[svn-commits] rizzo: branch 1.4 r44992 -
/branches/1.4/channels/chan_sip.c
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Thu Oct 12 15:07:12 MST 2006
Author: rizzo
Date: Thu Oct 12 17:07:11 2006
New Revision: 44992
URL: http://svn.digium.com/view/asterisk?rev=44992&view=rev
Log:
merge formatting and minor code simplifications from trunk
Modified:
branches/1.4/channels/chan_sip.c
Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?rev=44992&r1=44991&r2=44992&view=diff
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Thu Oct 12 17:07:11 2006
@@ -181,10 +181,10 @@
below EXPIRY_GUARD_LIMIT */
#define DEFAULT_EXPIRY 900 /*!< Expire slowly */
-static int min_expiry; /*!< Minimum accepted registration time */
-static int max_expiry; /*!< Maximum accepted registration time */
-static int default_expiry;
-static int expiry;
+static int min_expiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted registration time */
+static int max_expiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted registration time */
+static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
+static int expiry = DEFAULT_EXPIRY;
#ifndef MAX
#define MAX(a,b) ((a) > (b) ? (a) : (b))
@@ -1260,7 +1260,6 @@
static int add_sdp(struct sip_request *resp, struct sip_pvt *p);
/*--- Authentication stuff */
-static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req, char *header, char *respheader, int sipmethod, int init);
static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
static int build_reply_digest(struct sip_pvt *p, int method, char *digest, int digest_len);
static int clear_realm_authentication(struct sip_auth *authlist); /* Clear realm authentication list (at reload) */
@@ -1273,8 +1272,6 @@
int sipmethod, char *uri, enum xmittype reliable,
struct sockaddr_in *sin, struct sip_peer **authpeer);
static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin);
-static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req, char *header, char *respheader, int sipmethod, int init);
-static int build_reply_digest(struct sip_pvt *p, int method, char* digest, int digest_len);
/*--- Domain handling */
static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */
@@ -1413,7 +1410,6 @@
static int sip_reregister(void *data);
static int __sip_do_register(struct sip_registry *r);
static int sip_reg_timeout(void *data);
-static int do_register_auth(struct sip_pvt *p, struct sip_request *req, char *header, char *respheader);
static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
static void sip_send_all_registers(void);
@@ -1497,7 +1493,6 @@
/*------Response handling functions */
static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
static void handle_response_refer(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno);
-static int handle_response_peerpoke(struct sip_pvt *p, int resp, struct sip_request *req);
static int handle_response_register(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno);
static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int ignore, int seqno);
@@ -4537,6 +4532,8 @@
/*! \brief Process SIP SDP offer, select formats and activate RTP channels
If offer is rejected, we will not change any properties of the call
+ Return 0 on success, a negative value on errors.
+ Must be called after find_sdp().
*/
static int process_sdp(struct sip_pvt *p, struct sip_request *req)
{
@@ -11341,10 +11338,7 @@
/* RFC3261 says we must treat every 1xx response (but not 100)
that we don't recognize as if it was 183.
*/
- if ((resp > 100) &&
- (resp < 200) &&
- (resp != 180) &&
- (resp != 183))
+ if (resp > 100 && resp < 200 && resp != 180 && resp != 183)
resp = 183;
switch (resp) {
@@ -11353,6 +11347,7 @@
sip_cancel_destroy(p);
check_pendings(p);
break;
+
case 180: /* 180 Ringing */
if (!ast_test_flag(req, SIP_PKT_IGNORE))
sip_cancel_destroy(p);
@@ -11372,6 +11367,7 @@
ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
+
case 183: /* Session progress */
if (!ast_test_flag(req, SIP_PKT_IGNORE))
sip_cancel_destroy(p);
@@ -11386,6 +11382,7 @@
ast_set_flag(&p->flags[0], SIP_CAN_BYE);
check_pendings(p);
break;
+
case 200: /* 200 OK on invite - someone's answering our call */
if (!ast_test_flag(req, SIP_PKT_IGNORE))
sip_cancel_destroy(p);
@@ -11505,6 +11502,7 @@
}
}
break;
+
case 403: /* Forbidden */
/* First we ACK */
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
@@ -11514,22 +11512,26 @@
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
break;
+
case 404: /* Not found */
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);
break;
+
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, FALSE);
break;
+
case 491: /* Pending */
/* we have to wait a while, then retransmit */
/* Transmission is rescheduled, so everything should be taken care of.
We should support the retry-after at some point */
break;
+
case 501: /* Not implemented */
if (p->owner)
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
@@ -11717,57 +11719,50 @@
}
/*! \brief Handle qualification responses (OPTIONS) */
-static int handle_response_peerpoke(struct sip_pvt *p, int resp, struct sip_request *req)
-{
- struct sip_peer *peer;
- int pingtime;
- struct timeval tv;
-
- if (resp != 100) {
- int statechanged = 0;
- int newstate = 0;
- peer = p->relatedpeer;
- gettimeofday(&tv, NULL);
- pingtime = ast_tvdiff_ms(tv, peer->ps);
- if (pingtime < 1)
- pingtime = 1;
- if ((peer->lastms < 0) || (peer->lastms > peer->maxms)) {
- if (pingtime <= peer->maxms) {
- ast_log(LOG_NOTICE, "Peer '%s' is now REACHABLE! (%dms / %dms)\n", peer->name, pingtime, peer->maxms);
- statechanged = 1;
- newstate = 1;
- }
- } else if ((peer->lastms > 0) && (peer->lastms <= peer->maxms)) {
- if (pingtime > peer->maxms) {
- ast_log(LOG_NOTICE, "Peer '%s' is now TOO LAGGED! (%dms / %dms)\n", peer->name, pingtime, peer->maxms);
- statechanged = 1;
- newstate = 2;
- }
- }
- if (!peer->lastms)
- statechanged = 1;
- peer->lastms = pingtime;
- peer->call = NULL;
- if (statechanged) {
- ast_device_state_changed("SIP/%s", peer->name);
- if (newstate == 2) {
- manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Lagged\r\nTime: %d\r\n", peer->name, pingtime);
- } else {
- manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "Peer: SIP/%s\r\nPeerStatus: Reachable\r\nTime: %d\r\n", peer->name, pingtime);
- }
- }
-
- if (peer->pokeexpire > -1)
- ast_sched_del(sched, peer->pokeexpire);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
-
- /* Try again eventually */
- if ((peer->lastms < 0) || (peer->lastms > peer->maxms))
- peer->pokeexpire = ast_sched_add(sched, DEFAULT_FREQ_NOTOK, sip_poke_peer_s, peer);
- else
- peer->pokeexpire = ast_sched_add(sched, DEFAULT_FREQ_OK, sip_poke_peer_s, peer);
- }
- return 1;
+static void handle_response_peerpoke(struct sip_pvt *p, int resp, struct sip_request *req)
+{
+ struct sip_peer *peer = p->relatedpeer;
+ int statechanged, is_reachable, was_reachable;
+ int pingtime = ast_tvdiff_ms(ast_tvnow(), peer->ps);
+
+ /*
+ * Compute the response time to a ping (goes in peer->lastms.)
+ * -1 means did not respond, 0 means unknown,
+ * 1..maxms is a valid response, >maxms means late response.
+ */
+ if (pingtime < 1) /* zero = unknown, so round up to 1 */
+ pingtime = 1;
+
+ /* Now determine new state and whether it has changed.
+ * Use some helper variables to simplify the writing
+ * of the expressions.
+ */
+ was_reachable = peer->lastms > 0 && peer->lastms <= peer->maxms;
+ is_reachable = pingtime <= peer->maxms;
+ statechanged = peer->lastms == 0 /* yes, unknown before */
+ || was_reachable != is_reachable;
+
+ peer->lastms = pingtime;
+ peer->call = NULL;
+ if (statechanged) {
+ const char *s = is_reachable ? "Reachable" : "Lagged";
+
+ ast_log(LOG_NOTICE, "Peer '%s' is now %s. (%dms / %dms)\n",
+ peer->name, s, pingtime, peer->maxms);
+ ast_device_state_changed("SIP/%s", peer->name);
+ manager_event(EVENT_FLAG_SYSTEM, "PeerStatus",
+ "Peer: SIP/%s\r\nPeerStatus: %s\r\nTime: %d\r\n",
+ peer->name, s, pingtime);
+ }
+
+ if (peer->pokeexpire > -1)
+ ast_sched_del(sched, peer->pokeexpire);
+ ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+
+ /* Try again eventually */
+ peer->pokeexpire = ast_sched_add(sched,
+ is_reachable ? DEFAULT_FREQ_OK : DEFAULT_FREQ_NOTOK,
+ sip_poke_peer_s, peer);
}
/*! \brief Immediately stop RTP, VRTP and UDPTL as applicable */
@@ -11819,8 +11814,8 @@
/* We don't really care what the response is, just that it replied back.
Well, as long as it's not a 100 response... since we might
need to hang around for something more "definitive" */
-
- res = handle_response_peerpoke(p, resp, req);
+ if (resp != 100)
+ handle_response_peerpoke(p, resp, req);
} else if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
switch(resp) {
case 100: /* 100 Trying */
@@ -13904,72 +13899,72 @@
transmit_response(p, "404 Not Found", req);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
return 0;
- } else {
- /* XXX reduce nesting here */
- /* Initialize tag for new subscriptions */
- if (ast_strlen_zero(p->tag))
- make_our_tag(p->tag, sizeof(p->tag));
-
- if (!strcmp(event, "presence") || !strcmp(event, "dialog")) { /* Presence, RFC 3842 */
-
- /* Header from Xten Eye-beam Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/xpidf+xml */
- /* Polycom phones only handle xpidf+xml, even if they say they can
- handle pidf+xml as well
- */
- if (strstr(p->useragent, "Polycom")) {
- p->subscribed = XPIDF_XML;
- } else if (strstr(accept, "application/pidf+xml")) {
- p->subscribed = PIDF_XML; /* RFC 3863 format */
- } else if (strstr(accept, "application/dialog-info+xml")) {
- p->subscribed = DIALOG_INFO_XML;
- /* IETF draft: draft-ietf-sipping-dialog-package-05.txt */
- } else if (strstr(accept, "application/cpim-pidf+xml")) {
- p->subscribed = CPIM_PIDF_XML; /* RFC 3863 format */
- } else if (strstr(accept, "application/xpidf+xml")) {
- p->subscribed = XPIDF_XML; /* Early pre-RFC 3863 format with MSN additions (Microsoft Messenger) */
- } else {
- /* Can't find a format for events that we know about */
- transmit_response(p, "489 Bad Event", req);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
- return 0;
- }
- } else if (!strcmp(event, "message-summary")) {
- if (!ast_strlen_zero(accept) && strcmp(accept, "application/simple-message-summary")) {
- /* Format requested that we do not support */
- transmit_response(p, "406 Not Acceptable", req);
- if (option_debug > 1)
- ast_log(LOG_DEBUG, "Received SIP mailbox subscription for unknown format: %s\n", accept);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
- return 0;
- }
- /* Looks like they actually want a mailbox status
- This version of Asterisk supports mailbox subscriptions
- The subscribed URI needs to exist in the dial plan
- In most devices, this is configurable to the voicemailmain extension you use
- */
- if (!authpeer || ast_strlen_zero(authpeer->mailbox)) {
- transmit_response(p, "404 Not found (no mailbox)", req);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
- ast_log(LOG_NOTICE, "Received SIP subscribe for peer without mailbox: %s\n", authpeer->name);
- return 0;
- }
-
- p->subscribed = MWI_NOTIFICATION;
- if (authpeer->mwipvt && authpeer->mwipvt != p) /* Destroy old PVT if this is a new one */
- /* We only allow one subscription per peer */
- sip_destroy(authpeer->mwipvt);
- authpeer->mwipvt = p; /* Link from peer to pvt */
- p->relatedpeer = authpeer; /* Link from pvt to peer */
- } else { /* At this point, Asterisk does not understand the specified event */
+ }
+
+ /* Initialize tag for new subscriptions */
+ if (ast_strlen_zero(p->tag))
+ make_our_tag(p->tag, sizeof(p->tag));
+
+ if (!strcmp(event, "presence") || !strcmp(event, "dialog")) { /* Presence, RFC 3842 */
+
+ /* Header from Xten Eye-beam Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/xpidf+xml */
+ /* Polycom phones only handle xpidf+xml, even if they say they can
+ handle pidf+xml as well
+ */
+ if (strstr(p->useragent, "Polycom")) {
+ p->subscribed = XPIDF_XML;
+ } else if (strstr(accept, "application/pidf+xml")) {
+ p->subscribed = PIDF_XML; /* RFC 3863 format */
+ } else if (strstr(accept, "application/dialog-info+xml")) {
+ p->subscribed = DIALOG_INFO_XML;
+ /* IETF draft: draft-ietf-sipping-dialog-package-05.txt */
+ } else if (strstr(accept, "application/cpim-pidf+xml")) {
+ p->subscribed = CPIM_PIDF_XML; /* RFC 3863 format */
+ } else if (strstr(accept, "application/xpidf+xml")) {
+ p->subscribed = XPIDF_XML; /* Early pre-RFC 3863 format with MSN additions (Microsoft Messenger) */
+ } else {
+ /* Can't find a format for events that we know about */
transmit_response(p, "489 Bad Event", req);
+ ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ return 0;
+ }
+ } else if (!strcmp(event, "message-summary")) {
+ if (!ast_strlen_zero(accept) && strcmp(accept, "application/simple-message-summary")) {
+ /* Format requested that we do not support */
+ transmit_response(p, "406 Not Acceptable", req);
if (option_debug > 1)
- ast_log(LOG_DEBUG, "Received SIP subscribe for unknown event package: %s\n", event);
- ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ ast_log(LOG_DEBUG, "Received SIP mailbox subscription for unknown format: %s\n", accept);
+ ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
return 0;
}
- if (p->subscribed != MWI_NOTIFICATION && !resubscribe)
- p->stateid = ast_extension_state_add(p->context, p->exten, cb_extensionstate, p);
- }
+ /* Looks like they actually want a mailbox status
+ This version of Asterisk supports mailbox subscriptions
+ The subscribed URI needs to exist in the dial plan
+ In most devices, this is configurable to the voicemailmain extension you use
+ */
+ if (!authpeer || ast_strlen_zero(authpeer->mailbox)) {
+ transmit_response(p, "404 Not found (no mailbox)", req);
+ ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ ast_log(LOG_NOTICE, "Received SIP subscribe for peer without mailbox: %s\n", authpeer->name);
+ return 0;
+ }
+
+ p->subscribed = MWI_NOTIFICATION;
+ if (authpeer->mwipvt && authpeer->mwipvt != p) /* Destroy old PVT if this is a new one */
+ /* We only allow one subscription per peer */
+ sip_destroy(authpeer->mwipvt);
+ authpeer->mwipvt = p; /* Link from peer to pvt */
+ p->relatedpeer = authpeer; /* Link from pvt to peer */
+ } else { /* At this point, Asterisk does not understand the specified event */
+ transmit_response(p, "489 Bad Event", req);
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "Received SIP subscribe for unknown event package: %s\n", event);
+ ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
+ return 0;
+ }
+
+ if (p->subscribed != MWI_NOTIFICATION && !resubscribe)
+ p->stateid = ast_extension_state_add(p->context, p->exten, cb_extensionstate, p);
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p)
p->lastinvite = seqno;
@@ -14001,48 +13996,47 @@
ASTOBJ_UNLOCK(p->relatedpeer);
}
} else {
+ struct sip_pvt *p_old;
+
if ((firststate = ast_extension_state(NULL, p->context, p->exten)) < 0) {
ast_log(LOG_ERROR, "Got SUBSCRIBE for extension %s@%s from %s, but there is no hint for that extension\n", p->exten, p->context, ast_inet_ntoa(p->sa.sin_addr));
transmit_response(p, "404 Not found", req);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
return 0;
- } else {
- /* XXX reduce nesting here */
- struct sip_pvt *p_old;
-
- transmit_response(p, "200 OK", req);
- transmit_state_notify(p, firststate, 1); /* Send first notification */
- append_history(p, "Subscribestatus", "%s", ast_extension_state2str(firststate));
- /* hide the 'complete' exten/context in the refer_to field for later display */
- ast_string_field_build(p, subscribeuri, "%s@%s", p->exten, p->context);
-
- /* remove any old subscription from this peer for the same exten/context,
- as the peer has obviously forgotten about it and it's wasteful to wait
- for it to expire and send NOTIFY messages to the peer only to have them
- ignored (or generate errors)
- */
- ast_mutex_lock(&iflock);
- for (p_old = iflist; p_old; p_old = p_old->next) {
- if (p_old == p)
- continue;
- if (p_old->initreq.method != SIP_SUBSCRIBE)
- continue;
- if (p_old->subscribed == NONE)
- continue;
- ast_mutex_lock(&p_old->lock);
- if (!strcmp(p_old->username, p->username)) {
- if (!strcmp(p_old->exten, p->exten) &&
- !strcmp(p_old->context, p->context)) {
- ast_set_flag(&p_old->flags[0], SIP_NEEDDESTROY);
- ast_mutex_unlock(&p_old->lock);
- break;
- }
+ }
+
+ transmit_response(p, "200 OK", req);
+ transmit_state_notify(p, firststate, 1); /* Send first notification */
+ append_history(p, "Subscribestatus", "%s", ast_extension_state2str(firststate));
+ /* hide the 'complete' exten/context in the refer_to field for later display */
+ ast_string_field_build(p, subscribeuri, "%s@%s", p->exten, p->context);
+
+ /* remove any old subscription from this peer for the same exten/context,
+ as the peer has obviously forgotten about it and it's wasteful to wait
+ for it to expire and send NOTIFY messages to the peer only to have them
+ ignored (or generate errors)
+ */
+ ast_mutex_lock(&iflock);
+ for (p_old = iflist; p_old; p_old = p_old->next) {
+ if (p_old == p)
+ continue;
+ if (p_old->initreq.method != SIP_SUBSCRIBE)
+ continue;
+ if (p_old->subscribed == NONE)
+ continue;
+ ast_mutex_lock(&p_old->lock);
+ if (!strcmp(p_old->username, p->username)) {
+ if (!strcmp(p_old->exten, p->exten) &&
+ !strcmp(p_old->context, p->context)) {
+ ast_set_flag(&p_old->flags[0], SIP_NEEDDESTROY);
+ ast_mutex_unlock(&p_old->lock);
+ break;
}
- ast_mutex_unlock(&p_old->lock);
}
- ast_mutex_unlock(&iflock);
+ ast_mutex_unlock(&p_old->lock);
}
+ ast_mutex_unlock(&iflock);
}
if (!p->expiry)
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
@@ -15685,10 +15679,6 @@
allow_external_domains = DEFAULT_ALLOW_EXT_DOM; /* Allow external invites */
global_regcontext[0] = '\0';
expiry = DEFAULT_EXPIRY;
- min_expiry = DEFAULT_MIN_EXPIRY; /*!< Minimum accepted registration time */
- max_expiry = DEFAULT_MAX_EXPIRY; /*!< Maximum accepted registration time */
- default_expiry = DEFAULT_DEFAULT_EXPIRY;
- expiry = DEFAULT_EXPIRY;
global_notifyringing = DEFAULT_NOTIFYRINGING;
global_alwaysauthreject = 0;
global_allowsubscribe = FALSE;
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