Overview of Voice Over Internet Protocol Purva R. Rajkotia ...Presentation Transcript
Overview of Voice Over Internet Protocol
Purva R. Rajkotia,
November 4 ,2004
Overview of Voice Over Internet Protocol
• Presentation Outline
– History of VoIP
– What is VoIP?
– Components of VoIP
– VoIP Protocols
– Basic VoIP Call
– VoIP Standards
– Regulatory Environment
– Benefits of VoIP
– Challenges for VoIP
– VoIP on CDMA2000
– Business Applications Benefits
– Technology Adoption
Some Interesting IP facts
•50% of all enterprise telephone stations sold in US were IP based
•More than 30% of US and UK population will subscribe to VoIP
service in the next 3 years
-Mercer Management Group
•Worldwide revenue from the corporate Internet Protocol (IP) telephony
business will grow from $ 1 billion by the end of this year to $ 5.5 billion
History of VoIP
• VoIP came into existence as a result of work done by few
hobbyists in Israel in the year 1995,when only
PC-to-PC communication was in vogue.
• Later on during 1995, Vocaltec Inc. released Internet Phone
• The software used to compress the voice signal, convert it
into voice packet and finally ship it over the internet. The
sound quality was not even close to that of the standard
equipment in use at that point of time. This attempt was the
first IP phone that came into existence.
• VoIP has made great progress since then.
So, what is VoIP?
• Voice Over Internet Protocol (VoIP) is the assembly of voice
into IP data. This data can be transmitted over an IP
network to an addressable (IP address) destination.
• Voice calls look like data calls.
• VoIP calls are packet switched and analog calls are circuit
• Packet switched is data that can be routed through different
routes on a network to reach a destination.
• Circuit Switched is a connection where a physical path is
dedicated between two end points.
So, what is VoIP?
Analog CODEC Digital
Components of VoIP
Components of VoIP
• The gateway converts the signals from the traditional
telephony interfaces (POTS, T1/E1,ISDN) to VoIP.
• An IP phone is a terminal that has native VoIP support and
can connect directly to an IP network.
• The server provides management and administrative
functions to support the routing of calls across they
• In a system based on H.323, the server is known as a
gatekeeper. In SIP/SDP, the server is a SIP server.
In a system based on MEGACO or MGCP, the server is call
• The IP network can be a private network, an Intranet or the
VoIP Protocol Stack
ADDRESSING AUDIO CODEC DTMF ADDRESSING
RAS DNS RTP/RT H.245 Q.931 DNS
(H.225 CP (H.225)
UNRELIABLE RELIABLE TRANSPORT
TRANSPORT (UDP) (TCP)
• Multicast IP – The objective here is to send one packet and have it
received at many destinations. In order to propagate these
datagrams to multiple destinations, the routers within the network
infrastructure operates with modified routing protocols. These
multicast routing protocols construct logical spanning tree, which
describes how the multicast traffic flows to the end stations.
• Real Time Transport protocol provides end-to-end delivery service
for data that requires real time support, such as interactive
audio and video. The services provided by RTP includes payload
type identification, sequence numbering, time stamping and
• Real Time control Protocol monitors the quality of service and
conveys information about the participants in an ongoing session.
It is based on the periodic transmission of control packets to all
participants in the session, using the same distribution
mechanism as the data packets.
• Resource Reservation Protocol ( RSVP) defines the QoS and the
mechanisms to provide the QoS. RSVP is a control protocol,
therefore it works in collaboration with-not instead of- traditional
• Session Description Protocol describes multimedia session for
the purposes of session announcement, session invitation, and
other forms of session related initiation. It conveys information
about media streams in multimedia sessions to allow the recipients
of a session description to participate in that session. Session
Description Protocol information may be transported using SAP,
SIP, RTSP, MIME or HTTP.
• SIP is a control or signaling protocol used for creating, modifying,
and terminating sessions between participants. These sessions
may include multimedia conferences, telephone calls, distance
learning, or other types of multimedia distribution. SIP is a part of
the internet multimedia protocol architecture which includes other
protocols like RTP, SAP and SDP.
Basic VoIP Call Flow
Basic VoIP Call
1.The caller picks up the handset. This signals an off-hook condition
to the VoIP signaling.
2. The session application issues a dial tone and waits for the caller to
dial a telephone number.
3. The caller dials the telephone number. The session application
stores the dialed digits.
4. After enough digits are accumulated to match a configured
destination pattern, the telephone number is mapped to an IP host via
the dial plan mapper. The IP host has a direct connection to either the
destination telephone number or a private branch exchange (PBX) that
is responsible for completing the call to the configured destination
5. The session application runs the session protocol to establish
transmission and reception channels for each direction over the IP net
Basic VoIP Call
If Resource Reservation Protocol (RSVP) has been configured,
RSVP reservations are put into effect to achieve the desired quality
of service (QoS) over the IP network.
6. The coder-decoder compression schemes (codecs) are enabled for
both ends of the connection using Real-Time Transport Protocol/
User Datagram Protocol/Internet Protocol (RTP/UDP/IP) as the
7. Any call-progress indications are cut through the voice path as
soon as an end-to-end audio channel is established. Signaling
carried over the IP network is encapsulated in Real-Time Transport
Control Protocol (RTCP) using the RTCP application- defined (APP)
8. When either person hangs up the phone, RSVP reservations are
torn down (if RSVP is used) and the session ends. Each end
becomes idle, waiting for the next off-hook condition to trigger
another call setup.
– ITU-T Standards and Recommendation
• H.323(Packet-based multimedia communication Systems)
• H.225.0 (Call Signaling protocols and media stream
packetization for packet based multimedia)
• H.225.0 ( Annex G Gatekeeper to gatekeeper –
• H.245 ( Control protocol for multimedia communications)
• H.235 ( Security and encryption for H- series multimedia
• H.450.x Supplementary services for multimedia
VoIP Standards (contd)
• Signaling ( contd)
• H.323 Annex D Real time fax using T.38
• H.323 Annex E Call connection over UDP
• H.323 Annex F Single Use device
• T.120 series Data protocols for multimedia conferencing
– IETF RFCs and Drafts
• RFC 2543 –SIP ( Session Initiation Protocol)
• RFC 2327 – SDP ( Session Description Protocol)
• Internet Draft MGCP ( Media Gateway Control Protocol)
• Internet Draft MEGACO protocol
• RFC 1889 – RTP, RTCP
• CALEA and Law Enforcement Issue
– FBI wants the FCC to bring internet calling under provisions of
the 1994 Communications of Law Enforcement Act ( CALEA),
which requires phone carriers to provide them with the direct
access to phone lines. But, in case of VoIP, the packet
information travels in digital packets which is relatively easy
to encrypt or to use “secure tunnels” making them
inaccessible to law enforcement.
Benefits of VoIP
• VoIP has extremely efficient use of bandwidth as well as
• Hardware and protocols for VoIP are largely off the shelf,
interchangeable and developed rapidly
• VoIP can also take advantage of the same functionality that
is driving the internet, allowing providers to take advantage
of the equipment at a higher level of productivity and cost
• IP is the driving force for NGN( Next Generation Networks)
with Convergence being an important requirement. VoIP is
probably the simplest and easiest path to convergence.
Challenges for VoIP
• The current IP telephony services in the current state do not offer
carrier grade standards.
• Worrisome is the fact that the callers’ addresses don’t show up
on emergency operators’ screens when they call 911.
• For VoIP on cable, the HSD (High Speed Data) connection is
required, risking the caller to network and power outages.
• While the future is very promising, VoIP still has numerous
technology challenges ahead, including integration, back office
applications, OSS support, powering soft switches, return path
and Packet cable.
VoIP on cdma2000
• Service Option 60/61 to support the VoIP connection
• Allow Service Option 33 to set up the VoIP call
• Allow smaller packet sizes on the PHY layer to allow for fast
er delivery of the VoIP packets
• Faster call set-up
• Header compression (1x: LLA-ROHC, Header
Removal, HRPD- ROHC)
• QoS support ( QoS parameter negotiation and QoS
• Segment based framing ( for HRPD).
Business Applications Benefits
• VoIP yields two types of benefits:
- Soft Benefits
• Replacing a PBX with VoIP server may save a
company specific amount of money every year.
• Soft benefits don’t necessarily save money, or if
they, do , they don’t always save an easily calculated
amount of money. But they do affect the overall
bottom line in the future, like unified messaging etc.
– Hard Benefits
• Clearly defined cost savings.
• It is estimated that operator can reap cost savings by
leveraging the investments made for the made for
the same network to support HSD. If network
powering along with the NIUs (Network Interface
Units) and related functions and equipment are
excluded then its possible to see considerable
cost savings. 21/48
Technology Adoption Process
Early Early Late
Innovators Adopters Majority Majority
Introduction Growth Maturity Decline