[Ekiga-list] connection, but no sound

[Christoph Groth]

Hi all,

I would like to call a friend with ekiga.  Both of us sit behind a
router/firewall: Mine is configured to forward UDP ports 5000-5100 to
my machine.  I do not know anything about my friend's firewall.

We both have accounts with at the German sip-phone-service "web.de
freephone".  Using this service communication works.  Both of us can
call other (traditional) phones via the web.de gateway.  Also I can
call my friend.  I think web.de is routing all the media data, because
when we connect via web.de we are both forced to PCMU and the service
quality is quite bad.  Furthermore it is often necessary to try
setting up a call many times until it actually succeeds.

That is why we tried using ekiga.net accounts.  This time we can
reliably achieve a connection either way at the first try.  SPEEX is
being negotiated as audio codec, however, no one can hear anything.

After receiving ACK from the other computer my machine keeps logging
things like:

2006/11/15 22:46:30.623   0:24.037        SIP Transport:848c368 SIP     Queueing PDU: 1 ACK sip:cwg at 86.83.156.104:5065;transport=udp
2006/11/15 22:46:30.623   0:24.037        SIP Transport:848c368 SIP     Waiting for PDU on udp$213.186.62.145:5060<if=udp$192.168.1.80:5065>
2006/11/15 22:46:30.623   0:24.037          SIP Handler:84acae0 SIP     Handling PDU 1 ACK sip:cwg at 86.83.156.104:5065;transport=udp
2006/11/15 22:46:30.623   0:24.038          SIP Handler:84acae0 SIP     ACK received: ConnectedPhase
2006/11/15 22:46:30.624   0:24.038          SIP Handler:84acae0 GMSIPEndpoint    SIP connection established
2006/11/15 22:46:30.634   0:24.048          SIP Handler:84acae0 RTP     Found existing session 1
2006/11/15 22:46:30.634   0:24.048          SIP Handler:84acae0 RTP     Found existing session 2
2006/11/15 22:46:30.634   0:24.048          SIP Handler:84acae0 GMManager        Will establish the connection
2006/11/15 22:46:30.634   0:24.048          SIP Handler:84acae0 OpalMan OnEstablished Call[1]-EP<sip>[e6f8f96b-6073-db11-95ca-0003c909570f at spheniscus]
2006/11/15 22:46:30.634   0:24.048          SIP Handler:84acae0 Call    OnEstablished Call[1]-EP<sip>[e6f8f96b-6073-db11-95ca-0003c909570f at spheniscus]
2006/11/15 22:46:30.637   0:24.051          SIP Handler:84acae0 OpalCon Media stream threads started.
2006/11/15 22:46:30.637   0:24.051          SIP Handler:84acae0 SIP     Awaiting next PDU.
2006/11/15 22:46:30.688   0:24.103          Media Patch:8396180 Silence Threshold increased to: 7
2006/11/15 22:46:31.008   0:24.423          Media Patch:8396180 Silence Threshold increased to: 8
2006/11/15 22:46:31.328   0:24.742          Media Patch:8396180 Silence Threshold increased to: 9
2006/11/15 22:46:31.637   0:25.051                  Housekeeper RTP     Found existing session 1
2006/11/15 22:46:31.637   0:25.051                  Housekeeper RTP     Found existing session 2
2006/11/15 22:46:31.648   0:25.062          Media Patch:8396180 Silence Threshold increased to: 10
2006/11/15 22:46:31.968   0:25.382          Media Patch:8396180 Silence Threshold increased to: 11
2006/11/15 22:46:32.288   0:25.702          Media Patch:8396180 Silence Threshold increased to: 12
2006/11/15 22:46:32.349   0:25.763          Media Patch:8396180 RTP     Transmit statistics:  packets=101 octets=5252 avgTime=20 maxTime=40 minTime=16
2006/11/15 22:46:32.608   0:26.022          Media Patch:8396180 Silence Threshold increased to: 13

I wonder if someone has an idea how to solve this problem?  I
especially wonder why RTP works with the web.de service but not with
ekiga.net.

Thank you in advance

Christoph