(1) VoIP benefits
Some important benefits of VoIP implementation are as follows:
l Cost savings --- By moving voice traffic to IP-based networks, companies (or users) can
reduce the toll charges associated with transporting calls over PSTN. VoIP service providers
and companies can also conserve by investing in additional capacity only when it is needed.
This is made possible by the distributed nature of VoIP and by reduced operations costs as
companies combine voice and data traffic onto one network.
l Open standards and multi vendor interoperability --- By adopting open standards, both
companies and service providers can purchase equipment from multi vendors and eliminate
their dependency on proprietary solutions.
l Integrated voice and data networks --- Companies (or users) can build truly integrated
networks for voice and data. These integrated networks not only provide the quality and
reliability equivalent to PSTN, they also enable companies to quickly and flexibly take
advantage of new opportunities within the changing world of communications.
(2) Total VoIP and PSTN Traffic 1997 - 2001
The chart shows the global PSTN traffic
growth. The data of 2001 recently updated is
that the PSTN voice traffic grew just over 10%
– the slowest rate of growth in 20 years. On the
other hand, VoIP traffic has exceeded more than
this chart in 2001, about 10% of global market.
VoIP traffic here is all cross-border voice calls
carried on IP networks but terminated on public
switched telephone networks.
VoIP Migration for Private Network
Voice Network Migration
4.1 VoIP Migration for Private Network
At the early stage, VoIP was applied to the private business networks to get clear benefits
of operation cost reduction by integrating with voice (including FAX) and data traffics into
Traditional private networks have PBX (Private Branch Exchange) and connected to
PSTN (Public Switched Telephone Network) for analog telephones. When VoIP is
introduced, the interface to PBX is required to bridge the existing network and VoIP
network with Gateway.
When the usage of ordinary telephones becomes obsolete, PBX is eliminated and only
VoIP network will work.
VoIP Application for PSTN
•Signal Conversion H.323 H.323
•Packeting of Voice Gateway Gateway
ITSP IP Network ITSP
• Conversion between
Telephone Number & H.323
ITSP=IP Telephony Service Provider
4.2 VoIP Application for PSTN
The network configuration for VoIP service by ITSP is shown here. As the access
networks, the existing subscriber lines are used and the lines are connected through the
existing local exchanges to Internet by ITSP.
Features of the application of VoIP to PSTN are described as follows:
² Internet prosperity
Because of Internet prosperity, VoIP has been applied for public telephone use.
Although the existing analog telephone subscriber line and Local PSTN are used
initially, the main transmission is relied on Internet.
² Breakaway from POTS (Plain Old Telephone Service)
Because of competitive de facto standardized facilities and operations, Internet can
carry the voice traffic much cheaper than POTS, the traditional telephone network
and service are being broken away and even old players are entering in the VoIP
² Competition among Carriers
New comers having VoIP technologies have emerged on the telephone service
market. The operator called Greenfield Operator in general who have no existing
network, provides VoIP service by use of entirely IP-based backbone network as
shown below. Severe telephone competition decreases the telephone charge
drastically, so that it becomes clear to keep the infrastructure on healthy financial
background for most of network providers.
Customer Greenfield Greenfield Customer
IP Network Broadband
IP Access IP Access
IP Telephone Type
Soft Phone Type PC Net Phone
Soft Phone Call Channel
TA Type Multi Ports Type
Router Device Router Device
VoIP PSTN PSTN
4.3 VoIP Terminal
There are several types of VoIP terminal depending on usage of resources and
configurations. In the case of IP Telephone, entire VoIP functions are accommodated in the
set, of which block diagram is shown below. The voice coding methods are summarized in
the table below.
Compression Sample Typical end-to-end delay MOS
Rate (kbps) Size (ms) Score
(excluding channel delay)
G.711 PCM 64 0.125 <<1 4.1
G.726 ADPCM 32 0.125 60 3.85
G.728 LD-CELP 16 0.625 <<2 3.61
G.729 CS-ACELP 8 10 25 – 35 3.92
G.729a CS-ACELP 8 10 25 - 35 3.7
G.723.1 MP-MLQ 6.3 30 97 3.9
G.723.1 ACELP 5.3 30 67 3.65
ADPCM: Adaptive Differential Pulse Code Modulation, LD-CELP: Low Delay Code Excited Linear Predictive
CS-ACELP: Conjugates Structure Algebraic Code Excited Linear Predictive, MP-MLQ: Multi Pulse, Multi
Call Control and Signaling Signaling and Media
H.323 Audio / Video
H.245 Q.931 RAS SIP MGCP RTP RTCP RTSP
H.323 Version 1 and 2 supports H.245 over TCP, Q.931 over TCP and RAS over UDP.
H.323 Version 3 and 4 supports H.245 over UDP/TCP and Q.931 over UDP/TCP and RAS over UDP.
SIP supports TCP and UDP.
4.4 VoIP Protocols
For VoIP control protocols, there are two categories of protocols; one for call control (e.g.
H.323 and SIP) and the other is gateway control (e.g. MGCP/Megaco, H.248). The details
of those protocols are shown in the table below.
H.323 SIP MGCP/H.248/Megaco
Standards body ITU IETF H.248—ITU
Architecture Distributed (Peer-to-peer) Distributed (Peer-to-peer) Centralized (Master/Slave)
Current version H.323v4 RFC2543-bis07 MGCP 1.0, Megaco, H.248
Call agent/media•@ gateway
Call control Gatekeeper Proxy/Redirect Server controller
Endpoints Gateway, terminal User agent Media gateway
Transmission Control MGCP—UDP;•@
Signaling transport Protocol (TCP) or User TCP or UDP
Datagram Protocol (UDP) Megaco/H.248—both
Multimedia capable Yes Yes Yes
H.245 (signaling) or RFC RFC 2833 (media) or•@Signaling or RFC 2833 (media)
DTMF-relay transport 2833 (media) INFO (signaling)
Fax-relay transport T.38 T.38 T.38
Supplemental services Provided by endpoints or Provided all control
by endpoints or
call control •@
c Provided by call agent
SIP: Session Initiation Protocol
MGCP: Media Gateway Control Protocol, also known as Megaco and H.248, for handling the
signaling and session management during a multimedia conference. The protocol defines a means of
communication between a media gateway, which converts data from the format required for a circuit-
switched network to that required for a packet-switched network and the media gateway controller.
RTP: Real-Time Transport Protocol that specifies a way for programs to manage the real-time
transmission of multimedia data over either unicast or multicast network services.
RTCP: Real-Time Transport Control Protocol, RTSP: Real Time Streaming Protocol
H.323 & SIP Protocol Stack
OSI References H.323 Protocol Suite SIP Protocol Stack
6. Presentation RAS H.245 Q.931
5. Session RTP/RTCP SIP
4. Transport UDP TCP UDP TCP
3. Network IP IP
2. Data Link
4.5 H.323 & SIP Protocol Stack
Both SIP and H.323 define mechanisms for call routing, call signaling, capabilities
exchange, media control, and supplementary services. SIP is a new protocol that promises
scalability, flexibility and ease of implementation when building complex systems. H.323 is
an established protocol that has been widely used because of its manageability, reliability
and interoperability with PSTN.
SIP session setup for VoIP is illustrated as follows:
(SIP Server) (SIP Server)
Softphon SIP Phone
100 ( Trying) INVITE
100 ( Trying)
180 ( Ringing) Pick-up
180 ( Ringing) handset
180 ( Ringing) 200 ( OK)
200 ( OK)
200 ( OK)
Media Session (Telephone Conversation)
200 ( OK)
SIP session setup example with SIP trapezoid