[Asterisk-cvs] asterisk/channels chan_sip.c,1.791,1.792
kpfleming at lists.digium.com
kpfleming at lists.digium.com
Mon Jul 25 19:20:10 CDT 2005
Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv17948/channels
Modified Files:
chan_sip.c
Log Message:
clean up SIP response and pedantic REGISTER matching (bug #3663)
Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.791
retrieving revision 1.792
diff -u -d -r1.791 -r1.792
--- chan_sip.c 25 Jul 2005 21:58:32 -0000 1.791
+++ chan_sip.c 25 Jul 2005 23:26:47 -0000 1.792
@@ -123,21 +123,21 @@
static const char config[] = "sip.conf";
static const char notify_config[] = "sip_notify.conf";
-#define SIP_REGISTER 1
-#define SIP_OPTIONS 2
-#define SIP_NOTIFY 3
-#define SIP_INVITE 4
-#define SIP_ACK 5
-#define SIP_PRACK 6
-#define SIP_BYE 7
-#define SIP_REFER 8
-#define SIP_SUBSCRIBE 9
-#define SIP_MESSAGE 10
-#define SIP_UPDATE 11
-#define SIP_INFO 12
-#define SIP_CANCEL 13
-#define SIP_PUBLISH 14
-#define SIP_RESPONSE 100
+#define SIP_RESPONSE 1
+#define SIP_REGISTER 2
+#define SIP_OPTIONS 3
+#define SIP_NOTIFY 4
+#define SIP_INVITE 5
+#define SIP_ACK 6
+#define SIP_PRACK 7
+#define SIP_BYE 8
+#define SIP_REFER 9
+#define SIP_SUBSCRIBE 10
+#define SIP_MESSAGE 12
+#define SIP_UPDATE 13
+#define SIP_INFO 14
+#define SIP_CANCEL 15
+#define SIP_PUBLISH 16
#define RTP 1
#define NO_RTP 0
@@ -147,6 +147,7 @@
char *text;
} sip_methods[] = {
{ 0, RTP, "-UNKNOWN-" },
+ { SIP_RESPONSE, NO_RTP, "SIP/2.0" },
{ SIP_REGISTER, NO_RTP, "REGISTER" },
{ SIP_OPTIONS, NO_RTP, "OPTIONS" },
{ SIP_NOTIFY, NO_RTP, "NOTIFY" },
@@ -384,6 +385,7 @@
int lines; /* SDP Content */
char *line[SIP_MAX_LINES];
char data[SIP_MAX_PACKET];
+ int debug; /* Debug flag for this packet */
};
struct sip_pkt;
@@ -586,6 +588,7 @@
struct sip_pkt {
struct sip_pkt *next; /* Next packet */
int retrans; /* Retransmission number */
+ int method; /* SIP method for this packet */
int seqno; /* Sequence number */
unsigned int flags; /* non-zero if this is a response packet (e.g. 200 OK) */
struct sip_pvt *owner; /* Owner call */
@@ -787,7 +790,7 @@
static int expire_register(void *data);
static int callevents = 0;
-static struct ast_channel *sip_request(const char *type, int format, void *data, int *cause);
+static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause);
static int sip_devicestate(void *data);
static int sip_sendtext(struct ast_channel *ast, const char *text);
static int sip_call(struct ast_channel *ast, char *dest, int timeout);
@@ -803,6 +806,7 @@
static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno); /* Add realm authentication in list */
static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, char *realm); /* Find authentication for a specific realm */
static void append_date(struct sip_request *req); /* Append date to SIP packet */
+static int determine_firstline_parts(struct sip_request *req);
/* Definition of this channel for channel registration */
static const struct ast_channel_tech sip_tech = {
@@ -810,7 +814,7 @@
.description = "Session Initiation Protocol (SIP)",
.capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
.properties = AST_CHAN_TP_WANTSJITTER,
- .requester = sip_request,
+ .requester = sip_request_call,
.devicestate = sip_devicestate,
.call = sip_call,
.hangup = sip_hangup,
@@ -830,8 +834,13 @@
int find_sip_method(char *msg)
{
int i, res = 0;
+
+ if (!msg || ast_strlen_zero(msg))
+ return 0;
+
/* Strictly speaking, SIP methods are case SENSITIVE, but we don't check */
- for (i=1;(i < (sizeof(sip_methods) / sizeof(sip_methods[0]))) && !res; i++) {
+ /* following Jon Postel's rule: Be gentle in what you accept, strict with what you send */
+ for (i=1; (i < (sizeof(sip_methods) / sizeof(sip_methods[0]))) && !res; i++) {
if (!strcasecmp(sip_methods[i].text, msg))
res = sip_methods[i].id;
}
@@ -1071,7 +1080,7 @@
}
/*--- __sip_reliable_xmit: transmit packet with retransmits ---*/
-static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal)
+static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod)
{
struct sip_pkt *pkt;
pkt = malloc(sizeof(struct sip_pkt) + len + 1);
@@ -1079,6 +1088,7 @@
return -1;
memset(pkt, 0, sizeof(struct sip_pkt));
memcpy(pkt->data, data, len);
+ pkt->method = sipmethod;
pkt->packetlen = len;
pkt->next = p->packets;
pkt->owner = p;
@@ -1091,8 +1101,9 @@
pkt->retransid = ast_sched_add(sched, DEFAULT_RETRANS, retrans_pkt, pkt);
pkt->next = p->packets;
p->packets = pkt;
- __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
- if (!strncasecmp(pkt->data, "INVITE", 6)) {
+
+ __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); /* Send packet */
+ if (sipmethod == SIP_INVITE) {
/* Note this is a pending invite */
p->pendinginvite = seqno;
}
@@ -1226,7 +1237,7 @@
return res;
}
-static void parse(struct sip_request *req);
+static void parse_request(struct sip_request *req);
static char *get_header(struct sip_request *req, char *name);
static void copy_request(struct sip_request *dst,struct sip_request *src);
@@ -1236,7 +1247,7 @@
memset(dst, 0, sizeof(*dst));
memcpy(dst->data, src->data, sizeof(dst->data));
dst->len = src->len;
- parse(dst);
+ parse_request(dst);
}
/*--- send_response: Transmit response on SIP request---*/
@@ -1246,6 +1257,7 @@
char iabuf[INET_ADDRSTRLEN];
struct sip_request tmp;
char tmpmsg[80];
+
if (sip_debug_test_pvt(p)) {
if (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE)
ast_verbose("%sTransmitting (NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr), ntohs(p->recv.sin_port), req->data);
@@ -1258,7 +1270,7 @@
snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq"));
append_history(p, "TxRespRel", tmpmsg);
}
- res = __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable > 1));
+ res = __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable > 1), req->method);
} else {
if (recordhistory) {
parse_copy(&tmp, req);
@@ -1292,7 +1304,7 @@
snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq"));
append_history(p, "TxReqRel", tmpmsg);
}
- res = __sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable > 1));
+ res = __sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable > 1), req->method);
} else {
if (recordhistory) {
parse_copy(&tmp, req);
@@ -2879,27 +2891,18 @@
struct sip_pvt *p;
char *callid;
char tmp[256] = "";
- char iabuf[INET_ADDRSTRLEN];
- char *cmd;
char *tag = "", *c;
callid = get_header(req, "Call-ID");
if (pedanticsipchecking) {
- /* In principle Call-ID's uniquely identify a call, however some vendors
- (i.e. Pingtel) send multiple calls with the same Call-ID and different
- tags in order to simplify billing. The RFC does state that we have to
- compare tags in addition to the call-id, but this generate substantially
- more overhead which is totally unnecessary for the vast majority of sane
- SIP implementations, and thus Asterisk does not enable this behavior
- by default. Short version: You'll need this option to support conferencing
- on the pingtel */
- ast_copy_string(tmp, req->header[0], sizeof(tmp));
- cmd = tmp;
- c = strchr(tmp, ' ');
- if (c)
- *c = '\0';
- if (!strcasecmp(cmd, "SIP/2.0"))
+ /* In principle Call-ID's uniquely identify a call, but with a forking SIP proxy
+ we need more to identify a branch - so we have to check branch, from
+ and to tags to identify a call leg.
+ For Asterisk to behave correctly, you need to turn on pedanticsipchecking
+ in sip.conf
+ */
+ if (req->method == SIP_RESPONSE)
ast_copy_string(tmp, get_header(req, "To"), sizeof(tmp));
else
ast_copy_string(tmp, get_header(req, "From"), sizeof(tmp));
@@ -2913,15 +2916,16 @@
}
- if (ast_strlen_zero(callid)) {
- ast_log(LOG_WARNING, "Call missing call ID from '%s'\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr));
- return NULL;
- }
ast_mutex_lock(&iflock);
p = iflist;
while(p) {
- if (!strcmp(p->callid, callid) &&
- (!pedanticsipchecking || !tag || ast_strlen_zero(p->theirtag) || !strcmp(p->theirtag, tag))) {
+ int found = 0;
+ if (req->method == SIP_REGISTER)
+ found = (!strcmp(p->callid, callid));
+ else
+ found = (!strcmp(p->callid, callid) &&
+ (!pedanticsipchecking || !tag || ast_strlen_zero(p->theirtag) || !strcmp(p->theirtag, tag))) ;
+ if (found) {
/* Found the call */
ast_mutex_lock(&p->lock);
ast_mutex_unlock(&iflock);
@@ -3055,12 +3059,13 @@
return t;
}
-/*--- parse: Parse a SIP message ----*/
-static void parse(struct sip_request *req)
+/*--- parse_request: Parse a SIP message ----*/
+static void parse_request(struct sip_request *req)
{
/* Divide fields by NULL's */
char *c;
int f = 0;
+
c = req->data;
/* First header starts immediately */
@@ -3070,16 +3075,15 @@
/* We've got a new header */
*c = 0;
-#if 0
- printf("Header: %s (%d)\n", req->header[f], strlen(req->header[f]));
-#endif
+ if (option_debug > 3)
+ ast_log(LOG_DEBUG, "Header: %s (%d)\n", req->header[f], (int) strlen(req->header[f]));
if (ast_strlen_zero(req->header[f])) {
/* Line by itself means we're now in content */
c++;
break;
}
if (f >= SIP_MAX_HEADERS - 1) {
- ast_log(LOG_WARNING, "Too many SIP headers...\n");
+ ast_log(LOG_WARNING, "Too many SIP headers. Ignoring.\n");
} else
f++;
req->header[f] = c + 1;
@@ -3100,11 +3104,10 @@
if (*c == '\n') {
/* We've got a new line */
*c = 0;
-#if 0
- printf("Line: %s (%d)\n", req->line[f], strlen(req->line[f]));
-#endif
+ if (option_debug > 3)
+ ast_log(LOG_DEBUG, "Line: %s (%d)\n", req->line[f], (int) strlen(req->line[f]));
if (f >= SIP_MAX_LINES - 1) {
- ast_log(LOG_WARNING, "Too many SDP lines...\n");
+ ast_log(LOG_WARNING, "Too many SDP lines. Ignoring.\n");
} else
f++;
req->line[f] = c + 1;
@@ -3120,6 +3123,8 @@
req->lines = f;
if (*c)
ast_log(LOG_WARNING, "Odd content, extra stuff left over ('%s')\n", c);
+ /* Split up the first line parts */
+ determine_firstline_parts(req);
}
/*--- process_sdp: Process SIP SDP and activate RTP channels---*/
@@ -3649,6 +3654,7 @@
ast_log(LOG_WARNING, "Request already initialized?!?\n");
return -1;
}
+ req->method = SIP_RESPONSE;
req->header[req->headers] = req->data + req->len;
snprintf(req->header[req->headers], sizeof(req->data) - req->len, "SIP/2.0 %s\r\n", resp);
req->len += strlen(req->header[req->headers]);
@@ -4218,10 +4224,9 @@
add_sdp(&req, p);
/* Use this as the basis */
copy_request(&p->initreq, &req);
- parse(&p->initreq);
+ parse_request(&p->initreq);
if (sip_debug_test_pvt(p))
ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
- determine_firstline_parts(&p->initreq);
p->lastinvite = p->ocseq;
ast_set_flag(p, SIP_OUTGOING);
return send_request(p, &req, 1, p->ocseq);
@@ -4368,6 +4373,7 @@
{
struct sip_request req;
+ req.method = sipmethod;
if (init) {
/* Bump branch even on initial requests */
p->branch ^= rand();
@@ -4450,10 +4456,9 @@
if (!p->initreq.headers) {
/* Use this as the basis */
copy_request(&p->initreq, &req);
- parse(&p->initreq);
+ parse_request(&p->initreq);
if (sip_debug_test_pvt(p))
ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
- determine_firstline_parts(&p->initreq);
}
p->lastinvite = p->ocseq;
return send_request(p, &req, init ? 2 : 1, p->ocseq);
@@ -4562,7 +4567,7 @@
if (!p->initreq.headers) { /* Use this as the basis */
copy_request(&p->initreq, &req);
- parse(&p->initreq);
+ parse_request(&p->initreq);
if (sip_debug_test_pvt(p))
ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
determine_firstline_parts(&p->initreq);
@@ -4577,7 +4582,7 @@
if (!p->initreq.headers) {
/* Use this as the basis */
copy_request(&p->initreq, req);
- parse(&p->initreq);
+ parse_request(&p->initreq);
if (sip_debug_test_pvt(p))
ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
determine_firstline_parts(&p->initreq);
@@ -4608,7 +4613,7 @@
if (!p->initreq.headers) {
/* Use this as the basis */
copy_request(&p->initreq, &req);
- parse(&p->initreq);
+ parse_request(&p->initreq);
if (sip_debug_test_pvt(p))
ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
determine_firstline_parts(&p->initreq);
@@ -4884,7 +4889,7 @@
add_header_contentLength(&req, 0);
add_blank_header(&req);
copy_request(&p->initreq, &req);
- parse(&p->initreq);
+ parse_request(&p->initreq);
if (sip_debug_test_pvt(p)) {
ast_verbose("REGISTER %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
}
@@ -9026,7 +9031,7 @@
}
}
} else if (debug)
- ast_verbose("Ignoring this request\n");
+ ast_verbose("Ignoring this INVITE request\n");
if (!p->lastinvite && !ignore && !p->owner) {
/* Handle authentication if this is our first invite */
res = check_user(p, req, SIP_INVITE, e, 1, sin, ignore);
@@ -9552,7 +9557,8 @@
/* Save useragent of the client */
useragent = get_header(req, "User-Agent");
- ast_copy_string(p->useragent, useragent, sizeof(p->useragent));
+ if (!ast_strlen_zero(useragent))
+ ast_copy_string(p->useragent, useragent, sizeof(p->useragent));
/* Find out SIP method for incoming request */
if (!strcasecmp(cmd, "SIP/2.0")) { /* Response to our request */
@@ -9584,18 +9590,20 @@
/* New SIP request coming in
(could be new request in existing SIP dialog as well...)
*/
- p->method = find_sip_method(cmd); /* Find out which SIP method they are using */
+ p->method = req->method; /* Find out which SIP method they are using */
if (option_debug > 2)
ast_log(LOG_DEBUG, "**** Received %s (%d) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd);
if (p->icseq && (p->icseq > seqno)) {
ast_log(LOG_DEBUG, "Ignoring too old SIP packet packet %d (expecting >= %d)\n", seqno, p->icseq);
return -1;
- } else if (p->icseq && (p->icseq == seqno) && (strcasecmp(cmd, "CANCEL") || ast_test_flag(p, SIP_ALREADYGONE))) {
+ } else if (p->icseq && (p->icseq == seqno) && (p->method != SIP_CANCEL|| ast_test_flag(p, SIP_ALREADYGONE))) {
/* ignore means "don't do anything with it" but still have to
respond appropriately. We do this if we receive a repeat of
the last sequence number */
- ignore=1;
+ ignore=2;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Ignoring SIP message because of retransmit (%s Seqno %d, ours %d)\n", sip_methods[p->method].text, p->icseq, seqno);
}
if (seqno >= p->icseq)
@@ -9697,7 +9705,6 @@
socklen_t len;
int nounlock;
int recount = 0;
- int debug;
char iabuf[INET_ADDRSTRLEN];
len = sizeof(sin);
@@ -9715,13 +9722,14 @@
}
req.data[res] = '\0';
req.len = res;
- debug = sip_debug_test_addr(&sin);
+ req.debug = sip_debug_test_addr(&sin);
if (pedanticsipchecking)
- req.len = lws2sws(req.data, req.len);
- if (debug)
+ req.len = lws2sws(req.data, req.len); /* Fix multiline headers */
+ if (req.debug)
ast_verbose("\n<-- SIP read from %s:%d: \n%s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), req.data);
- parse(&req);
- if (debug) {
+ parse_request(&req);
+ req.method = find_sip_method(req.rlPart1);
+ if (req.debug) {
ast_verbose("--- (%d headers %d lines)", req.headers, req.lines);
if (req.headers + req.lines == 0)
ast_verbose(" Nat keepalive ");
@@ -9733,9 +9741,6 @@
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:
@@ -10100,7 +10105,7 @@
/*--- sip_request: PBX interface function -build SIP pvt structure ---*/
/* SIP calls initiated by the PBX arrive here */
-static struct ast_channel *sip_request(const char *type, int format, void *data, int *cause)
+static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause)
{
int oldformat;
struct sip_pvt *p;
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