Summary:	ASTERISK-09170: [patch] Invite with codecs not supported by the caller in Early Bridge
Reporter:	Marcel Barbulescu (marcelbarbulescu)	Labels:
Date Opened:	2007-04-03 03:21:45	Date Closed:	2007-07-11 19:58:59
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/CodecHandling
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) patch.diff ( 1) verbosedebug.txt
Description:	When directrptsetup=yes and the Asterisk perform early bridge, it doesn't make sense to send an INVITE to the destination with the IP of the source for the RTP and codecs that the source doesn't support, as it is doing right now. A patch is attached. ****** ADDITIONAL INFORMATION ****** For the sip trace: the Asterisk is configured with g729 and the destination supports g729 but the source doesn't. When the destination is invited with the IP of the source, it gets invited with g729 too. Please note that the IPs and phone numbers in the trace file are not real.
Comments:	By: Serge Vecher (serge-v) 2007-04-03 10:22:11 could you please get a disclaimer on file? see bottom of main page. By: Marcel Barbulescu (marcelbarbulescu) 2007-04-03 11:21:29 Disclaimer in the mail. By: Serge Vecher (serge-v) 2007-04-04 08:26:13 would you please attach verbosedebug.txt from a patched asterisk? By: Marcel Barbulescu (marcelbarbulescu) 2007-04-04 09:45:17 The difference is that the Asterisk drops the G729 offer from the SDP of the initial invite to the provider (only PCMU is left) and because of that the g729 is also missing from the SDP of the "183 Session Progress" messages from the provider. If it's necessary I'll recreate the test environment later on and I'll post a trace but what I stated above is exactly what's happening with a patched system. By: Joshua C. Colp (jcolp) 2007-05-17 11:12:02 Fixed in 1.4 as of revision 64754 and trunk as of revision 64755. Thanks!