Hello All,
I am working on SIP and how to pass the Audio through it.

I'll brief on what have been achieved so far

1.> Register to an Asterisk server which is associated with a PSTN gateway.

2.> Send an invite (which sends back a httpdigest challenge ) and send back user name and password and make a call to a PSTN number by inviting it. eg. " SIP:9958******@172.16.*.* ". This actions sends a ring to the phone.

And as soon as the phone is picked up after receiving a ok i want the Audio streaming should start and the Audio should be passed between both the peers.

Any suggestions towards what should i use so as to pass the Audio stream and what path should i take to achieve this.

many thanks and Regards.