hi, i have create sip session. now i want to use it for VOIP using j2m. how can i do this. i mean which APIs are helpful in developing it. i hav following questions
1. how can i get sound from mic to use in app and in which format
2. is it ok to buffer sound n thn transfer it over network?
3. voice bufer direclty goes to other client mobile or processed at sipserver before reaching the clinet mobile.
4. audio streaming is possible or not for S60 3rd edition.
if u hav some docs and guidlines other thn these, plz tell me.