VoIP and Unified Communications services depend on signaling protocols to establish a call or connection. These protocols performs basic call setup functions, number translation, feature negotiation, and call management. If the receiving device is available, the call server contacts it and instructs both the caller and called devices to establish a peer-to-peer UDP path to carry RTP audio/video traffic (Real-time Transport Protocol – RFC 3550). Call signaling protocols can be either standards-based or vendor / proprietary-based.

Standard-based Signaling Protocols

  • H.323 - International Telecommunications Union (ITU) designed the H.323 protocol suite to define how multimedia, such as video and audio travel over a packet-switched network. Although widely deployed due to its early availability, it is now being replaced in many situations by SIP.
  • MGCP - Media Gateway Control Protocol, an IETF standard (RFC 3435), operates at the backbone of the network and typically used by network elements like call agents which routes calls between gateways and the PSTN (public switched telephone network).
  • SIP - Session Initiation Protocol was produced by the IETF as an IP standard (RFC 3261) for call/session set up, modification and termination. SIP has broad applicability in enabling voice and video connectivity, as well as instant messaging, across the IP based Internet, enables better interoperability among vendors, easier application development, and easier operation through firewalls.

Proprietary Signaling Protocols

Several VoIP vendors offer proprietary signaling protocols in addition to supporting one or more standards-based protocols. A couple of the most commonly deployed include:

  • SCCP - The Skinny Client Control Protocol is a proprietary signaling protocol used by Cisco Systems.
  • Extended H.323 - Avaya’s proprietary signaling protocol which is based on the standard H.323.  

The time required for the Call Manager Servers using signaling protocols to successfully establish the call can have significant impact on user experience. To manage this part of a VoIP implementation requires tools that closely monitor performance of signaling protocol-related metrics such as call set-up times, incomplete/failed calls and errors.