The Internet Encyclopedia (Volume 3)

P1: IML

Morris WL040/Bidgoli-Vol III-Ch-53 September 15, 2003 12:23 Char Count= 0

VOICE OVERINTERNETPROTOCOL 655

Calling

Telephone

Called

IP Network Telephone

IP Router

Call

Control

Gateway

1. Dial
   2. Look Up
   Dialed #


2. Generate


3. Ring


4. Off-
   Hook


5. Open
   Channel


6. Open
   Channel


7. RTP
   (Voice Stream)

IP Packets

Voice

Mail

5. Called Phone
   Does Not Answer

Audible Ring

Figure 8: Called telephone unavailable with reroute to voice mail with H.323-type signaling.

the far end. RTCP provides a separate signaling chan-

nel between the end devices (e.g., telephones) to allow

exchange of information about packet loss, packet jitter,

and packet delays, as well as additional information, such

as the source’s name, e-mail, phone, and identification.

(Real-Time Transport Protocol [RTP], 2001, August), and

(Streaming Video Over the Internet, 2002).

Additional requirements for a commercial VOIP sys-

tem include being able to produce information for billing

and/or accounting for calls. Similarly, today’s users de-

mand that it provide other features, such as Caller ID and

voicemail. These capabilities may reside in the call con-

troller, IP telephones, and/or other devices in the IP net-

work. Figure 8 shows how voicemail can be provided with

H.232.

H.323 vs. SIP

In addition to H.323, SIP is the major competing standard

for VOIP signaling. They have both evolved to offer very

similar feature capabilities.

H.323 takes a more telecommunications-oriented ap-

proach than SIP. SIP takes an Internet-oriented approach.

H.323 is the older of the two (Doron, 2001; Paketizer,

2002). It was developed under the International Telecom-

munications Union (ITU), a telecommunications stan-

dards group, and has gone through various revisions

(ITU-T, Recommendation H.323, 2000). The latest version

of H.323 standard (H.323 v3) is very robust in that it cov-

ers many possible implementations. However, H.323 is

considered more difficult to implement than SIP due to

its use of binary encoded signaling commands.

H.323 v.3 can be implemented with or without a call

control server. Thus, an H.323 v.3 end device (e.g., a tele-

phone) can be designed to either set up a call through a

call control server, or set it up directly with another end

device without using an intervening call control server.

An H.323 v.3 call control server can be set up to relay the

communications stream during the call, or the end devices

can directly establish the communications RTP streaming

channel between themselves. The call control server can

be either stateless (i.e., not track a transaction or a session

state) or stateful (i.e., track a transaction and/or call ses-

sion state). The significance of this is that in the stateful

configuration, H.323 v.3 is not as scalable. Finally, H.323

v.3 employs signaling protocols that can easily be mapped

through a gateway for routing calls between the VOIP net-

work and the public switched network.

SIP is a Web-based architecture that was developed

under the Internet Engineering Task Force (IETF). Like

URLs and Web e-mail, SIP’s messages are in ASCII text

format that follow the HTTP programming model, i.e.,

using a grammar similar to that used to create basic Web

pages—resulting in slightly lower efficiency in transmit-

ting signaling information, as compared to the more ef-

ficiently encoded binary H.323 signaling messages. Also,

SIP is very extensible, leading many vendors to implement

variations that may be somewhat incompatible (IETF, SIP,

2003; IETF, SIP RFC2543bis, 2002).

An address used for routing a SIP messages is of the

form [email protected]. As shown in Figure 9, when

a phone wishes to originate a call, it transmits an ASCII

SIP “INVITE” message (1) addressed to the SIP address of

the called phone (e.g., [email protected]., where “xyz.com”

is the domain name of the SIP proxy server for the called

phone). Using conventional domain name server (DNS)

lookup, the internet routes this e-mail-type message to

the SIP call control server, which may act as either a

proxy server or a redirect server for the domain of the

dialed called telephone address (here, xyz.com). In order

to determine where the telephone is located, the proxy

server or redirect server will query a location server (2),

which will return the routing directions. What happens

next depends upon whether the call controller is act-

ing as a proxy server or a redirect server for the called

telephone.