Asterisk and SIP Call Path Optimization

Hi all,

Asterisk support natively the Call Path Optimization if using IAX2 protocol.

In this way Asterisk allows two endpoint to exchange their signalling/media directly.

I would like to make the same thing in SIP, but I found no way to do it via sip.conf or extensions.conf .

Is there a way to do it?

If it is not allowed Asterisk developpers should think to implement it.

Any opinion about this matter would be appreciated,

thank you very much in advance for your attention,


Hi, if i’ve understood u want two endpoints to exchange media directy, right?

I know that writing in sip.conf canreinvite=yes, let the endpoint to perform a reinvite


“When SIP initiates the call, the INVITE message contains the information on where to send the media streams. Asterisk uses itself as the end-points of media streams when setting up the call. Once the call has been accepted, Asterisk sends another (re)INVITE message to the clients with the information necessary to have the two clients send the media streams directly to each other”

See … anreinvite for more explanation

Hi Nadia_IE,

what you say is interesting but does not solve my question, I would like to Call Path Optimize also the signalling flow, so to totally remove Asterisk from the path.

Did anyone make something similar in Asterisk with two SIP calls?