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

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=18797 
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Reported By:                imp
Assigned To:                
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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:              
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Date Submitted:             2011-02-11 10:44 CST
Last Modified:              2011-02-11 12:22 CST
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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.
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 (0131863) lmadsen (administrator) - 2011-02-11 12:22
 https://issues.asterisk.org/view.php?id=18797#c131863 
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Not enough information provided to proceed.

Per the bug guidelines:

* SIP trace from the Asterisk console
* Enable debug level logging and SIP history via sip.conf
* Provide a configuration which can be used to duplicate the situation
* What is the topology in use and what piece of equipment

The configuration and the SIP trace are the most important. 

Issue History 
Date Modified    Username       Field                    Change               
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2011-02-11 12:22 lmadsen        Note Added: 0131863                          
2011-02-11 12:22 lmadsen        Status                   new => feedback     
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