G722 (HDVoice) is a new Codec that is supported by most current VOIP Phones, but NOT by most VOIP Providers.
When I enable G722 on my system and make a call using a phone which supports G722, FreePBX places all calls in G722 mode, regardless of whether they are internal or external calls. For external calls, this means that Asterisk must transcode the call from G722 to G711, which introduces another latency step.
I'd prefer to have some way to set the Codec by route, i.e. so that when I make internal calls, the calls use G722, but when I make an external call using a SIP Trunk, the call goes out on G711 (avoiding the transcoding/latency issues).
I suggest that FreePBX add a feature that allows a user to select an internal vs. external Codec, or to set an outbound route Codec that will override the general Asterisk SIP codecs.