[asterisk-bugs] [Asterisk 0011801]: mobile to asterisk audio stability strongly depends on asterisk to mobile audio activity

[Asterisk Bug Tracker]

The following issue has been CLOSED 
====================================================================== 
http://bugs.digium.com/view.php?id=11801 
====================================================================== 
Reported By:                manouchk
Assigned To:                mnicholson
====================================================================== 
Project:                    Asterisk
Issue ID:                   11801
Category:                   Addons/chan_mobile
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 98514 
Request Review:              
Resolution:                 open
Fixed in Version:           
====================================================================== 
Date Submitted:             2008-01-20 12:47 CST
Last Modified:              2009-04-21 13:13 CDT
====================================================================== 
Summary:                    mobile to asterisk audio stability strongly depends
on asterisk to mobile audio activity
Description: 
In a simple testing configuration with a remote mobile (mobile R), a remote
connected to asterisk by bluetooth (mobile A) and a sip phone (I 'm using
x-lite for the test), I found that the stability of the audio flux from
mobile to asterisk strongly depends on the activity asterisk to mobile
volume in a connexion between the sip phone and the remote mobile.

It means that the lag can be very high about 8 seconds and that some audio
parts from the mobile are lost (if no sound from asterisk to mobile)

If in the contrary there is sound made on the sip phone side, this sound
is firstly perfectly transmitted to the mobile and the lag is only about 1
or 2 seconds for the audio coming from the mobile to asterisk (and then the
sip phone).

======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0011556 "No audio" on incoming blueto...
has duplicate       0012768 Multipile issues with chan_mobile
related to          0011368 chan_mobile does not recognize dtmf tog...
child of            0012567 Big latency (up to 3 sec) when call wai...
====================================================================== 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-21 13:13 mnicholson     Status                   ready for testing =>
closed
======================================================================