[asterisk-commits] branch oej/securertp-trunk r35124 - in
/team/oej/securertp-trunk: ./ channels/
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Jun 20 08:16:09 MST 2006
Author: oej
Date: Tue Jun 20 10:16:09 2006
New Revision: 35124
URL: http://svn.digium.com/view/asterisk?rev=35124&view=rev
Log:
Reset automerge
Modified:
team/oej/securertp-trunk/ (props changed)
team/oej/securertp-trunk/Makefile
team/oej/securertp-trunk/channels/chan_sip.c
Propchange: team/oej/securertp-trunk/
------------------------------------------------------------------------------
automerge = http://edvina.net/training/
Propchange: team/oej/securertp-trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.2-merged' - no diff available.
Propchange: team/oej/securertp-trunk/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Jun 20 10:16:09 2006
@@ -1,1 +1,1 @@
-/trunk:1-35084
+/trunk:1-35123
Modified: team/oej/securertp-trunk/Makefile
URL: http://svn.digium.com/view/asterisk/team/oej/securertp-trunk/Makefile?rev=35124&r1=35123&r2=35124&view=diff
==============================================================================
--- team/oej/securertp-trunk/Makefile (original)
+++ team/oej/securertp-trunk/Makefile Tue Jun 20 10:16:09 2006
@@ -358,8 +358,6 @@
else
HAVEDOT=no
endif
-
-INSTALL=install
_all: all
@echo " +--------- Asterisk Build Complete ---------+"
Modified: team/oej/securertp-trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/securertp-trunk/channels/chan_sip.c?rev=35124&r1=35123&r2=35124&view=diff
==============================================================================
--- team/oej/securertp-trunk/channels/chan_sip.c (original)
+++ team/oej/securertp-trunk/channels/chan_sip.c Tue Jun 20 10:16:09 2006
@@ -4561,7 +4561,6 @@
char protocol[5] = "";
len = -1;
- numberofmediastreams++;
if (p->vrtp)
ast_rtp_pt_clear(newvideortp); /* Must be cleared in case no m=video line exists */
@@ -4579,6 +4578,7 @@
continue;
}
audio = TRUE;
+ numberofmediastreams++;
/* Found audio stream in this media definition */
portno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
@@ -4606,6 +4606,7 @@
}
/* If it is not audio - is it video ? */
ast_clear_flag(&p->flags[0], SIP_NOVIDEO);
+ numberofmediastreams++;
vportno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
@@ -4621,6 +4622,7 @@
if (debug)
ast_verbose("Got T.38 offer in SDP in dialog %s\n", p->callid);
udptlportno = x;
+ numberofmediastreams++;
if (p->owner && p->lastinvite) {
p->t38.state = T38_PEER_REINVITE; /* T38 Offered in re-invite from remote party */
@@ -4659,7 +4661,7 @@
return -2;
if (numberofmediastreams > 2)
- /* We have too many media streams, fail this offer */
+ /* We have too many fax, audio and/or video media streams, fail this offer */
return -3;
/* RTP addresses and ports for audio and video */
@@ -4706,6 +4708,45 @@
iterator = req->sdp_start;
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
char *mimeSubtype = ast_strdupa(a); /* ensures we have enough space */
+ if (option_debug > 1) {
+ int breakout = FALSE;
+
+ /* If we're debugging, check for unsupported sdp options */
+ if (!strcasecmp(a, "inactive")) {
+ /* Inactive media streams: Not supported */
+ if (debug)
+ ast_verbose("Got unsupported a:inactive in SDP offer \n");
+ breakout = TRUE;
+ } else if (!strncasecmp(a, "rtcp:", (size_t) 5)) {
+ if (debug)
+ ast_verbose("Got unsupported a:rtcp in SDP offer \n");
+ breakout = TRUE;
+ } else if (!strncasecmp(a, "fmtp:", (size_t) 5)) {
+ /* Format parameters: Not supported */
+ /* Note: This is used for codec parameters, like bitrate for
+ G722 and video formats for H263 and H264
+ See RFC2327 for an example */
+ if (debug)
+ ast_verbose("Got unsupported a:fmtp in SDP offer \n");
+ breakout = TRUE;
+ } else if (!strncasecmp(a, "framerate:", (size_t) 10)) {
+ /* Video stuff: Not supported */
+ if (debug)
+ ast_verbose("Got unsupported a:framerate in SDP offer \n");
+ breakout = TRUE;
+ } else if (!strncasecmp(a, "maxprate:", (size_t) 9)) {
+ /* Video stuff: Not supported */
+ if (debug)
+ ast_verbose("Got unsupported a:maxprate in SDP offer \n");
+ breakout = TRUE;
+ } else if (!strncasecmp(a, "ptime:", (size_t) 6)) {
+ if (debug)
+ ast_verbose("Got unsupported a:ptime in SDP offer \n");
+ breakout = TRUE;
+ }
+ if (breakout) /* We have a match, skip to next header */
+ continue;
+ }
if (!strcasecmp(a, "sendonly")) {
sendonly = 1;
continue;
@@ -4715,40 +4756,6 @@
} else if (!strcasecmp(a, "sendrecv")) {
sendonly = 0;
continue;
- } else if (option_debug) {
- /* If we're debugging, check for unsupported sdp options */
- if (!strcasecmp(a, "inactive")) {
- /* Inactive media streams: Not supported */
- if (debug)
- ast_verbose("Got unsupported a:inactive in SDP offer \n");
- continue;
- } else if (!strncasecmp(a, "rtcp:", (size_t) 5)) {
- if (debug)
- ast_verbose("Got unsupported a:rtcp in SDP offer \n");
- continue;
- } else if (!strncasecmp(a, "fmtp:", (size_t) 5)) {
- /* Format parameters: Not supported */
- /* Note: This is used for codec parameters, like bitrate for
- G722 and video formats for H263 and H264
- See RFC2327 for an example */
- if (debug)
- ast_verbose("Got unsupported a:fmtp in SDP offer \n");
- continue;
- } else if (!strncasecmp(a, "framerate:", (size_t) 10)) {
- /* Video stuff: Not supported */
- if (debug)
- ast_verbose("Got unsupported a:framerate in SDP offer \n");
- continue;
- } else if (!strncasecmp(a, "maxprate:", (size_t) 9)) {
- /* Video stuff: Not supported */
- if (debug)
- ast_verbose("Got unsupported a:maxprate in SDP offer \n");
- continue;
- } else if (!strncasecmp(a, "ptime:", (size_t) 6)) {
- if (debug)
- ast_verbose("Got unsupported a:ptime in SDP offer \n");
- continue;
- }
} else if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2)
continue;
/* We have a rtpmap to handle */
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