we developed a VOIP application on symbian 3rd Edition FP2 using symbian c++ . Now we started testing the application in the device(E72). On the other end we are using X-lite on the system where Brekeke SIP server is installed. We are able to send and receive SIP requests and when we start RTP, RTP is transfered from Device to X-lite and vice versa. On X-lite ,when we play RTP packets ,it's very clear. But after receiving RTP packet from X-lite,we were unable to play RTP packet in the E72 device.
There is no problem in execution,but voice is not routed.Is there any problem in the conversions? or is there any problem in E72 audio routing? we were struck here from many days and still didn't got any solution.
Thanks in Advance