Author: rizzo
Date: Sun Jul 29 03:19:19 2007
New Revision: 77668
URL: http://svn.digium.com/view/asterisk?view=rev&rev=77668
Log:
more documentation on internal representation of incoming SIP messages.
Remove definitions for now-unused flags, and add references to print
routines for other flags.
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL:
http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=77668&r1=77667&r2=77668
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Sun Jul 29 03:19:19 2007
@@ -654,13 +654,31 @@
#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 @@
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 @@
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 @@
#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 @@
#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 \
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