Compare various VoIP signaling protocols

Voice Over IP:Voice-over-IP (VoIP) implementations enables users to carry voice traffic (for
example, telephone calls and faxes) over an IP network. There are 3 main causes for the evolution of the Voice over IP market:

Low cost phone calls Add-on services and unified messaging Merging of data/voice infrastructures

A VoIP system consists of a number of different components: Gateway/Media Gateway, Gatekeeper, Call agent, Media Gateway Controller, Signaling Gateway and a Call manager The following VoIP protocols are described here: Megaco H.248 MGCP MIME RVP over IP SAPv2 SDP SGCP SIP Skinny Gateway Control Protocol Media Gateway Control Protocol Remote Voice Protocol Over IP Specification Session Announcement Protocol Session Description Protocol Simple Gateway Control Protocol Session Initiation Protocol Skinny Client Control Protocol (SCCP)

Megaco (H.248): The Media Gateway Control Protocol, (Megaco) is a result of joint efforts of the IETF and the ITU-T Study Group 16. The protocol definition of this protocol is common text with ITU-T Recommendation H.248. The Megaco protocol is used between elements of a physically decomposed multimedia gateway. There are no functional differences from a system view between a decomposed gateway, with distributed sub-components potentially on more than one physical device, and a monolithic gateway such as described in H.246. This protocol creates a general framework suitable for gateways, multipoint control units and interactive voice response units (IVRs). MGCP: Media Gateway Control Protocol (MGCP) is used for controlling telephony gateways from external call control elements called media gateway controllers or call agents. A telephony gateway is a network element that provides conversion between the audio signals carried on telephone circuits and data packets carried over the Internet or over other packet networks. MGCP assumes a call control architecture where the call control intelligence is outside the gateways and handled by external call control elements. The MGCP assumes that these call control elements, or Call Agents, will synchronize with each other to send coherent commands to the gateways under their control. MGCP is, in essence, a master/slave protocol, where the gateways are expected to execute commands sent by the Call Agents. MGCP Commands MGC --> MG CreateConnection: Creates a connection between

MGC --> MG MGC <--> MG

two endpoints; uses SDP to define the receive capabilities of the paricipating endpoints. ModifyConnection: Modifies the properties of a connection; has nearly the same parameters as the CreateConnection command. DeleteConnection: Terminates a connection and collects statistics on the execution of the connection.

MIME: This set of standards, collectively called the Multipurpose Internet Mail Extensions, or MIME, redefine the format of messages to allow for textual message bodies in character sets other than US-ASCII, an extensible set of different formats for non-textual message bodies, multi-part message bodies, and textual header information in character sets other than US-ASCII. The initial standard in this set, RFC 2045, specifies the various headers used to describe the structure of MIME messages. RFC 2046 defines the general structure of the MIME media typing system and defines an initial set of media types. The third standard, RFC 2047, describes extensions to RFC 822 to allow non-US-ASCII text data in Internet mail header fields. The fourth standard, RFC 2048, specifies various IANA registration procedures for MIME-related facilities. The fifth and final standard, RFC 2049, describes MIME conformance criteria as well as providing some illustrative examples of MIME message formats, acknowledgements, and the bibliography.

RVP over IP: Remote Voice Protocol (RVP) is MCK Communications' protocol for transporting digital telephony sessions over packet or circuit based data networks. The protocol is used primarily in MCK's Extender product family, which extends PBX services over Wide Area Networks (WANs). RVP provides facilities for connection establishment and configuration between a client (or remote station set) device and a server (or phone switch) device. RVP/IP uses TCP to transport signalling and control data, and UDP to transport voice data. SAPv2: SAP is an announcement protocol that is used by session directory clients. A SAP announcer periodically multicasts an announcement packet to a well-known multicast address and port. The announcement is multicast with the same scope as the session it is announcing, ensuring that the recipients of the announcement can also be potential recipients of the session the announcement describes (bandwidth and other such constraints permitting). This is also important for the scalability of the protocol, as it keeps local session announcements local. SDP: The Session Description Protocol (SDP) describes multimedia sessions for the purpose of session announcement, session invitation and other forms of multimedia session initiation. On Internet Multicast backbone (Mbone) a session directory tool is used to advertise multimedia conferences and communicate the conference addresses

and conference tool-specific information necessary for participation. The SDP does this. It communicates the existence of a session and conveys sufficient information to enable participation in the session. Many of the SDP messages are sent by periodically multicasting an announcement packet to a well-known multicast address and port using SAP (session announcement protocol). These messages are UDP packets with a SAP header and a text payload. The text payload is the SDP session description. Messages can also be sent using email or the WWW (World Wide Web). SIP: Session Initiation Protocol (SIP) is a application layer control simple signalling protocol for VoIP implementations using the Redirect Mode. SIP is a textual client-server base protocol and provides the necessary protocol mechanisms so that the end user systems and proxy servers can provide different services: 1. Call forwarding in several scenarios: no answer, busy , unconditional, address manipulations (as 700, 800 , 900- type calls). 2. Callee and calling number identification 3. Personal mobility 4. Caller and callee authentication 5. Invitations to multicast conference 6. Basic Automatic Call Distribution (ACD) SGCP: Simple Gateway Control Protocol (SGCP) is used to control telephony gateways from external call control elements. A telephony gateway is a network element that provides conversion between the audio signals carried on telephone circuits and data packets carried over the Internet or over other packet networks. The SGCP assumes a call control architecture where the call control intelligence is outside the gateways and is handled by external call control elements. The SGCP assumes that these call control elements, or Call Agents, will synchronize with each other to send coherent commands to the gateways under their control. Skinny: Skinny Client Control Protocol (SCCP). Telephony systems are moving to a common wiring plant. The end station of a LAN or IP- based PBX must be simple to use, familiar and relatively cheap. The H.323 recommendations are quite an expensive system. An H.323 proxy can be used to communicate with the Skinny Client using the SCCP. In such a case the telephone is a skinny client over IP, in the context of H.323. A proxy is used for the H.225 and H.245 signalling.