[sip-comm-dev] applet based on sip comm


#1

Hi guys,
I'm trying to implement an applet based on your sip communicator
services, which will allow calling. I'm using only these services:
configuration,media, netaddr and sip protocol.
Please could someone of you look at my applet init() and start()
methods? Whether I call all the necessary methods to register on
server and initialize call?

I could connect to the server and initialize the call, I can hear the
second participant, but when I try to tell him something, he can't
hear me. I'm sure that this is not problem of Nat or somefirewalls,
because we were trying it over VPN.
We tried to check differences in console output between sip
communicator and our applet and the first difference is that after
message 'We will be able to transmit in:' are in sip communicator
listed these lines:
Audio=[1]=dvi/rtp; sdp=5
Audio=[2]=ilbc/rtp; sdp=97
Audio=[4]=speex/rtp; sdp=110
Audio=[6]=g723/rtp; sdp=4
Audio=[7]=gsm/rtp; sdp=3
Audio=[8]=ULAW/rtp; sdp=0
Audio=[9]=ALAW/rtp; sdp=8

and in our applet there is no one.
Could this be the cause?

Thx
Zdenek

SipApplet.java (9.58 KB)