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Clean up chan_sip

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This clean up was broken out from
https://reviewboard.asterisk.org/r/1976/ and addresses the following:
 - struct sip_refer converted to use the stringfields API.
 - sip_{refer|notify}_allocate -> sip_{notify|refer}_alloc to match
   other *alloc functions.
 - Replace get_msg_text, get_msg_text2 and get_pidf_body -> No, not
   get_pidf_msg_text_body3 but get_content, to match add_content.
 - get_body doesn't get the request body, renamed to get_content_line.
 - get_body_by_line doesn't get the body line, and is just a simple if
   test. Moved code inline and removed function.
 - Remove camelCase in struct sip_peer peer state variables,
   onHold -> onhold, inUse -> inuse, inRinging -> ringing.
 - Remove camelCase in struct sip_request rlPart1 -> rlpart1,
   rlPart2 -> rlpart2.
 - Rename instances of pvt->randdata to pvt->nonce because that is what
   it is, no need to update struct sip_pvt because _it already has a
   nonce field_.
 - Removed struct sip_pvt randdata stringfield.
 - Remove useless (and inconsistent) 'header' suffix on variables in
   handle_request_subscribe.
 - Use ast_strdupa on Event header in handle_request_subscribe to avoid
   overly complicated strncmp calls to find the event package.
 - Move get_destination check in handle_request_subscribe to avoid
   duplicate checking for packages that don't need it.
 - Move extension state callback management in handle_request_subscribe
   to avoid duplicate checking for packages that don't need it.
 - Remove duplicate append_date prototype.
 - Rename append_date -> add_date to match other add_xxx functions.
 - Added add_expires helper function, removed code that manually added
   expires header.
 - Remove _header suffix on add_diversion_header (no other header adding
   functions have this).
 - Don't pass req->debug to request handle_request_XXXXX handlers if req
   is also being passed.
 - Don't pass req->ignore to check_auth as req is already being passed.
 - Don't create a subscription in handle_request_subscribe if
   p->expiry == 0.
 - Don't walk of the back of referred_by_name when splitting string in
   get_refer_info
 - Remove duplicate check for no dialog in handle_incoming when
   sipmethod == SIP_REFER, handle_request_refer checks for that.

Review: https://reviewboard.asterisk.org/r/1993/
Patch-by: gareth


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
certified/11.2
Kinsey Moore 13 years ago
parent b83500ab61
commit e5210366e4
3 changed files with 379 additions and 460 deletions
Show Stats Download Patch File Download Diff File

775
channels/chan_sip.c
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File diff suppressed because it is too large Load Diff

58
channels/sip/include/sip.h
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@ -785,14 +785,14 @@ struct sip_socket {
* Then call parse_request() and req.method = find_sip_method();
* to initialize the other fields. The \r\n at the end of each line is
* replaced by \0, so that data[] is not a conforming SIP message anymore.
* After this processing, rlPart1 is set to non-NULL to remember
* After this processing, rlpart1 is set to non-NULL to remember
* that we can run get_header() on this kind of packet.
*
* parse_request() splits the first line as follows:
* Requests have in the first line method uri SIP/2.0
* rlPart1 = method; rlPart2 = uri;
* rlpart1 = method; rlpart2 = uri;
* Responses have in the first line SIP/2.0 NNN description
* rlPart1 = SIP/2.0; rlPart2 = NNN + description;
* rlpart1 = SIP/2.0; rlpart2 = NNN + description;
*
* For outgoing packets, we initialize the fields with init_req() or init_resp()
* (which fills the first line to "METHOD uri SIP/2.0" or "SIP/2.0 code text"),
@ -802,8 +802,8 @@ struct sip_socket {
* \endverbatim
*/
struct sip_request {
ptrdiff_t rlPart1; /*!< Offset of the SIP Method Name or "SIP/2.0" protocol version */
ptrdiff_t rlPart2; /*!< Offset of the Request URI or Response Status */
ptrdiff_t rlpart1; /*!< Offset of the SIP Method Name or "SIP/2.0" protocol version */
ptrdiff_t rlpart2; /*!< Offset of the Request URI or Response Status */
int headers; /*!< # of SIP Headers */
int method; /*!< Method of this request */
int lines; /*!< Body Content */
@ -824,7 +824,7 @@ struct sip_request {
/* \brief given a sip_request and an offset, return the char * that resides there
*
* It used to be that rlPart1, rlPart2, and the header and line arrays were character
* It used to be that rlpart1, rlpart2, and the header and line arrays were character
* pointers. They are now offsets into the ast_str portion of the sip_request structure.
* To avoid adding a bunch of redundant pointer arithmetic to the code, this macro is
* provided to retrieve the string at a particular offset within the request's buffer
@ -918,26 +918,27 @@ struct _map_x_s {
const char *s;
};
/*! \brief Structure to handle SIP transfers. Dynamically allocated when needed
\note OEJ: Should be moved to string fields */
/*! \brief Structure to handle SIP transfers. Dynamically allocated when needed */
struct sip_refer {
char refer_to[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO extension */
char refer_to_domain[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO domain */
char refer_to_urioption[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO uri options */
char refer_to_context[AST_MAX_EXTENSION]; /*!< Place to store REFER-TO context */
char referred_by[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
char referred_by_name[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
char refer_contact[AST_MAX_EXTENSION]; /*!< Place to store Contact info from a REFER extension */
char replaces_callid[SIPBUFSIZE]; /*!< Replace info: callid */
char replaces_callid_totag[SIPBUFSIZE/2]; /*!< Replace info: to-tag */
char replaces_callid_fromtag[SIPBUFSIZE/2]; /*!< Replace info: from-tag */
struct sip_pvt *refer_call; /*!< Call we are referring. This is just a reference to a
* dialog owned by someone else, so we should not destroy
* it when the sip_refer object goes.
*/
int attendedtransfer; /*!< Attended or blind transfer? */
int localtransfer; /*!< Transfer to local domain? */
enum referstatus status; /*!< REFER status */
AST_DECLARE_STRING_FIELDS(
AST_STRING_FIELD(refer_to); /*!< Place to store REFER-TO extension */
AST_STRING_FIELD(refer_to_domain); /*!< Place to store REFER-TO domain */
AST_STRING_FIELD(refer_to_urioption); /*!< Place to store REFER-TO uri options */
AST_STRING_FIELD(refer_to_context); /*!< Place to store REFER-TO context */
AST_STRING_FIELD(referred_by); /*!< Place to store REFERRED-BY extension */
AST_STRING_FIELD(referred_by_name); /*!< Place to store REFERRED-BY extension */
AST_STRING_FIELD(refer_contact); /*!< Place to store Contact info from a REFER extension */
AST_STRING_FIELD(replaces_callid); /*!< Replace info: callid */
AST_STRING_FIELD(replaces_callid_totag); /*!< Replace info: to-tag */
AST_STRING_FIELD(replaces_callid_fromtag); /*!< Replace info: from-tag */
);
struct sip_pvt *refer_call; /*!< Call we are referring. This is just a reference to a
* dialog owned by someone else, so we should not destroy
* it when the sip_refer object goes.
*/
int attendedtransfer; /*!< Attended or blind transfer? */
int localtransfer; /*!< Transfer to local domain? */
enum referstatus status; /*!< REFER status */
};
/*! \brief Struct to handle custom SIP notify requests. Dynamically allocated when needed */
@ -1006,7 +1007,6 @@ struct sip_pvt {
AST_STRING_FIELD(callid); /*!< Global CallID */
AST_STRING_FIELD(initviabranch); /*!< The branch ID from the topmost Via header in the initial request */
AST_STRING_FIELD(initviasentby); /*!< The sent-by from the topmost Via header in the initial request */
AST_STRING_FIELD(randdata); /*!< Random data */
AST_STRING_FIELD(accountcode); /*!< Account code */
AST_STRING_FIELD(realm); /*!< Authorization realm */
AST_STRING_FIELD(nonce); /*!< Authorization nonce */
@ -1304,9 +1304,9 @@ struct sip_peer {
struct sip_auth_container *auth;/*!< Realm authentication credentials */
int amaflags; /*!< AMA Flags (for billing) */
int callingpres; /*!< Calling id presentation */
int inUse; /*!< Number of calls in use */
int inRinging; /*!< Number of calls ringing */
int onHold; /*!< Peer has someone on hold */
int inuse; /*!< Number of calls in use */
int ringing; /*!< Number of calls ringing */
int onhold; /*!< Peer has someone on hold */
int call_limit; /*!< Limit of concurrent calls */
int t38_maxdatagram; /*!< T.38 FaxMaxDatagram override */
int busy_level; /*!< Level of active channels where we signal busy */

6
channels/sip/security_events.c
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@ -125,7 +125,7 @@ void sip_report_inval_password(const struct sip_pvt *p, const char *response_cha
},
.common.session_id = session_id,
.challenge = p->randdata,
.challenge = p->nonce,
.received_challenge = response_challenge,
.received_hash = response_hash,
};
@ -206,7 +206,7 @@ void sip_report_failed_challenge_response(const struct sip_pvt *p, const char *r
},
.common.session_id = session_id,
.challenge = p->randdata,
.challenge = p->nonce,
.response = response,
.expected_response = expected_response,
};
@ -242,7 +242,7 @@ void sip_report_chal_sent(const struct sip_pvt *p)
},
.common.session_id = session_id,
.challenge = p->randdata,
.challenge = p->nonce,
};
if (!ast_strlen_zero(p->from)) { /* When dialing, show account making call */

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