Intro To VoIP

Introduction to VoIP
VoIP Defined
 Routing of voice traffic over internet protocols

 Based on packet switching vs. traditional circuit
switching (PSTN)
 VoIP is NOT:
 IP Telephony: refers to any "telephone" type service
carried over IP, which could include faxing, or text
messaging (VoIP is voice over IP only).
 ‘True’ data convergence: refers to sending both voice
and data (such as LAN traffic) over any data network
(typically an IP network, frame relay network or ATM
network)
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Circuit Switching
 Based on T1 or E1 circuits (SS7 and ISDN)

 Maintains connection path
 Bandwidth reserved through circuit life
 Dedicated channel
PBX PBX
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Packet Switching
 Payload broken up & sent individually as packets

 Each Packet individually addressed
 Connection path different for each packet
 Packets only sent when there is payload
Call Call
Server Server
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Why is VoIP an attractive solution?
 Real-time application of voice / video
 Cost
 Low-cost infrastructure
 Scalability / routing flexibility
 Control of policy routing based on non-deterministic
parameters (e.g.: cost, least number of ‘hops’)
 Simultaneous multimedia transmission
 Efficient use of bandwidth
 Discreet states (on / off) vs. dedicated circuits (always
on)
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Circuit Switch vs Packet Switched
Circuit Switched Packet Switched
Bandwidth Fixed – 56K or 64Kbps Variable – up to Gbps
Addressing Required (phone number) Required per packet
Latency Minimal Minimal to high
Payload during Yes No

silence
Intelligence In Network In CPE
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VoIP Elements: Codecs
 Device / program that converts sampled analog

signals into digital signals for transmission
(compression), then converts the digital signals
back to analog for playing (decompression)
 Operate on a set ‘standardized’ algorithms designed to
yield specific bandwidth
Note: Customer quality calls normally

need a MOS in the range 3.5 to 4.2
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Bandwidth Consumption: Codecs
Compression Bit Rate (kbps) MOS Score Compression Delay (ms)
G.711 PCM 64 4.1 0.75
G.726 ADPCM 32 3.85 1
G.728 LD-CELP 16 3.61 3 to 5
G.729 CS-ACELP 8 3.92 10
G.729 x 2 Encodings 8 3.27 10
G.729 x 3 Encodings 8 2.68 10
G.729a CS-ACELP 8 3.7 10
G.723.1 MP-MLQ 6.3 3.9 30

G.723.1 ACELP 5.3 3.65 30
T38 Fax Transmission
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VoIP Elements: Quality Factors
 Latency
 The amount of time from the source sending a packet to the source
receiving a response
 Jitter (Delay Variation)
 The difference in packet arrival from origination to termination
 Packet Loss
 Lost packets of information from origination to termination
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VoIP Elements: Signaling
Main signaling protocols:

 H323
 SIP
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VoIP Elements: Media
 Media and Signaling paths can be separate
 Utilizes established IP technology
 Resilient to packet loss
SMTP, TELNET, ICMP
 Delay intolerant Application Layer
HTTP, FTTP, RTP etc.
 Uses RTP
Transport Layer TCP and UDP
IP addressing, routing
Internet Layer
and masking
Ethernet, Token Ring,

Network Access Layer
FDDI etc
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VoIP Elements – IP Phones
 IP Phones really need no introduction or little

explanation. They are simple devices we are
familiar with but have a VoIP signaling capability
that enables them to interact directly with other
VoIP devices
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VoIP Elements – Analog Terminal Adapter
(ATA)
 Adaptors that interface legacy analog telephones, fax machines, analog conference
telephones and other analog devices to IP based telephony networks thereby
allowing companies to protect their investments in analog phones, fax machines,
and speakerphones, and migrate to IP at their own pace. These products address
the needs of enterprises, small-office environments, and the emerging VoIP
managed voice services and local services market by helping companies cost-
effectively turn their analog devices into IP devices.
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VoIP Elements – Session Border Controller
 New breed of networking technology that
provides layer 5 routing and control to manage
real-time traffic flows in IP networks.
 Technology addresses issues of:
 Session routing
 Signaling interoperability
 Network security
 Call admission control
 Service quality
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Market Requiremnts
 As VoIP is on IP, it is therefore inherently non-
deterministic.
 VoIP signaling has to be reliably handled
 The two protocols for signaling the VoIP
community uses mainly are H.323 and SIP for
end-points.
 Session Controllers contend with several issues
that impede the progress of VoIP networks for
carriers
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VoIP Elements – Session Border Controller
Used at the edge of the network (border) to route

signaling and/or media between endpoints
Integrates signaling / media streams seamlessly
Can act as:

Peering agent
Border Element from network core
Interoperates with various VoIP protocols

H323 / SIP
RTP
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Session Border Controller – Benefits
For Private IPs:

Enable calls securely to and from a protected enterprise
network
Used with firewall (network security)
For Public IPs:

NAT
Interoperability (H323 / SIP)
Without an SBC, the media traffic travels directly between the

VoIP phones, which consumes bandwidth unnecessarily
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Call Flow Example 1: Basic Network
IP Header: IP Header:
Source IP: 100.1.1.1 Source IP 200.2.2.2
Dest. IP: 200.2.2.2 Dest. IP 100.1.1.1
Signaling From IP 100.1.1.1
Signaling
To IP From IP 200.2.2.2 To IP 100.1.1.1
A 200.2.2.2 B
100.1.1.1 200.2.2.2
Media can be established to each other
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Call Flow Example 2: Basic Network with Firewall
IP Header: IP Header:
Source IP 100.1.1.1 Source IP 200.2.2.2
Dest. IP 200.2.2.2 Dest. IP 10.10.1.1
A Signaling From IP 10.10.1.1 To IP 200.2.2.2
Signaling From IP 200.2.2.2 To IP 10.10.1.1
10.10.1.1
B
X 200.2.2.2
FW
100.1.1.1
Media cannot be established to each other
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Call Flow Example 3: Proxy Network
IP Header:
Source IP 10.10.1.1
Dest. IP 200.2.2.2
10.10.1.1 IP Header:
Source IP 100.1.1.1
Dest. IP 200.2.2.2 X
Signaling From IP 10.10.1.1 To IP
IP Header:
Source IP 200.2.2.2
Dest. IP 100.1.1.1
B
200.2.2.2
200.2.2.2 Signaling From IP 200.2.2.2 To IP 100.1.1.1
Signaling only
Proxy IP Header:
100.1.1.1 Source IP 200.2.2.2
Dest. IP 100.10.1.1
Media cannot be established to each other
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Call Flow Example 4: SBC Network
IP Header:
Source IP 10.10.1.1
Dest. IP 200.2.2.2
10.10.1.1 IP Header:
Source IP 100.1.1.1 IP Header: B
Dest. IP 200.2.2.2 Source IP 200.2.2.2
Dest. IP 100.1.1.1 200.2.2.2
SBC IP Header:
100.1.1.1 Source IP 200.2.2.2
Dest. IP 100.10.1.1
Media can be established to each other
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Session Controller
Follow the packets - Packet
3 (coming back)
X
Follow the packets - Packet Follow the packets SA: - Packet
190.18.18.100
1 (before the firewall) 2 (after the firewall)
DA: 10.1.1.3
SA: 10.1.1.3 SA: 200.200.200.200
SIP Info Based on 10.1.1.3 –
DA: 190.18.18.100 DA: 190.18.18.100190.18.18.100 SIP UA
Userid: User2
SIP Info Based on 10.1.1.3 – SIP Info Based on 10.1.1.3 – IP: 190.18.18.100
190.18.18.100 190.18.18.100
SIP UA “Internet”
Userid: User1
IP: 10.1.1.3
Firewall
Private side IP 10.1.1.1
Public side IP 200.200.200.200
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Challenges of VoIP
 Call Quality (QoS) –

 Is network transmission / routing reliable between
endpoints?
 Interoperability –
 How does VoIP work between various telecom
vendors? (ex: Cisco, Clarent, Broadsoft)
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Challenges of VoIP
 Interworking Function (IWF) –

 Are existing IP / ‘VoIP-specific’ protocols compatible?
 Network Security –
 How does VoIP protect customers and businesses
within public and private IP space?
 Session Border Controllers: do they ‘help’ or ‘hurt’ in
terms of efficient call routing / security?
 KUMAR
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Introduction to VoIP

 Routing of voice traffic over internet protocols

 Based on T1 or E1 circuits (SS7 and ISDN)

 Payload broken up & sent individually as packets

Bandwidth Fixed – 56K or 64Kbps Variable – up to Gbps

Addressing Required (phone number) Required per packet

Latency Minimal Minimal to high

Payload during Yes No

Intelligence In Network In CPE

 Device / program that converts sampled analog

Note: Customer quality calls normally

G.711 PCM 64 4.1 0.75

G.726 ADPCM 32 3.85 1

G.728 LD-CELP 16 3.61 3 to 5

G.729 CS-ACELP 8 3.92 10

G.729 x 2 Encodings 8 3.27 10

G.729 x 3 Encodings 8 2.68 10

G.729a CS-ACELP 8 3.7 10

G.723.1 MP-MLQ 6.3 3.9 30

Main signaling protocols:

Ethernet, Token Ring,

 IP Phones really need no introduction or little

Used at the edge of the network (border) to route

Can act as:

Interoperates with various VoIP protocols

For Private IPs:

For Public IPs:

Without an SBC, the media traffic travels directly between the

Media can be established to each other

Media cannot be established to each other

Media cannot be established to each other

Media can be established to each other

 Call Quality (QoS) –

 Interworking Function (IWF) –

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