[asterisk-bugs] [Asterisk 0014299]: RTP delayed by 30 seconds when SIP calls is bridged via two LOCAL channel.
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Jan 21 11:42:31 CST 2009
The following issue has been SUBMITTED.
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http://bugs.digium.com/view.php?id=14299
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Reported By: chris-mac
Assigned To:
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Project: Asterisk
Issue ID: 14299
Category: Channels/chan_local
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: SVN
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1
SVN Revision (number only!): 169673
Request Review:
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Date Submitted: 2009-01-21 11:42 CST
Last Modified: 2009-01-21 11:42 CST
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Summary: RTP delayed by 30 seconds when SIP calls is bridged
via two LOCAL channel.
Description:
My scenario is as follows:
1a. Call is originated via .call file on a 'Caller' LOCAL channel.
1b. 'Caller' LOCAL channel dials a SIP phone.
2a. When 'Caller' SIP phone answers, it is connected to 'Called' LOCAL
channel.
2b. 'Called' LOCAL channel dials another SIP phone.
Problem: RTP stream from 'Called' SIP phone is not relayed to the 'Caller'
for the first 30 seconds.
This can be seen in 'full.log' (attached) line 1768 (when Called phone
answers) and line 1769 (30 seconds later - when audio is finally delivered
to the Caller).
How to reproduce:
1. Edit line 4 of extensions.conf (attached).
2. Copy test.call (attached) to /var/spool/asterisk/outgoing/
3. Answer 'Called' phone.
4. There will be silence for the first 30 seconds.
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Issue History
Date Modified Username Field Change
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2009-01-21 11:42 chris-mac New Issue
2009-01-21 11:42 chris-mac Asterisk Version => SVN
2009-01-21 11:42 chris-mac Regression => No
2009-01-21 11:42 chris-mac SVN Branch (only for SVN checkouts, not tarball
releases) => 1.6.1
2009-01-21 11:42 chris-mac SVN Revision (number only!) => 169673
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