VoIP

Voice Over Internet Protocol

 VoIPis
forVoiceoverInternetProtocol.

short

Voice over Internet Protocol is a category

of hardware and software that enables
people to use theInternetas the
transmission medium for telephone calls
by sending voice data in packets usingIP
rather than by traditional circuit
transmissions of thePSTN.

Gateways allow PCs to also

reach phones

Initially, PC to PC voice
calls over the Internet

Public Switched
Telephone Network

PSTN (NY)

Gatewa
y

Multimedia
PC
IP Network

Gatewa
y
PSTN (DC)

or
phones to
reach

Multimedia
PC

Circuit-Switched Telephony
Traditional PSTN Approach

SCP

Signaling Network

Signaling

Class 5
Switch

Typically analog
loop,
conversion to
digital at local
switch

Class 4
Switch

Circuit-based Trunks

Most service logic in local

switches, rest in SCPs

Class 5
Switch

64 kb/s
digital
voice
Data travels over a parallel (but separate)
network

Media
stream

PSTN vs. VoIP

Voice network use circuit
switching.
Dedicated path between calling
and called party.
Bandwidth reserved in advance.

Cost is based on distance and

time.

Data network use packet

switching.
No dedicated path between
sender and receiver.
It acquires and releases
bandwidth, as it needed.
Cost is not based on distance
and time.

How does it work?

1. Compression voice is compressed typically
with one of the following codecs, G7.11 64k,
G7.29AB 8k, G723.1 6.3k
2. Encapsulation the digitized voice is wrapped
in an IP packet
3. Routing the voice packet is routed thru the
network to its final destination

VoIP protocols are:

H.323 protocol suite
Media Gateway Control Protocol (MGCP
)
Session Initiation Protocol (SIP)
H.248 (Media Gateway Control
(Megaco))
Real-time Transport Protocol (RTP)
Real-time Transport Control Protocol
(RTCP)
Secure Real-time Transport Protocol
(SRTP)
Session Description Protocol (SDP)

Device Control
Protocolslike H.248
(more popularly known
as Megaco),
Media Gateway
Control Protocol
(MGCP), NCP, Real-time
Transport Protocol (RTP)
Access Service
Signalling
protocolslike Session
Initiation Protocol (SIP)
and H.323
Network Service
Signalling
Protocolslike SIP, SIPT, CMSS, BICC etc.

Call Agent SIP

Server
Service Broker
Application
Server
Media Server
Signaling
Gateway
Trunking
gateway
Access Gateway
and subscriber
gateway
Access

International Voice Market

Calls Terminated on PSTN

Source: Telegeography 2010

(2001 figures were

H.323 Architecture
ITU-T

H.323 Gatekeeper

H.323
Termin
al

H.323 Zone

PSTN

3 stages of signaling:
RAS to Gatekeeper
H.225 call signaling
H.245 media stream
control
(can be simplified for
VoIP)

H.323
Gatewa
y

H.323
Multipoint Control Unit

Telco-centric multimedia, multiparty conferencing (initially for LANs)

Gatekeeper for network control, heavy-weight protocols
Widely deployed in first wave of VoIP standardization

 RAS (Registration/Admission/Status): defined as H.225 (RAS) protocol in the standard. RAS

messages are originating by the gateway at the moment a terminal initiates a call. These
messages are transported using UDP (user datagram protocol) to the gatekeeper.

H.225: also known as H.225.0 call control signaling. This protocol specifies the use and
support of Q.931 signaling messages. This protocol is responsible for establishing and
releasing connections.

H.245: used after establishment of the connection, between the two gateways, to negotiate
parameters of the call such as codecs to be used, video or conference support, timer values,
etc. Moreover, IP logical channel ports are exchanged for the RTP sessions and eventually
the transmission of data and/or voice traffic.

RTP is the protocol used for transporting voice packets and it is managed by RTCP.

SIP (Session Initiation Protocol)

The SIP consists of the
following entities:
User Agent Client (UAC): Caller application
that initiates and sends SIP requests.
User Agent Server (UAS):Receives and
responds to SIP requests: accepts, redirects, or
refuses calls.
Proxy Server:Contacts one or more clients or
next-hop servers and passes the call requests
further. It contains UAC and UAS.
Redirect Server:Does not initiate SIP requests
or accept calls. Accepts SIP requests, maps the
address into new addresses and returns those
addresses to the client.

Methods of SIP are:

INVITE:User or service is
invited to participate in a
session.
ACK:Client has received a final
response to an INVITE request.
OPTIONS:Server being queried
about capabilities.
REGISTER:Client registers
address with a SIP server.
BYE:User Agent Client indicates
to server to release the call.

Advantages of VOIP (Voice over Internet Protocol)

Decrease the cost of telephone calls in long distance.

Multi-functionality of the communication link to the user.
Alternatives to Internet telephony are DSL technology and
ISDN technology.

Disadvantages of VOIP

Quality of service (Latency, Jitter and packet loss )

Number portability
Emergency calls
Fax support
Power requirements
Security
Caller ID