[asterisk-commits] rizzo: trunk r77739 - /trunk/channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Sun Jul 29 16:24:56 CDT 2007
Author: rizzo
Date: Sun Jul 29 16:24:56 2007
New Revision: 77739
URL: http://svn.digium.com/view/asterisk?view=rev&rev=77739
Log:
move some dialog-only flags to proper variables, namely
SIP_NOVIDEO, SIP_DIALOG_ANSWEREDELSEWHERE, SIP_PAGE2_NOTEXT,
SIP_PAGE2_OUTGOING_CALL
These are seldom used so the diff is relatively small.
Note that 'OUTGOING_CALL' is dangerously similar to another
dialog flag, 'SIP_OUTGOING', so the description will need to
clarify the different meaning of the two.
Also note that the description of NOTEXT is a bit unclear - does
it mean we don't support it, or 'not requested or not supported' ?
On passing fix a comment referring to video instead of text.
Finally, mark with XXX a possibly misleading debugging message.
(maybe the latter is worth backporting).
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=77739&r1=77738&r2=77739
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Sun Jul 29 16:24:56 2007
@@ -775,7 +775,6 @@
they have a common layout so it is easy to copy them.
*/
#define SIP_OUTGOING (1 << 0) /*!< D: Direction of the last transaction in this dialog */
-#define SIP_NOVIDEO (1 << 1) /*!< D: Didn't get video in invite, don't offer */
#define SIP_RINGING (1 << 2) /*!< D: Have sent 180 ringing */
#define SIP_PROGRESS_SENT (1 << 3) /*!< D: Have sent 183 message progress */
#define SIP_NEEDREINVITE (1 << 4) /*!< D: Do we need to send another reinvite? */
@@ -784,7 +783,6 @@
#define SIP_CALL_LIMIT (1 << 7) /*!< D: Call limit enforced for this call */
#define SIP_INC_COUNT (1 << 8) /*!< D: Did this dialog increment the counter of in-use calls? */
#define SIP_INC_RINGING (1 << 9) /*!< D: Did this connection increment the counter of in-use calls? */
-#define SIP_DIALOG_ANSWEREDELSEWHERE (1 << 10) /*!< D: This call is cancelled due to answer on another channel */
#define SIP_DEFER_BYE_ON_TRANSFER (1 << 11) /*!< D: Do not hangup at first ast_hangup */
#define SIP_PROMISCREDIR (1 << 12) /*!< DP: Promiscuous redirection */
@@ -855,9 +853,7 @@
#define SIP_PAGE2_RFC2833_COMPENSATE (1 << 25) /*!< DP: Compensate for buggy RFC2833 implementations */
#define SIP_PAGE2_BUGGY_MWI (1 << 26) /*!< DP: Buggy CISCO MWI fix */
-#define SIP_PAGE2_NOTEXT (1 << 27) /*!< GDP: Text not supported */
#define SIP_PAGE2_TEXTSUPPORT (1 << 28) /*!< GDP: Global text enable */
-#define SIP_PAGE2_OUTGOING_CALL (1 << 30) /*!< D: Is this an outgoing call? */
#define SIP_PAGE2_FLAGS_TO_COPY \
(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
@@ -1049,6 +1045,10 @@
char do_history; /*!< Set if we want to record history */
char alreadygone; /*!< already destroyed by our peer */
char needdestroy; /*!< need to be destroyed by the monitor thread */
+ char outgoing_call; /*!< this is an outgoing call */
+ char answered_elsewhere; /*!< This call is cancelled due to answer on another channel */
+ char novideo; /*!< Didn't get video in invite, don't offer */
+ char notext; /*!< Text not supported (?) */
int timer_t1; /*!< SIP timer T1, ms rtt */
unsigned int sipoptions; /*!< Supported SIP options on the other end */
@@ -3596,7 +3596,7 @@
{
char name[256];
int *inuse = NULL, *call_limit = NULL, *inringing = NULL;
- int outgoing = ast_test_flag(&fup->flags[1], SIP_PAGE2_OUTGOING_CALL);
+ int outgoing = fup->outgoing_call;
struct sip_user *u = NULL;
struct sip_peer *p = NULL;
@@ -3883,7 +3883,7 @@
if (option_debug)
ast_log(LOG_DEBUG, "This call was answered elsewhere");
append_history(p, "Cancel", "Call answered elsewhere");
- ast_set_flag(&p->flags[0], SIP_DIALOG_ANSWEREDELSEWHERE);
+ p->answered_elsewhere = TRUE;
}
if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
@@ -4347,7 +4347,7 @@
ast_moh_stop(ast);
break;
case AST_CONTROL_VIDUPDATE: /* Request a video frame update */
- if (p->vrtp && !ast_test_flag(&p->flags[0], SIP_NOVIDEO)) {
+ if (p->vrtp && !p->novideo) {
transmit_info_with_vidupdate(p);
/* ast_rtcp_send_h261fur(p->vrtp); */
} else
@@ -5451,11 +5451,12 @@
return -1;
}
vhp = hp; /* Copy to video address as default too */
- thp = hp; /* Copy to video address as default too */
+ thp = hp; /* Copy to text address as default too */
iterator = req->sdp_start;
- ast_set_flag(&p->flags[0], SIP_NOVIDEO);
- ast_set_flag(&p->flags[1], SIP_PAGE2_NOTEXT);
+ /* default: novideo and notext set */
+ p->novideo = TRUE;
+ p->notext = TRUE;
if (p->vrtp)
ast_rtp_pt_clear(newvideortp); /* Must be cleared in case no m=video line exists */
@@ -5490,7 +5491,7 @@
} else if ((sscanf(m, "video %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
(sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) {
video = TRUE;
- ast_clear_flag(&p->flags[0], SIP_NOVIDEO);
+ p->novideo = FALSE;
numberofmediastreams++;
vportno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
@@ -5506,7 +5507,7 @@
} else if ((sscanf(m, "text %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
(sscanf(m, "text %d RTP/AVP %n", &x, &len) == 1)) {
text = TRUE;
- ast_clear_flag(&p->flags[1], SIP_PAGE2_NOTEXT);
+ p->notext = FALSE;
numberofmediastreams++;
tportno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
@@ -6960,8 +6961,9 @@
capability = p->jointcapability;
+ /* XXX note, Video and Text are negated - 'true' means 'no' */
ast_debug(1, "** Our capability: %s Video flag: %s Text flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability),
- ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False", ast_test_flag(&p->flags[1], SIP_PAGE2_NOTEXT) ? "True" : "False");
+ p->novideo ? "True" : "False", p->notext ? "True" : "False");
ast_debug(1, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
#ifdef WHEN_WE_HAVE_T38_FOR_OTHER_TRANSPORTS
@@ -6972,7 +6974,7 @@
#endif
/* Check if we need video in this call */
- if ((capability & AST_FORMAT_VIDEO_MASK) && !ast_test_flag(&p->flags[0], SIP_NOVIDEO)) {
+ if ((capability & AST_FORMAT_VIDEO_MASK) && !p->novideo) {
if (p->vrtp) {
needvideo = TRUE;
ast_debug(2, "This call needs video offers!\n");
@@ -6999,7 +7001,7 @@
}
/* Check if we need text in this call */
- if((capability & AST_FORMAT_TEXT_MASK) && !ast_test_flag(&p->flags[1], SIP_PAGE2_NOTEXT)) {
+ if((capability & AST_FORMAT_TEXT_MASK) && !p->notext) {
if (sipdebug_text)
ast_verbose("We think we can do text\n");
if (p->trtp) {
@@ -8334,7 +8336,7 @@
p->invitestate = INV_CONFIRMED;
reqprep(&resp, p, sipmethod, seqno, newbranch);
- if (sipmethod == SIP_CANCEL && ast_test_flag(&p->flags[0], SIP_DIALOG_ANSWEREDELSEWHERE))
+ if (sipmethod == SIP_CANCEL && p->answered_elsewhere)
add_header(&resp, "Reason:", "SIP;cause=200;text=\"Call completed elsewhere\"");
add_header_contentLength(&resp, 0);
@@ -16592,7 +16594,7 @@
return NULL;
}
- ast_set_flag(&p->flags[1], SIP_PAGE2_OUTGOING_CALL);
+ p->outgoing_call = TRUE;
if (!(p->options = ast_calloc(1, sizeof(*p->options)))) {
sip_destroy(p);
More information about the asterisk-commits
mailing list