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merge formatting and minor code simplifications from trunk

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@44992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
Luigi Rizzo 19 years ago
parent 0c4d8edca6
commit 0e230d79d8
1 changed files with 147 additions and 157 deletions
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304
channels/chan_sip.c
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@ -181,10 +181,10 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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 void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_
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 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
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 char *regstate2str(enum sipregistrystate regstate) attribute_const;
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 @@ static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *
/*------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 @@ static int find_sdp(struct sip_request *req)
/*! \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 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
/* 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 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
}
}
break;
case 403: /* Forbidden */
/* First we ACK */
transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
@ -11514,22 +11512,26 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
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 @@ static int handle_response_register(struct sip_pvt *p, int resp, char *rest, str
}
/*! \brief Handle qualification responses (OPTIONS) */
static int handle_response_peerpoke(struct sip_pvt *p, int resp, struct sip_request *req)
static void 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);
}
}
struct sip_peer *peer = p->relatedpeer;
int statechanged, is_reachable, was_reachable;
int pingtime = ast_tvdiff_ms(ast_tvnow(), peer->ps);
if (peer->pokeexpire > -1)
ast_sched_del(sched, peer->pokeexpire);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
/*
* 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;
/* 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);
/* 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);
}
return 1;
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 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
/* 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 */
@ -13904,73 +13899,73 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req,
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 */
/* Initialize tag for new subscriptions */
if (ast_strlen_zero(p->tag))
make_our_tag(p->tag, sizeof(p->tag));
/* 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;
}
if (!strcmp(event, "presence") || !strcmp(event, "dialog")) { /* Presence, RFC 3842 */
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 */
/* 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;
if (p && !ast_test_flag(&p->flags[0], SIP_NEEDDESTROY)) {
@ -14001,48 +13996,47 @@ static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req,
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 @@ static int reload_config(enum channelreloadreason reason)
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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