[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 12:17 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.
====================================================================== 

---------------------------------------------------------------------- 
 falves11 - 05-08-08 12:17  
---------------------------------------------------------------------- 
The ssh connection freezes and also the CDR does not get written to the
database (cdr_odbc). It only happens when I call from behind NAT, otherwise
works fine. Also, if I hangup in the origin, the destination leg remains
connected. But I get a reply to my "BYE" message. I might be getting it
because I set the DMZ to my computer, otherwise I would not. It is like it
does not understand how to handle NAT anymore.
What kind of debugging ca we do if it freezes? However, I disconnect and
reconnect again and Asteriksk works,

-- Called 19544447408 at 4.71.122.130
    -- SIP/4.71.122.130-b4054d48 is making progress passing it to
SIP/45990-0366b878
    -- SIP/4.71.122.130-b4054d48 answered SIP/45990-0366b878
    -- Packet2Packet bridging SIP/45990-0366b878 and
SIP/4.71.122.130-b4054d48
We could NOT get the channel lock for SIP/207.155.147.30-03677968!
SIP transaction failed: 682c66d823d8ecfe1f03aa4378768a0e at minixel.com
We could NOT get the channel lock for SIP/207.155.147.30-03677968!
SIP transaction failed: 682c66d823d8ecfe1f03aa4378768a0e at minixel.com
We could NOT get the channel lock for SIP/207.155.147.30-03677968!
SIP transaction failed: 682c66d823d8ecfe1f03aa4378768a0e at minixel.com
We could NOT get the channel lock for SIP/207.155.147.30-03677968!
SIP transaction failed: 682c66d823d8ecfe1f03aa4378768a0e at minixel.com
We could NOT get the channel lock for SIP/207.155.147.30-03677968!
SIP transaction failed: 682c66d823d8ecfe1f03aa4378768a0e at minixel.com
We could NOT get the channel lock for SIP/207.155.147.30-03677968!
SIP transaction failed: 682c66d823d8ecfe1f03aa4378768a0e at minixel.com
We could NOT get the channel lock for SIP/207.155.147.30-03677968!
SIP transaction failed: 682c66d823d8ecfe1f03aa4378768a0e at minixel.com
Maximum retries exceeded on transmission
682c66d823d8ecfe1f03aa4378768a0e at minixel.com for seqno 103 (Critical
Request) 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
05-08-08 12:17  falves11       Note Added: 0086622                          
======================================================================