MEDIA GATEWAY CONTROLLER AND ITS PROTOCOLS
of status and control. This protocol information is embedded in UDP, which is reliably
transported by TCP.
Other VoIP entities are gateways and there is a gatekeeper that is optional.
The leading issue in VoIP implementation is guaranteed quality of service (QoS).
H.323 is based on RTP (real-time protocol) which is comparatively new. RTP-compliant
equipment includes control mechanisms for synchronizing different traffic streams. On
the other side of the coin, RTP has no mechanisms for ensuring on-time delivery of
traffic signals or for recovering lost packets. It does not address the QoS issue related to
guaranteed bit rate availability for specific applications. The IEC  reports that there
is a draft signaling proposal to strengthen the Internet's ability to handle real-time traffic
reliably. This would dedicate end-to-end transport paths for specific sessions much like
the circuit-switched PSTN does. This is the resource reservation protocol (RSVP). It will
be implemented in routers to establish and maintain requested transmission paths and
Session Initiation Protocol (SIP)
SIP is based on RFC 2543  and is an application layer signaling protocol. It deals with
interactive multimedia communication sessions between end-users, called user agents.
It defines their initiation, modification, and termination. SIP calls may be terminal-to-
terminal, or they may require a server to intercede. If a server is to be involved, it is only
required to locate the called party. For interworking with non-IP networks, Megaco and
H.323 are required. Often vendors of VoIP equipment integrate all three protocols on a
SIP is closely related to IP. SIP borrows most of its syntax and semantics from the
familiar HTTP (hypertext transfer protocol). An SIP message looks very much like an
HTTP message, especially with message formatting, header, and multipurpose Internet
mail extension support. It uses addresses that are very similar to URLs (uniform resource
locators) and to email. For example, a call may be made to so-in-so@such-and-such. SIP
messages are text-based rather than binary. This makes writing easier and the debugging
of software more straightforward.
There are two modes with which a caller can set up a call with SIP. These are called
redirect and proxy, and servers are designed to handle these modes. Both modes issue
an "invite" message for another user to participate in a call. The redirect server is used
to supply the address (URL) of an unknown called addressee. In this case the "invite"
message is sent to the redirect server, which consults the location server for address
information. Once this address information is sent to the calling user, a second "invite"
message is issued, now with the correct address.
One specific type of SIP is called SIP-T (T for telephone). This is a function that allows
calls from CCITT Signaling System 7 (SS7) to interface with a telephone in an IP-based
network. The particular user part of SS7 for this application is ISUP (see Chapter 14, SS
Media Gateway Control Protocol (MGCP)
This protocol was the predecessor to Megaco (see Section 15.4.4) and still holds sway with
a number of carriers and other VoIP users. MGCP  assumes a call control architecture
where the call control "intelligence" is outside the gateways (i.e., at the network edge) and
handled by external call control elements. Thus, the MGCP assumes that these call control
elements, or "call agents," will synchronize with each other to send coherent commands