I'm trying to write a VoIP client using SIP and RTP on an E60 device. The message flow for creating a session from my phone works perfect, but on the receipt of initial INVITE Requests from a remote peer they are passed to the built in VoIP client (a dialog "incoming internet call" pops up) and my application wont start.
As I read in the SDK documentation "Clients must create a class derived from CSIPResolvedClient to receive requests outside SIP dialogs. The receipt of a SIP request might require the launch of the resolved target client if the client is not running."
I am supporting a Plugin, could it be there is something wrong with this or is there some other problem? Why can't my client receive initial SIP Requests?
Furthermore: I'm aware that for the audio connection I'll need to set up a full duplex rtp connection. I saw some comments on the "Audio Proxy Server" that might be of assistance in this case. Where can I find more information on what this is and how it might help me?
Any hints would be highly appreciated, thanks a lot in advance!!!!!