "Unable to build sip pvt data for
notify\n");
return RESULT_FAILURE;
@@ -9055,10 +9061,6 @@
/* Get the command */
cseq += len;
- /* Determine the request URI for sip, sips or tel URIs */
- if( determine_firstline_parts( req ) < 0 ) {
- return -1;
- }
cmd = req->rlPart1;
e = req->rlPart2;
@@ -9066,7 +9068,6 @@
useragent = get_header(req, "User-Agent");
strncpy(p->useragent, useragent, sizeof(p->useragent)-1);
-
/* Find out SIP method for incoming request */
if (!strcasecmp(cmd, "SIP/2.0")) { /* Response to our request */
p->method = SIP_RESPONSE;
@@ -9246,10 +9247,15 @@
/* Must have at least two headers */
return 1;
}
+
+ /* Determine the request URI for sip, sips or tel URIs */
+ if (determine_firstline_parts(&req) < 0)
+ return 1;
+
/* Process request, with netlock held */
retrylock:
ast_mutex_lock(&netlock);
- p = find_call(&req, &sin);
+ p = find_call(&req, &sin, find_sip_method(req.rlPart1));
if (p) {
/* Go ahead and lock the owner if it has one -- we may need it
*/
if (p->owner && ast_mutex_trylock(&p->owner->lock)) {
@@ -9298,7 +9304,7 @@
return 0;
}
- p = sip_alloc(NULL, NULL, 0);
+ p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY);
if (!p) {
ast_log(LOG_WARNING, "Unable to build sip pvt data for MWI\n");
return -1;
@@ -9523,7 +9529,7 @@
ast_log(LOG_NOTICE, "Still have a call...\n");
sip_destroy(peer->call);
}
- p = peer->call = sip_alloc(NULL, NULL, 0);
+ p = peer->call = sip_alloc(NULL, NULL, 0, SIP_OPTIONS);
if (!peer->call) {
ast_log(LOG_WARNING, "Unable to allocate call for poking peer
'%s'\n", peer->name);
return -1;
@@ -9626,7 +9632,7 @@
ast_log(LOG_NOTICE, "Asked to get a channel of unsupported
format %s while capability is %s\n", ast_getformatname(oldformat),
ast_getformatname(global_capability));
return NULL;
}
- p = sip_alloc(NULL, NULL, 0);
+ p = sip_alloc(NULL, NULL, 0, SIP_INVITE);
if (!p) {
ast_log(LOG_WARNING, "Unable to build sip pvt data for '%s'\n",
(char *)data);
return NULL;
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