This thread has gotten rather confused....
1) VoIP generally implied voice band. Tieline, Comrex, APT, and Barix are better described as Audio over IP to distinguish that they are intended for voice-with-backround-noise, and music, as well as voice.
2) The mU Law codec (part of ITU G.711) is familiar to all as it is the codec used for voice telephone calls in the US. It is limited to 300-3.4 kHz and using a whopping 64 kbps to do so. It is most certainly not similar to G.722 in quality - G.722 can achieve 7 to 7.5 kHz bandwidth at 56 kbps! It is low delay.
3) Delay is more critical when using IP since packetization and routing ad substantial delay
For more on this and other factors involved with Audio over IP check out our the new third edition of our Audio over IP guide here (or contact me off line and I can send you the printed version):
http://www.aptx.com/Admin/Editor/Assets/PDF/IP Audio Networking.pdf
4) The worse the network, the more buffering will be required. This delay becomes a bigger issue. Enhanced apt-X, included in all of our codecs, and licensed to many codec manufacturers worldwide, has an encode/decode delay of less than 3 msec. It the same 64 k bits per second that mU law uses for mono phone quality audio, it can give you two channels of 20-3.5 kHz pro-grade (low noise and distortion) audio.
I hope this helps untangle this thread.
Rolf Taylor
Applications/Support Engineer
APT North America
800-955-APTX (2789)