With most business that integrate VoIP into their network, they not use VoIP technology within their own LAN but also across WAN's to remote sites, other businesses, ISP's, etc. There are a few popular voice trunking protocols to pick from and I'm starting to learn the history and details behind each. There are four voice gateway protocols that I hear and see configured quite a bit in the voice networks I've seen so far.
1. H.323: This is a pretty mature and older voice protocol that was created primarily for connecting to other networks using the ISDN connection. This doesn't surprise me as I see this used mainly on PSTN connections to ISP's. While the protocol is pretty stable, there are a lot of features newer protocols handle better along with being easier to manage. When I run the debug command debug isdn q931 on a gateway, the information is usually pretty cryptic and hard to undertstand.
2. MGCP: I see this protocol used all the time in the Cisco VoIP networks, apparently Cisco is really the only vendor that uses this protocol even though it's an open standard. MGCP is a centralized based configuration.
3. SCCP: Again this is only on Cisco VoIP networks because this is Cisco's proprietary protocol. This protocol is used specifically for communication with Cisco IP Phones and other Cisco endpoints such as ATA's.
4. SIP: This is the new kid on the block and it's still being developed constantly. SIP is basically a lighter and more feature rich version of H.323. Except that it is based on VoIP instead of ISDN, it uses a lot of the same standards that HTTP uses which makes it a lot simpler to develop. I haven't touched SIP yet but I've heard a LOT about it. There's actually a forum that is very involved with this new protocol, check it out at www.sipforum.org