Re: [GnomeMeeting-devel-list] Audio Connection needs more than 10s to be established

[Damien Sandras <dsandras seconix com>]

Hello,


* What version of Asterisk is it?
* What are you calling?
* Does it happen if you call the echo machine of Asterisk?
* Can you provide a debug 4 output? (-d 4)

Thanks,

Le mardi 18 octobre 2005 à 20:38 +0200, Christoph Lukas a écrit :
> Hi,
> 
> I have followed the gnomemeeting development for quite some month now
> and have enjoyed the Debian testing snapshot packages.
> 
> I am using the gnomemeeting snapshot version as a softphone (running on
> my laptop) registered to an asterisk server running on my local
> network's router and am experiencing the following problem:
> 
> * When I place a call to the outside world, the call is established, I
> can receive audio from the remote party but it seems that no audio is
> transmitted to the remote party. 
> * Gnomemeeting shows 'State: Standby, Incoming Codec: GSM, Outgoing
> Codec: GSM' and no statistics graph.
> * After several seconds ( from 1 - 30! ) gnomemeeting switches to
> 'State: Connected', Audio is transmitted to the remote party and
> everything works like expected.
> 
> * This behaviour does not happen on receiving calls.
> * This does not happen if I replace gnomemeeting with kphone.
> * This does also not happen if I register gnomemeeting to one of my
> SIP-Providers directly (1&1, sipgate) instead of my local asterisk
> server.
> * The local asterisk server does not show any unexpected log messages.
> 
> * I have configured gnomemeeting to operate without sip proxy, without
> stun and without NAT address translation ( both laptop and router are in
> the same subnet).
> 
> * Running ethereal one can see the following SIP communication:
> 
> No.  Time      Source       Destination  Protocol Info
>    3 1.653333  192.168.0.61 192.168.0.1  SIP/SDP  Request: INVITE sip:0XXXXXXXXXX sipgw, with session description
>    4 1.653960  192.168.0.1  192.168.0.61 SIP      Status: 407 Proxy Authentication Required
>    5 1.659131  192.168.0.61 192.168.0.1  SIP      Request: ACK sip:0XXXXXXXXXX sipgw
>    6 1.689936  192.168.0.61 192.168.0.1  SIP/SDP  Request: INVITE sip:0XXXXXXXXXX sipgw, with session description
>    7 1.690447  192.168.0.1  192.168.0.61 SIP      Status: 100 Trying
>    8 5.091282  192.168.0.1  192.168.0.61 SIP/SDP  Status: 183 Session Progress, with session description
>  184 10.580831 192.168.0.1  192.168.0.61 SIP/SDP  Status: 200 OK, with session description
>  186 10.586421 192.168.0.61 192.168.0.1  SIP      Request: ACK sip:0XXXXXXXXXX 192 168 0 1
> 1372 32.545089 192.168.0.1  192.168.0.61 SIP      Request: OPTIONS sip:cluk 192 168 0 61:5060;transport=udp
> 1373 32.546095 192.168.0.61 192.168.0.1  SIP      Status: 200 OK
> 
> The first RTP packet is received at 7.17, the first RTP packet is send
> at 30.84.
> Using kphone in the same scenario the last two SIP packets seem to be
> missing, the first RTP packet is send immediately after receiving the
> '183 Session Progress' packet.
> 
> I would really like to use gnomemeeting instead of kphone so what can I
> do to help debugging this problem? 
> 
> Thanks in advance for any help,
> Christoph
> 
> 
> _______________________________________________
> Gnomemeeting-devel-list mailing list
> Gnomemeeting-devel-list gnome org
> http://mail.gnome.org/mailman/listinfo/gnomemeeting-devel-list
-- 
 _      Damien Sandras
(o-     GnomeMeeting: http://www.gnomemeeting.org/
//\     FOSDEM 2005 : http://www.fosdem.org
v_/_    H.323 phone : callto:ils.seconix.com/dsandras seconix com