Using Asterisk as SIP registrar

Hi All,

I have an Audiocodes TDM to SIP converter (Model: M1000)…I would like to use it to terminate TDM calls to IP and have the call send to a SIP user.

In this case Audiocodes, Asterisk and SIP users are all behind NAT… Audiocode and Asterisk on one NAT and clients are behind multiple NAT.

I also came across OpenSIPs… which one is better to use in my scenario? Can I use Asterisk to be a SIP registrar only?

and can someone please clarify this for me as well: if I use Asterisk as SIP registrar, which one will handle the Audio? I think it’s the audiocodes that does everything…no? to be clear, if the user is using G729, will the CPU on Asterisk be used or Audiocodes?


Can someone answer this question please?

Can I use Asterisk as SIP Registrar only?