Summary:	ASTERISK-02288: Inbound SIP call Status
Reporter:	flydoc (flydoc)	Labels:
Date Opened:	2004-08-28 14:21:42	Date Closed:	2011-06-07 14:10:11
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Core/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	extensions.conf: wait(5) Ringing Wait(5) Answer Do Something that generates RTP 1) AST_STATE is set to RINGING before 180/183 is sent to the calling sip device 2) AST_STATE is set to UP before the  ACK to the 200OK is received This is the bigger problem as media can be sent before the receiving device is ready. DTMF sent at this stage can be lost
Comments:	By: flydoc (flydoc) 2004-08-28 14:30:11 I have fixed 1) 2) I don't know enough about * to make ast_answer block the channel until the 200OK is received (while still being able to process SIP messages) By: Mark Spencer (markster) 2004-08-28 15:28:59 Last paragraph of section 5.1 of RFC3264:   Once the offerer has sent the offer, it MUST be prepared to receive   media for any recvonly streams described by that offer.  It MUST be   prepared to send and receive media for any sendrecv streams in the   offer, and send media for any sendonly streams in the offer (of   course, it cannot actually send until the peer provides an answer   with the needed address and port information).  In the case of RTP,   even though it may receive media before the answer arrives, it will   not be able to send RTCP receiver reports until the answer arrives.