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VoIP FundamentalsTech 160

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Agenda Tech 160

Voice Communication Voice over PSTN Voice over IP Quality of Service VoIP Security References

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Voice Communication

The Traditional Way – Voice over PSTN Migration from the entirely analogue PSTN to the digital

technology starting in 1975 yielded many advantages such as: Greater bandwidth Reduced error rates Enhanced management and control

For converting the analogue signal into a digital format and

reconvert it back into an analogue one, a matching pair of

codecs (coder/decoder) is used (PCM = Pulse Code Modulation,

standardized by the ITU-T as G.711).

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Voice over PSTN

Needed bandwidth in kbit/ s using the Codecs: Packet Size in msec.

G.711u G.711a

G.726-16 G.726-24 G.726-32 G.726-40 G.729a G723-5.3 G723-6.3 iLBC-13.33

10 105 58 66 73 81 50 - - -

20 84 37 45 52 60 29 - - -

30 77 30 38 45 53 22 19 20 27

40 73 26 34 42 50 18 - - -

50 71 24 32 40 48 16 - - -

60 70 23 30 38 46 15 12 13 60

Required Bandwidth for Standard Packets:

Required Bandwidth for Encrypted Packets (in the case that a VPN is used):Needed bandwidth in kbit/ s using the Codecs: Packet

Size in msec.

G.711u G.711a

G.726-16 G.726-24 G.726-32 G.726-40 G.729a G723-5.3 G723-6.3 iLBC-13.33

10 148 98 105 117 123 92 - - -

20 105 59 65 74 80 49 - - -

30 91 43 52 60 66 35 33 34 41

40 84 37 45 53 61 29 - - -

50 80 33 41 48 56 25 - - -

60 77 30 37 46 53 22 19 20 27

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Voice over PSTN

PSTN Advantages Reliable Good voice quality and minimal delays Worldwide phone coverage

PSTN Drawbacks Inefficient use of the available bandwidth Costly long distance service Worldwide phone coverage Complex provisioning of the available services

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VoIP

Definition:Voice over IP (VoIP), also called Internet telephony or IP telephony, is The transmission of voice telephony services over IP, the Internet Protocol. It

is a general term that refers to any means of converting voice calls into voice data packets that are transmitted over an IP network, either public or private. 

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VoIP

TimingBecause IP does not, by default, provide any QoS mechanism, parameters or limits had to be placed on latency, jitter and loss in order to achieve acceptable voice quality. 

Compression

VoIP often uses DSP that compress and process the signal, thereby requiring less than the 64 Kbps required by PCM. The result is encapsulated inside an IP packet, along with a UDP header for purposes of multiplexing, header error control, and identification of the application by port number. RTP is run for end-to-end delivery services such as payload type identification, packet sequence numbering, time stamping, and delivery monitoring. On the receiving end, these Internet Protocol processes are then reversed and the voice is extracted and played to the listener.

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VoIP

Packet Loss, Latency and JitterA major complication of VoIP is that these packets are not delivered at the same pace they entered the network. Additionally, some packets may be lost in transit, due to the IP network being a highly shared packet network characterized by unpredictable levels of congestion. There is variability in the latency, jitter and loss of the packets.

The use of intelligent algorithms is applied to resolve these packet issues to minimize their impact on the quality of voice. These algorithms are designed to fill in the voids by stretching the voice frames received earlier and blending them with those received later.

Echo CancellationVoIP also makes use of various techniques for echo cancellation, as echo becomes perceptible when delays exceed 15ms-20ms.

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VoIP

VoIP Advantages:

Use of the same network for data and voice resulting in: Reduced cabling and infrastructure costs Simplified network administration and control Voice and data staff merge Productivity and service delivery improvements

Highly cost-effective and possibly free long distance telephony

compared to PSTN (depending on whether a public or a private

PSTN/VoIP gateway is used)

Means to consolidate and unify phone services and features from

different locations

Future feature enhancement possibilities

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VoIP

VoIP Drawbacks: Some loss of voice data under heavy data traffic due to traffic

volume unpredictability over the Internet, resulting in packet loss, delays or jitter. For the majority of the connections this will not be experienced.

Some countries may regulate the use of VoIP to protect local PSTN monopolies, which could lead to the introduction of VoIP call fees over the public Internet. Currently there are no specific rates established by any country.

Limited gateway coverage to bridge the PSTN with the data network in order to reach any telephone around the world. This limitation is non-existent if the VoIP device has PSTN connectivity in addition to VoIP capability.

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Quality of Service

Quality of Service (QoS)In packet-switched networks, quality of service is a mechanism used to prioritize

network traffic to ensure that network devices handle high-priority traffic first. In

VoIP, QoS mechanisms guarantee that the values for packet latency, jitter, loss of

packets and other parameters affecting the voice quality are within an

acceptable range.

Service Level Agreement (SLA)Service-level agreement is an agreement between a customer and a service

provider that guarantees basic performance benchmarks, usually in exchange

for a commitment to spend a fixed amount with the provider over a time period.

Some metrics that SLAs may specify include: the percentage of the time services

will be available, the number of users that can be served simultaneously, etc.

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Quality of Service

MoS (Mean Opinion Score)VoIP service SLAs tend to be judged on their Mean Opinion Scores (MOS). They can include such factors as call completion rates and the length of time required for a user to hear a dial tone or to connect to the dialed party. Various measurement techniques are used in association with SLAs, including active network tests made at regular time intervals as well as passive measurements that are based on actual calls placed across network.

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VoIP Signaling Protocols

As packet telephony networks grew in the late 90s and interconnection dependencies emerged, it became clear that the industry needed standard VoIP protocols.

Several groups developed independent standards, each with its own unique characteristics.

There are four main VoIP call-control protocols (also known as signaling protocols):

Session Initiation Protocol (SIP) is an RFC standard (RFC 3261) from the Internet Engineering Task Force(IETF)

H.323 (an ITU recommendation defining "packet-based multimedia communications systems")

Media Gateway Control Protocol (MGCP) H.248/Megaco (an ITU recommendation defining a "Gateway Control

Protocol")

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SIP

The Session Initiation Protocol (SIP) Is a signalling protocol used for establishing sessions in an IP network.

A simple session could be a two-way telephone call or it could be a video and messaging communication between two devices.

Is a request-response protocol that closely resembles two other Internet protocols, HTTP and SMTP (the protocols that power the world wide web and email); consequently, SIP sits comfortably alongside Internet applications.

There are two basic components within SIP: The SIP user agent = the end system component for the call The SIP network server = the network device that handles the signaling associated

with multiple calls. SIP is implemented by installing servers on the data network, either private or on the public internet, that run SIP software that allows users to register the location (in the

form of an IP address) of their phone or device. The SIP address for a device is similar to an email address and takes the form user@location.com (i.e. 30112345@sip.epygi.com). Once registered on the SIP server users can call other registered users.

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VoIP Security

VoIP security is a major issue in VoIP communication and should never be underestimated. Fraudulent users can easily take advantage of an insecure VoIP system and place calls, causing extra traffic and toll charges. Eavesdropping is another threat to the security of VoIP communication.

VoIP Security Tips

VPN: Run VoIP traffic over a VPN.

Secure RTP (SRTP): Use SRTP to encrypt RTP payload.

Secure SIP (SSIP): Use SSIP to encrypt the SIP messaging port.

Firewall: Allow traffic from trusted network addresses only.

Call Filtering: Allow calls from trusted numbers only.

Authentication: Authenticate users prior to providing VoIP services.

Quadro System Security Management: Quadro SIP IDS and Security Audit

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SIP Resources on the Internet:

General Information Sites: http://www.voip-info.org/tiki-index.php

VoIP Service Provider List: http://www.voipproviderslist.com/

SIP Training and Certifications: http://www.thesipschool.com/

Free news and magazines: http://www.tmcnet.com/

Internet Engineering Task Force: http://www.ietf.org/ http://www.ietf.org/rfc/rfc3550.txt

(RTP- Real Time Protocol) http://www.ietf.org/html.charters/

sip-charter.html (SIP)

International Telecommunication Union

(ITU): http://www.itu.int/home/index.html

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The end