Building a Voice Over IP (VOIP) Service

VOIP services are needed by telecommunication company since many subscribers has many purposes by doing that such business activity, contacting their family etc. Telecommunication Company prefer to provide international call service with VOIP rather than International Direct Dial (IDD) service which need a license from the regulatory institution beside it gives cheaper tariff. From subscribers’ view, the different between both VOIP and IDD service is while they make a dial session. VOIP service is valid to their subscribers only and subscribers should dial with additional prefix. Meanwhile, IDD service is valid to any subscribers from any telecommunication company with certainty prefix number which already given by regulatory institution.

It is different with IDD service, while Telecommunication Company decides to build a VOIP service, there will be some network infrastructures needed to be inserted to the current one. VOIP service is running on the existing network, telecommunication company just need to make sure that the call was delivered to the destination at abroad and conversation happen between calling and called number. Generally, the new network infrastructures that needed are;

  1. VOIP gateway
  2. International Link/Bandwidth
  3. International VOIP partner

The international call from subscriber will forwarded from Mobile Switching Centre (MSC) to VOIP gateway and digit translation is happening here. This new digit number will be encapsulated to the IP base packet and will be delivered to the International VOIP partner. VOIP gateway responsible to covert traditional call bases (ISDN and Signaling) coming from subscriber to the IP base packet.

Since VOIP service is international call base, telecommunication company should deploy international link or bandwidth to deliver and retrieve a packet. It is like a bandwidth needed by mobile internet service.The international link could be dedicated E1 transmission that will be terminated from Telecommunication Company to the international partner. Usually, Telecommunication Company prefers to provide internet bandwidth to deliver the VOIP traffic since they can insert it to another service which requires internet bandwidth as well such as mobile internet service .

Telecommunication Company did not deliver the VOIP traffic to the destination number directly. The international call will be forwarded to their international VOIP partner at abroad and let them deliver it to the proper destination number. International VOIP partner may forward the call to the destination number directly through VOIP service as well or they will forward it to their partner as well. It is depending on the level of international VOIP partner has been chosen by Telecommunication Company. Telecommunication Company also can connect their international call to several International VOIP partner to distribute various traffic scenario. This scenario usually was applied to give various tariff calls to certainty countries

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30 comments to Building a Voice Over IP (VOIP) Service

  • Ocim

    lengkap tutorialnya

    keep posting om

  • Syarief

    Salam kenal sobat… ditunggu komentarnya

  • mas doyok

    lengkap banget nih
    seneng postingannya pake alur

  • findtechno

    Nice Article
    Thanks.

  • FITA

    mampir ya ke website ku, nti aq kunjung balik…thanx..

  • Kesuksesan Hidup | Renungan | Motivator | Kata Bijak

    Thanks for sharing this great article :)

    Hope success will always be yours…

  • riz

    Building a Voice Over IP (VOIP) Service

    waw very interesting yor post n good soltion

  • lusi

    nice post bro

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  • Sell Your Phone

    Nice post. It was helpful and informative. hope to see more greats posts like this.

  • Free International VOIP Calls

    Telco indonesia is not a good VOIP company there prices are not good sir.

  • iphin cow

    good info sir………..

  • xander

    nice info bro

  • shaz

    nice information

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    wah bagus banget eung postingannya

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    nice info sobat

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  • Matthew C. Kriner

    Great information! Thanks!

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  • Amie Eynon

    Hey! Thank you, this has helped my research a lot!. Ironically, I have been reading about voice over IP for the past 5-6 days but I didn’t see any product which lets one to dial free calls to Russia, I saved 4 or 5 sites that gave free calls to the United States but couldn?t find any for the the opposite so have you found any product that allows you to make unlimited calls? Please email me!

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  • Mark Haynes

    Nice blog, I found you while searching on Yahoo. Does anyone know where to find a ? I really would like to get started using a voip service.

  • Imran Malik

    Impressive piece of information, let me elaborate more on VoIP. Voice over Internet Protocol has been around since many years. But due to lack of sufficient and affordable bandwidth it was not possible to carry carrier grade voice over Internet Protocol. But since the arrival of low cost internet bandwidth and new speech codecs such as G.729, G.723 which utilizes very low payload to carry carrier class voice it has recently been possible to leverage the true benefits of VoIP. G.723 codec utilizes only 6 Kbps (Kilo Bytes/sec) which is capable of maintaining a constant stream of data between peers and deliver carrier grade voice quality. Lets put this way if you have 8 Mbps internet connection, by using G.723 codec you can run upto 100 telephone lines with crystal clear and carrier grade voice quality. I am also a user of VoIP and have setup a small PBX at home. Since I have discovered VoIP I have never used traditional PSTN service.

    Dear readers, if you have not yet tried VoIP I suggest that you try VoIP technology and I bet you will never want to use the traditional PSTN phone service ever again. VoIP has far more superior features to offer which traditional PSTN sadly cannot offer.

    Also It has recently been possile to carry Video alongwith VoIP by using low payload video codecs. I cannot resist to tell you that by using T.38 passthrough and disabling VAD VoIP can carry FAX transmission, but beaware FAX T.38 passthrough will only work when using wide band protocols such as G.711, a-Law and u-Law.

    By using ATA (Analog Telephone Adapter) which converts VoIP signals into traditional PSTN you can also using Dial-up modems to connect to various dialup services. I wont go in to the details what VoIP can offer, to cut my story short VoIP is a must to have product for every business and individual.

    How VoIP Works

    When we make a VoIP call, a communication channel is established between caller and called party over IP (Internet Protocol) which runs on top of computer data networks. A telephony conversation that takes place over VoIP are converted into binary data packets streams in real time and transmitted over data network, when these data packets arrive at the destination these are again converted into standard telephony conversation. This whole process of voice conversion into data, transmission and data conversion into back voice conversation takes place within less than few milliseconds. That is how a VoIP is call is transmitted over data networks. I hope that now you understand basics of how a VoIP call takes place.

    What are speech codec’s and what role codec plays in VoIP?

    Speech codec play a vital role in VoIP and codec determines the quality and cost of the call. Let me explain you what exactly VoIP codec’s are and how they work. You may have heard about data compression, or probably you have heard about air compressor which compresses a volume of air in enclosed container, VoIP codec’s are no different than a air compressor. Speech codec’s compresses voice into data packets and decompresses it upon arrival at destination. Some VoIP codec’s can compress huge amount of voice while maintaining QoS which means use this type of codec will cost less because it will consume just a fraction of data network. Some codec’s are just not capable of encoding huge amount of voice they simply consume huge amount of data networks bandwidth hence the cost goes up.

    Following is a list of VoIP codec’s along with how much data network bandwidth they consume.

    * AMR Codec
    * BroadVoice Codec 16Kbps narrowband, and 32Kbps wideband
    * GIPS Family – 13.3 Kbps and up
    * GSM – 13 Kbps (full rate), 20ms frame size
    * iLBC – 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
    * ITU G.711 – 64 Kbps, sample-based Also known as alaw/ulaw
    * ITU G.722 – 48/56/64 Kbps ADPCM 7Khz audio bandwidth
    * ITU G.722.1 – 24/32 Kbps 7Khz audio bandwidth (based on Polycom’s SIREN codec)
    * ITU G.722.1C – 32 Kbps, a Polycom extension, 14Khz audio bandwidth
    * ITU G.722.2 – 6.6Kbps to 23.85Kbps. Also known as AMR-WB. CELP 7Khz audio bandwidth
    * ITU G.723.1 – 5.3/6.3 Kbps, 30ms frame size
    * ITU G.726 – 16/24/32/40 Kbps
    * ITU G.728 – 16 Kbps
    * ITU G.729 – 8 Kbps, 10ms frame size
    * Speex – 2.15 to 44.2 Kbps
    * LPC10 – 2.5 Kbps
    * DoD CELP – 4.8 Kbps

    Switch to VoIP Today and you will never want to use traditional PSTN ever again.

    Thanks

    -Imran

  • Joline Mou

    Im surprised to have found this website, gave you guys a bookmark, just awesome information I found here.

  • my sister is very good in installing VOIP equipments and i really admire her for that.;”`

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