VoIP Functions
In the traditional PSTN telephony network, all the elements required to complete a call
are transparent to an end user. Migration to VoIP requires an awareness of these required
elements and a thorough understanding of the protocols and components that provide the
same functionality in an IP network.
Required VoIP functionality includes these functions:
■ Signaling: Signaling is the capability to generate and exchange control information
that will be used to establish, monitor, and release connections between two endpoints.
Voice signaling requires the capability to provide supervisory, address, and
alerting functionality between nodes. The PSTN network uses Signaling System 7
(SS7) to transport control messages. SS7 uses out-of-band signaling, which, in this
case, is the exchange of call control information in a separate dedicated channel.
VoIP presents several options for signaling, including H.323, Session Initiation
Protocol (SIP), H.248, Media Gateway Control Protocol (MGCP), and Skinny Client
Control Protocol (SCCP). Some VoIP gateways are also capable of initiating SS7 signaling
directly to the PSTN network. Signaling protocols are classified as either peerto-
peer or client/server protocols.
SIP and H.323 are examples of peer-to-peer signaling protocols where the end
devices or gateways contain the intelligence to initiate and terminate calls and interpret
call control messages. H.248, SCCP, and MGCP are examples of client/server
protocols where the endpoints or gateways do not contain call control intelligence
but send or receive event notifications to a server commonly referred to as a call
agent. For example, when an MGCP gateway detects a telephone that has gone off
hook, it does not know to automatically provide a dial tone. The gateway sends an
event notification to the call agent, telling the agent that an off-hook condition has
been detected. The call agent notifies the gateway to provide a dial tone.
■ Database services: Access to services, such as toll-free numbers or caller ID,
requires the capability to query a database to determine whether the call can be
placed or information can be made available. Database services include access to
billing information, caller name delivery (CNAM), toll-free database services, and
calling-card services. VoIP service providers can differentiate their services by providing
access to many unique database services. For example, to simplify fax access
to mobile users, a provider can build a service that converts fax to e-mail. Another
example is providing a call notification service that places outbound calls with prerecorded
messages at specific times to notify users of such events as school closures,
wake-up calls, or appointments.
■ Bearer control: Bearer channels are the channels that carry voice calls. Proper supervision
of these channels requires that appropriate call connect and call disconnect
signaling be passed between end devices. Correct signaling ensures that the channel
is allocated to the current voice call and that a channel is properly deallocated when
either side terminates the call. Connect and disconnect messages are carried by SS7
in the PSTN network. Connect and disconnect message are carried by SIP, H.323,
H.248, or MGCP within the IP network.
■ Codecs: Codecs provide the coding and decoding translation between analog and
digital facilities. Each codec type defines the method of voice coding and the compression
mechanism that is used to convert the voice stream. The PSTN uses TDM
to carry each voice call. Each voice channel reserves 64 kbps of bandwidth and uses
the G.711 codec to convert an analog voice wave to a 64-kbps digitized voice stream.
In VoIP design, codecs might compress voice beyond the 64-kbps voice stream to
allow more efficient use of network resources. The most widely used codec in the
WAN environment is G.729, which compresses the voice stream to 8 kbps.