Group 3

Shobhan Garg – 205113003Rajesh Sethi – 205113013Richa Choudhary – 205113023Akash Hirke – 205113033Ayaz Qureshi – 205113043Kaushal Varshney – 205113053Rohit Gurjar – 205113063Jitendra Nagar – 205113073Arun Kumar Meena - 205113077Arpit Gupta - 205113083

VoIP Security

What is VoIP

Voice Over Internet Protocol (VoIP)

• A methodology for the delivery of Voice Communications over Internet Protocol Networks, such as the Internet

• Also called as IP Telephony, Internet Telephony, Broadband Telephony or Broadband Phone Service

Voice over Internet Protocol (VoIP)

contd…

• Similar to Traditional Digital Telephony

• Involve Signaling, Digitization of the Analog Voice Signals, and Encoding

• Traditional Digital Telephony sends the Digital Signals over a Circuit Switched Network

• In VoIP, the digital information is packetized, and transmission occurs as IP Packets over a Packet Switched Network

VoIP Architectures

• PC to PC

• Phone to Phone Via Internet

• PC to Phone

Session Initiation Protocol (SIP)

• Can be used for Two Party (Unicast) or Multiple Party (Multicast) Sessions

• Each resource of a SIP Network is identified by a Uniform Resource Identifier (URI)

• The URI is of the form

• sip:username:password@host:port

SIP Network Elements

• User Agent

• Proxy Server

• Registrar

• Redirect Server

• Session Border Controller

• Gateway

Quality of Service (QoS)

• Less Reliable as there is no mechanism to ensure that the Data Packets are not lost and are arriving in order

• A Best Effort Network

• Latency can be introduced that may exceed the permissible values

• Latency can be minimized by marking Voice Packets as being delay-sensitive

PSTN vs. INTERNET

• PSTN

• Voice network use circuitswitching.

• Dedicated path betweencalling and called party.

• Bandwidth reserved in advance.

• Cost is based on distanceand time.

INTERNET

• 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.

Overcoming the Challenges

Latency

Packet loss

Scalability

Jitter

Bandwidth

Reliability

Security

Interoperability

Latency

Latency is the time taken for a packet to arrive at its destination

Packet switching overhead

Congestion

Latency may result in voice synchronization problems

Packet Loss

Packet loss in unavoidable

It can be minimally tolerated in voice transmission

It should not, in the first place, distort the audio

ScalabilityAbility to add more telephony equipment as the company grows

Network bandwidth and other issues may have an effect on scalability

JitterJitter is the delay experienced in receiving a packet when a packet is expected to arrive at the end point at a certain time

BandwidthWhen bandwidth is shared between voice and computer data, certain bandwidth may have to be allocated for voice communication on a network

Reliability

Because the computer network is used, the reliability of the network will have an impact on the telephony service

In the analog telephone industry, reliability of 99

.999 percent uptime is required

The above is known as five nines

VoIP networks can achieve over 98 percent reliability ?

Security

As VoIP uses the Internet, for example, it is vulnerable to the same type as security risks

Hacking

Denial of service

Interoperability

• IP telephony equipment manufactu

red by different vendors must be ab

le to talk to each other

– Standardized protocols are needed

How VoIP Works:

With VoIP, analog voice calls are converted into packets of data. The packets travel like any other type of data, such as e-mail, over the public Internet and any private Internet Protocol (IP) network.

Using a VoIP service, you can call landline or cell phones. You can also call computer-to-computer, with both parties speaking into a computer microphone and listening through computer speakers or headsets.

• Converting the voice signal– ADC (analog to digital)

– DAC (digital to analog)

Voice (source) - - ADC - - - Internet - - - DAC - - Voice (dest)

• Transmission of voice traffic in packets

• The 1-2-3s of VoIP

• 1. Compression – voice is compressed typic

ally with one of the following codecs, G7.11 64

k, G7.29AB 8k, G723.1 6.3k

• 2. Encapsulation – the digitized voice is wra

pped in an IP packet

• 3. Routing – the voice packet is routed thru t

he network to its final destination

Components

• VoIP Protocols

• VoIP Gateway

• VoIP Codecs

1. VOIP Gateway

Voice over Internet Protocol (VoIP) gateway is a

device that converts analog telephony signals to

digital.

A network device that converts voice and Fax

calls, in real time, between the public switched telephone network (PSTN) and an IP network.

Type of Gateway

• Analog

- FXS gateway

- FXO gateway

• Digital

Features

• Call routing, packetization and control signaling management.

• Voice and fax compression/decompression.

• External controller interfaces.

VOIP Codecs

• A codec, which stands for coder-decoder, converts an audio signal into compressed digital form for transmission and then back into an uncompressed audio signal for replay. It's the essence of VoIP. It converts each tiny sample into digitized data and compresses it for transmission.

• Common VoIP Codec:

• G.711 - Delivers precise speech transmission. G.711 uses a logarithmic compression. It squeezes each 16-bit sample to 8 bits, thus it achieves a compression ratio of 1:2. The resulting bitrate is 64 kbit/s for one direction, so a call consumes 128 kbit/s.

• This codec can be used freely in VoIP applications as there are no licensing fees. It works best in local area networks where we have a lot of bandwidth available.

• G.722 - Adapts to varying compressions and bandwidth is conserved with network congestion.

• G.729 - G.729 is a codec that has low bandwidth requirements but provides good audio quality (MOS = 4.0). The codec encodes audio in frames, each frame is 10 milliseconds long. Given the sampling frequency of 8 kHz, the 10 ms frame contains 80 audio samples. G.729 is a licensed codec.

•

• G.723.1 - High compression with high quality audio. Lot of processor power. It is a licensed codec.

•

• G.726 – An improved version of G.721 and G.723 (different from G.723.1)

PBX

Yesterday’s Networks

Circuit Switched Networks (Voice)

CO

PBX

COCO

Packet Switched Networks (Data)Router

Router

Router

Router

Router

• Separated networks

• Separated applications/services

PBX

IP Phone

Converged Network

PSTN

CO

Gateway

Router

Router

Router

Router

• Converged network• Separated or integrated applications

PBX

IP Phone

IP Network

Multimedia PC

Multimedia PC

Initially, PC to PC v

oice calls over the I

nternet

VoIP Architecture?

PSTN

(DC)

Gateway

PSTN

(NY)

Gateway

Public Switched Tele

phone Network

Gateways allow PCs

to also reach phone

s

…or phones to reac

h phones

VoIP Network Model

SIP

RTP, RTCP, RTSP

Transport Layer (UDP, TCP)

Network Layer (IP, IP Multicast)

Data Link Layer

Physical Layer

• The Session Initiation Protocol (SIP) is a communications protocol for signaling and controlling multimedia communication sessions. The most common applications of SIP are in Internet telephony for voice and video calls, as well as instant messaging all over Internet Protocol (IP) networks.

• The Real-time Transport Protocol (RTP) is a network protocol for delivering audio and video over IP networks. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and web-based push-to-talk features.

• The Real Time Streaming Protocol (RTSP) is a network control protocol designed for use in entertainment and communications systems to control streaming media servers. The protocol is used for establishing and controlling media sessions between end points.

• Higher overhead of TCP does not make sense for telephone call. Because audio must stream! No wait for missing packets. Play missing part as silence.

• UDP Offerrs best-effort delivery. to handle duplication, delay, out-of-order delivery, each RTP message contains

IP Protocol Layering

Physical Transport (e.g, Cable Modem)

IP (Internet Protocol)

TCP UDP

Applications (e.g., email, web pages)

Email Data (1000 bytes)TCP Header

(20 bytes)IP Header (20 bytes)

A Typical IP Datagram

VoIP SIP

Advantages of VoIP

• Cheaper than the Traditional Telephone System

• Calls can be made from anywhere to anywhere using the single account

• Images, Videos and Text can also be sent along with the Voice

• The Network need not be of a particular Topology

Disadvantages of VoIP

• Packet Loss and Jitter can be there

• Calls cannot be made if the Internet Connection is down

• Calling Emergency Numbers using VoIP will not provide your location to the Emergency Response Services

VoIP Security

SECURITY BASICS

• AUTHENTICATION

• AUTHORIZATION

• AVAILABILITY (Use of different segment for VoIPs)

• ENCRYPTION

ATTACK VECTORS

A local subnet, such as an internal network, where VoIP is used By

unplugging and/or sharing a VoIP hard phone’s Ethernet connection

(usually sitting on one’s desk), an attacker can connect to the voice

network.

A local network that is using wireless technology with untrusted users,

such as a coffee shop, hotel room, or conference center An attacker

can simply connect to the wireless network, reroute traffic, and capture VoIP calls

A public or non-trusted network, such as the Internet, where VoIP

communication is used An attacker who has access to a public network

can simply sniff the communication and capture telephone calls.

• Compromising the VOIPs phone’s configuration file

• Uploading a malicious configuration file

UNCONVENTIONAL VOIP SECURITY

THREATS

• VoIP Phishing

• Caller ID Spoofing

• Anonymous Eavesdropping and Call Redirection

• Spam Over Internet Telephony