[asterisk-bugs] [Asterisk 0012566]: Loss of audio after 2 seconds of P2P RTP bridging

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12566 
====================================================================== 
Reported By:                falves11
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   12566
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 114912 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             05-01-2008 13:23 CDT
Last Modified:              05-08-2008 14:46 CDT
====================================================================== 
Summary:                    Loss of audio after 2 seconds of P2P RTP bridging
Description: 
I have a peer like this

[federico]
type=peer
host=72.187.243.97
context=default
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=g729
allow=ulaw
nat=yes
insecure=port,invite

but when I type "sip show peers"
ame/username              Host            Dyn Nat ACL Port     Status
federico                   72.187.243.97        N      5060    
Unmonitored

and in fact the calls connect and there is second of audio and then the
audio stops. I am behind a nat.
====================================================================== 

---------------------------------------------------------------------- 
 elguero - 05-08-08 14:46  
---------------------------------------------------------------------- 
I think I am seeing the same issue and maybe I can shed some more light on
this with what I am experiencing.  I am using trunk, r115537

My setup: Asterisk is on my router.  Therefore it has a public IP.  On
asterisk I have a SIP registration to a SIP provider.  I then have on the
LAN side a Polycom 430.  Phone calls from SIP provider dial the Polycom. 
(I do have realtime setup so the Polycom is registered to Asterisk using
realtime, not sure if that matters).

file:

1.  Destination in my case is a Polycom

2.  I am using trunk r115537

3.  Core show locks attached

4.  Core debug attached

Just in case, RTP debug attached.

Now, outbound calling seems to be working fine.  This only affects
incoming calls.  I have tried to stop P2P bridging (directrtp off) but that
doesn't seem to work and asterisk is still trying to switch to P2P.

Let me know if I can provide any further information to help track this
down. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
05-08-08 14:46  elguero        Note Added: 0086632                          
======================================================================