[asterisk-commits] branch oej/02-labarea r21238 -
/team/oej/02-labarea/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Apr 18 16:47:59 MST 2006
Author: oej
Date: Tue Apr 18 18:47:56 2006
New Revision: 21238
URL: http://svn.digium.com/view/asterisk?rev=21238&view=rev
Log:
Update
Modified:
team/oej/02-labarea/channels/chan_sip.c
Modified: team/oej/02-labarea/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/02-labarea/channels/chan_sip.c?rev=21238&r1=21237&r2=21238&view=diff
==============================================================================
--- team/oej/02-labarea/channels/chan_sip.c (original)
+++ team/oej/02-labarea/channels/chan_sip.c Tue Apr 18 18:47:56 2006
@@ -228,10 +228,11 @@
enum xmittype {
XMIT_CRITICAL = 2, /*!< Transmit critical SIP message reliably, with re-transmits.
- If it fails, it's critical and will cause a teardown of the session */
+ If it fails, it's critical and will cause a teardown of the session */
XMIT_RELIABLE = 1, /*!< Transmit SIP message reliably, with re-transmits */
XMIT_UNRELIABLE = 0, /*!< Transmit SIP message without bothering with re-transmits */
};
+
enum subscriptiontype {
NONE = 0,
@@ -317,6 +318,11 @@
{ SIP_INFO, NO_RTP, "INFO" },
{ SIP_CANCEL, NO_RTP, "CANCEL" },
{ SIP_PUBLISH, NO_RTP, "PUBLISH" }
+};
+
+enum sip_header {
+ SIP_HEADER_FROM,
+ SIP_HEADER_TO,
};
/*! Define SIP option tags, used in Require: and Supported: headers
@@ -1095,7 +1101,7 @@
static void sip_dump_history(struct sip_pvt *dialog); /* Dump history to LOG_DEBUG at end of dialog, before destroying data */
static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
static int transmit_state_notify(struct sip_pvt *p, int state, int full);
-static const char *gettag(const struct sip_request *req, char *header, char *tagbuf, int tagbufsize);
+static const char *gettag(const struct sip_request *req, enum sip_header header, char *tagbuf, int tagbufsize);
static int find_sip_method(const char *msg);
static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported);
static void sip_destroy(struct sip_pvt *p);
@@ -2758,7 +2764,7 @@
__sip_pretend_ack(p);
/* Send a new request: CANCEL */
- transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, XMIT_RELIABLE, TRUE);
+ 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);
@@ -3491,11 +3497,6 @@
char totag[128];
char fromtag[128];
- req->callid = get_header(req, "Call-ID");
- req->from = get_header(req, "From");
- req->to = get_header(req, "To");
- req->cseq = get_header(req, "Cseq");
- req->via = get_header(req, "Via");
//SKREP
ast_log(LOG_DEBUG, "---Find_call::: Call-ID: %s\n", req->callid);
@@ -3512,9 +3513,9 @@
For Asterisk to behave correctly, you need to turn on pedanticsipchecking
in sip.conf
*/
- if (gettag(req, "To", totag, sizeof(totag)))
+ if (gettag(req, SIP_HEADER_TO, totag, sizeof(totag)))
ast_set_flag(req, SIP_PKT_WITH_TOTAG); /* Used in handle_request/response */
- gettag(req, "From", fromtag, sizeof(fromtag));
+ gettag(req, SIP_HEADER_FROM, fromtag, sizeof(fromtag));
tag = (req->method == SIP_RESPONSE) ? totag : fromtag;
@@ -3760,6 +3761,11 @@
ast_log(LOG_WARNING, "Odd content, extra stuff left over ('%s')\n", c);
/* Split up the first line parts */
determine_firstline_parts(req);
+ req->callid = get_header(req, "Call-ID");
+ req->from = get_header(req, "From");
+ req->to = get_header(req, "To");
+ req->cseq = get_header(req, "Cseq");
+ req->via = get_header(req, "Via");
}
/*! \brief Process SIP SDP and activate RTP channels*/
@@ -3847,6 +3853,10 @@
vportno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
+ /* XXX Crashes on
+ m=video 7160/2 RTP/AVP 34
+ (RADvision)
+ */
if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
return -1;
@@ -4420,7 +4430,7 @@
snprintf(p->lastmsg, sizeof(p->lastmsg), "Tx: %s", sip_methods[sipmethod].text);
if (!seqno) {
- p->ocseq++;
+ p->ocseq++; /* Increment our Cseq */
seqno = p->ocseq;
}
@@ -10291,7 +10301,7 @@
if (ast_strlen_zero(p->theirtag) || (resp >= 200)) {
char tag[128];
- gettag(req, "To", tag, sizeof(tag));
+ gettag(req, SIP_HEADER_TO, tag, sizeof(tag));
ast_string_field_set(p, theirtag, tag);
}
if (p->relatedpeer && p->method == SIP_OPTIONS) {
@@ -10498,12 +10508,12 @@
if (ast_test_flag(req, SIP_PKT_DEBUG))
ast_verbose("SIP Response message for INCOMING dialog %s arrived\n", msg);
- if (resp == 200) {
+ if (resp == 200) {
/* Tags in early session is replaced by the tag in 200 OK, which is
the final reply to our INVITE */
- char tag[128];
-
- gettag(req, "To", tag, sizeof(tag));
+ char tag[BUFSIZ];
+
+ gettag(req, SIP_HEADER_TO, tag, sizeof(tag));
ast_string_field_set(p, theirtag, tag);
}
@@ -10750,24 +10760,25 @@
return 0;
}
-/*! \brief Get tag from packet */
-static const char *gettag(const struct sip_request *req, char *header, char *tagbuf, int tagbufsize)
-{
- const char *thetag;
+/*! \brief Get tag from incoming SIP packet */
+static const char *gettag(const struct sip_request *req, enum sip_header header, char *tagbuf, int tagbufsize)
+{
+ char *thetag;
if (!tagbuf)
return NULL;
tagbuf[0] = '\0'; /* reset the buffer */
- thetag = get_header(req, header);
- thetag = strcasestr(thetag, ";tag=");
- if (thetag) {
+ if (header == SIP_HEADER_FROM)
+ thetag = ast_strdupa(req->from);
+ else
+ thetag = ast_strdupa(req->to);
+ if ((thetag = strcasestr(thetag, ";tag="))) {
char *sep;
thetag += 5;
- ast_copy_string(tagbuf, thetag, tagbufsize);
- sep = strchr(tagbuf, ';');
- if (sep)
+ if ((sep = strchr(thetag, ';')))
*sep = '\0';
}
+ ast_copy_string(tagbuf, thetag, tagbufsize);
return thetag; /* XXX maybe this should be tagbuf instead ? */
}
@@ -11733,29 +11744,6 @@
if (option_debug > 3)
ast_log(LOG_DEBUG, "**** Received %s (%d) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd);
- /* Check for loop in INVITE */
- if (p->icseq && p->icseq == seqno && req->method == SIP_INVITE) {
- char *branch = ast_strdupa(req->via);
- char *tag;
- char *s;
- int ok = FALSE;
- if (ast_strlen_zero(branch)) {
- ast_log(LOG_DEBUG, "**** NO VIA HEADER in req->via\n");
- branch = ast_strdupa(get_header(req, "Via"));
- }
- tag = strcasestr(branch, ";branch=");
- if (tag) {
- if ((s = strchr(tag, ';')))
- *s = '\0';
- tag += 8;
- ast_log(LOG_DEBUG, "### Comparing our branch %s with request branch %s\n", p->viabranchtag, tag);
-
- }
- ok = TRUE;
- //SKREP
- ast_log(LOG_DEBUG, "**** Our initial VIA %s - This req VIA %s\n", p->initreq.via, req->via);
- //if (strcasecmp(p->viabranchtag, ???? )
- }
if (p->icseq && (p->icseq > seqno)) {
if (option_debug)
@@ -11783,11 +11771,37 @@
increasing */
p->icseq = seqno;
+ /* Check for loop in INVITE */
+ if (p->icseq && p->icseq == seqno && req->method == SIP_INVITE) {
+ char *branch = ast_strdupa(req->via);
+ char *tag;
+ char *s;
+ int ok = FALSE;
+ if (ast_strlen_zero(branch)) {
+ ast_log(LOG_DEBUG, "**** NO VIA HEADER in req->via\n");
+ branch = ast_strdupa(get_header(req, "Via"));
+ }
+ tag = strcasestr(branch, ";branch=");
+ if (tag) {
+ if ((s = strchr(tag, ';')))
+ *s = '\0';
+ tag += 8;
+ ast_log(LOG_DEBUG, "### Comparing our branch %s with request branch %s\n", p->viabranchtag, tag);
+
+ }
+ ok = TRUE;
+ //SKREP
+ ast_log(LOG_DEBUG, "**** Our initial VIA %s - This req VIA %s\n", p->initreq.via, req->via);
+ //if (strcasecmp(p->viabranchtag, ???? )
+ } else {
+ ast_log(LOG_DEBUG, "**** Different: p->icseq %d secno %d \n", p->icseq, seqno);
+ }
+
/* Find their tag if we haven't got it */
if (ast_strlen_zero(p->theirtag)) {
- char tag[128];
-
- gettag(req, "From", tag, sizeof(tag));
+ char tag[BUFSIZ];
+
+ gettag(req, SIP_HEADER_FROM, tag, sizeof(tag));
ast_string_field_set(p, theirtag, tag);
}
snprintf(p->lastmsg, sizeof(p->lastmsg), "Rx: %s", cmd);
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