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

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11801 
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Reported By:                manouchk
Assigned To:                dbowerman
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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:              
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Date Submitted:             01-20-2008 12:47 CST
Last Modified:              05-18-2008 02:35 CDT
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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).

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---------------------------------------------------------------------- 
 ughnz - 05-18-08 02:35  
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I can confirm that this is happening and can re-produce it 100%

Using BOL Sip phone a software based SIP client and turning the mic gain
right down will stop all audio from the cellphone. Turning the mic gain up
and making a sound will start the audio from the cellphone until you turn
the gain down again.

Watching traffic to <> from the SIP client with the mic gain turned right
down in is not sending any RTP traffic which results in no RTP traffic from
the cellphone.

With all other channels having the mic gain right down does not effect the
RTP stream.

Using asterisk 1.4.19-1 with chan_mobile rev454 & bluez 3.31 

Issue History 
Date Modified   Username       Field                    Change               
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05-18-08 02:35  ughnz          Note Added: 0086984                          
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