[Ekiga-list] connection, but no sound
Damien Sandras
dsandras at seconix.com
Sat Nov 18 13:01:24 UTC 2006
Le vendredi 17 novembre 2006 à 20:09 +0100, Christoph Groth a écrit :
> 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.
>
Please post a full output.
Again, I would say that for some obscure reason your router is not
relaying the media.
--
Damien Sandras
Ekiga Softphone : http://www.ekiga.org
NOVACOM : http://www.novacom.be
FOSDEM : http://www.fosdem.org
SIP Phone : sip:dsandras at ekiga.net
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