Summary:	ASTERISK-05000: rtptimeout counts when using app_record.
Reporter:	Brian West (bkw918)	Labels:
Date Opened:	2005-09-05 23:51:27	Date Closed:	2008-01-15 15:53:16.000-0600
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Core/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	Asterisk sends NO rtp back when using app record.  If the otherside happens to be asterisk and you have the rtptimeout set to say 60 seconds you'll get hung up on because of this.  This in my opinion is a bug.  It should send back null/empty frames or something to keep the timeout happy.
Comments:	By: Tilghman Lesher (tilghman) 2005-09-06 00:24:35 Then we need something in chan_sip.c:sip_write to send AST_FRAME_NULL frames to rtp.c:ast_rtp_write, and we need ast_rtp_write() to know how to send null frames in a way that makes sense and won't cause problems on the SIP client end. By: Olle Johansson (oej) 2005-09-06 05:15:19 Can we use the comfort noise stuff in rtp.c? By: Kevin P. Fleming (kpfleming) 2005-09-13 18:49:20 Ugh... Yes, I think the right thing to do here would be to send AST_FRAME_CNG. We'll need to determine if all the codecs are ready to support that, though. It would also not be necessary to send CNG in response to every frame we receive, only once per second (or so). By: Olle Johansson (oej) 2005-10-20 02:18:58 Any updates? /O By: Ted Cabeen (secabeen) 2005-10-24 15:42:59 If you need help testing or debugging code to do this, I'd be happy to help, as it's affecting one of our SIP inbound providers. By: Kevin P. Fleming (kpfleming) 2005-11-01 12:31:27.000-0600 Fixed in CVS HEAD. There is now a global option (in asterisk.conf) called 'transmit_silence_during_record' that will cause this to happen. For now it sends SLINEAR silence, which will be transcoded into whatever format the channel is using. In the future if we have VAD-aware codecs they could choose to translate this into CNG and save bandwidth. By: Digium Subversion (svnbot) 2008-01-15 15:53:16.000-0600 Repository: asterisk Revision: 6925 U   trunk/app.c U   trunk/apps/app_record.c U   trunk/asterisk.c U   trunk/channel.c U   trunk/doc/README.asterisk.conf U   trunk/include/asterisk/channel.h U   trunk/include/asterisk/options.h ------------------------------------------------------------------------ r6925 | kpfleming | 2008-01-15 15:53:16 -0600 (Tue, 15 Jan 2008) | 2 lines optionally send silence during recording (issue ASTERISK-5000) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=6925