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

noreply at bugs.digium.com noreply at bugs.digium.com
Mon Apr 7 20:24:45 CDT 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=11801 
====================================================================== 
Reported By:                manouchk
Assigned To:                dbowerman
====================================================================== 
Project:                    Asterisk
Issue ID:                   11801
Category:                   Addons/chan_mobile
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     assigned
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 98514 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             01-20-2008 12:47 CST
Last Modified:              04-07-2008 20:24 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).

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

---------------------------------------------------------------------- 
 zaterio - 04-07-08 20:24  
---------------------------------------------------------------------- 
With the sony ericcson  k510 the problem are similar:

when I calling from a sip client: if in the sip client side there are no
sound i cant hear anithing from another side, for example when i say
"start" in the another side the person begins to count from 1 to 10 (1 seg
steps) , i can hear 1...2.. and no more. when i restart talking i can hear
again. I planig a one generator in the asterisk to maintaing the
comunication.

the configs:
usb dongles: cambride chipset
asterisk-addons svn version: 576
asterisk trunk svn: 112765
kernel 2.6.18
debian OS 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
04-07-08 20:24  zaterio        Note Added: 0085114                          
======================================================================




More information about the asterisk-bugs mailing list