[asterisk-commits] irroot: trunk r338435 - in /trunk: ./ channels/ channels/sip/include/
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Thu Sep 29 07:22:50 CDT 2011
Author: irroot
Date: Thu Sep 29 07:22:43 2011
New Revision: 338435
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=338435
Log:
Merged revisions 338417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................
r338417 | irroot | 2011-09-29 14:16:42 +0200 (Thu, 29 Sep 2011) | 19 lines
Merged revisions 338416 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
........
r338416 | irroot | 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) | 12 lines
The rtptimeout setting is ignored on a per peer basis.
Not only is the rtptimeout ignored in some cases but
rtpkeepalive and rtpholdtimeout is affected.
this commit also removes rtptimeout/rtpholdtimeout on
text rtp.
(closes issue ASTERISK-18559)
Review: https://reviewboard.asterisk.org/r/1452
........
................
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
trunk/channels/sip/include/sip.h
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-10-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=338435&r1=338434&r2=338435
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Sep 29 07:22:43 2011
@@ -5089,9 +5089,9 @@
if (!(dialog->vrtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) {
return -1;
}
- ast_rtp_instance_set_timeout(dialog->vrtp, global_rtptimeout);
- ast_rtp_instance_set_hold_timeout(dialog->vrtp, global_rtpholdtimeout);
- ast_rtp_instance_set_keepalive(dialog->vrtp, global_rtpholdtimeout);
+ ast_rtp_instance_set_timeout(dialog->vrtp, dialog->rtptimeout);
+ ast_rtp_instance_set_hold_timeout(dialog->vrtp, dialog->rtpholdtimeout);
+ ast_rtp_instance_set_keepalive(dialog->vrtp, dialog->rtpkeepalive);
ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1);
}
@@ -5100,16 +5100,15 @@
if (!(dialog->trtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) {
return -1;
}
- ast_rtp_instance_set_timeout(dialog->trtp, global_rtptimeout);
- ast_rtp_instance_set_hold_timeout(dialog->trtp, global_rtpholdtimeout);
- ast_rtp_instance_set_keepalive(dialog->trtp, global_rtpholdtimeout);
+ /* Do not timeout text as its not constant*/
+ ast_rtp_instance_set_keepalive(dialog->trtp, dialog->rtpkeepalive);
ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, 1);
}
- ast_rtp_instance_set_timeout(dialog->rtp, global_rtptimeout);
- ast_rtp_instance_set_hold_timeout(dialog->rtp, global_rtpholdtimeout);
- ast_rtp_instance_set_keepalive(dialog->rtp, global_rtpkeepalive);
+ ast_rtp_instance_set_timeout(dialog->rtp, dialog->rtptimeout);
+ ast_rtp_instance_set_hold_timeout(dialog->rtp, dialog->rtpholdtimeout);
+ ast_rtp_instance_set_keepalive(dialog->rtp, dialog->rtpkeepalive);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_RTCP, 1);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
@@ -5170,6 +5169,9 @@
ast_string_field_set(dialog, engine, peer->engine);
+ dialog->rtptimeout = peer->rtptimeout;
+ dialog->rtpholdtimeout = peer->rtpholdtimeout;
+ dialog->rtpkeepalive = peer->rtpkeepalive;
if (dialog_initialize_rtp(dialog)) {
return -1;
}
@@ -5177,22 +5179,9 @@
if (dialog->rtp) { /* Audio */
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_timeout(dialog->rtp, peer->rtptimeout);
- ast_rtp_instance_set_hold_timeout(dialog->rtp, peer->rtpholdtimeout);
- ast_rtp_instance_set_keepalive(dialog->rtp, peer->rtpkeepalive);
/* Set Frame packetization */
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(dialog->rtp), dialog->rtp, &dialog->prefs);
dialog->autoframing = peer->autoframing;
- }
- if (dialog->vrtp) { /* Video */
- ast_rtp_instance_set_timeout(dialog->vrtp, peer->rtptimeout);
- ast_rtp_instance_set_hold_timeout(dialog->vrtp, peer->rtpholdtimeout);
- ast_rtp_instance_set_keepalive(dialog->vrtp, peer->rtpkeepalive);
- }
- if (dialog->trtp) { /* Realtime text */
- ast_rtp_instance_set_timeout(dialog->trtp, peer->rtptimeout);
- ast_rtp_instance_set_hold_timeout(dialog->trtp, peer->rtpholdtimeout);
- ast_rtp_instance_set_keepalive(dialog->trtp, peer->rtpkeepalive);
}
/* XXX TODO: get fields directly from peer only as they are needed using dialog->relatedpeer */
@@ -5221,7 +5210,6 @@
ast_copy_string(dialog->zone, peer->zone, sizeof(dialog->zone));
dialog->allowtransfer = peer->allowtransfer;
dialog->jointnoncodeccapability = dialog->noncodeccapability;
- dialog->rtptimeout = peer->rtptimeout;
/* Update dialog authorization credentials */
ao2_lock(peer);
@@ -5334,10 +5322,13 @@
dialog->relatedpeer = sip_ref_peer(peer, "create_addr: setting dialog's relatedpeer pointer");
sip_unref_peer(peer, "create_addr: unref peer from sip_find_peer hashtab lookup");
return res;
- }
-
- if (dialog_initialize_rtp(dialog)) {
- return -1;
+ } else {
+ dialog->rtptimeout = global_rtptimeout;
+ dialog->rtpholdtimeout = global_rtpholdtimeout;
+ dialog->rtpkeepalive = global_rtpkeepalive;
+ if (dialog_initialize_rtp(dialog)) {
+ return -1;
+ }
}
ast_string_field_set(dialog, tohost, hostport.host);
@@ -15922,6 +15913,9 @@
else
p->noncodeccapability &= ~AST_RTP_DTMF;
p->jointnoncodeccapability = p->noncodeccapability;
+ p->rtptimeout = peer->rtptimeout;
+ p->rtpholdtimeout = peer->rtpholdtimeout;
+ p->rtpkeepalive = peer->rtpkeepalive;
if (!dialog_initialize_rtp(p)) {
if (p->rtp) {
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs);
@@ -16043,6 +16037,9 @@
/* Finally, apply the guest policy */
if (sip_cfg.allowguest) {
get_rpid(p, req);
+ p->rtptimeout = global_rtptimeout;
+ p->rtpholdtimeout = global_rtpholdtimeout;
+ p->rtpkeepalive = global_rtpkeepalive;
if (!dialog_initialize_rtp(p)) {
res = AUTH_SUCCESSFUL;
} else {
Modified: trunk/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/trunk/channels/sip/include/sip.h?view=diff&rev=338435&r1=338434&r2=338435
==============================================================================
--- trunk/channels/sip/include/sip.h (original)
+++ trunk/channels/sip/include/sip.h Thu Sep 29 07:22:43 2011
@@ -1072,6 +1072,8 @@
time_t lastrtprx; /*!< Last RTP received */
time_t lastrtptx; /*!< Last RTP sent */
int rtptimeout; /*!< RTP timeout time */
+ int rtpholdtimeout; /*!< RTP timeout time on hold*/
+ int rtpkeepalive; /*!< RTP send packets for keepalive */
struct ast_ha *directmediaha; /*!< Which IPs are allowed to interchange direct media with this peer - copied from sip_peer */
struct ast_sockaddr recv; /*!< Received as */
struct ast_sockaddr ourip; /*!< Our IP (as seen from the outside) */
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