SIP works in conjunction with Resource Reservation Protocol (RSVP),
Real-time Transport Protocol (RTP), Real-time Control Protocol (RTCP),
Real-time Streaming Protocol (RTSP), Session Announcement Protocol (SAP), and
Session Description Protocol (SDP).
H.323 also works in conjunction with RTP and RTCP. Modern voice gateways
usually have two parts. The first part is the signaling gateway and the second
is the media gateway. The signaling gateway communicates with the media gateway
using MGCP. MGCP can interoperate with both SIP and H.323. Figure
shows the
signaling and transport protocols used for delivering VoIP in a SIP or H.323
network.
Signaling System 7 (SS7)
Another protocol that
should be mentioned is Signaling System 7 (SS7). SS7 is the Common Channel
Signaling (CCS) system that is used with circuit-switched networks, such as
Integrated Services Digital Network (ISDN) and the PSTN. Bellcore developed
SS7. It separates signaling information from user data. A specified channel,
called the D channel, is used exclusively to carry signaling information for
all other channels in the system. This type of signaling is called out-of-band
because it does not use the user bandwidth.
For VoIP to be able to route
calls to the PSTN, it must be able to interface with SS7, which the PSTN
uses.
ENUM
The IETF Telephone Number Resolution working group,
known as ENUM, is devising a scheme to map E.164 telephone numbers to IP
addresses using the Internet DNS. The objective is to allow any application,
including a SIP application, to discover resources associated with a unique
phone number. A SIP phone or proxy server would use number domain translation
and DNS resolution to discover a DNS resource. These DNS resources would
provide a SIP address at which the dialed number could be reached.