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more documentation on internal representation of incoming SIP messages.

Remove definitions for now-unused flags, and add references to print
routines for other flags.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77668 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
rizzo 2007-07-29 08:19:19 +00:00
parent 7a29f51638
commit fa819e4866

View file

@ -654,13 +654,31 @@ static int *sipsock_read_id; /*!< ID of IO entry for sipsock FD */
#define INC_CALL_RINGING 3
/*! \brief sip_request: The data grabbed from the UDP socket
* data[] contains the packet itself, additional fields are set
* after parsing.
*
* Incoming messages: we first store the data from the socket in data[],
* adding a trailing \0 to make string parsing routines happy.
* 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
* 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;
* Responses have in the first line SIP/2.0 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"),
* and then fill the rest with add_header() and add_line().
* The \r\n at the end of the line are still there, so the get_header()
* and similar functions don't work on these packets.
*/
struct sip_request {
char *rlPart1; /*!< SIP Method Name or "SIP/2.0" protocol version */
char *rlPart2; /*!< The Request URI or Response Status */
int len; /*!< Length */
int len; /*!< bytes used in data[], excluding trailing '\0'. Rarely used. */
int headers; /*!< # of SIP Headers */
int method; /*!< Method of this request */
int lines; /*!< Body Content */
@ -674,25 +692,6 @@ struct sip_request {
char data[SIP_MAX_PACKET];
};
/*
* A sip packet is stored into the data[] buffer, with the header followed
* by an empty line and the body of the message.
* On outgoing packets, data is accumulated in data[] with len reflecting
* the next available byte, headers and lines count the number of lines
* in both parts. There are no '\0' in data[0..len-1].
*
* On received packet, the input read from the socket is copied into data[],
* len is set and the string is NUL-terminated. Then a parser fills up
* the other fields -header[] and line[] to point to the lines of the
* message, rlPart1 and rlPart2 parse the first lnie as below:
*
* Requests have in the first line METHOD URI SIP/2.0
* rlPart1 = method; rlPart2 = uri;
* Responses have in the first line SIP/2.0 code description
* rlPart1 = SIP/2.0; rlPart2 = code + description;
*
*/
/*! \brief structure used in transfers */
struct sip_dual {
struct ast_channel *chan1; /*!< First channel involved */
@ -766,48 +765,52 @@ struct sip_auth {
When flags are used by multiple structures, it is important that
they have a common layout so it is easy to copy them.
*/
#define __SIP_ALREADYGONE (1 << 0) /*!< D: Whether or not we've already been destroyed by our peer */
#define __SIP_NEEDDESTROY (1 << 1) /*!< D: if we need to be destroyed by the monitor thread */
#define SIP_NOVIDEO (1 << 2) /*!< D: Didn't get video in invite, don't offer */
#define SIP_RINGING (1 << 3) /*!< D: Have sent 180 ringing */
#define SIP_PROGRESS_SENT (1 << 4) /*!< D: Have sent 183 message progress */
#define SIP_NEEDREINVITE (1 << 5) /*!< D: Do we need to send another reinvite? */
#define SIP_PENDINGBYE (1 << 6) /*!< D: Need to send bye after we ack? */
#define SIP_GOTREFER (1 << 7) /*!< D: Got a refer? */
#define SIP_PROMISCREDIR (1 << 8) /*!< DP: Promiscuous redirection */
#define SIP_TRUSTRPID (1 << 9) /*!< DP: Trust RPID headers? */
#define SIP_USEREQPHONE (1 << 10) /*!< DP: Add user=phone to numeric URI. Default off */
#define __SIP_REALTIME (1 << 11) /*!< P: Flag for realtime users */
#define SIP_USECLIENTCODE (1 << 12) /*!< DP: Trust X-ClientCode info message */
#define SIP_OUTGOING (1 << 13) /*!< D: Direction of the last transaction in this dialog */
#define SIP_DIALOG_ANSWEREDELSEWHERE (1 << 14) /*!< D: This call is cancelled due to answer on another channel */
#define SIP_DEFER_BYE_ON_TRANSFER (1 << 15) /*!< D: Do not hangup at first ast_hangup */
/* DTMF flags - see str2dtmfmode() and dtmfmode2str() */
#define SIP_DTMF (3 << 16) /*!< DP: DTMF Support: four settings, uses two bits */
#define SIP_DTMF_RFC2833 (0 << 16) /*!< DP: DTMF Support: RTP DTMF - "rfc2833" */
#define SIP_DTMF_INBAND (1 << 16) /*!< DP: DTMF Support: Inband audio, only for ULAW/ALAW - "inband" */
#define SIP_DTMF_INFO (2 << 16) /*!< DP: DTMF Support: SIP Info messages - "info" */
#define SIP_DTMF_AUTO (3 << 16) /*!< DP: DTMF Support: AUTO switch between rfc2833 and in-band DTMF */
/* NAT settings */
/* NAT settings - see nat2str() */
#define SIP_NAT (3 << 18) /*!< DP: four settings, uses two bits */
#define SIP_NAT_NEVER (0 << 18) /*!< DP: No nat support */
#define SIP_NAT_RFC3581 (1 << 18) /*!< DP: NAT RFC3581 */
#define SIP_NAT_ROUTE (2 << 18) /*!< DP: NAT Only ROUTE */
#define SIP_NAT_ALWAYS (3 << 18) /*!< DP: NAT Both ROUTE and RFC3581 */
/* re-INVITE related settings */
#define SIP_REINVITE (7 << 20) /*!< DP: three bits used */
#define SIP_CAN_REINVITE (1 << 20) /*!< DP: allow peers to be reinvited to send media directly p2p */
#define SIP_CAN_REINVITE_NAT (2 << 20) /*!< DP: allow media reinvite when new peer is behind NAT */
#define SIP_REINVITE_UPDATE (4 << 20) /*!< DP: use UPDATE (RFC3311) when reinviting this peer */
/* "insecure" settings */
/* "insecure" settings - see insecure2str() */
#define SIP_INSECURE (3 << 23) /*!< DP: two bits used */
#define SIP_INSECURE_PORT (1 << 23) /*!< DP: don't require matching port for incoming requests */
#define SIP_INSECURE_INVITE (1 << 24) /*!< DP: don't require authentication for incoming INVITEs */
/* Sending PROGRESS in-band settings */
#define SIP_PROG_INBAND (3 << 25) /*!< DP: three settings, uses two bits */
#define SIP_PROG_INBAND_NEVER (0 << 25)
#define SIP_PROG_INBAND_NO (1 << 25)
#define SIP_PROG_INBAND_YES (2 << 25)
#define __SIP_NO_HISTORY (1 << 27) /*!< D: Suppress recording request/response history */
#define SIP_CALL_LIMIT (1 << 28) /*!< D: Call limit enforced for this call */
#define SIP_SENDRPID (1 << 29) /*!< DP: Remote Party-ID Support */
#define SIP_INC_COUNT (1 << 30) /*!< D: Did this dialog increment the counter of in-use calls? */
@ -828,9 +831,6 @@ struct sip_auth {
#define SIP_PAGE2_RTSAVE_SYSNAME (1 << 5) /*!< G: Save system name at registration? */
/* Space for addition of other realtime flags in the future */
#define SIP_PAGE2_IGNOREREGEXPIRE (1 << 10) /*!< G: Ignore expiration of peer */
#define __SIP_PAGE2_DEBUG (3 << 11) /*!< G: Debug flags */
#define __SIP_PAGE2_DEBUG_CONFIG (1 << 11) /*!< G: Debug flags */
#define __SIP_PAGE2_DEBUG_CONSOLE (1 << 12) /*!< G: Debug flags */
#define SIP_PAGE2_DYNAMIC (1 << 13) /*!< P: Dynamic Peers register with Asterisk */
#define SIP_PAGE2_SELFDESTRUCT (1 << 14) /*!< P: Automatic peers need to destruct themselves */
#define SIP_PAGE2_VIDEOSUPPORT (1 << 15) /*!< DP: Video supported if offered? */
@ -850,7 +850,6 @@ struct sip_auth {
#define SIP_PAGE2_BUGGY_MWI (1 << 26) /*!< DP: 26: Buggy CISCO MWI fix */
#define SIP_PAGE2_NOTEXT (1 << 27) /*!< GPD: 27: Text not supported */
#define SIP_PAGE2_TEXTSUPPORT (1 << 28) /*!< GPD: 28: Global text enable */
#define __SIP_PAGE2_DEBUG_TEXT (1 << 29) /*!< GPD: 29: Global text debug */
#define SIP_PAGE2_OUTGOING_CALL (1 << 30) /*!< D: 30: Is this an outgoing call? */
#define SIP_PAGE2_FLAGS_TO_COPY \