[asterisk-commits] file: trunk r105675 - in /trunk: ./ channels/ include/asterisk/ main/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Mar 4 12:08:42 CST 2008
Author: file
Date: Tue Mar 4 12:08:42 2008
New Revision: 105675
URL: http://svn.digium.com/view/asterisk?view=rev&rev=105675
Log:
Merged revisions 105674 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines
When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.
(closes issue #10355)
Reported by: wdecarne
Patches:
10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson
........
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
trunk/include/asterisk/rtp.h
trunk/main/rtp.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=105675&r1=105674&r2=105675
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Mar 4 12:08:42 2008
@@ -4824,8 +4824,10 @@
if (p->t38.state == T38_PEER_DIRECT) {
change_t38_state(p, T38_ENABLED);
res = transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
- } else
+ } else {
+ ast_rtp_new_source(p->rtp);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
+ }
}
sip_pvt_unlock(p);
return res;
@@ -4858,6 +4860,7 @@
if ((ast->_state != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+ ast_rtp_new_source(p->rtp);
transmit_response_with_sdp(p, "183 Session Progress", &p->initreq, XMIT_UNRELIABLE, FALSE);
ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
}
@@ -5095,9 +5098,11 @@
res = -1;
break;
case AST_CONTROL_HOLD:
+ ast_rtp_new_source(p->rtp);
ast_moh_start(ast, data, p->mohinterpret);
break;
case AST_CONTROL_UNHOLD:
+ ast_rtp_new_source(p->rtp);
ast_moh_stop(ast);
break;
case AST_CONTROL_VIDUPDATE: /* Request a video frame update */
Modified: trunk/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/rtp.h?view=diff&rev=105675&r1=105674&r2=105675
==============================================================================
--- trunk/include/asterisk/rtp.h (original)
+++ trunk/include/asterisk/rtp.h Tue Mar 4 12:08:42 2008
@@ -170,6 +170,8 @@
int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc);
+void ast_rtp_new_source(struct ast_rtp *rtp);
+
/*! \brief Setting RTP payload types from lines in a SDP description: */
void ast_rtp_pt_clear(struct ast_rtp* rtp);
/*! \brief Set payload types to defaults */
Modified: trunk/main/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/main/rtp.c?view=diff&rev=105675&r1=105674&r2=105675
==============================================================================
--- trunk/main/rtp.c (original)
+++ trunk/main/rtp.c Tue Mar 4 12:08:42 2008
@@ -175,6 +175,8 @@
enum strict_rtp_state strict_rtp_state; /*!< Current state that strict RTP protection is in */
struct sockaddr_in strict_rtp_address; /*!< Remote address information for strict RTP purposes */
+
+ int set_marker_bit:1; /*!< Whether to set the marker bit or not */
};
/* Forward declarations */
@@ -2324,6 +2326,12 @@
return ast_netsock_set_qos(rtp->s, tos, cos, desc);
}
+void ast_rtp_new_source(struct ast_rtp *rtp)
+{
+ rtp->set_marker_bit = 1;
+ return;
+}
+
void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
{
rtp->them.sin_port = them->sin_port;
@@ -3036,6 +3044,13 @@
}
}
}
+
+ /* If we have been explicitly told to set the marker bit do so */
+ if (rtp->set_marker_bit) {
+ mark = 1;
+ rtp->set_marker_bit = 0;
+ }
+
/* If the timestamp for non-digit packets has moved beyond the timestamp
for digits, update the digit timestamp.
*/
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