[asterisk-bugs] [Asterisk 0018797]: RTP Early Media not Passed to Caller

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=18797 
====================================================================== 
Reported By:                imp
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   18797
Category:                   Core/RTP
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           Older 1.6.2 - please test a newer version 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2011-02-11 10:44 CST
Last Modified:              2011-02-14 02:38 CST
====================================================================== 
Summary:                    RTP Early Media not Passed to Caller
Description: 
In Switzerland the price of value added numbers is announced via early
audio before the connection is established to allow the caller to hang up
without generating costs.

This early media is not passed if the call is routed via asterisk.

Traces show:

Asterisk sends invite with sdp to carrier SIP GW.

SIP GW starts sending RTP early media to RTP endpoint specified by
invite.

SIP GW signals '100 trying'
SIP GW signals '180 ringing'

No Audio is passed to the client connected to the asterisk.

Depending on the 'progressinband' and 'rematuremedia' settings, either the
client is told to 'ring' or inband 'ringing' is generated by asterisk, but
no media from the carrier forwarded.

SIP GW signals '200 OK' + sdp

Call is established and two way audio is working. The caller has missed
the early media announcement.

According my interpretation of RFC3960 asterisk should forward early audio
if it is receiving early audio.
====================================================================== 

---------------------------------------------------------------------- 
 (0131918) imp (reporter) - 2011-02-14 02:38
 https://issues.asterisk.org/view.php?id=18797#c131918 
---------------------------------------------------------------------- 
Equipment used:

Asterisk 2.6.2.15 on FreeBSD as SIP Gateway to 'Trunk' Customers for
screening and routing purposes which cannot be achived by the Carrier PBX.

Alcatel OmniPCX Enterprise R9.0 h1.301.40 as 'Trunk' Client atteched to
Asterisk. But problem can also be reproduced with Grandstream, or ZyXEL
V2000W or N900 Sip Client (and probaly many more, as the pcap shows
asterisk is not sending rtp).

Carrier PBX: IPGallery Release: 4.1-15_SS (www.ipgallery.com).

PS: I did play around with sending 'Progressing' or not, setting
'progressinboand' to no, never, yes and  prematuremedia to 'no' or 'yes'.
The resulty I got vary, but none of them cause asterisk to actualy forward
the early audio it got from the carrier to the client. Asterisk either
generated early audio 'ringing' itself or told the client to generate
'ringing'. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2011-02-14 02:38 imp            Note Added: 0131918                          
======================================================================