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

[Christoph Lukas <christoph lukas gmx net>]

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