[asterisk-bugs] [Asterisk 0018797]: RTP Early Media not Passed to Caller
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Mar 9 14:10:44 CST 2011
The following issue has been UPDATED.
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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: acknowledged
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: 2011-03-09 14:10 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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Issue History
Date Modified Username Field Change
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2011-03-09 14:10 lmadsen Description Updated
2011-03-09 14:10 lmadsen Additional Information Updated
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