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Morris WL040/Bidgoli-Vol III-Ch-53 September 15, 2003 12:23 Char Count= 0
656 VOICE OVERINTERNETPROTOCOL(IP)xyz.comCalling
TelephoneCalled
TelephoneIP Network
IP Router
or SwitchProxy &
Location
Servers- Look Up Gateway
[email protected] - RING to
[email protected] - OK to
[email protected] - ACK to
IP of xyz - OK
- RTP
(Voice Stream)
IP PacketsVoice
Mail[email protected]
[email protected]- RING to
[email protected] - INVITE to
[email protected] - INVITE to
[email protected]
xyz.comFigure 9: VOIP call routed using SIP signaling.For a proxy server, the location server will return the
IP address for the called telephone. Using that IP address,
the SIP INVITE message will be directed to the called tele-
phone (3) along with the calling telephone’s IP address.
The called telephone will then return a “RING” message
to the proxy server (4), which then forwards that RING
message to the originating telephone (5). When the called
phone answers, an OK message is returned (6, 7), which
includes the called telephone’s IP address. Finally, the
originating telephone (which now knows the IP address
of the destination phone from the INVITE/OK message
exchange) sends an “ACK” message to the IP address of
the called telephone (8). The originating telephone then
establishes an RTP communications link with the called
telephone (9).
For a redirect server, the location server will re-
turn the redirected (i.e., forwarding) e-mail-style SIP ad-
dress of the called telephone (e.g., [email protected], or
[email protected]). The redirect server will then forward the
INVITE request to the proxy or redirect server associated
with that redirected address, and then the steps enumer-
ated above will take place.
Each location server must track the IP address of each
of the telephones in its SIP domain. Thus, each SIP tele-
phone must register with its domain’s location server via
a registration server of its telephone service provider. An
individual telephone can be registered with any registra-
tion server with which the user has a service arrangement.
Registration binds each SIP telephone’s IP address to its
SIP address in its service provider’s domain.
Note that a SIP call control server’s primary purpose
is to handle the routing of initial supervisory and ad-
dress signaling information. Also note that, after the ini-
tial exchange of supervisory and address information, SIP
end devices establish and maintain the communicationschannel without involvement of the SIP call control server.
Like H.323 v.3, the SIP packets that carry the signaling
messages almost always follow a different path from the
path taken by the communications data. Finally, with SIP
most of the intelligence resides in the end devices, as com-
pared to being in the network, as is the case with conven-
tional telephone networks and with H.323.
Because SIP proxy and redirect servers typically do
not track a call’s status after the call is set up, SIP is of-
ten viewed as being more scalable than H.323. When a
call control server is configured to track call status, its re-
sources must bear the added burden of such monitoring.
SIP’s ASCII encoding is considered more extensible
and open than the binary encoded signaling of H.323 v.3.
SIP uses a very generic syntax for messages, which can
be customized to fit the needs of different applications
resident on end devices. For analysis of the various (and
somewhat controversial) comparisons of SIP and H.323,
see Dalgica and Fang (1999).Integrating VOIP Into Conventional
Circuit-Switched Telephony Networks
As previously noted, the value of a telephony network is
a direct function of how many telephones are directly or
indirectly connected to it; thus a VOIP network must be
able to exchange calls (i.e., internetwork) with the PSTN
such that a VOIP user can originate telephone calls to
and receive telephone calls from a telephone user who is
connected to the PSTN. In addition, for conventional tele-
phone providers to deploy VOIP technology inside their
networks, VOIP technology’s presence must be impercep-
tible to their existing base of telephone users.
Figure 10 illustrates how a VOIP call can make a con-
nection from a phone connected to an IP network to