CCNA Voice 05

For a call to be set up, maintained, and torn down, a signaling protocol must work behind the scenes. There are various signaling protocols in use by VoIP systems. The industry standard signaling protocols are MGCP, H.323 and SIP. Cisco has its own proprietary protocol known as the SCCP.
Of the protocols, H.323 and SIP are peer-to-peer while MGCP and SCCP are client-server based. A peer-to-peer signaling protocol doesn't require a central server to communicate properly. On the other hand, the client-server signaling protocol requires a central server to communicate. A client-server protocol cannot directly communicate with another peer, and it has limited intelligence. The client acts as a dumb terminal and the server tells it what to do when it picks up the phone, and when it dials a number, and whether it should play a tone or start ringing. It does not have its own dial plan. On the other hand, peer-to-peer clients have their own dial plan and is intelligent enough to set up calls independently of a central server.

SCCP is a Cisco-proprietary client-server protocol that supports voice and video transmission. The messages are sent in TCP first. After the call is set up, RTP is used for the actual call.

H.323 is an ITU standard based on Q.931 (ISDN). It is peer-to-peer and uses a set of protocols for various tasks:
H.225 - Registration, Admission and Status as well as call signaling
H.245 - Exchange featuresets and open channels
RTP - Used to transmit the actual data

Like SCCP, H323 supports both Voice, Video and Data transmission.

MGCP is an IEFT standard that operates in a client-server fashion. Like SCCP, it requires the server to instruct the phone to perform every operation. MGCP does not have its own dialplan and it relies on the Call Manager to do routing.

Another IETF standard is the SIP. SIP is peer-to-peer and is easier to troubleshoot due to the ASCII messages sent between the peers. SIP can be used for voice, video, and data. SIP operates on port 5060 and can use either UDP or TCP. Like most other protocols, RTP is used for the actual transmission of voice once the call is set up. SIP has four major components:
User Agent - Endpoint devices such as phones that initiate sessions
Registrar Server - Stores the location of all User Agent
Proxy Server - Accepts requests and queries the Registrar
Redirect Server - Provides routing information for the clients

The CCME or CUCME supports only H.323, SCCP and SIP. It does not support MGCP. Only the non-express version supports it MGCP.