Hi, Am a SIP newbie and I need some help, this isn't an opensips question, I just need some guidance and pointers and I hope someone can help me.
I am developing a web-basedsip softphone using Opensips as the proxy. Eventually I want to integrate the proxy with Asterisk for PSTN calling. My client is working fine and a 3 way handshake is successfully done. What do I need to do to carry the call from the UAC to the UAS once the session has been setup? Will I need to license codecs to handle the call? As I mentioned, this isn't about Opensips, just need some pointers and guidance. Any help will be appreaciated. regards, james
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