[asterisk-commits] lmadsen: trunk r283458 - in /trunk: ./ channels/ res/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Aug 24 13:58:50 CDT 2010
Author: lmadsen
Date: Tue Aug 24 13:58:46 2010
New Revision: 283458
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=283458
Log:
Merged revisions 283457 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r283457 | lmadsen | 2010-08-24 13:56:29 -0500 (Tue, 24 Aug 2010) | 9 lines
Fix issue where TOS is no longer set on RTP packets.
Fix issue where the tos is no longer being set on RTP packets through res_rtp_asterisk.
(closes issue #17890)
Reported by: elguero
Patches:
qos_18.diff uploaded by elguero (license 37)
Review: https://reviewboard.asterisk.org/r/868
........
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
trunk/res/res_rtp_asterisk.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.
Modified: trunk/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/chan_sip.c?view=diff&rev=283458&r1=283457&r2=283458
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Aug 24 13:58:46 2010
@@ -4789,7 +4789,7 @@
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
- ast_rtp_instance_set_qos(dialog->rtp, global_tos_audio, 0, "SIP RTP");
+ ast_rtp_instance_set_qos(dialog->rtp, global_tos_audio, global_cos_audio, "SIP RTP");
do_setnat(dialog);
Modified: trunk/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/trunk/res/res_rtp_asterisk.c?view=diff&rev=283458&r1=283457&r2=283458
==============================================================================
--- trunk/res/res_rtp_asterisk.c (original)
+++ trunk/res/res_rtp_asterisk.c Tue Aug 24 13:58:46 2010
@@ -270,6 +270,7 @@
static int ast_rtp_dtmf_compatible(struct ast_channel *chan0, struct ast_rtp_instance *instance0, struct ast_channel *chan1, struct ast_rtp_instance *instance1);
static void ast_rtp_stun_request(struct ast_rtp_instance *instance, struct ast_sockaddr *suggestion, const char *username);
static void ast_rtp_stop(struct ast_rtp_instance *instance);
+static int ast_rtp_qos_set(struct ast_rtp_instance *instance, int tos, int cos, const char* desc);
/* RTP Engine Declaration */
static struct ast_rtp_engine asterisk_rtp_engine = {
@@ -293,6 +294,7 @@
.dtmf_compatible = ast_rtp_dtmf_compatible,
.stun_request = ast_rtp_stun_request,
.stop = ast_rtp_stop,
+ .qos = ast_rtp_qos_set,
};
static inline int rtp_debug_test_addr(struct ast_sockaddr *addr)
@@ -2549,6 +2551,13 @@
ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
}
+static int ast_rtp_qos_set(struct ast_rtp_instance *instance, int tos, int cos, const char *desc)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+ return ast_set_qos(rtp->s, tos, cos, desc);
+}
+
static char *rtp_do_debug_ip(struct ast_cli_args *a)
{
char *arg = ast_strdupa(a->argv[4]);
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