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1 Voice over Internet Protocol (VoIP)
1.1 Introduction to VoIP
VoIP (Voice over Internet Protocol) is a method of voice transport per Internet Protocol used
in packet oriented networks. VoIP is defined in the recommendations ITU-T H.32x and RFC
2443. VoIP can use accelerating hardware to achieve this purpose and can also be used in a PC
environment.
2 Principles of VoIP
VoIP works like that. First the A/DC (Analog to Digital Converter) to convert analog voice
to digital signals. Now the bits have to be compressed in a good format for transmission. Then
we have to insert our voice packets in data packets using a real-time protocol (RTP over UDP
over IP). For signaling between customers terminal unit we need a signaling protocol ITU-T
H.323 or SIP(Session Initiate Protocol). At Rx site we have to disassemble packets, extract
data, then convert them to analog voice signals and send them to sound card (or phone). All
that must be done in a real time fashion cause we cannot waiting for too long a vocal answer.
On voice transport via packet oriented network could be respect requirements of QoS (Quality
of Service) See below article 3.
2.1 Analog to digital conversion
For conversion analog voice to digital signal is very popular use standard PC soundcards or
similar A/D converters. This cards sampling with 16bit a band of 22,050 kHz with sampling
freq 44,100 kHz like that throughput of 2 bytes * 44100 (samples per second) = 88200 Bytes/s,
176.4 kBytes/s for stereo stream.
2.2 Algorithm for processing of digital signal and Compression
For VoIP we needn't such a throughput (176 kBytes/s) to send voice packet. Digitalized voice
data we can compress it, route it, convert it to a new better format that could be quickly
transmitted. The survey of format is in Tab. 1.
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Inhaltsverzeichnis

Seite 1 - 2 Principles of VoIP

1 Voice over Internet Protocol (VoIP) 1.1 Introduction to VoIP VoIP (Voice over Internet Protocol) is a method of voice transport per Internet P

Seite 2 - Num. of layer

5.2.3 QoS-plot Qosplot is a tool that takes as input a set of text log files created by CRUDE [9] and produces as output the data and command files fo

Seite 3

5.3.1 The fidelity of Measuring The fidelity of measuring depend on synchronization of time between sides of measuring [12]. The CRUDE/RUDE can use s

Seite 4 - 1990 1995 1995 1996 1996

5.4.1 Simulation of VoIP flows With simulation utilities like CRUDE/RUDE we can not hollow exactly simulate VoIP flows (with signalization etc.) but w

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Behind the test we measured a loss of 3 packets and delay just about 180ms. This relative big delay is caused using of WiFi equipment. The WiFi use

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Fig. 7 - Delay of flow Fig. 8 - IP packet transfer delay variation (IPDV) of flow Through recommendation ITU-T Y.1541 are defined availabilit

Seite 7 - 3 Quality of Service (QoS)

Recommendation ITU-T Y.1541 IP QoS Classes and ObjectivesQoS Classes and ObjectivesNetwork Performance ParameterNature of ObjectiveClass 0 Class 1 Cla

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6 Voice network with Asterisk PBX Now we will unreel a cheap, powerful and modular solution of local Voice network with Asterisk PBX described in

Seite 9 - 5.2 About utilities

interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with

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Power test PII 400, 256 MB RAM with average time of calls tcs=10 s, time of test ttst=120 ms, G.729 codecNew calls during 1 sec.Concurent callsTot

Seite 11 - 5.4 Solution of QoS test

6.1.2 Installation and configuration The source code is available from ftp.dignum.com as a tarball. After download and unpack the tarball we can r

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Tab. 1 - Compression in VoIP2.3 Transport and signaling protocol in VoIP network For telephony over IP based network we need not only transmit aud

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Most important files are extensions.conf, modem.conf, and channel configuration files like sip.conf if we use soft sip phones. Extensions.conf

Seite 14 - Fig. 7 - Delay of flow

; Configuration for isdn4linux [interfaces];; By default, incoming calls should come in on the "remote" context;context=remote;; Modem Drive

Seite 15 - QoS Classes and Objectives

Where xxx is the username associated with the SIP client, or is an arbitrary name used by other configuration files to refer to this SIP device. Typ

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6.2.1 iFon soft-phoneiFon™ supports both audio and video with Asterisk as well as Free World Dialup. It also interoperates with all major SIP phones,

Seite 17 - Pentium 4, 2.6 GHz, 1 GB RAM

The X-Lite soft-phones provide standard PBX functionality. You can Touch-tones [DTMF], Multiple Proxies, Line Hold, Inbound Call 'Ignore&apo

Seite 18

6.2.4 Kphone softp-hone KPhone is a SIP (Session Initiation Protocol) user agent for Linux, with which you can initiate VoIP (Voice over IP) co

Seite 19

Phone Protocol Codec OS PBX functionsiFone SIP/H.323 G.729A,G.723.1, GSM/AMR, G.722Lin, Mac, Win NoSJphone SIP/H.323 G.711au, GSM Lin, Mac, Win NoX-Li

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6.4 Part of local interconnected ISDN phones At least two cards are required to interconnect internal telephones with external telephone lines. In

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Fig. 16 - VoIP network with SIP soft-phone clients in FER network6.5.1 Configuration of ISDN modem for outgoing calls to ISDN network In connec

Seite 22 - 6.2 Part of clients

If module was not install we can recompile kernel with ISDN HiSax driver for passive ISDN cards and Windbond 6692 based card. See bellow. Fig. 17 -

Seite 23 - Fig. 11 - iPhone client

Firstly, VoIP doesn't use TCP because it is too heavy for real time applications, so instead a UDP (datagram) is used. Secondly, UDP

Seite 24 - Fig. 13 - SJphone

6.5.1.2 Testing the card and the physical connection For testing of physical connection we can use Minicom or another program with similar functi

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; and the line is considered up. "Ring" means we wait until the ring cadence; occurs at least once. "Answer" means we wait until

Seite 26 - 6.3 Part of ISDN connections

secret=xlite2callerid="2102 xlite2" <2102>host=dynamicnat=yes canreinvite=no disallow=allallow=gsm ; GSM con

Seite 27

6.5.4 Configuration of SIP client From compare of soft VoIP clients we can see that the best solution is take the Xlite. First is a Freeware. Sec

Seite 28 - $ modprobe hisax type=13

Fig. 20 -example of network setupFig. 21 -user name and password setup

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Appendix AD-Link AirPlus DWL-900 AP+ The D-Link AirPlus DWL-900 AP+ is an enhanced 802.11b+ Wireless Network Access Point featuring a

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HP ProCurve Switch 2626 Switch Designed for mid-size enterprises that require a smaller, more cost effective switch without sacrificing

Seite 31

[10] QoS-plot a Cesnet project: Tool for computing QoS characteristics, Available <URL: http://www.ces.net/project/qosip/>[11] GNU-Plot: Toll

Seite 32

H.320 H.321 H.322 H.323 H.324Approval1990 1995 1995 1996 1996NetworkISDN-BRA ISDN-PRAATMLANGuaranteedBandwidthPacket switched networksNon-Guaranteedba

Seite 33 - Fig. 19 -xlite client

The H.323 encapsulated 4 primary components:• terminal - Clients that initialize VoIP connection. Although terminals could talk together without any

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o call forwardingo callee and calling ``number'' delivery, where numbers can be any (preferably unique) naming scheme; o personal mobility,

Seite 35 - Appendix A

3 Quality of Service (QoS) We said many times that VoIP applications require a real-time data streaming cause we expect an interactive data voic

Seite 36 - Reference:

4 Simple solution of VoIP without especially HW for testing transmit of voice over IP based networkFig. 3 -Simple solution of VoIP on switched netw

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Hence we have to measure below stated parameters for test quality of IP networks for real-time depend services like as VoIP. QoS parameters are define

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