[svn-commits] dvossel: trunk r244597 - in /trunk/channels: ./ sip/ sip/include/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Feb 3 14:33:36 CST 2010
Author: dvossel
Date: Wed Feb 3 14:33:32 2010
New Revision: 244597
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=244597
Log:
-----Changes -----
New files
- channels/sip/sip.h – A new header for shared #define, enum, and struct
definitions.
- channels/sip/include/sip_utils.h – sip util functions shared among
the all the sip APIs
- channels/sip/include/config_parser.h – sip config-parser API
- channels/sip/config_parser.c – Contains sip.conf parsing helper functions
with unit tests.
- channels/sip/include/reqresp_parser.h – sip request response parser API
- channels/sip/reqresp_parser.c – Contains sip request and response parsing
helper functions with unit tests.
New Unit Tests
- sip_parse_uri_test
- sip_parse_host_test
- sip_parse_register_line_test
Code Refactoring
- All reusable #define, enum, and struct definitions were moved out of chan_sip.c
into sip.h. During this process formatting changes were made to comments
in both sip.h and chan_sip.c in order to better adhere to the coding guidelines.
- The beginnings of three new sip APIs, sip-utils.h, config-parser.h,
reqresp-parser.h using existing chan_sip.c functions.
- parse_uri() and get_calleridname() were moved from chan_sip.c to request-parser.c
along with unit tests for both functions.
- sip_parse_host() and sip_parse_register_line() were moved from chan_sip.c to
config-parser.c along with unit tests for both functions.
Changes to parse_uri()
-removal of the options parameter. It was never used and did not behave correctly.
-additional check for [?header] field. When this field was present, the transport
type was not being set correctly.
----- Overview -----
This patch is introduced with the hope that unit tests for all our sip parsing
functions will be written soon. chan_sip is a huge file, and with the addition of
each unit test chan_sip is going to grow larger and harder to maintain. I'm proposing
we begin refactoring chan_sip, starting with the parsing functions. With each parsing
function we move into a separate helper file, a unit test should accompany it. I've
attempted to lay down the ground work for this change by creating two new parser
helper files (config-parser.c and reqresp-parser.c) and moving all shared structs,
enums, and defines from chan_sip.c into a shared sip.h file. We can't verify everything
in Asterisk using unit tests, but string parsing is one area where unit tests make
the most sense. By beginning to restructure the code in this way, chan_sip not only
becomes less bloated, but Asterisk as a whole will become more stable.
Review: https://reviewboard.asterisk.org/r/477/
Added:
trunk/channels/sip/
trunk/channels/sip/config_parser.c (with props)
trunk/channels/sip/include/
trunk/channels/sip/include/config_parser.h (with props)
trunk/channels/sip/include/reqresp_parser.h (with props)
trunk/channels/sip/include/sip.h (with props)
trunk/channels/sip/include/sip_utils.h (with props)
trunk/channels/sip/reqresp_parser.c (with props)
Modified:
trunk/channels/Makefile
trunk/channels/chan_sip.c
Modified: trunk/channels/Makefile
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/Makefile?view=diff&rev=244597&r1=244596&r2=244597
==============================================================================
--- trunk/channels/Makefile (original)
+++ trunk/channels/Makefile Wed Feb 3 14:33:32 2010
@@ -69,6 +69,7 @@
rm -f h323/Makefile
$(if $(filter chan_iax2,$(EMBEDDED_MODS)),modules.link,chan_iax2.so): iax2-parser.o iax2-provision.o
+$(if $(filter chan_sip,$(EMBEDDED_MODS)),modules.link,chan_sip.so): sip/config_parser.o sip/reqresp_parser.o
$(if $(filter chan_dahdi,$(EMBEDDED_MODS)),modules.link,chan_dahdi.so): sig_analog.o sig_pri.o
ifneq ($(filter chan_h323,$(EMBEDDED_MODS)),)
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=244597&r1=244596&r2=244597
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Feb 3 14:33:32 2010
@@ -218,7 +218,6 @@
#include "asterisk/paths.h" /* need ast_config_AST_SYSTEM_NAME */
#include "asterisk/lock.h"
-#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
@@ -230,7 +229,6 @@
#include "asterisk/manager.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
-#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/dsp.h"
#include "asterisk/features.h"
@@ -239,8 +237,6 @@
#include "asterisk/causes.h"
#include "asterisk/utils.h"
#include "asterisk/file.h"
-#include "asterisk/astobj.h"
-#include "asterisk/test.h"
/*
Uncomment the define below, if you are having refcount related memory leaks.
With this uncommented, this module will generate a file, /tmp/refs, which contains
@@ -256,8 +252,6 @@
#include "asterisk/astobj2.h"
#include "asterisk/dnsmgr.h"
#include "asterisk/devicestate.h"
-#include "asterisk/linkedlists.h"
-#include "asterisk/stringfields.h"
#include "asterisk/monitor.h"
#include "asterisk/netsock.h"
#include "asterisk/localtime.h"
@@ -266,10 +260,12 @@
#include "asterisk/translate.h"
#include "asterisk/ast_version.h"
#include "asterisk/event.h"
-#include "asterisk/tcptls.h"
#include "asterisk/stun.h"
#include "asterisk/cel.h"
-#include "asterisk/strings.h"
+#include "sip/include/sip.h"
+#include "sip/include/config_parser.h"
+#include "sip/include/reqresp_parser.h"
+#include "sip/include/sip_utils.h"
/*** DOCUMENTATION
<application name="SIPDtmfMode" language="en_US">
@@ -546,82 +542,10 @@
</manager>
***/
-#ifndef FALSE
-#define FALSE 0
-#endif
-
-#ifndef TRUE
-#define TRUE 1
-#endif
-
-/* Arguments for find_peer */
-#define FINDUSERS (1 << 0)
-#define FINDPEERS (1 << 1)
-#define FINDALLDEVICES (FINDUSERS | FINDPEERS)
-
-#define SIPBUFSIZE 512 /*!< Buffer size for many operations */
-
-#define XMIT_ERROR -2
-
-#define SIP_RESERVED ";/?:@&=+$,# " /*!< Reserved characters in the username part of the URI */
-
-#define DEFAULT_DEFAULT_EXPIRY 120
-#define DEFAULT_MIN_EXPIRY 60
-#define DEFAULT_MAX_EXPIRY 3600
-#define DEFAULT_MWI_EXPIRY 3600
-#define DEFAULT_REGISTRATION_TIMEOUT 20
-#define DEFAULT_MAX_FORWARDS "70"
-
-/* guard limit must be larger than guard secs */
-/* guard min must be < 1000, and should be >= 250 */
-#define EXPIRY_GUARD_SECS 15 /*!< How long before expiry do we reregister */
-#define EXPIRY_GUARD_LIMIT 30 /*!< Below here, we use EXPIRY_GUARD_PCT instead of
- EXPIRY_GUARD_SECS */
-#define EXPIRY_GUARD_MIN 500 /*!< This is the minimum guard time applied. If
- GUARD_PCT turns out to be lower than this, it
- will use this time instead.
- This is in milliseconds. */
-#define EXPIRY_GUARD_PCT 0.20 /*!< Percentage of expires timeout to use when
- below EXPIRY_GUARD_LIMIT */
-#define DEFAULT_EXPIRY 900 /*!< Expire slowly */
-
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 mwi_expiry = DEFAULT_MWI_EXPIRY;
-
-#define DEFAULT_QUALIFY_GAP 100
-#define DEFAULT_QUALIFY_PEERS 1
-
-
-#define CALLERID_UNKNOWN "Anonymous"
-#define FROMDOMAIN_INVALID "anonymous.invalid"
-
-#define DEFAULT_MAXMS 2000 /*!< Qualification: Must be faster than 2 seconds by default */
-#define DEFAULT_QUALIFYFREQ 60 * 1000 /*!< Qualification: How often to check for the host to be up */
-#define DEFAULT_FREQ_NOTOK 10 * 1000 /*!< Qualification: How often to check, if the host is down... */
-
-#define DEFAULT_RETRANS 1000 /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
-#define MAX_RETRANS 6 /*!< Try only 6 times for retransmissions, a total of 7 transmissions */
-#define DEFAULT_TIMER_T1 500 /*!< SIP timer T1 (according to RFC 3261) */
-#define SIP_TRANS_TIMEOUT 64 * DEFAULT_TIMER_T1 /*!< SIP request timeout (rfc 3261) 64*T1
- \todo Use known T1 for timeout (peerpoke)
- */
-#define DEFAULT_TRANS_TIMEOUT -1 /*!< Use default SIP transaction timeout */
-#define PROVIS_KEEPALIVE_TIMEOUT 60000 /*!< How long to wait before retransmitting a provisional response (rfc 3261 13.3.1.1) */
-#define MAX_AUTHTRIES 3 /*!< Try authentication three times, then fail */
-
-#define SIP_MAX_HEADERS 64 /*!< Max amount of SIP headers to read */
-#define SIP_MAX_LINES 64 /*!< Max amount of lines in SIP attachment (like SDP) */
-#define SIP_MIN_PACKET 4096 /*!< Initialize size of memory to allocate for packets */
-#define MAX_HISTORY_ENTRIES 50 /*!< Max entires in the history list for a sip_pvt */
-
-#define INITIAL_CSEQ 101 /*!< Our initial sip sequence number */
-
-#define DEFAULT_MAX_SE 1800 /*!< Session-Timer Default Session-Expires period (RFC 4028) */
-#define DEFAULT_MIN_SE 90 /*!< Session-Timer Default Min-SE period (RFC 4028) */
-
-#define SDP_MAX_RTPMAP_CODECS 32 /*!< Maximum number of codecs allowed in received SDP */
/*! \brief Global jitterbuffer configuration - by default, jb is disabled */
static struct ast_jb_conf default_jbconf =
@@ -631,47 +555,13 @@
.resync_threshold = -1,
.impl = ""
};
-static struct ast_jb_conf global_jbconf; /*!< Global jitterbuffer configuration */
-
-static const char config[] = "sip.conf"; /*!< Main configuration file */
-static const char notify_config[] = "sip_notify.conf"; /*!< Configuration file for sending Notify with CLI commands to reconfigure or reboot phones */
-
-#define RTP 1
-#define NO_RTP 0
-
-/*! \brief Authorization scheme for call transfers
-
-\note Not a bitfield flag, since there are plans for other modes,
- like "only allow transfers for authenticated devices" */
-enum transfermodes {
- TRANSFER_OPENFORALL, /*!< Allow all SIP transfers */
- TRANSFER_CLOSED, /*!< Allow no SIP transfers */
-};
-
-
-/*! \brief The result of a lot of functions */
-enum sip_result {
- AST_SUCCESS = 0, /*!< FALSE means success, funny enough */
- AST_FAILURE = -1, /*!< Failure code */
-};
-
-/*! \brief States for the INVITE transaction, not the dialog
- \note this is for the INVITE that sets up the dialog
-*/
-enum invitestates {
- INV_NONE = 0, /*!< No state at all, maybe not an INVITE dialog */
- INV_CALLING = 1, /*!< Invite sent, no answer */
- INV_PROCEEDING = 2, /*!< We got/sent 1xx message */
- INV_EARLY_MEDIA = 3, /*!< We got 18x message with to-tag back */
- INV_COMPLETED = 4, /*!< Got final response with error. Wait for ACK, then CONFIRMED */
- INV_CONFIRMED = 5, /*!< Confirmed response - we've got an ack (Incoming calls only) */
- INV_TERMINATED = 6, /*!< Transaction done - either successful (AST_STATE_UP) or failed, but done
- The only way out of this is a BYE from one side */
- INV_CANCELLED = 7, /*!< Transaction cancelled by client or server in non-terminated state */
-};
+static struct ast_jb_conf global_jbconf; /*!< Global jitterbuffer configuration */
+
+static const char config[] = "sip.conf"; /*!< Main configuration file */
+static const char notify_config[] = "sip_notify.conf"; /*!< Configuration file for sending Notify with CLI commands to reconfigure or reboot phones */
/*! \brief Readable descriptions of device states.
- \note Should be aligned to above table as index */
+ * \note Should be aligned to above table as index */
static const struct invstate2stringtable {
const enum invitestates state;
const char *desc;
@@ -686,52 +576,11 @@
{INV_CANCELLED, "Cancelled"}
};
-/*! \brief When sending a SIP message, we can send with a few options, depending on
- type of SIP request. UNRELIABLE is moslty used for responses to repeated requests,
- where the original response would be sent RELIABLE in an INVITE transaction */
-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 */
- XMIT_RELIABLE = 1, /*!< Transmit SIP message reliably, with re-transmits */
- XMIT_UNRELIABLE = 0, /*!< Transmit SIP message without bothering with re-transmits */
-};
-
-/*! \brief Results from the parse_register() function */
-enum parse_register_result {
- PARSE_REGISTER_DENIED,
- PARSE_REGISTER_FAILED,
- PARSE_REGISTER_UPDATE,
- PARSE_REGISTER_QUERY,
-};
-
-/*! \brief Type of subscription, based on the packages we do support, see \ref subscription_types */
-enum subscriptiontype {
- NONE = 0,
- XPIDF_XML,
- DIALOG_INFO_XML,
- CPIM_PIDF_XML,
- PIDF_XML,
- MWI_NOTIFICATION
-};
-
-/*! \brief The number of media types in enum \ref media_type below. */
-#define OFFERED_MEDIA_COUNT 4
-
-/*! \brief Media types generate different "dummy answers" for not accepting the offer of
- a media stream. We need to add definitions for each RTP profile. Secure RTP is not
- the same as normal RTP and will require a new definition */
-enum media_type {
- SDP_AUDIO, /*!< RTP/AVP Audio */
- SDP_VIDEO, /*!< RTP/AVP Video */
- SDP_IMAGE, /*!< Image udptl, not TCP or RTP */
- SDP_TEXT, /*!< RTP/AVP Realtime Text */
-};
-
/*! \brief Subscription types that we support. We support
- - dialoginfo updates (really device status, not dialog info as was the original intent of the standard)
- - SIMPLE presence used for device status
- - Voicemail notification subscriptions
-*/
+ * - dialoginfo updates (really device status, not dialog info as was the original intent of the standard)
+ * - SIMPLE presence used for device status
+ * - Voicemail notification subscriptions
+ */
static const struct cfsubscription_types {
enum subscriptiontype type;
const char * const event;
@@ -747,237 +596,42 @@
{ MWI_NOTIFICATION, "message-summary", "application/simple-message-summary", "mwi" } /* RFC 3842: Mailbox notification */
};
-
-/*! \brief Authentication types - proxy or www authentication
- \note Endpoints, like Asterisk, should always use WWW authentication to
- allow multiple authentications in the same call - to the proxy and
- to the end point.
-*/
-enum sip_auth_type {
- PROXY_AUTH = 407,
- WWW_AUTH = 401,
-};
-
-/*! \brief Authentication result from check_auth* functions */
-enum check_auth_result {
- AUTH_DONT_KNOW = -100, /*!< no result, need to check further */
- /* XXX maybe this is the same as AUTH_NOT_FOUND */
-
- AUTH_SUCCESSFUL = 0,
- AUTH_CHALLENGE_SENT = 1,
- AUTH_SECRET_FAILED = -1,
- AUTH_USERNAME_MISMATCH = -2,
- AUTH_NOT_FOUND = -3, /*!< returned by register_verify */
- AUTH_FAKE_AUTH = -4,
- AUTH_UNKNOWN_DOMAIN = -5,
- AUTH_PEER_NOT_DYNAMIC = -6,
- AUTH_ACL_FAILED = -7,
- AUTH_BAD_TRANSPORT = -8,
- AUTH_RTP_FAILED = 9,
-};
-
-/*! \brief States for outbound registrations (with register= lines in sip.conf */
-enum sipregistrystate {
- REG_STATE_UNREGISTERED = 0, /*!< We are not registered
- * \note Initial state. We should have a timeout scheduled for the initial
- * (or next) registration transmission, calling sip_reregister
- */
-
- REG_STATE_REGSENT, /*!< Registration request sent
- * \note sent initial request, waiting for an ack or a timeout to
- * retransmit the initial request.
- */
-
- REG_STATE_AUTHSENT, /*!< We have tried to authenticate
- * \note entered after transmit_register with auth info,
- * waiting for an ack.
- */
-
- REG_STATE_REGISTERED, /*!< Registered and done */
-
- REG_STATE_REJECTED, /*!< Registration rejected
- * \note only used when the remote party has an expire larger than
- * our max-expire. This is a final state from which we do not
- * recover (not sure how correctly).
- */
-
- REG_STATE_TIMEOUT, /*!< Registration timed out
- * \note XXX unused */
-
- REG_STATE_NOAUTH, /*!< We have no accepted credentials
- * \note fatal - no chance to proceed */
-
- REG_STATE_FAILED, /*!< Registration failed after several tries
- * \note fatal - no chance to proceed */
-};
-
-/*! \brief Modes in which Asterisk can be configured to run SIP Session-Timers */
-enum st_mode {
- SESSION_TIMER_MODE_INVALID = 0, /*!< Invalid value */
- SESSION_TIMER_MODE_ACCEPT, /*!< Honor inbound Session-Timer requests */
- SESSION_TIMER_MODE_ORIGINATE, /*!< Originate outbound and honor inbound requests */
- SESSION_TIMER_MODE_REFUSE /*!< Ignore inbound Session-Timers requests */
-};
-
-/*! \brief The entity playing the refresher role for Session-Timers */
-enum st_refresher {
- SESSION_TIMER_REFRESHER_AUTO, /*!< Negotiated */
- SESSION_TIMER_REFRESHER_UAC, /*!< Session is refreshed by the UAC */
- SESSION_TIMER_REFRESHER_UAS /*!< Session is refreshed by the UAS */
-};
-
-/*! \brief Define some implemented SIP transports
- \note Asterisk does not support SCTP or UDP/DTLS
-*/
-enum sip_transport {
- SIP_TRANSPORT_UDP = 1, /*!< Unreliable transport for SIP, needs retransmissions */
- SIP_TRANSPORT_TCP = 1 << 1, /*!< Reliable, but unsecure */
- SIP_TRANSPORT_TLS = 1 << 2, /*!< TCP/TLS - reliable and secure transport for signalling */
-};
-
-/*! \brief definition of a sip proxy server
+/*! \brief The core structure to setup dialogs. We parse incoming messages by using
+ * structure and then route the messages according to the type.
*
- * For outbound proxies, a sip_peer will contain a reference to a
- * dynamically allocated instance of a sip_proxy. A sip_pvt may also
- * contain a reference to a peer's outboundproxy, or it may contain
- * a reference to the sip_cfg.outboundproxy.
+ * \note Note that sip_methods[i].id == i must hold or the code breaks
*/
-struct sip_proxy {
- char name[MAXHOSTNAMELEN]; /*!< DNS name of domain/host or IP */
- struct sockaddr_in ip; /*!< Currently used IP address and port */
- time_t last_dnsupdate; /*!< When this was resolved */
- enum sip_transport transport;
- int force; /*!< If it's an outbound proxy, Force use of this outbound proxy for all outbound requests */
- /* Room for a SRV record chain based on the name */
-};
-
-/*! \brief argument for the 'show channels|subscriptions' callback. */
-struct __show_chan_arg {
- int fd;
- int subscriptions;
- int numchans; /* return value */
-};
-
-
-/*! \brief States whether a SIP message can create a dialog in Asterisk. */
-enum can_create_dialog {
- CAN_NOT_CREATE_DIALOG,
- CAN_CREATE_DIALOG,
- CAN_CREATE_DIALOG_UNSUPPORTED_METHOD,
-};
-
-/*! \brief SIP Request methods known by Asterisk
-
- \note Do _NOT_ make any changes to this enum, or the array following it;
- if you think you are doing the right thing, you are probably
- not doing the right thing. If you think there are changes
- needed, get someone else to review them first _before_
- submitting a patch. If these two lists do not match properly
- bad things will happen.
-*/
-
-enum sipmethod {
- SIP_UNKNOWN, /*!< Unknown response */
- SIP_RESPONSE, /*!< Not request, response to outbound request */
- SIP_REGISTER, /*!< Registration to the mothership, tell us where you are located */
- SIP_OPTIONS, /*!< Check capabilities of a device, used for "ping" too */
- SIP_NOTIFY, /*!< Status update, Part of the event package standard, result of a SUBSCRIBE or a REFER */
- SIP_INVITE, /*!< Set up a session */
- SIP_ACK, /*!< End of a three-way handshake started with INVITE. */
- SIP_PRACK, /*!< Reliable pre-call signalling. Not supported in Asterisk. */
- SIP_BYE, /*!< End of a session */
- SIP_REFER, /*!< Refer to another URI (transfer) */
- SIP_SUBSCRIBE, /*!< Subscribe for updates (voicemail, session status, device status, presence) */
- SIP_MESSAGE, /*!< Text messaging */
- SIP_UPDATE, /*!< Update a dialog. We can send UPDATE; but not accept it */
- SIP_INFO, /*!< Information updates during a session */
- SIP_CANCEL, /*!< Cancel an INVITE */
- SIP_PUBLISH, /*!< Not supported in Asterisk */
- SIP_PING, /*!< Not supported at all, no standard but still implemented out there */
-};
-
-/*! \brief Settings for the 'notifycid' option, see sip.conf.sample for details. */
-enum notifycid_setting {
- DISABLED = 0,
- ENABLED = 1,
- IGNORE_CONTEXT = 2,
-};
-
-/*! \brief The core structure to setup dialogs. We parse incoming messages by using
- structure and then route the messages according to the type.
-
- \note Note that sip_methods[i].id == i must hold or the code breaks */
static const struct cfsip_methods {
enum sipmethod id;
int need_rtp; /*!< when this is the 'primary' use for a pvt structure, does it need RTP? */
char * const text;
enum can_create_dialog can_create;
} sip_methods[] = {
- { SIP_UNKNOWN, RTP, "-UNKNOWN-", CAN_CREATE_DIALOG },
- { SIP_RESPONSE, NO_RTP, "SIP/2.0", CAN_NOT_CREATE_DIALOG },
- { SIP_REGISTER, NO_RTP, "REGISTER", CAN_CREATE_DIALOG },
- { SIP_OPTIONS, NO_RTP, "OPTIONS", CAN_CREATE_DIALOG },
- { SIP_NOTIFY, NO_RTP, "NOTIFY", CAN_CREATE_DIALOG },
- { SIP_INVITE, RTP, "INVITE", CAN_CREATE_DIALOG },
- { SIP_ACK, NO_RTP, "ACK", CAN_NOT_CREATE_DIALOG },
- { SIP_PRACK, NO_RTP, "PRACK", CAN_NOT_CREATE_DIALOG },
- { SIP_BYE, NO_RTP, "BYE", CAN_NOT_CREATE_DIALOG },
- { SIP_REFER, NO_RTP, "REFER", CAN_CREATE_DIALOG },
- { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE", CAN_CREATE_DIALOG },
- { SIP_MESSAGE, NO_RTP, "MESSAGE", CAN_CREATE_DIALOG },
- { SIP_UPDATE, NO_RTP, "UPDATE", CAN_NOT_CREATE_DIALOG },
- { SIP_INFO, NO_RTP, "INFO", CAN_NOT_CREATE_DIALOG },
- { SIP_CANCEL, NO_RTP, "CANCEL", CAN_NOT_CREATE_DIALOG },
- { SIP_PUBLISH, NO_RTP, "PUBLISH", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD },
- { SIP_PING, NO_RTP, "PING", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
+ { SIP_UNKNOWN, RTP, "-UNKNOWN-",CAN_CREATE_DIALOG },
+ { SIP_RESPONSE, NO_RTP, "SIP/2.0", CAN_NOT_CREATE_DIALOG },
+ { SIP_REGISTER, NO_RTP, "REGISTER", CAN_CREATE_DIALOG },
+ { SIP_OPTIONS, NO_RTP, "OPTIONS", CAN_CREATE_DIALOG },
+ { SIP_NOTIFY, NO_RTP, "NOTIFY", CAN_CREATE_DIALOG },
+ { SIP_INVITE, RTP, "INVITE", CAN_CREATE_DIALOG },
+ { SIP_ACK, NO_RTP, "ACK", CAN_NOT_CREATE_DIALOG },
+ { SIP_PRACK, NO_RTP, "PRACK", CAN_NOT_CREATE_DIALOG },
+ { SIP_BYE, NO_RTP, "BYE", CAN_NOT_CREATE_DIALOG },
+ { SIP_REFER, NO_RTP, "REFER", CAN_CREATE_DIALOG },
+ { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE",CAN_CREATE_DIALOG },
+ { SIP_MESSAGE, NO_RTP, "MESSAGE", CAN_CREATE_DIALOG },
+ { SIP_UPDATE, NO_RTP, "UPDATE", CAN_NOT_CREATE_DIALOG },
+ { SIP_INFO, NO_RTP, "INFO", CAN_NOT_CREATE_DIALOG },
+ { SIP_CANCEL, NO_RTP, "CANCEL", CAN_NOT_CREATE_DIALOG },
+ { SIP_PUBLISH, NO_RTP, "PUBLISH", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD },
+ { SIP_PING, NO_RTP, "PING", CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
};
-
-static unsigned int chan_idx;
-
-/*! 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
-
-/* SIP options */
-#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)
-#define SIP_OPT_NOREFERSUB (1 << 14)
-#define SIP_OPT_HISTINFO (1 << 15)
-#define SIP_OPT_RESPRIORITY (1 << 16)
-#define SIP_OPT_FROMCHANGE (1 << 17)
-#define SIP_OPT_RECLISTINV (1 << 18)
-#define SIP_OPT_RECLISTSUB (1 << 19)
-#define SIP_OPT_OUTBOUND (1 << 20)
-#define SIP_OPT_UNKNOWN (1 << 21)
-
/*! \brief List of well-known SIP options. If we get this in a require,
we should check the list and answer accordingly. */
static const struct cfsip_options {
- int id; /*!< Bitmap ID */
- int supported; /*!< Supported by Asterisk ? */
- char * const text; /*!< Text id, as in standard */
+ int id; /*!< Bitmap ID */
+ int supported; /*!< Supported by Asterisk ? */
+ char * const text; /*!< Text id, as in standard */
} sip_options[] = { /* XXX used in 3 places */
/* RFC3262: PRACK 100% reliability */
{ SIP_OPT_100REL, NOT_SUPPORTED, "100rel" },
@@ -1050,175 +704,29 @@
{ AST_REDIRECTING_REASON_CALL_FWD_DTE, "unknown"}
};
-static enum AST_REDIRECTING_REASON sip_reason_str_to_code(const char *text)
-{
- enum AST_REDIRECTING_REASON ast = AST_REDIRECTING_REASON_UNKNOWN;
- int i;
-
- for (i = 0; i < ARRAY_LEN(sip_reason_table); ++i) {
- if (!strcasecmp(text, sip_reason_table[i].text)) {
- ast = sip_reason_table[i].code;
- break;
- }
- }
-
- return ast;
-}
-
-static const char *sip_reason_code_to_str(enum AST_REDIRECTING_REASON code)
-{
- if (code >= 0 && code < ARRAY_LEN(sip_reason_table)) {
- return sip_reason_table[code].text;
- }
-
- return "unknown";
-}
-
-/*! \brief SIP Methods we support
- \todo This string should be set dynamically. We only support REFER and SUBSCRIBE if we have
- allowsubscribe and allowrefer on in sip.conf.
-*/
-#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO"
-
-/*! \brief SIP Extensions we support
- \note This should be generated based on the previous array
- in combination with settings.
- \todo We should not have "timer" if it's disabled in the configuration file.
-*/
-#define SUPPORTED_EXTENSIONS "replaces, timer"
-
-/*! \brief Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS */
-#define STANDARD_SIP_PORT 5060
-/*! \brief Standard SIP TLS port from RFC 3261. DO NOT CHANGE THIS */
-#define STANDARD_TLS_PORT 5061
-
-/*! \note in many SIP headers, absence of a port number implies port 5060,
- * and this is why we cannot change the above constant.
- * There is a limited number of places in asterisk where we could,
- * in principle, use a different "default" port number, but
- * we do not support this feature at the moment.
- * You can run Asterisk with SIP on a different port with a configuration
- * option. If you change this value in the source code, the signalling will be incorrect.
- *
- */
-
-/*! \name DefaultValues Default values, set and reset in reload_config before reading configuration
-
- These are default values in the source. There are other recommended values in the
- sip.conf.sample for new installations. These may differ to keep backwards compatibility,
- yet encouraging new behaviour on new installations
- */
-/*@{*/
-#define DEFAULT_CONTEXT "default" /*!< The default context for [general] section as well as devices */
-#define DEFAULT_MOHINTERPRET "default" /*!< The default music class */
-#define DEFAULT_MOHSUGGEST ""
-#define DEFAULT_VMEXTEN "asterisk" /*!< Default voicemail extension */
-#define DEFAULT_CALLERID "asterisk" /*!< Default caller ID */
-#define DEFAULT_MWI_FROM ""
-#define DEFAULT_NOTIFYMIME "application/simple-message-summary"
-#define DEFAULT_ALLOWGUEST TRUE
-#define DEFAULT_RTPKEEPALIVE 0 /*!< Default RTPkeepalive setting */
-#define DEFAULT_CALLCOUNTER FALSE /*!< Do not enable call counters by default */
-#define DEFAULT_SRVLOOKUP TRUE /*!< Recommended setting is ON */
-#define DEFAULT_COMPACTHEADERS FALSE /*!< Send compact (one-character) SIP headers. Default off */
-#define DEFAULT_TOS_SIP 0 /*!< Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions. */
-#define DEFAULT_TOS_AUDIO 0 /*!< Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions. */
-#define DEFAULT_TOS_VIDEO 0 /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
-#define DEFAULT_TOS_TEXT 0 /*!< Text packets should be marked as XXXX XXXX, but the default is 0 to be compatible with previous versions. */
-#define DEFAULT_COS_SIP 4 /*!< Level 2 class of service for SIP signalling */
-#define DEFAULT_COS_AUDIO 5 /*!< Level 2 class of service for audio media */
-#define DEFAULT_COS_VIDEO 6 /*!< Level 2 class of service for video media */
-#define DEFAULT_COS_TEXT 5 /*!< Level 2 class of service for text media (T.140) */
-#define DEFAULT_ALLOW_EXT_DOM TRUE /*!< Allow external domains */
-#define DEFAULT_REALM "asterisk" /*!< Realm for HTTP digest authentication */
-#define DEFAULT_DOMAINSASREALM FALSE /*!< Use the domain option to guess the realm for registration and invite requests */
-#define DEFAULT_NOTIFYRINGING TRUE /*!< Notify devicestate system on ringing state */
-#define DEFAULT_NOTIFYCID DISABLED /*!< Include CID with ringing notifications */
-#define DEFAULT_PEDANTIC FALSE /*!< Avoid following SIP standards for dialog matching */
-#define DEFAULT_AUTOCREATEPEER FALSE /*!< Don't create peers automagically */
-#define DEFAULT_MATCHEXTERNIPLOCALLY FALSE /*!< Match extern IP locally default setting */
-#define DEFAULT_QUALIFY FALSE /*!< Don't monitor devices */
-#define DEFAULT_CALLEVENTS FALSE /*!< Extra manager SIP call events */
-#define DEFAULT_ALWAYSAUTHREJECT FALSE /*!< Don't reject authentication requests always */
-#define DEFAULT_REGEXTENONQUALIFY FALSE
-#define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
-#define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
-#ifndef DEFAULT_USERAGENT
-#define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
-#define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
-#define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
-#define DEFAULT_ENGINE "asterisk" /*!< Default RTP engine to use for sessions */
-#define DEFAULT_CAPABILITY (AST_FORMAT_ULAW | AST_FORMAT_TESTLAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263);
-#endif
-/*@}*/
/*! \name DefaultSettings
Default setttings are used as a channel setting and as a default when
configuring devices
*/
/*@{*/
-static char default_language[MAX_LANGUAGE]; /*!< Default language setting for new channels */
-static char default_callerid[AST_MAX_EXTENSION]; /*!< Default caller ID for sip messages */
-static char default_mwi_from[80]; /*!< Default caller ID for MWI updates */
-static char default_fromdomain[AST_MAX_EXTENSION]; /*!< Default domain on outound messages */
-static char default_notifymime[AST_MAX_EXTENSION]; /*!< Default MIME media type for MWI notify messages */
-static char default_vmexten[AST_MAX_EXTENSION]; /*!< Default From Username on MWI updates */
-static int default_qualify; /*!< Default Qualify= setting */
+static char default_language[MAX_LANGUAGE]; /*!< Default language setting for new channels */
+static char default_callerid[AST_MAX_EXTENSION]; /*!< Default caller ID for sip messages */
+static char default_mwi_from[80]; /*!< Default caller ID for MWI updates */
+static char default_fromdomain[AST_MAX_EXTENSION]; /*!< Default domain on outound messages */
+static char default_notifymime[AST_MAX_EXTENSION]; /*!< Default MIME media type for MWI notify messages */
+static char default_vmexten[AST_MAX_EXTENSION]; /*!< Default From Username on MWI updates */
+static int default_qualify; /*!< Default Qualify= setting */
static char default_mohinterpret[MAX_MUSICCLASS]; /*!< Global setting for moh class to use when put on hold */
-static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
+static char default_mohsuggest[MAX_MUSICCLASS]; /*!< Global setting for moh class to suggest when putting
* a bridged channel on hold */
-static char default_parkinglot[AST_MAX_CONTEXT]; /*!< Parkinglot */
-static char default_engine[256]; /*!< Default RTP engine */
-static int default_maxcallbitrate; /*!< Maximum bitrate for call */
-static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
-static unsigned int default_transports; /*!< Default Transports (enum sip_transport) that are acceptable */
-static unsigned int default_primary_transport; /*!< Default primary Transport (enum sip_transport) for outbound connections to devices */
-
+static char default_parkinglot[AST_MAX_CONTEXT]; /*!< Parkinglot */
+static char default_engine[256]; /*!< Default RTP engine */
+static int default_maxcallbitrate; /*!< Maximum bitrate for call */
+static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
+static unsigned int default_transports; /*!< Default Transports (enum sip_transport) that are acceptable */
+static unsigned int default_primary_transport; /*!< Default primary Transport (enum sip_transport) for outbound connections to devices */
/*@}*/
-
-/*! \name GlobalSettings
- Global settings apply to the channel (often settings you can change in the general section
- of sip.conf
-*/
-/*@{*/
-/*! \brief a place to store all global settings for the sip channel driver
-
- These are settings that will be possibly to apply on a group level later on.
- \note Do not add settings that only apply to the channel itself and can't
- be applied to devices (trunks, services, phones)
-*/
-struct sip_settings {
- int peer_rtupdate; /*!< G: Update database with registration data for peer? */
- int rtsave_sysname; /*!< G: Save system name at registration? */
- int ignore_regexpire; /*!< G: Ignore expiration of peer */
- int rtautoclear; /*!< Realtime ?? */
- int directrtpsetup; /*!< Enable support for Direct RTP setup (no re-invites) */
- int pedanticsipchecking; /*!< Extra checking ? Default off */
- int autocreatepeer; /*!< Auto creation of peers at registration? Default off. */
- int srvlookup; /*!< SRV Lookup on or off. Default is on */
- int allowguest; /*!< allow unauthenticated peers to connect? */
- int alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
- int compactheaders; /*!< send compact sip headers */
- int allow_external_domains; /*!< Accept calls to external SIP domains? */
- int callevents; /*!< Whether we send manager events or not */
- int regextenonqualify; /*!< Whether to add/remove regexten when qualifying peers */
- int matchexterniplocally; /*!< Match externip/externhost setting against localnet setting */
- char regcontext[AST_MAX_CONTEXT]; /*!< Context for auto-extensions */
- unsigned int disallowed_methods; /*!< methods that we should never try to use */
- int notifyringing; /*!< Send notifications on ringing */
- int notifyhold; /*!< Send notifications on hold */
- enum notifycid_setting notifycid; /*!< Send CID with ringing notifications */
- enum transfermodes allowtransfer; /*!< SIP Refer restriction scheme */
- int allowsubscribe; /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE
- the global setting is in globals_flags[1] */
- char realm[MAXHOSTNAMELEN]; /*!< Default realm */
- int domainsasrealm; /*!< Use domains lists as realms */
- struct sip_proxy outboundproxy; /*!< Outbound proxy */
- char default_context[AST_MAX_CONTEXT];
- char default_subscribecontext[AST_MAX_CONTEXT];
- struct ast_ha *contact_ha; /*! \brief Global list of addresses dynamic peers are not allowed to use */
- format_t capability; /*!< Supported codecs */
-};
static struct sip_settings sip_cfg; /*!< SIP configuration data.
\note in the future we could have multiple of these (per domain, per device group etc) */
@@ -1229,65 +737,63 @@
ast_uri_decode(str); \
} \
-static int global_match_auth_username; /*!< Match auth username if available instead of From: Default off. */
-
-static int global_relaxdtmf; /*!< Relax DTMF */
-static int global_prematuremediafilter; /*!< Enable/disable premature frames in a call (causing 183 early media) */
-static int global_rtptimeout; /*!< Time out call if no RTP */
-static int global_rtpholdtimeout; /*!< Time out call if no RTP during hold */
-static int global_rtpkeepalive; /*!< Send RTP keepalives */
-static int global_reg_timeout; /*!< Global time between attempts for outbound registrations */
-static int global_regattempts_max; /*!< Registration attempts before giving up */
-static int global_shrinkcallerid; /*!< enable or disable shrinking of caller id */
-static int global_callcounter; /*!< Enable call counters for all devices. This is currently enabled by setting the peer
- call-limit to INT_MAX. When we remove the call-limit from the code, we can make it
- with just a boolean flag in the device structure */
-static unsigned int global_tos_sip; /*!< IP type of service for SIP packets */
-static unsigned int global_tos_audio; /*!< IP type of service for audio RTP packets */
-static unsigned int global_tos_video; /*!< IP type of service for video RTP packets */
-static unsigned int global_tos_text; /*!< IP type of service for text RTP packets */
-static unsigned int global_cos_sip; /*!< 802.1p class of service for SIP packets */
-static unsigned int global_cos_audio; /*!< 802.1p class of service for audio RTP packets */
-static unsigned int global_cos_video; /*!< 802.1p class of service for video RTP packets */
-static unsigned int global_cos_text; /*!< 802.1p class of service for text RTP packets */
-static unsigned int recordhistory; /*!< Record SIP history. Off by default */
-static unsigned int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
-static char global_useragent[AST_MAX_EXTENSION]; /*!< Useragent for the SIP channel */
-static char global_sdpsession[AST_MAX_EXTENSION]; /*!< SDP session name for the SIP channel */
-static char global_sdpowner[AST_MAX_EXTENSION]; /*!< SDP owner name for the SIP channel */
-static int global_authfailureevents; /*!< Whether we send authentication failure manager events or not. Default no. */
-static int global_t1; /*!< T1 time */
-static int global_t1min; /*!< T1 roundtrip time minimum */
-static int global_timer_b; /*!< Timer B - RFC 3261 Section 17.1.1.2 */
-static unsigned int global_autoframing; /*!< Turn autoframing on or off. */
-static int global_qualifyfreq; /*!< Qualify frequency */
[... 4628 lines stripped ...]
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