[asterisk-commits] twilson: branch 1.6.2 r253158 - in /branches: 1.4/channels/ 1.4/include/aster...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Mar 17 11:25:58 CDT 2010
Author: twilson
Date: Wed Mar 17 11:25:52 2010
New Revision: 253158
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=253158
Log:
Revert API change in release branches
This re-renames ast_rtp_update_source to ast_rtp_new_source
Modified:
branches/1.4/channels/chan_h323.c
branches/1.4/channels/chan_mgcp.c
branches/1.4/channels/chan_sip.c
branches/1.4/channels/chan_skinny.c
branches/1.4/include/asterisk/rtp.h
branches/1.4/main/rtp.c
branches/1.6.0/channels/chan_h323.c
branches/1.6.0/channels/chan_mgcp.c
branches/1.6.0/channels/chan_sip.c
branches/1.6.0/channels/chan_skinny.c
branches/1.6.0/include/asterisk/rtp.h
branches/1.6.0/main/rtp.c
branches/1.6.1/channels/chan_h323.c
branches/1.6.1/channels/chan_mgcp.c
branches/1.6.1/channels/chan_sip.c
branches/1.6.1/channels/chan_skinny.c
branches/1.6.1/include/asterisk/rtp.h
branches/1.6.1/main/rtp.c
branches/1.6.2/channels/chan_h323.c
branches/1.6.2/channels/chan_mgcp.c
branches/1.6.2/channels/chan_sip.c
branches/1.6.2/channels/chan_skinny.c
branches/1.6.2/include/asterisk/rtp.h
branches/1.6.2/main/rtp.c
Modified: branches/1.4/channels/chan_h323.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_h323.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.4/channels/chan_h323.c (original)
+++ branches/1.4/channels/chan_h323.c Wed Mar 17 11:25:52 2010
@@ -918,7 +918,7 @@
res = 0;
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(pvt->rtp);
+ ast_rtp_new_source(pvt->rtp);
res = 0;
break;
case AST_CONTROL_SRCCHANGE:
Modified: branches/1.4/channels/chan_mgcp.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_mgcp.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.4/channels/chan_mgcp.c (original)
+++ branches/1.4/channels/chan_mgcp.c Wed Mar 17 11:25:52 2010
@@ -1442,7 +1442,7 @@
ast_moh_stop(ast);
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(sub->rtp);
+ ast_rtp_new_source(sub->rtp);
break;
case AST_CONTROL_SRCCHANGE:
ast_rtp_change_source(sub->rtp);
Modified: branches/1.4/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Wed Mar 17 11:25:52 2010
@@ -3864,7 +3864,7 @@
if (option_debug)
ast_log(LOG_DEBUG, "SIP answering channel: %s\n", ast->name);
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
}
@@ -3899,7 +3899,7 @@
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_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
if (!global_prematuremediafilter) {
p->invitestate = INV_EARLY_MEDIA;
transmit_provisional_response(p, "183 Session Progress", &p->initreq, 1);
@@ -4147,11 +4147,11 @@
res = -1;
break;
case AST_CONTROL_HOLD:
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
ast_moh_start(ast, data, p->mohinterpret);
break;
case AST_CONTROL_UNHOLD:
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
ast_moh_stop(ast);
break;
case AST_CONTROL_VIDUPDATE: /* Request a video frame update */
@@ -4162,7 +4162,7 @@
res = -1;
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
break;
case AST_CONTROL_SRCCHANGE:
ast_rtp_change_source(p->rtp);
Modified: branches/1.4/channels/chan_skinny.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/channels/chan_skinny.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.4/channels/chan_skinny.c (original)
+++ branches/1.4/channels/chan_skinny.c Wed Mar 17 11:25:52 2010
@@ -2868,7 +2868,7 @@
case AST_CONTROL_PROCEEDING:
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(sub->rtp);
+ ast_rtp_new_source(sub->rtp);
break;
case AST_CONTROL_SRCCHANGE:
ast_rtp_change_source(sub->rtp);
Modified: branches/1.4/include/asterisk/rtp.h
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/include/asterisk/rtp.h?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.4/include/asterisk/rtp.h (original)
+++ branches/1.4/include/asterisk/rtp.h Wed Mar 17 11:25:52 2010
@@ -180,7 +180,7 @@
int ast_rtp_settos(struct ast_rtp *rtp, int tos);
/*! \brief Indicate that we need to set the marker bit */
-void ast_rtp_update_source(struct ast_rtp *rtp);
+void ast_rtp_new_source(struct ast_rtp *rtp);
/*! \brief Indicate that we need to set the marker bit and change the ssrc */
void ast_rtp_change_source(struct ast_rtp *rtp);
Modified: branches/1.4/main/rtp.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.4/main/rtp.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.4/main/rtp.c (original)
+++ branches/1.4/main/rtp.c Wed Mar 17 11:25:52 2010
@@ -2082,7 +2082,7 @@
return res;
}
-void ast_rtp_update_source(struct ast_rtp *rtp)
+void ast_rtp_new_source(struct ast_rtp *rtp)
{
if (rtp) {
rtp->set_marker_bit = 1;
Modified: branches/1.6.0/channels/chan_h323.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.0/channels/chan_h323.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.0/channels/chan_h323.c (original)
+++ branches/1.6.0/channels/chan_h323.c Wed Mar 17 11:25:52 2010
@@ -919,7 +919,7 @@
res = 0;
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(pvt->rtp);
+ ast_rtp_new_source(pvt->rtp);
res = 0;
break;
case AST_CONTROL_SRCCHANGE:
Modified: branches/1.6.0/channels/chan_mgcp.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.0/channels/chan_mgcp.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.0/channels/chan_mgcp.c (original)
+++ branches/1.6.0/channels/chan_mgcp.c Wed Mar 17 11:25:52 2010
@@ -1477,7 +1477,7 @@
ast_moh_stop(ast);
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(sub->rtp);
+ ast_rtp_new_source(sub->rtp);
break;
case AST_CONTROL_SRCCHANGE:
ast_rtp_change_source(sub->rtp);
Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Wed Mar 17 11:25:52 2010
@@ -5315,7 +5315,7 @@
ast_setstate(ast, AST_STATE_UP);
ast_debug(1, "SIP answering channel: %s\n", ast->name);
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
}
@@ -5350,7 +5350,7 @@
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_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
if (!global_prematuremediafilter) {
p->invitestate = INV_EARLY_MEDIA;
transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
@@ -5670,11 +5670,11 @@
res = -1;
break;
case AST_CONTROL_HOLD:
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
ast_moh_start(ast, data, p->mohinterpret);
break;
case AST_CONTROL_UNHOLD:
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
ast_moh_stop(ast);
break;
case AST_CONTROL_VIDUPDATE: /* Request a video frame update */
@@ -5693,7 +5693,7 @@
}
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
break;
case AST_CONTROL_SRCCHANGE:
ast_rtp_change_source(p->rtp);
Modified: branches/1.6.0/channels/chan_skinny.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.0/channels/chan_skinny.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.0/channels/chan_skinny.c (original)
+++ branches/1.6.0/channels/chan_skinny.c Wed Mar 17 11:25:52 2010
@@ -3769,7 +3769,7 @@
case AST_CONTROL_PROCEEDING:
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(sub->rtp);
+ ast_rtp_new_source(sub->rtp);
break;
case AST_CONTROL_SRCCHANGE:
ast_rtp_change_source(sub->rtp);
Modified: branches/1.6.0/include/asterisk/rtp.h
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.0/include/asterisk/rtp.h?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.0/include/asterisk/rtp.h (original)
+++ branches/1.6.0/include/asterisk/rtp.h Wed Mar 17 11:25:52 2010
@@ -188,7 +188,7 @@
int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc);
/*! \brief Indicate that we need to set the marker bit */
-void ast_rtp_update_source(struct ast_rtp *rtp);
+void ast_rtp_new_source(struct ast_rtp *rtp);
/*! \brief Indicate that we need to set the marker bit and change the ssrc */
void ast_rtp_change_source(struct ast_rtp *rtp);
Modified: branches/1.6.0/main/rtp.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.0/main/rtp.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.0/main/rtp.c (original)
+++ branches/1.6.0/main/rtp.c Wed Mar 17 11:25:52 2010
@@ -2401,7 +2401,7 @@
return ast_netsock_set_qos(rtp->s, tos, cos, desc);
}
-void ast_rtp_update_source(struct ast_rtp *rtp)
+void ast_rtp_new_source(struct ast_rtp *rtp)
{
if (rtp) {
rtp->set_marker_bit = 1;
Modified: branches/1.6.1/channels/chan_h323.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.1/channels/chan_h323.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.1/channels/chan_h323.c (original)
+++ branches/1.6.1/channels/chan_h323.c Wed Mar 17 11:25:52 2010
@@ -914,7 +914,7 @@
res = 0;
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(pvt->rtp);
+ ast_rtp_new_source(pvt->rtp);
res = 0;
break;
case AST_CONTROL_SRCCHANGE:
Modified: branches/1.6.1/channels/chan_mgcp.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.1/channels/chan_mgcp.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.1/channels/chan_mgcp.c (original)
+++ branches/1.6.1/channels/chan_mgcp.c Wed Mar 17 11:25:52 2010
@@ -1480,7 +1480,7 @@
ast_moh_stop(ast);
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(sub->rtp);
+ ast_rtp_new_source(sub->rtp);
break;
case AST_CONTROL_SRCCHANGE:
ast_rtp_change_source(sub->rtp);
Modified: branches/1.6.1/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.1/channels/chan_sip.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.1/channels/chan_sip.c (original)
+++ branches/1.6.1/channels/chan_sip.c Wed Mar 17 11:25:52 2010
@@ -5785,7 +5785,7 @@
ast_setstate(ast, AST_STATE_UP);
ast_debug(1, "SIP answering channel: %s\n", ast->name);
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
}
@@ -5820,7 +5820,7 @@
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_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
if (!global_prematuremediafilter) {
p->invitestate = INV_EARLY_MEDIA;
transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
@@ -6143,11 +6143,11 @@
res = -1;
break;
case AST_CONTROL_HOLD:
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
ast_moh_start(ast, data, p->mohinterpret);
break;
case AST_CONTROL_UNHOLD:
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
ast_moh_stop(ast);
break;
case AST_CONTROL_VIDUPDATE: /* Request a video frame update */
@@ -6166,7 +6166,7 @@
}
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
break;
case AST_CONTROL_SRCCHANGE:
ast_rtp_change_source(p->rtp);
Modified: branches/1.6.1/channels/chan_skinny.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.1/channels/chan_skinny.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.1/channels/chan_skinny.c (original)
+++ branches/1.6.1/channels/chan_skinny.c Wed Mar 17 11:25:52 2010
@@ -3918,7 +3918,7 @@
case AST_CONTROL_PROCEEDING:
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(sub->rtp);
+ ast_rtp_new_source(sub->rtp);
break;
case AST_CONTROL_SRCCHANGE:
ast_rtp_change_source(sub->rtp);
Modified: branches/1.6.1/include/asterisk/rtp.h
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.1/include/asterisk/rtp.h?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.1/include/asterisk/rtp.h (original)
+++ branches/1.6.1/include/asterisk/rtp.h Wed Mar 17 11:25:52 2010
@@ -211,7 +211,7 @@
int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc);
/*! \brief Indicate that we need to set the marker bit */
-void ast_rtp_update_source(struct ast_rtp *rtp);
+void ast_rtp_new_source(struct ast_rtp *rtp);
/*! \brief Indicate that we need to set the marker bit and change the ssrc */
void ast_rtp_change_source(struct ast_rtp *rtp);
Modified: branches/1.6.1/main/rtp.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.1/main/rtp.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.1/main/rtp.c (original)
+++ branches/1.6.1/main/rtp.c Wed Mar 17 11:25:52 2010
@@ -2616,7 +2616,7 @@
return ast_netsock_set_qos(rtp->s, type_of_service, class_of_service, desc);
}
-void ast_rtp_update_source(struct ast_rtp *rtp)
+void ast_rtp_new_source(struct ast_rtp *rtp)
{
if (rtp) {
rtp->set_marker_bit = 1;
Modified: branches/1.6.2/channels/chan_h323.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/channels/chan_h323.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.2/channels/chan_h323.c (original)
+++ branches/1.6.2/channels/chan_h323.c Wed Mar 17 11:25:52 2010
@@ -914,7 +914,7 @@
res = 0;
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(pvt->rtp);
+ ast_rtp_new_source(pvt->rtp);
res = 0;
break;
case AST_CONTROL_SRCCHANGE:
Modified: branches/1.6.2/channels/chan_mgcp.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/channels/chan_mgcp.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.2/channels/chan_mgcp.c (original)
+++ branches/1.6.2/channels/chan_mgcp.c Wed Mar 17 11:25:52 2010
@@ -1454,7 +1454,7 @@
ast_moh_stop(ast);
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(sub->rtp);
+ ast_rtp_new_source(sub->rtp);
break;
case AST_CONTROL_SRCCHANGE:
ast_rtp_change_source(sub->rtp);
Modified: branches/1.6.2/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/channels/chan_sip.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.2/channels/chan_sip.c (original)
+++ branches/1.6.2/channels/chan_sip.c Wed Mar 17 11:25:52 2010
@@ -6188,7 +6188,7 @@
ast_setstate(ast, AST_STATE_UP);
ast_debug(1, "SIP answering channel: %s\n", ast->name);
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, FALSE);
ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
}
@@ -6223,7 +6223,7 @@
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_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
if (!global_prematuremediafilter) {
p->invitestate = INV_EARLY_MEDIA;
transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
@@ -6546,11 +6546,11 @@
res = -1;
break;
case AST_CONTROL_HOLD:
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
ast_moh_start(ast, data, p->mohinterpret);
break;
case AST_CONTROL_UNHOLD:
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
ast_moh_stop(ast);
break;
case AST_CONTROL_VIDUPDATE: /* Request a video frame update */
@@ -6569,7 +6569,7 @@
}
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(p->rtp);
+ ast_rtp_new_source(p->rtp);
break;
case AST_CONTROL_SRCCHANGE:
ast_rtp_change_source(p->rtp);
Modified: branches/1.6.2/channels/chan_skinny.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/channels/chan_skinny.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.2/channels/chan_skinny.c (original)
+++ branches/1.6.2/channels/chan_skinny.c Wed Mar 17 11:25:52 2010
@@ -4256,7 +4256,7 @@
case AST_CONTROL_PROCEEDING:
break;
case AST_CONTROL_SRCUPDATE:
- ast_rtp_update_source(sub->rtp);
+ ast_rtp_new_source(sub->rtp);
break;
case AST_CONTROL_SRCCHANGE:
ast_rtp_change_source(sub->rtp);
Modified: branches/1.6.2/include/asterisk/rtp.h
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/include/asterisk/rtp.h?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.2/include/asterisk/rtp.h (original)
+++ branches/1.6.2/include/asterisk/rtp.h Wed Mar 17 11:25:52 2010
@@ -217,7 +217,7 @@
int ast_rtp_setqos(struct ast_rtp *rtp, int tos, int cos, char *desc);
/*! \brief Indicate that we need to set the marker bit */
-void ast_rtp_update_source(struct ast_rtp *rtp);
+void ast_rtp_new_source(struct ast_rtp *rtp);
/*! \brief Indicate that we need to set the marker bit and change the ssrc */
void ast_rtp_change_source(struct ast_rtp *rtp);
Modified: branches/1.6.2/main/rtp.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.6.2/main/rtp.c?view=diff&rev=253158&r1=253157&r2=253158
==============================================================================
--- branches/1.6.2/main/rtp.c (original)
+++ branches/1.6.2/main/rtp.c Wed Mar 17 11:25:52 2010
@@ -2657,7 +2657,7 @@
return ast_netsock_set_qos(rtp->s, type_of_service, class_of_service, desc);
}
-void ast_rtp_update_source(struct ast_rtp *rtp)
+void ast_rtp_new_source(struct ast_rtp *rtp)
{
if (rtp) {
rtp->set_marker_bit = 1;
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