[asterisk-bugs] [Asterisk 0016941]: SIP RTP audio delay

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16941 
====================================================================== 
Reported By:                sharvanek
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16941
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           1.4.29.1 
JIRA:                       SWP-990 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 suspended
Fixed in Version:           
====================================================================== 
Date Submitted:             2010-03-01 20:42 CST
Last Modified:              2010-03-23 12:50 CDT
====================================================================== 
Summary:                    SIP RTP audio delay
Description: 
This is *not* a latency issue.

Here's a full description:
http://forums.digium.com/viewtopic.php?f=1&t=73267&start=0&sid=404b57776485f8f2e376bfaf1d394dd5

This is a very very odd issue, when a caller comes in through two asterisk
boxes via SIP the IVR is heard fine, but when transferred from the IVR the
caller and the person being called experience about 2-3 seconds of silence
before audio begins to flow.

This has been reported numerous times across the net but no real solution,
some examples are:

http://www.mail-archive.com/asterisk@uc.org/msg05938.html
http://www.trixbox.org/forums/trixbox-forums/help/3-second-delay-answering-calls

sadly there isn't much debug here, a packet capture shows RTP flowing etc
the same in both circumstances, please advise.


====================================================================== 

---------------------------------------------------------------------- 
 (0119753) lmadsen (administrator) - 2010-03-23 12:50
 https://issues.asterisk.org/view.php?id=16941#c119753 
---------------------------------------------------------------------- 
Suspended due to lack of feedback from the reporter. If you're able to
provide the information requested, then feel free to reopen the issue and
attach it. Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-03-23 12:50 lmadsen        Note Added: 0119753                          
2010-03-23 12:50 lmadsen        Status                   feedback => closed  
2010-03-23 12:50 lmadsen        Resolution               open => suspended   
======================================================================