I'm developing a SIP VOIP client on Symbain.The Sip example in the SDK quite good to understand the SIP basics.Now i'm looking for an example that demonstrate the RTP APIs.I spent good time to find out help for that but not able to get it.Please help me, i'm wondering how can i open a duplex channel between two terminals for transmitting a real voice data.
I came to know that i've to use RTP frame work for the transmission of voice data as i'm using SIP for Signals.would that be fine to use RTP Stack for Series60 or i need to implement my own RTP stack.
Please let me know ASAP incase u know any other better forum for this thread.
I am also working on that thing can you told me your sip client is able to receiving a INVITE request from Network or not?
If yes then gave me some idea how you have done this thing.
My sip client is able to sending INVITE and REGISTER request but could not
receive a INVITE.
plz help.
Thanks,
Navneetbond