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Options in voice communication By H Raghavan CCCF 7 Aug 2015
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Outline of the discussion Information on present voice infrastructure Facts about existing system Introduction to emerging technology Comparison of new v/s traditional system Mobile option Work done by CCCF on new technology Demo
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Information on present voice infrastructure Alcatel Omni PCX Enterprise 4400 system 1600 extensions: 1000 analog, 600 digital Voicemail facility to serve 1000 extensions ~300 ADSL ports for internet connection to colony
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Facts on present voice infrastructure Purchased and installed 10 years back System is robust and no major breakdown since installation ADSL is integrated with the system A team of three manage the service Support by service provider is good 1300 extensions in use currently
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Introduction to emerging technology VoIP Abstract What is VoIP How VoIP works VoIP architecture VoIP components
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Abstract VoIP is a technology that enables users to make and receive calls over a data circuit (internet) instead of circuit switched network (Public Switched Telephone Network, PSTN)
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What is VoIP VoIP is also called internet telephony or IP telephony. VoIP is a service / software application which digitizes the analog audio signals to IP packets and sends it over an IP network. VoIP service allow users to make / receive calls to / from any VoIP device / PSTN / mobile number.
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How VoIP works The main components of VoIP (Hardware and software) are: ● VoIP a pplication software ● Protocol ● Hard / Soft phones ● Media gateway ● Codec
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VoIP application software Open source and proprietary VoIP application software is available for Linux and Windows Works as server to VoIP clients (IP Phone, softphone etc.) Functions as soft switch for call processing and call establishment. It acts as a gateway for external calls It also maintains database of users and provide flexibility to define dial plan Billing feature and usage logs for each extension Provides security features to secure the server Gives all the features we get from a EPABX Video session is possible if phone supports
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List of VoIP software Open source Asterix PBX Free Switch Free PBX GNU Gatekeeper Yate Proprietary 3CX PBX Avaya Application Server Cisco Call Manager Alcatel-Lucent
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List of VoIP clients List of open source VoIP clients for desktop / laptop o Ventrilo o X-Lite o MicroSIP o Linphone o Empathy List of open source VoIP soft clients for mobile users o RedPhone o Sipdroid o Yuilop o iCall Mobile
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VoIP Protocols SIP (Session Initiation Protocol) and H.323 signaling protocols are available for VoIP deployment Both are non-proprietary and handle registration, signaling, establishing channels and call termination SIP is widely used and similar to http SIP supports audio and video SIP works at application layer SIP protocol used for signaling during call establishment and tear down RTP (Real Time Protocol) used for transporting voice data UDP (User Datagram Protocol) used for transport of voice packets SIP id’s are easy to remember like e-mail id’s sip:user@voip.tifr.res.in:port
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VoIP Network Model SIP Network Layer IP Transport Layer UDP, TCP RTP, RTCP Datalink Layer Physical Layer
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IP Hard and Soft phones IP hard phones are available from Rs 3,000/= to Rs 40,000/= Some hard phone models comes with built-in camera High end phones allow users to check mails. and SMS Open source and proprietary soft phones are available for different OS and mobile phones. IP hard phones resemble traditional phones No user training required to use the phone
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IP hard phones
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VoIP phone interface
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Media gateway An active device for interfacing PSTN lines to VoIP server. Available in 8, 16 and 24 port denominations Converts analog signals to digital and vice versa H.323 protocol used between VoIP server and media gateway for call handling
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Codec Acronym for coder-decoder Separate codec used for audio and video One of the function is to compress the raw audio / video to save bandwidth Efficiency of codec determines the quality of audio / video Popular audio codecs G711, G722, G723, G726, G729 Popular video codecs H261, H263, H264
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PSTN: Public Switched Telephone Network
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Schematic of a VoIP setup
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Signal flow in VoIP
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How a VoIP phone is connected Do not need additional data and power point Single data point serve both PC & Phone POE (Power Over Ethernet) provide power for VoIP phone VoIP POE enabled
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Comparison of new v/s traditional system Total expenses incurred till date for the EPABX Cost of Alcatel EPABX INR 5300000 Upgrade Cost of Software & Digital phones INR 2275000 AMC for 10 Years @ Rs.4.0 Lacs/annum INR 4000000 Cabling cost for building colony network INR 2500000 Incidental Expense INR 500000 Total expense over a period of 10 Yrs for 1600 ports INR 14575000 Per port expense over a period of 10 Years INR 9100
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Comparison… VoIP (Open source)EPABX Disruptive technologyFading technology Stable and open source technology available Gets locked to a proprietary technology Single network for voice and dataSeparate voice network to be maintained Less initial investment, approx. 1/3rd the cost of EPABX Incur high initial investment and recurring maintenance charges No cost for expansion, cost is only towards hard phone purchase Expansion costs towards purchase of prop. hardware Upgrade at no costUpgrade costs are substantial Mobility is built in to the technologyMobility not possible
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Comparison … VoIP (Open source)EPABX After complete integration with SIP trunking, overseas calls are cheaper Overseas calls are expensive Voice quality a step below compared to EPABX level Very high quality Data network centricDedicated network Power outage can disturb the service Service continues even under long hours of power outage Prone to network congestionDo not get affected by external factors Needs less of power and spaceNeeds more power and space Single team can maintain both the service Separate team is required
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Completely converged Network
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Integrated TIFR voice communication system
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Telecom norms in India for VoIP communication Telecom regulator of India permits voice communication over IP. Framework of regulator permit communication under the conditions ◦ a) PC to PC (Both within as well as outside India). b) PC to Telephone (PC in India to Telephone outside India) c) IP based H.323/SIP Terminals in India to similar terminals both in India and abroad, employing IP addressing scheme of ‘IANA’
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Mobile option (Non-VoIP ) Mobile communication by availing CUG (Closed User Group) feature. Calls within CUG is free. This option needs a low investment and implementation is quick Considerable saving as there will be no AMC charges. No voice network maintenance cost to TIFR, service will be maintained by mobile operator Yearly voice communication cost of TIFR Colaba campus is around Rs 22 L to Rs 24 L. (Excluding official mobile charges) Each user will be provided by a mobile phone instead of wired phone as extension. Guaranteed uptime and mobility feature built-in To be proposed to CCCF committee after detail exploration List price for each connection is Rs 99/= per month includes free 200 mins talk time and 100 SMS
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Work done by CCCF on VoIP Using open source VoIP software, platform is built and ready for testing. Tested VoIP features and call quality using hard and soft phones. Work to be done by CCCF on VoIP Request to be submitted to CCCF committee and seek its approval for beta testing. Acquire skills for building redundant VoIP system. System to be tested on IPv6.
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Acknowledgement I thank my colleagues Sri Vijay Naik and Sri Shisheer Teli for their contribution
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Live demonstration of VoIP features Call a VoIP extension and check for voice quality Calling TIFR VoIP number from outside Three party conference Voice mail Calling a user on his mobile having a VoIP extension Receiving call over TIFR wireless network Multiple extensions ringing simultaneously Video call over VoIP softphone
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Thank you for your participation H Raghavan CCCF
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