Asterisk staying in the path of the media


We all know that Asterisk always stays in the Media RTP path for many reasons including billing. That’s, I guess, the main reason why Asterisk doesn’t scale very well.

I just want to know how other SIP servers actually perform almost the same functions, including billing, without having to be involved with RTP.

Anyone can explain this dilemma?


The asterisk should not need to be in the media path, definitely not for billing purposes. As long as the SIP messaging is going through the asterisk, it should be well aware of all call details.

I’m pretty sure there’s a configuration setting to actually remove the asterisk from the media path, even in a LAN environment. I’m sorry I can’t tell you the exact command, but I’m pretty new with PBX configs.

Do some reading on canreinvite=yes

Download this book FREE from Digium start reading at the bottom of page 82 … 510480.pdf

This is a great book, I have a copy.

I know about SIP “canreinvite” but what puzzles me is what the book says “where media initially goes through Asterisk untill a re INVITE happens”.

When and what makes IP phones send a reINVITE?? I understand that phones do that in case of enduser putting another on HOLD or forwarding the call to a thrid party etc… other than that there will be no reINVITEs.

Am I correct in that?


Here are a couple of links that may help you out; … media+path … anreinvite

Other things that may stop invites (canreinvite=yes) working.

  1. You have it disabled the phone
  2. You have it disabled in sip.conf
  3. You have an option configured that keeps asterisk in the call path like dial(SIP\301,30,T) it is the T in this case.

With invites configured and working IP Phone A can communicate with IP Phone B that is the RTP stream goes direct it is not passing through two Asterisk boxes so it will have less delay.

IP Phone A—Asterisk--------------------------------Asterisk—IP Phone B