[Asterisk-cvs] asterisk/channels chan_sip.c,1.767,1.768
kpfleming at lists.digium.com
kpfleming at lists.digium.com
Mon Jun 20 21:27:03 CDT 2005
Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv2571/channels
Modified Files:
chan_sip.c
Log Message:
begin properly supporting 'Supported' and 'Required' headers (bug #4351)
Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.767
retrieving revision 1.768
diff -u -d -r1.767 -r1.768
--- chan_sip.c 21 Jun 2005 01:07:31 -0000 1.767
+++ chan_sip.c 21 Jun 2005 01:27:54 -0000 1.768
@@ -183,12 +183,81 @@
{ "Via", "v" },
};
+/* Define SIP option tags, used in Require: and Supported: headers */
+/* We need to be aware of these properties in the phones to use
+ the replace: header. We should not do that without knowing
+ that the other end supports it...
+ This is nothing we can configure, we learn by the dialog
+ Supported: header on the REGISTER (peer) or the INVITE
+ (other devices)
+ We are not using many of these today, but will in the future.
+ This is documented in RFC 3261
+*/
+#define SUPPORTED 1
+#define NOT_SUPPORTED 0
-#define DEFAULT_SIP_PORT 5060 /* From RFC 3261 (former 2543) */
-#define SIP_MAX_PACKET 4096 /* Also from RFC 3261 (2543), should sub headers tho */
+#define SIP_OPT_REPLACES (1 << 0)
+#define SIP_OPT_100REL (1 << 1)
+#define SIP_OPT_TIMER (1 << 2)
+#define SIP_OPT_EARLY_SESSION (1 << 3)
+#define SIP_OPT_JOIN (1 << 4)
+#define SIP_OPT_PATH (1 << 5)
+#define SIP_OPT_PREF (1 << 6)
+#define SIP_OPT_PRECONDITION (1 << 7)
+#define SIP_OPT_PRIVACY (1 << 8)
+#define SIP_OPT_SDP_ANAT (1 << 9)
+#define SIP_OPT_SEC_AGREE (1 << 10)
+#define SIP_OPT_EVENTLIST (1 << 11)
+#define SIP_OPT_GRUU (1 << 12)
+#define SIP_OPT_TARGET_DIALOG (1 << 13)
+
+/* List of well-known SIP options. If we get this in a require,
+ we should check the list and answer accordingly. */
+const struct cfsip_options {
+ int id; /* Bitmap ID */
+ int supported; /* Supported by Asterisk ? */
+ char *text; /* Text id, as in standard */
+} sip_options[] = {
+ /* Replaces: header for transfer */
+ { SIP_OPT_REPLACES, SUPPORTED, "replaces" },
+ /* RFC3262: PRACK 100% reliability */
+ { SIP_OPT_100REL, NOT_SUPPORTED, "100rel" },
+ /* SIP Session Timers */
+ { SIP_OPT_TIMER, NOT_SUPPORTED, "timer" },
+ /* RFC3959: SIP Early session support */
+ { SIP_OPT_EARLY_SESSION, NOT_SUPPORTED, "early-session" },
+ /* SIP Join header support */
+ { SIP_OPT_JOIN, NOT_SUPPORTED, "join" },
+ /* RFC3327: Path support */
+ { SIP_OPT_PATH, NOT_SUPPORTED, "path" },
+ /* RFC3840: Callee preferences */
+ { SIP_OPT_PREF, NOT_SUPPORTED, "pref" },
+ /* RFC3312: Precondition support */
+ { SIP_OPT_PRECONDITION, NOT_SUPPORTED, "precondition" },
+ /* RFC3323: Privacy with proxies*/
+ { SIP_OPT_PRIVACY, NOT_SUPPORTED, "privacy" },
+ /* Not yet RFC, but registred with IANA */
+ { SIP_OPT_SDP_ANAT, NOT_SUPPORTED, "sdp_anat" },
+ /* RFC3329: Security agreement mechanism */
+ { SIP_OPT_SEC_AGREE, NOT_SUPPORTED, "sec_agree" },
+ /* SIMPLE events: draft-ietf-simple-event-list-07.txt */
+ { SIP_OPT_EVENTLIST, NOT_SUPPORTED, "eventlist" },
+ /* GRUU: Globally Routable User Agent URI's */
+ { SIP_OPT_GRUU, NOT_SUPPORTED, "gruu" },
+ /* Target-dialog: draft-ietf-sip-target-dialog-00.txt */
+ { SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED, "target-dialog" },
+};
+
+/* SIP Methods we support */
#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY"
+/* SIP Extensions we support */
+#define SUPPORTED_EXTENSIONS "replaces"
+
+#define DEFAULT_SIP_PORT 5060 /* From RFC 3261 (former 2543) */
+#define SIP_MAX_PACKET 4096 /* Also from RFC 3261 (2543), should sub headers tho */
+
static char default_useragent[AST_MAX_EXTENSION] = DEFAULT_USERAGENT;
#define DEFAULT_CONTEXT "default"
@@ -417,6 +486,7 @@
ast_group_t pickupgroup; /* Pickup group */
int lastinvite; /* Last Cseq of invite */
unsigned int flags; /* SIP_ flags */
+ unsigned int sipoptions; /* Supported SIP sipoptions on the other end */
int capability; /* Special capability (codec) */
int jointcapability; /* Supported capability at both ends (codecs ) */
int peercapability; /* Supported peer capability */
@@ -539,7 +609,8 @@
struct ast_codec_pref prefs; /* codec prefs */
ast_group_t callgroup; /* Call group */
ast_group_t pickupgroup; /* Pickup Group */
- unsigned int flags; /* SIP_ flags */
+ unsigned int flags; /* SIP flags */
+ unsigned int sipoptions; /* Supported SIP options */
struct ast_flags flags_page2; /* SIP_PAGE2 flags */
int amaflags; /* AMA flags for billing */
int callingpres; /* Calling id presentation */
@@ -582,7 +653,8 @@
struct ast_codec_pref prefs; /* codec prefs */
int lastmsgssent;
time_t lastmsgcheck; /* Last time we checked for MWI */
- unsigned int flags; /* SIP_ flags */
+ unsigned int flags; /* SIP flags */
+ unsigned int sipoptions; /* Supported SIP options */
struct ast_flags flags_page2; /* SIP_PAGE2 flags */
int expire; /* When to expire this peer registration */
int expiry; /* Duration of registration */
@@ -693,6 +765,7 @@
static struct ast_frame *sip_read(struct ast_channel *ast);
static int transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req);
static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
+static int transmit_response_with_unsupported(struct sip_pvt *p, char *msg, struct sip_request *req, char *unsupported);
static int transmit_response_with_auth(struct sip_pvt *p, char *msg, struct sip_request *req, char *rand, int reliable, char *header, int stale);
static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, int reliable, int newbranch);
static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int inc, int reliable, int newbranch);
@@ -728,6 +801,7 @@
static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */
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 */
/* Definition of this channel for channel registration */
static const struct ast_channel_tech sip_tech = {
@@ -781,6 +855,52 @@
return c;
}
+/*--- parse_sip_options: Parse supported header in incoming packet */
+unsigned int parse_sip_options(struct sip_pvt *pvt, char *supported)
+{
+ char *next = NULL;
+ char *sep = NULL;
+ char *temp = ast_strdupa(supported);
+ int i;
+ unsigned int profile = 0;
+
+ if (!supported || ast_strlen_zero(supported) )
+ return 0;
+
+ if (option_debug > 2 && sipdebug)
+ ast_log(LOG_DEBUG, "Begin: parsing SIP \"Supported: %s\"\n", supported);
+
+ next = temp;
+ while (next) {
+ char res=0;
+ if ( (sep = strchr(next, ',')) != NULL) {
+ *sep = '\0';
+ sep++;
+ }
+ while (*next == ' ') /* Skip spaces */
+ next++;
+ if (option_debug > 2 && sipdebug)
+ ast_log(LOG_DEBUG, "Found SIP option: -%s-\n", next);
+ for (i=0; (i < (sizeof(sip_options) / sizeof(sip_options[0]))) && !res; i++) {
+ if (!strcasecmp(next, sip_options[i].text)) {
+ profile |= sip_options[i].id;
+ res = 1;
+ if (option_debug > 2 && sipdebug)
+ ast_log(LOG_DEBUG, "Matched SIP option: %s\n", next);
+ }
+ }
+ if (!res)
+ if (option_debug > 2 && sipdebug)
+ ast_log(LOG_DEBUG, "Found no match for SIP option: %s (Please file bug report!)\n", next);
+ next = sep;
+ }
+ if (pvt)
+ pvt->sipoptions = profile;
+
+ ast_log(LOG_DEBUG, "* SIP extension value: %d for call %s\n", profile, pvt->callid);
+ return(profile);
+}
+
/*--- sip_debug_test_addr: See if we pass debug IP filter */
static inline int sip_debug_test_addr(struct sockaddr_in *addr)
{
@@ -3685,7 +3805,17 @@
return __transmit_response(p, msg, req, 0);
}
-/*--- transmit_response: Transmit response, Make sure you get a reply */
+/*--- transmit_response_with_unsupported: Transmit response, no retransmits */
+static int transmit_response_with_unsupported(struct sip_pvt *p, char *msg, struct sip_request *req, char *unsupported)
+{
+ struct sip_request resp;
+ respprep(&resp, p, msg, req);
+ append_date(&resp);
+ add_header(&resp, "Unsupported", unsupported);
+ return send_response(p, &resp, 0, 0);
+}
+
+/*--- transmit_response_reliable: Transmit response, Make sure you get a reply */
static int transmit_response_reliable(struct sip_pvt *p, char *msg, struct sip_request *req, int fatal)
{
return __transmit_response(p, msg, req, fatal ? 2 : 1);
@@ -5079,6 +5209,7 @@
register_peer_exten(p, 0);
p->fullcontact[0] = '\0';
p->useragent[0] = '\0';
+ p->sipoptions = 0;
p->lastms = 0;
if (option_verbose > 2)
@@ -5156,6 +5287,9 @@
ast_verbose(VERBOSE_PREFIX_3 "Registered SIP '%s' at %s port %d expires %d\n", p->name, ast_inet_ntoa(iabuf, sizeof(iabuf), p->addr.sin_addr), ntohs(p->addr.sin_port), expiry);
register_peer_exten(p, 1);
}
+
+ /* Save SIP options profile */
+ p->sipoptions = pvt->sipoptions;
/* Save User agent */
useragent = get_header(req, "User-Agent");
@@ -6112,6 +6246,10 @@
if (!(res = check_auth(p, req, p->randdata, sizeof(p->randdata), user->name, user->secret, user->md5secret, sipmethod, uri, reliable, ignore))) {
sip_cancel_destroy(p);
ast_copy_flags(p, user, SIP_PROMISCREDIR | SIP_DTMF | SIP_REINVITE);
+ /* Copy SIP extensions profile from INVITE */
+ if (p->sipoptions)
+ user->sipoptions = p->sipoptions;
+
/* If we have a call limit, set flag */
if (user->incominglimit)
ast_set_flag(p, SIP_CALL_LIMIT);
@@ -6165,6 +6303,11 @@
ast_verbose("Found peer '%s'\n", peer->name);
/* Take the peer */
ast_copy_flags(p, peer, SIP_TRUSTRPID | SIP_USECLIENTCODE | SIP_NAT | SIP_PROG_INBAND | SIP_OSPAUTH);
+
+ /* Copy SIP extensions profile to peer */
+ if (p->sipoptions)
+ peer->sipoptions = p->sipoptions;
+
/* replace callerid if rpid found, and not restricted */
if (!ast_strlen_zero(rpid_num) && ast_test_flag(p, SIP_TRUSTRPID)) {
if (*calleridname)
@@ -6937,6 +7080,16 @@
ast_cli(fd, " Addr->IP : %s Port %d\n", peer->addr.sin_addr.s_addr ? ast_inet_ntoa(iabuf, sizeof(iabuf), peer->addr.sin_addr) : "(Unspecified)", ntohs(peer->addr.sin_port));
ast_cli(fd, " Defaddr->IP : %s Port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), peer->defaddr.sin_addr), ntohs(peer->defaddr.sin_port));
ast_cli(fd, " Def. Username: %s\n", peer->username);
+ ast_cli(fd, " SIP Options : ");
+ if (peer->sipoptions) {
+ for (x=0 ; (x < (sizeof(sip_options) / sizeof(sip_options[0]))); x++) {
+ if (peer->sipoptions & sip_options[x].id)
+ ast_cli(fd, "%s ", sip_options[x].text);
+ }
+ } else
+ ast_cli(fd, "(none)");
+
+ ast_cli(fd, "\n");
ast_cli(fd, " Codecs : ");
ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->capability);
ast_cli(fd, "%s\n", codec_buf);
@@ -7395,7 +7548,17 @@
ast_cli(fd, " Last Message: %s\n", cur->lastmsg);
ast_cli(fd, " Promiscuous Redir: %s\n", ast_test_flag(cur, SIP_PROMISCREDIR) ? "Yes" : "No");
ast_cli(fd, " Route: %s\n", cur->route ? cur->route->hop : "N/A");
- ast_cli(fd, " DTMF Mode: %s\n\n", dtmfmode2str(ast_test_flag(cur, SIP_DTMF)));
+ ast_cli(fd, " DTMF Mode: %s\n", dtmfmode2str(ast_test_flag(cur, SIP_DTMF)));
+ ast_cli(fd, " SIP Options : ");
+ if (cur->sipoptions) {
+ int x;
+ for (x=0 ; (x < (sizeof(sip_options) / sizeof(sip_options[0]))); x++) {
+ if (cur->sipoptions & sip_options[x].id)
+ ast_cli(fd, "%s ", sip_options[x].text);
+ }
+ } else
+ ast_cli(fd, "(none)\n");
+ ast_cli(fd, "\n\n");
found++;
}
cur = cur->next;
@@ -8721,6 +8884,27 @@
struct ast_channel *c=NULL;
int gotdest;
struct ast_frame af = { AST_FRAME_NULL, };
+ char *supported;
+ char *required;
+ unsigned int required_profile = 0;
+
+ /* Find out what they support */
+ if (!p->sipoptions) {
+ supported = get_header(req, "Supported");
+ if (supported)
+ parse_sip_options(p, supported);
+ }
+ required = get_header(req, "Required");
+ if (required && !ast_strlen_zero(required)) {
+ required_profile = parse_sip_options(NULL, required);
+ if (required_profile) { /* They require something */
+ /* At this point we support no extensions, so fail */
+ transmit_response_with_unsupported(p, "420 Bad extension", req, required);
+ ast_set_flag(p, SIP_NEEDDESTROY);
+ return -1;
+
+ }
+ }
/* Check if this is a loop */
/* This happens since we do not properly support SIP domain
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