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Merged revisions 323370 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r323370 | twilson | 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) | 10 lines
  
  Add rtpkeepalives back to 1.8
  
  The RTP-engine conversion left out support for handling rtpkeepalives.
  This patch adds them back.
  
  (closes issue ASTERISK-17304)
  Reported by: lmadsen
  
  Review: https://reviewboard.asterisk.org/r/1226/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Terry Wilson
2011-06-14 17:03:37 +00:00
parent 00181729b4
commit abd7ef817e
4 changed files with 131 additions and 1 deletions
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16
channels/chan_sip.c
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@@ -5061,6 +5061,7 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog)
}
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_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1);
}
@@ -5071,12 +5072,14 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog)
}
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);
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_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);
@@ -5144,6 +5147,7 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
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;
@@ -5151,10 +5155,12 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
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 */
@@ -25396,11 +25402,19 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
}
/* If we have no timers set, return now */
if (!ast_rtp_instance_get_timeout(dialog->rtp) && !ast_rtp_instance_get_hold_timeout(dialog->rtp)) {
if (!ast_rtp_instance_get_keepalive(dialog->rtp) && !ast_rtp_instance_get_timeout(dialog->rtp) && !ast_rtp_instance_get_hold_timeout(dialog->rtp)) {
dialog_unlink_rtpcheck(dialog);
return;
}
/* Check AUDIO RTP keepalives */
if (dialog->lastrtptx && ast_rtp_instance_get_keepalive(dialog->rtp) &&
(t > dialog->lastrtptx + ast_rtp_instance_get_keepalive(dialog->rtp))) {
/* Need to send an empty RTP packet */
dialog->lastrtptx = time(NULL);
ast_rtp_instance_sendcng(dialog->rtp, 0);
}
/*! \todo Check video RTP keepalives
Do we need to move the lastrtptx to the RTP structure to have one for audio and one