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:firstname.lastname@example.org.*.* ". 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.