Not quite. With SIP, once the call is established, the audio is supposed to go directly between phones whenever possible. This means that if both phones have real world IP addresses, with no NAT etc., then they will be able to communicate with whatever CODEC they choose. So, if both phones support G.722 and have it selected, then it can be used for that call. It's only when something, such as NAT, firewalls etc. prevent direct communication is the audio forced through a proxy or server. There is a method for VoIP PBXs to communicate with each other, for the purposes of setting up that phone to phone IP connection. This is covered in the Asterisk book*
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