Introducing VoIP Signaling Protocols

VoIP signaling protocols are responsible for call setup, maintenance, and teardown. A number of different protocols are in use—some standards-based, others proprietary, and each with advantages and disadvantages. The following sections introduce the signaling protocols you should know about, including SCCP, H.323, MGCP, and SIP.


VoIP Signaling Protocols

VoIP signaling protocols handle the call setup, maintenance, and teardown functions of VoIP calls. It is important to keep in mind that the signaling functions are an entirely separate packet stream from the actual voice bearer path (RTP). The signaling protocol in use must pass the supervisory, informational, and address information expected in any telephony system.

VoIP signaling protocols are either peer-to-peer or client-server; in the case of peer-to-peer protocols, the endpoints have the intelligence to perform the call-control signaling themselves, Client-server protocols send event notifications to the call agent (the Unified CM server) and receive instructions on what actions to perform in response. The following table summarizes the characteristics of the four signaling protocols dealt with here.


H.323
H.323 is not itself a protocol; it is an umbrella standard that defines several other related protocols for specific tasks. Originally conceived as a multimedia signaling protocol to emulate traditional telephony functionality in IP LAN environments, it is a long-established and stable protocol very suitable for intervendor compatibility. H.323 is supported by all Cisco voice gateways and CM platforms as well as some third-party video endpoints.


MGCP

Media Gateway Control Protocol is a lightweight client/server protocol for PSTN gateways and some clients. It is simple to configure and allows the call agent to control the MGCP gateway, eliminating the need for expensive gateways with intelligence and complex configurations. The gateway reports events such as a trunk going off-hook, and the call agent instructs the gateway on what to do; the gateway has no local dial plan because all call routing decisions are made at the call agent and relayed to the MGCP gateway. MGCP is not as widely implemented as SIP or H.323. MGCP is not supported by Unified CM Express or the Smart Business Communication System.


SIP

Session Initiation Protocol is an IETF standard that uses peer-to-peer signaling. It is very similar in structure and syntax to HTTP, and because it is text-based, it is relatively simple to debug and troubleshoot. SIP can use multiple transport layer protocols and can support security and proxy functions. SIP is an evolving standard that currently provides basic telephony functionality; further developments and extensions to the standard will soon make it feature-comparable with SCCP. One of its most important capabilities is creating SIP trunks to IP Telephony service providers, replacing or enhancing traditional TDM PSTN connections.


SCCP

Skinny Client Control Protocol is a Cisco-proprietary signaling protocol used in a client-server manner between Unified CM and Cisco IP Phones (and some Cisco gateways). SCCP uses TCP connections to the Unified CM to set up, maintain, and tear down voice and video calls. It is referred to as a stimulus protocol, meaning that it sends messages in response to events such as a phone going off-hook or a digit being dialed. SCCP is the default signaling protocol for all Cisco IP phones, although many also support SIP; SIP does not yet support the full feature set available to SCCP phones. All Cisco Unified Communications call agents (CM, CM Express, and the 500 Series) and some gateways support SCCP.