Meneropong Peranan ITU-T
Penulis : Abdul Salam Taba

Keberadaan dan peranan sub-organisasi telekomunikasi sedunia (International Telecommunication Union) yang khusus mengkaji dan membuat standar telekomunikasi secara global (ITU-T), diakui atau tidak, berperan signifikan dalam mendorong lahirnya berbagai jasa telekomunikasi khususnya dan teknologi komunikasi dan informasi (information and communication technology/ICT) pada umumnya. Sebagai contoh, sepanjang tahun 2005 telah ditetapkan 62 rekomendasi standar telekomunikasi dan ICT, termasuk 101 yang direvisi dan 89 yang diamendemen oleh ITU-T.
Sejak didirikan pertama kali pada tahun 1956 --dengan nama International Telegraph and Telepone Consultative Committee (CCITT)-- hingga bulan Juni 2006, ITU-T telah berhasil menetapkan sebanyak 3145 standar perangkat telekomunikasi dan ICT. Sebenarnya CCITT sendiri merupakan hasil penggabungan International Telephone Consultative Committee dan International Telegraph Consultative Committee yang masing-masing dibentuk pada tahun 1924 dan 1925 (ITU News; 6/2006).
Secara faktual, peranan dan fungsi ITU-T dapat dilihat dari berbagai upaya strategis yang dilakukan secara konsisten dan sistematis dalam pengembangan standar telekomunikasi dan ICT berbasis global. Berbagai upaya tersebut dilakukan sejak sejak tahun 1956, bahkan secara historis sejak tahun 1924, hingga sekarang. Secara kronologis berbagai upaya tersebut pada dasarnya dapat dibagi menjadi enam fase, dimana setiap fasenya menunjukkan peran konkret ITU-T yang berkesinambungan dari tahun ke tahun.

Fase pertama, upaya restrukturisasi organisasi yang ditandai dengan penggabungan Komite Konsultatif Telepon dan Telegrap Internasional. Fase ini berlangsung dari tahun 1956 sampai dengan tahun 1968 dengan kegiatan sebagai berikut.
• Tahun 1956, pembentukan CCITT yang dipicu oleh keberhasilan menciptakan standard perangkat telekomunikasi yang memungkinkan layanan telepon dan telegrap dapat disalurkan secara bersamaan dengan menggunakan transmisi kabel dan radio sirkit.
• Tahun 1958, rekomendasi mengenai tarif dan layanan telegram internasional disetujui oleh majelis Umum (Plenary Assembly) CCITT. Rekomendasi tersebut mengatur antara lain tentang pengiriman telegram dari dan ke berbagai negara, serta lalu lintas percakapan melalui telepon.
• Tahun 1960, melakukan kerjasama dengan organisasi-organisasi internasional lainnya seperti dengan International Electrotechnical Commission (IEC), misalnya, dalam penetapan standard kabel dalam bertelekomunikasi.
• Tahun 1964, upaya penomoran sambungan langsung internasional (SLI) yang berlangsung dari tahun 1960 hingga 1964. Upaya penomoran ini didahuluii dengan kajian mengenai semua aspek komunikasi antar benua yang melalui transmisi kabel laut, serta upaya routing and numbering untuk mengantisipasi layanan teleks dan telepon otomatik secara global.

Penomoran SLI ditetapkan oleh ITU-T dengan mengatur kode negara, kode area dan nomor lokal setiap Negara anggota.
• Tahun 1968, munculnya layanan faksimili secara global. Layanan ini mulai bisa digunakan setelah pertemuan Majelis Umum CCITT ke empat yang berlangsung di Argentina. Pertemuan ini berhasil menyetuji untuk pertama kalinya standar perangkat faksimili yang dapat dioperasikan secara global.

Fase kedua, pengaturan sambungan dan nomor telepon. Penomoran telepon merupakan suatu sistem buatan ITU yang memungkinkan orang dapat melakukan panggilan dan menerima panggilan lokal, SLJJ dan SLI. Sistem tersebut mengatur kode negara, kode area dan nomor lokal semua nomor telepon di seluruh dunia, yang memungkinkan proses pemanggilan dan penerimaan sambungan telepon dapat berlangsung secara otomatis. Proses ini bisa terjadi berkat sistem pensinyalan (signalling system) yang juga dibuat oleh ITU, dan standard/sistem pensinyalan ini sangat penting bagi beroperasinya layanan SLI.

• Tahun 1969, perbaikan sistem layanan telepon seluler. Layanan telepon seluler yang digunakan sekarang merupakan pengembangan dari pengembangan dari IMTS (improved mobile telephone service) yang diperkenalkan pertama kali pada taun 1969. Layanan IMTS ini dikenal sebagai generasi “0G” (oposan dari generasi ketiga seluler, 3G) yang merupakan telepon radio berbasis pra-seluler VHF/UHF yang terhubung dengan public switch telephone network (PSTN). IMTS merupakan telepon radio yang sama dengan layanan telepon land-dial, yang menawarkan layanan yang bisa tersambung langsung dan tanpa melalui operator.
• Tahun 1976, penggunaan jaringan packet switched (PS). Jaringan PS ini lahir berkat rekomendasi X.25 yang digunakan untuk penyediaan jaringan PS komersial dan internasional untuk pertama kalinya. Rekomendasi X.25 ini juga sesuai dengan standard ITU-T yang dimanfaatkan untuk jaringan WAN (wide area network). Jaringan PS digunakan secara luas di seluruh dunia pada era 1980 dan 1990, sebelum digantikan oleh teknologi-teknologi yang lbih baru seperti ISDN (integrated service digital, ADSl (asymmetric digital subscriber line), dan IP (internet protocol).
• Tahun 1981, dibuat sistem pensinyalan 7 (signalling system 7, SS7). Keberadaan SS& ini diakui pertama kali tahun 1981 di dalam rekomendasi seri Q.7XX. Sebelum diimplementasikan, sebagian anggota tidak menyetuji penggunaa sistem ini untuk layanan SLI. Keberadaan SS7 yang berkemampuan mengirim sinyal lewat saluran yang berbeda dan mendeteksi problem keamanan jaringan secara lebih dini ini, telah memungkinkan penggunaan jaringan internasional secara lebih efisien. SS7 ini juga bermanfaat untuk menghubungkan lalu lintas layanan VoIP (Voice over Internet Protocol) ke PSTN, dan mendukung berbagai layanan yang berbasis IN (intelligent network).
• Tahun 1984, lahirnya standard layanan SLI yang berbasis ISDN (integrated service digital network). ISDN ini merupakan sistem jaringan komunikasi yang memungkinkan data dan suara dapat dikirim secara bersamaan ke seluruh dunia, dengan menggunakan konektivitas digital end to end. Sistem yang berkemampuan mentransfer data 64 kbit per detik ini pertama kali digunakan sebagai cicuit-switched telephone system yang berbasis digital pada tahun 1984.
• Tahun 1984, munculnya standar komunikasi yang berbasis abstract syntax notation 1 (ASN.1). Kemunculan ASN.1 ini sangat mendukung perkembangan jaringan, karena dapat digunakan, misalnya, untuk melakukan verifikasi kartu kredit dan dokumen digital, termasuk berbagai program software yang ada sekarang. ASN.I merupakan notasi atau bahasa formal yang berfungsi untuk menampilkan struktur data untuk keperluan pengiriman, encoding dan decoding serta pengambilan data.

Fase ketiga, transisi ke era teknologi digital. Di akhir tahun 1970 hingga diawal tahun 1980, telekomunikasi mengalami perkembangan revolusioner ke arah teknologi digital. Ditandai dengan penyatuan teknologi telekomunikasi dan komputer, penurunan tarif SLJJ, peningkatan kapasitas transmisi akibat penggunaan sistem komunikasi satelit dan kabel laut (submarine cable), dan kemajuan pesat pada jaringan data berbasis public switched. Salah satu perubahan fundamental di masa tersebut ialah munculnya jaringan digital terpadu yang lebih dikenal dengan nama ISDN. Berbagai keberhasilan tersebut, membuat standardisasi yang dibuat ITU semakin penting.
• Tahun 1986, pembentukan Joint Photographic Expert Group (JPEG) yang didirikan oleh ITU, ISO dan IEC. Standar buatan JPEG banyak dimanfaatkan untuk penyimpanan dan pengiriman gambar (foto digital) secara on-line.
• Tahun 1988, kode identitas bagi pelanggan seluler SLI (international mobile subscriber identity, IMSI) ditemukan. Standar IMSI ini lahir melalui resolusi E.212 dari ITU-T dan merupakan suatu sistem yang dapat mengenali perangkat seluler (handheld) yang lagi bergerak dari satu jaringan ke jaringan lainnya. Selain itu, IMSI juga berfungsi mengidentifikasi perangkat seluler yang sedang bergerak untuk keperluan penagihan dan biaya berlangganan.
• Tahun 1988, standard PKI (public key infrastructure) berhasil dibuat berdasarkan rekomendasi X.509. Keberadaan PKI ini berfungsi untuk mengamankan jaringan public.
• Tahun 1988, standard untuk audio coding berhasil diciptakan. Teknologi telekomunikasi untuk layanan suara yang paling banyak digunakan saat ini ialah PSTN (public switch telecommunication network). Standar untuk PSTN ini dikembangkan oleh ITU-T (G.711 dan seri G.72x), dan edisi terakhir dibuat pada tahun 1990 yang banyak digunakan untuk pengembangan VoIP.
• Tahun 1988, standar untuk jaringan pengaturan layanan telekomunikasi (telecommunication management network, TMN) berhasil diciptakan. TMN yang banyak digunakan oleh para operator telekomunikasi terkemuka di dunia karena berfungsi mendukung pengatutan dan pengembangan layanan telekomunikasi lewat kemampuan interoperabilitasnya.
• Tahun 1989, penemuan standar teknologi untuk pengiriman informasi digital melalui SDH (synchronous digital hierarchy) yang berfungsi mengsingkroniasikan pengiriman data melalui jaringan serat optik. SDH ini banyak dimanfaatkan operator telekomunikasi sebagai tulang punggung jaringan radio dan serta optik, karena berkemampaun mengirim data berkecepatan gigabit dan memudahkan pengaturan jaringan.

Fase keempat, hadirnya beragam standar bagi layanan berbasis pita lebar (broadband) dan internet. Beberapa standar ITU yang telah berkontribusi positif bagi pengembangan akses dan kecepatan internet adalah standard seri-V untuk modem komputer, yang memungkinkan sebagian besar orang untuk pertama kalinya dapat menikmati layanan on-line; standard SDH untuk transmisi data melalui serat optik, yang banyak digunakan sebagai jaringan tulang punggung bagi beragam layanan data berpita lebar; serta standar untuk layanan yang berbasis teknologi DSL (digital subscriber line). Contoh lainnya ialah beragam standar untuk layanan gambar dan video, PKI (public key infrastructure), dan VoIP yang telah banyak mempengaruhi aktifitas dan kehidupan umat manusia.

• Tahun 1993, kelahiran ITU-T yang dipicu oleh hasil Konferensi Tingkat Tinggi ITU (ITU Plenipotentiary Conference) tahun 1992 berkekeputusan melakukan reformasi struktural, dengan cara memberikan kemudahan bagi ITU untuk beradaptasi dengan tuntutan perubahan lingkungan yang semakin kompleks. Berdasarkan hasil keputusan tersebut, maka pada tahun 1993 dibentuklah organisasi baru untuk mengurusi masalah-masalah standardisasi (ITU-T dan Telecommunication Standardization Advisory Group, TZAG), yang bertugas menggantikan tugas-tugas CCITT dan kegiatan penetapan standar sistem layanan seluler lain yang dilaksanakn oleh CCIR.
• Tahun 1993, penetapan standard bagi teknologi DSL (digital subscriber line). Dalam seri rekomendasi ITU-T G.922, ADSL merupakan suatu sistem berbasis DMT (discreate multi-tone technique) yang digunakan untuk menciptakan beragam jenis layanan dengan memaksimalkan jaringan telekomunikasi tradisional (copper). Jaringan DSL, yang beroperasi dengan memaksimalkan fungsi jaringan kabel tembaga milik incumbent, masih merupakan pilihan utama bagi operator dalam memberikan layanan yang bebasis teknologi berpita lebar kepada para pebisnis berskala kecil dan pelanggan perumahan. • Tahun 1996, standard internasional untuk layanan nomor telepon bebas ke seluruh dunia (universal international freephone numbers, AUIFN) diadopsi untuk pertama kalinya. AUFIN merupakan suatu layanan yang memungkinkan pemasar menggunakan nomor yang sama dalam berkomunikasi secara gratis di wilayah negara manapun dia berada, dimana biaya percakapannya nantinya akan ditagih ke rekening perusahaannya. Saat ini, layanan AUFIN secara global sudah berjumlah 29.000 nomor.
• Tahun 1996, penetapan standar yang dapat membantu pengembangan layan VoIP. Standar tersebut bernomor H.323 yang berfungsi mempermudah pengiriman layanan suara, gambar dan data melalui jaringan computer (internet). Standar seri H.323 berperan penting dalam pengembangan layanan VoIP karena didukung oleh pengembang perangkat dan berkemampaun interoperabilitas. Saat ini pengguna sistem H.323 diperkirakan sudah mengirim miliaran layanan komunikasi suara per menitnya.
• Tahun 1996, proses standardisasi teknologi PON (passive optical networks) yang berstandar G.983.1/2 memakan waktu 10 tahun, yakni dari tahun 1996-2006. Teknologi PON merupakan teknologi yang efektif dalam mengimplementasikan keterhubungan jaringan serat optik ke lokasi perumahan dan perusahaan, yang sebentar lagi akan berubah menjadi jaringan berbasis optik penuh. Selain dimanfaatkan operator untuk menghindari mahalnya biaya pengembangan jaringan, teknologi PON juga digunakan untuk menghubungkan local loop dengan perangkat pengguna yang beroperasi di semua jaringan serat optik.

Fase kelima, pengembangan standar teknologi jaringan NGN (next generation network). Secara teknologis, NGN merupakan teknologi yang dapat merubah sistem jaringan circuit switched ke sistem jaringan berbasis paket, yang diperkirakan akan banyak digunakan oleg operator di seluruh dunia dalam beberapa tahun mendatang. Karena selain dapat mengurangi biaya pengembangan bagi penyelenggaran jasa, NGN juga mampu menawarkan beragam jenis layanan. Untuk mendukung perkembangan NGN, ITU-T telah membentuk Focus Group yang bertugas membuat standar global bagi teknologi NGN.

• Tahun 1996, standar ATM (asynchronous transfer mode) ditetapakan. ATM merupakan teknologi jaringan yang memungkinkan transfer data berbentuk sel-sel (fixed-size packets), dan dapat dimanfaatkan untuk pengiriman data berskala kecil dan besar. Secara praksis, pengguna ADSL menggunakan teknologi ATM karena mampu mengirim data ratusan megabit per menit, serta dapat mengukur kualitas dan kinerja jaringan.
• Tahun 1997, rencana penomoran layanan SLI berstandar E.164 diterima secara global. Standar tersebut menawarkan empat kategori bagi struktur penomoran layanan SLI, yakni berdasarkan wilayah, layanan global, jaringan, dan kelompoka negara. Standar E.164 ini sangat bermanfaat bagi jaringan telepon umum, dan tanpa kehadirannya kita tidak melakukan SLI dengan baik dan mudah.
• Tahun 1998, standard modem sistem dial up berhasil dibuat. Tanpa kehadiran standar modem yang dibuat ITU-T, kemungkinan internet tidak berkembang seperti sekarang ini. Karena setiap orang yang mau mengakses internet, sebelum munculnya ISDN atau teknologi berpita lebar, harus menggunakan modem yang dibuat berdasarkan spesifikasi ITU. Bahkan penggunaan modem sebagai akses internet masih sangat penting hingga saat ini. Pada tahun 1998, standar V.90 merupakan standar modem dial up generasi baru dengan kecepatan akses 56 kilobit per menit. Standar modem V.92 ini mulai dikerjakan pada tahun 1999 dan ditetapkan pada tahun 2000, dengan kemampuan melipatgandakan kecepatan penerimaan incoming data.
• Tahun 1998, kemunculan akses layanan pita lebar yang berbasis kabel. Dewasa ini penggunaan kabel sebagai media dalam mengakses layanan berpita lebar lagi marak. Bagi anda yang memiliki modem kabel dapat menikmati layanan pita lebar dengan melakukan penyesuaian spesifikasi yang dibuat ITU-T. Standar J.112 yang diadopsi pada tahun 1998, misalnya, merupakan standar yang digunakan untuk beragam layanan televisi kabel interaktif. Demikian pula standard J.117 yang diaopsi pada tahun 1999, dapat diamanfaatkan untuk mengkonversi televisi kabel menjadi televisi digital dengan memanfaatkan HDTV (high-definition
• Tahun 1998, harmonisasi tarif interkoneksi. Seperti diketahui bahwa tarif interkoneksi merupakan tarif yang harus dibayar oleh penyelenggara jasa pada saat pelanggannya melakukan panggilan ke penyelenggara jasa lainnya. Sistem tarif interkoneksi ini semakin kompleks dengan munculnya liberalisasi dan globalisasi pasar. Untuk mengantisipasi masalah tersebut, ITU-T membuat prinsip-prinsip negosiasi tarif, dan berupaya membantu negara-negara berkembang melakukan penyesuaian dengan tuntutan perubahan pasar (rekomendasi D.140).

Selain ITU-T juga memperkenalkan konsep baru remunerasi internasional (rekomendasi D.140), dengan cara merubah sistem tarif akunting (accounting-rates system) ke sistem tarif terminasi (termination rate sistem).

Fase Keenam, kebangkitan era cybersecurity (“keamanan dunia maya”). Diyakini bahwa standardisasi merupakan upaya paling efektif dalam menyatukan berbagai kekuatan untuk menghadapi ancaman terhadap keamanan dunia maya, seperti phising, pencurian identitas secara on-line, dan spam. Ada 13 kelompok studi dan 70 lebih rekomendasi yang dibuat ITU-T terkait dengan masalah keamanan dunia maya. Sebagai contoh, rekomendasi X.805 yang memberikan kemampuan (spesifikasi) bagi para pemerintah, perusahaan dan penyelenggara jaringan telekomunikasi dalam mengatasi berbagai ancaman sistem jaringan dari dunia maya.
• Tahun 2000, protokol pensinyalan BICC (bearer independent call control) berhasil dibuat. Protokol BICC merupakan sebuah tonggak sejarah bagi pengembangan jaringan yang berbasis paket dan multi media pita lebar, karena memungkinkan migrasi tanpa batas dari jaringan sirkit switch ke jaringan paket berbasis multi media pita lebar yang berkapasitas besar. Protokol ini juga digunakan untuk mendukung beroperasinya layanan berbasis PSTN/ISDN melalui jaringan backbone berbasis paket (IP dan pita lebar), tanpa mengganggu interfaces penyelenggara jaringan dan jasa yang ada.
• Tahun 2002, persetujuan standar untuk video coding (H.264/AVC) yang berkemampuan menawarkan layanan video (film) berskala luas dan berkualitas tinggi, mulai dari layanan HDTV hingga videoconference dan seluler multimedia berbasis 3G. Standar H.264/AVC memiliki kualitas gambar yang lebih baik, kemampuan menyalurkan beragam jenis layanan, dan media penyimpanan data yang lebih besar.
• Tahun 2006, publikasi VDSL2 (very-high-bit-rate digital subscriber line) berdasarkan rekomendasi ITU-T. Standar ini memungkinkan para operator telekomunikasi berkompetisi dengan penyelenggara kabel dan satelit, dengan cara menawarkan beragam layanan antara lain seperti HDTV, video on demand, videoconferencing, akses internet berkecepatan tinggi, dan VoIP. Satndar VDSL2 yang baru mampu mengirim data hingga 100 megabit per menit baik secara up stream maupun down stream.

Berdasarkan berbagai upaya yang telah dan sedang dilakukan ITU-T, sebagaimana dipaparkan di atas, dapat disimpulkan bahwa sub-organisasi ITU tersebut berperan signifikan dalam mendorong inovasi teknologi dan kemajuan industri telekomunikasi (termasuk industri ICT), baik secara nasional dan regional maupun internasional. Pasalnya, standar yang dibuat ITU-T tidak hanya berkontribusi bagi pengembangan perangkat dan teknologi telekomunikasi, tetapi juga turut memicu lahirnya beragam bisnis dan jasa-jasa baru dalam industri telekomunikasi dan ICT secara global.

* Alumnus School of Economics The University of Newcastle, Australia. Published: Majalah Telematika, Vol.I Edisi 1, Februari 2007

Cyberoam Special Feature

To introduce Cyberoam, Cyberoam is the only UTM that offers identity-based security in addition to the complete set of security features. In doing so, it offers comprehensive security against external and internal threats. With insider threats accounting for over 50% of attacks, gaining visibility into individual user activity is critical for enterprise security. Cyberoam gives enterprises complete visibility into their network, showing Who is doing What. At the same time, it allows enterprises to create policies based on the user identity, reflecting the individual user's work profile.

In addition, Cyberoam Identity-based UTM appliances offer complete protection against existing and emerging Internet threats, including viruses, worms, Trojans, spyware, phishing, pharming and more.

CYBEROAM FEATURES
Stateful Inspection Firewall
VPN - Virtual Private Network
Gateway Anti-Virus &
Anti-Spyware

Gateway
Anti-Spam
Intrusion
Detection and Prevention

Content & Application
Filtering
Bandwidth Management

Multiple Link Management


CYBEROAM ADVANTAGE

Identity-based Security Identifies "Who is doing What?"

Single Administrative Interface for all functions

Secures even in DHCP & Wi-fi environments

Eliminates need for Technical Manpower

Reduced Capital & Operational Expenditures

Frees you from Multiple Vendors, Upgrades & Patches

Flexible, Easy-to-deploy & Easy-to-manage
500+ on-appliance reports


Cyberoam UTM is currently present in more than 60 countries worldwide. With
the UTM market growing at a pace of over 45% per annum, Cyberoam and RML
will drive the security appliance business through Cyberoam's unique
identity-based product offering and RML's channel partner network.

CERTIFICATIONS
Checkmark Certification - "UTM level 5" (Complete UTM Certification -
Certifies Firewall, VPN, Anti-Virus, Anti-Spyware, Anti-Spam, URL Filtering, IDP/IPS)
ICSA Labs Certified Firewall
Member of VPN Consortium (For Basic & AES Interoperability)

Cyberoam, with its unique Identity-based offering has secured

Whitepaper by IDC - featuring Cyberoam as a "Next Generation UTM"
Positive Rating in 2008 Gartner Marketscope of SMB Multi-function Firewalls
Winner of 2008 Frost & Sullivan "Emerging Company of the Year" Award
Star rating by SC magazine in UTM Reviews of 2007, 2008
2007 Global Excellence Awards in Integrated Security Appliance, Security
Solution for Education and Unified Security for Cyberoam
Infosecurity 'Tomorrows Technology Today' Award for Unified Security (2007)
http: com="" html="">Network Security Product's Guide - Best Content Filtering Solution (2008)

With all these great features, we assure you that Cyberoam solutions will be
beneficial to you and your organizations.

Benefits of Videoconferencing

With oil prices soaring, the pace of business faster than ever and a global imperative to reduce carbon emissions, videoconferencing is having a remarkable impact on the way we communicate and do business.
By replacing the need for in-person meetings, videoconferencing
* Eliminates travel expenses
* Saves time
* Increases productivity
* Provides real-time communication with small or large groups of people worldwide
* Enables remote training
* Strengthens relationships with remote colleagues, partners and customers
* Improves work-life balance for busy executives by eliminating the need to travel

The ROI for videoconferencing is clear and tangible and is felt almost immediately across-the-board.

PRESS RELEASE

Frost & Sullivan awards Cyberoam Emerging Company of the Year for Network Security in Asia-Pacific Success attributed to its Innovative Product Line – Identity-based integrated security appliances

(Ahmedabad, India 22 September, 2008) Cyberoam, the industry leading identity-based Unified Threat Management (UTM) range of appliances from Elitecore Technologies, announced today that Frost & Sullivan, the renowned global consulting company, has presented it with the 2008
Asia Pacific Emerging Company of the Year award for Network Security.

The prestigious award from Frost & Sullivan not only recognizes Cyberoam UTM’s unique identity-based security which effectively controls user-targeted external threats as well as insider threats, but also validates its architectural flexibility, targeted vertical focus, product innovation and R&D. Its strong emphasis on R&D is identified as a key factor behind Cyberoam’s growing success and global recognition in the network security space.
Arun Chandrasekaran, industry manager at Frost & Sullivan said, “One of the biggest strengths behind the success of Cyberoam is its innovative - identity-based integrated security appliances.” He adds, “Cyberoam has effectively used its sales and marketing resources, as well as technical superiority to gain a firm foothold in the market. Its products have found wide acceptance among businesses across various geographies, within a short span of time.

Cyberoam has successfully rolled out its enhanced multicore appliances with high performance, offering one of the best price/performance ratios in the industry.
Frost & Sullivan Awards are among the principal security awards in the industry. We are honored to receive this award,” said Harish Chib, Vice-President – New Business Development – Cyberoam. “This award is recognition of our ongoing commitment to meeting our customer needs by providing them with the best products and services that provide security against evolving threats.”

Cyberoam identity-based Unified Threat Management appliances provide the complete range of security features like Stateful Inspection Firewall, VPN, Gateway Anti-Virus, Gateway Anti-Spyware, Gateway Anti-Spam, Intrusion Prevention System, URL/Web Content Filtering in addition to Bandwidth Management and Multiple Link Management over a single platform.

Cyberoam has moved into 55 countries within a span of 2 years, including USA, UK, France, Switzerland, Australia and more, displaying speed in execution.


About Cyberoam
Cyberoam Identity-based UTM appliances offer comprehensive protection against existing and emerging Internet threats, including viruses, worms, Trojans, spyware, phishing, pharming and more. Cyberoam delivers the complete range of security features such as stateful inspection firewall, VPN, gateway anti-virus and anti-spyware, gateway anti-spam, intrusion prevention system, content filtering in addition to bandwidth management and multiple link management over a single platform. Cyberoam is certified by the West Coast Labs with CheckMark UTM Level 5 Certification, ICSA Labs, an independent division of Verizon Business, and is a member of the Virtual Private Network Consortium. Cyberoam has also received 5 star rating from SC Magazine in both 2007 and 2008, the 2007 Global Excellence Awards for Integrated Security Appliance, Security Solution for Education and Unified Security, and the 2007 Tomorrow’s Technology Today Award for Unified Security. Cyberoam has offices in the USA (Woburn, MA) and India. For more information, please visit www.cyberoam.com.

About Elitecore Technologies Limited
Elitecore Technologies Limited is the global provider of Cyberoam UTM appliances. Elitecore’s other divisions include Crestel Convergent Billing Solution that meets the voice, data, video billing and customer care requirements of Tier-1 service providers and 24online Billing and Bandwidth Management Solution for hotels, hotspots and Internet service providers. Elitecore has a strong R&D base and support center in India; it has sustained a healthy growth rate of over 75 % since inception. For more information, please visit www.elitecore.com

About Frost & Sullivan Frost & Sullivan, the Growth Partnership Company, partners with clients to accelerate their growth. The company's TEAM Research, Growth Consulting and Growth Team Membership empower clients to create a growth-focused culture that generates, evaluates and implements effective growth strategies. Frost & Sullivan employs over 45 years of experience in partnering with Global 1000 companies, emerging businesses and the investment community from more than 30 offices on six continents. For more information about Frost & Sullivan’s Growth Partnerships, visit http://www.frost.com

Comprehensive Internet Security System

Cyberoam’s integrated Security Appliances are purpose-built for comprehensive network protection and high performance needs of small, medium and large enterprises. Cyberoam’s Check Mark Level 5 certified, ICSA firewall certified identity-based Internet Security Appliances offer protection against external as well as internal threats.

Identity-based Security - Patent Pending Technology
Cyberoam’s unique user identity-based Internet Security Appliances solve today’s need to control individual user behavior to ensure comprehensive threat management. It gives complete visibility into “Who is doing What” in the network and allows policies to be created at the user level based on work profiles. With the finest level of controls and an unprecedented degree of control, flexibility and ease of management, the Cyberoam Internet Security Appliance is a highly effective UTM solution that reduces capital and operating expenses.

Architecture Flexibility
Cyberoam’s architectural flexibility can easily accommodate emerging applications like VoIP through own enhancements and easy third party plug-ins with no architectural changes, keeping enterprises in a state of constant threat-readiness in a rapidly evolving threat scenario. Cyberoam appliances balance performance and security by using technologies like multi core processors, regular expressions co processors and hardware based Advanced Cryptography Engine for accelerating key functions of the UTM appliance.

Data sheet CR100i Data sheet CR50i Data sheet CR25i

Konsep CIR (Committed Information Rate)

Bandwidth rasio (sharing)

Rasio bandwidth adalah besarnya bandwidth murni yang diberikan dari/ke pelanggan dibandingkan dengan banyaknya pelanggan. Misalnya: Bandwidth murni ke Internet yang diberikan adalah 64 kbps, bandwidth rasio / sharing adalah 1:2, berarti dari 1 bandwidth murni 64 kbps dibatasi untuk digunakan ke 2 pelanggan 64 kbps atau ke 3 pelanggan 64 kbps (untuk rasio/sharing 1:3). Akibatnya pelanggan akan berebut bandwidth dan akan sangat terasa kecepatan bandwidth menurun apabila pelanggan lainnya sedang melakukan transfer data secara besar-besaran dan terus menerus. Sistem ini TIDAK digunakan oleh ISP. CIR (Committed Information Rate) - yang digunakan ISP Analogi CIR diambil dari jaringan Frame Relay. CIR (dalam bits per detik) digunakan untuk mengatur flow data yang dikirimkan dari/ke pelanggan. Besar CIR normalnya adalah setengah dari besar bandwidth total. Jadi apabila menggunakan bandwidth 64kbps, maka CIR normalnya adalah 32kbps. Artinya, pada saat terjadi network congestion (jaringan jenuh / bandwidth habis — lihat penjelasan di bawah) jaminan kecepatan transfer data minimum adalah 32kbits (CIRnya bekerja) dengan catatan pada jaringan tidak terjadi gangguan apapun. Pada saat tidak terjadi network congestion, besar bandwidth ditentukan oleh EIR (Excess Information Rate) yaitu bandwidth total yang diberikan ke pelanggan (bandwidth yang disewa). Jadi pada sistem CIR, TIDAK ada pembatasan bandwidth maupun pembagian bandwidth / sharing ke pelanggan lain, kecepatan bandwidth dapat menurun hanya apabila terjadi network congestion.
Network Congestion (Jaringan jenuh)
Network congestion dapat terjadi apabila bandwidth-bandwidth pada gateway ISP penuh. Bisa bandwidth pada Access Point Wireless, bisa bandwidth pada Ethernet, maupun bandwidth gateway-gateway Internet ISP (OIX, INDONET, ADSL dll).

Bandwidth Meter http://myip.indo.net.id
NOTE:
Be carefull to be sure what bytes, bits, etc .. are …
If you are not sure, please read this:
1 byte = 8 bits.
1 kbits = 1024 bits.
1 mbits = 1024 kbits = 1048576 bits.
64 kbits/sec = 8 kbytes/sec.
The “bit” unity is used for speeds : kbits/sec or mbits/sec.
The “bytes” unity is used for data (traffic) : kbytes or mbytes.
thx,

Ervin Taufik

Bandwidth

1) In computer networks, bandwidth is often used as a synonym for data transfer rate - the amount of data that can be carried from one point to another in a given time period (usually a second). This kind of bandwidth is usually expressed in bits (of data) per second (bps). Occasionally, it's expressed as bytes per second (Bps). A modem that works at 57,600 bps hastwice the bandwidth of a modem that works at 28,800 bps. In general, a link with a high bandwidth is one that may be able to carry enough information to sustain the succession of images in a video presentation.

It should be remembered that a real communications path usually consists of a succession of links, each with its own bandwidth. If one of these is much slower than the rest, it is said to be a bandwidth bottleneck.

2) In electronic communication, bandwidth is the width of the range (or band) of frequencies that an electronic signal uses on a given transmission medium. In this usage, bandwidth is expressed in terms of the difference between the highest-frequency signal component and the lowest-frequency signal component. Since the frequency of a signal is measured in hertz (the number of cycles of change per second), a given bandwidth is the difference in hertz between the highest frequency the signal uses and the lowest frequency it uses. A typical voice signal has a bandwidth of approximately three kilohertz (3 kHz); an analog television (TV) broadcast video signal has a bandwidth of six megahertz (6 MHz) -- some 2,000 times as wide as the voice signal.

Source : http://searchnetworking.techtarget.com

Video Conference "xPoint"

Type perangkat Video Conference ini berupa Hardware. User hanya tinggal plug in perangkat ini saja dan perangkat siap di jalankan. Sangant mudah dalam pengoperasiannya. Dalam hal setting alat ini, tidak banyak settingan yang membuat bingung atau ribet, sangat user friendly.
Untuk kamera yang ada dalam perangkat ini, sangatlah flexibel karena antara kamera dan perangkat Video Conference tidak menjadi satu casing. Atau disebut juga Split System. Hal ini akan sangat memudahkan user untuk mengatur keberadaan kamera tanpa harus merubah posisi dari perangkat tersebut. Untuk kemampuan dari perangkat ini sangat tidak diragukan lagi. Kemampuan dari perangkat ini diatas kamera standar Video Conference yang ada di pasaran. Standar yang ada hanya mempunyai kemampuan 10x optical zoom. Akan tetapi erangkat kamera xPoint mempunyai kemampuan kamera 18x optical zoom. Sangat jauh bila dibandiingkan denga standar yang lain.

Advantages of Conferencing With xPoint™

• Provides TV-like video quality using advanced video standards.
• Various network types: IP (up to 4MB) and ISDN (up to 512Kbps)
• H.264 video support up to 4 Mbps
  
• Advanced data sharing capabilities with multiple user options
• Native 16:9 wide screen display support Dual monitor support
• CD-quality audio using 20 KHz ultra-wide band audio
• Web-based management for easier control and administration
  

(389 KB)

Brochure xPoint

High Definition Room Videoconferencing System "HD7000pro"

The HD7000pro is Emblaze-VCON's latest 1080p-ready, high definition room videoconferencing system. Delivering crisp, high definition video for sharper, wider -angle images and CD-quality audio using the latest audio codecs, the HD7000pro is a complete room system solution. With its 6-site multipoint calling ability, DualStream™ technology and session recording capabilities, the HD7000pro also enables presentations, spreadsheets, documents and video clips to be shared via VGA and USB inputs. The HD7000pro is ideal for integration into vertical markets such as educational institutions, healthcare organizations, financial institutions and government agencies.
Advantages of Conferencing with HD7000pro
* High definition videoconferencing system
* H.264 video support up to 4 Mbps and ISDN up to 512 Kbps (4xBRI)
* Multiparty videoconferencing via the embedded MCU
* Advanced data-sharing capabilities
* Session recording, storing and forwarding
* Dual monitor support
* Native 16:9 wide screen display support
* CD-quality audio
* Supports encrypted conferences using the H.235 encryption standard
* Web-based management
* 1080p ready

Brochure HD7000pro(649 KB)

Video Conference HD4000

Perangkat Video Conference selama ini identik dengan Hardware yang membutuhkan cost besar untuk pengadaannya. Sebenernya dengan perkembangan Technology seperti sekarang, hal itu bukan suatu hal yang menjadi keharusan. Perkembangan Software untuk Video Conference sudah sangat bagus baik segi kualitas, dan fitur. Untuk harga sudah pasti sangat jauh bila dibandingkan dengan perangkat yang Hardware base.

HD4000 merupakan Software Video Conference yang saat ini sudah bisa di sejajarkan kemampuannya dengan perangkat yang Hardware base. Kalau pada artikel sebelumnya ada Software Video Conference dengan type vPointHD, HD4000 memiliki beberapa kelebihan. Antara lain Codec Video yang akan sangat bagus karena di peruntukan untuk Multimedia Video Conference dimana sangata cocok untuk di terapkan di suatu meeting room yang memiliki kapasitas oeserta yang banyak. Sedangkan yang type vPoint, lebih di peruntun untuk personal Video Conference.
Selain itu HD4000 sangat mudah dalam pengoperasinnya, kualitas yang maksimal, dan didukung dengan fitur yang lengkap. Inilah yang menjadi HD4000 menjadi software Video Conference yang sangat siap berkompetisi di persaingan Teknology Video Conference sekarang ini.

Advantage of Conferencing with HD4000

• Provides TV-like video quality using advanced video standards.
• Advanced data sharing capabilities with multiple user options
• H.264 video support up to 4 Mbps
• Native 16:9 wide screen display support
• Dual monitor support
• CD-quality audio using 20 KHz ultra-wide band audio
• Call forward, call pickup, call transfer, ad-hoc conferencing
• Web-based management for easier control and administrationH.239 and HD DualStreamTM for simultaneously sending and receiving video and data streams
• Supports encrypted conferences using the H.235 encryption standard

Brochure HD4000(876Kb)

Konsep dari vPointHD ini adalah Video Conference dengan system software dan lebih di peruntukan ke personal Video COnference. Hanya dengan installasi di PC/Notebook maka kita bisa ber-Video Conference dengan full fiture bahkan bisa mengirim & menampilkan presentasi file dengan format ppt, pdf, jpg, doc. Bahkan lebih dari itu, bisa mengirim file video dengan format WMA, AVI, MP4, dll. Adapun file Audio yang bisa di kirim meliputi WAV, MP3. Masih banyak fitur dan kualitas yag sangant sesuai untuk diaplikasikan sehari-hari. Saat ini salah satu fitur yang hanya dimiliki oleh vPointHD EMBLAZE Vcon adalah Multicast Interactive dimana kita bisa mengirimkan multicast dan partisipan juga bisa ber-interactive dalam Conference tersebut

Advantages of Conferencing with vPoint HD

• High definition video and audio quality videoconferencing
• Supports ISDN networking up to 4 BRI lines
• Multiple viewing modes from mini-mode to full-screen
• Video Standards : H.261, H.263, H.263+/++, H.264 - up to 2Mbps outgoin
• Incorporates H.239, HD DualStream™ for simultaneously sending and receiving video + data streams
• Supports encrypted conferences using the H.235 encryption standard
• Emblaze-VCON PacketAssist™ architecture for advanced Quality of Service (QoS) over IP
• Remote software upgrades from network administrator guarantee uniformity across the organization (with MXM)
• Fully compatible with Windows Vista SP1
• Variety of models
• *1080p Full HD incoming - Executive Model
  

Brocure vpointHDv8.0

(536 KB)

MCU ( Multi Conference Unit )

VCB combines traditional multipoint videoconferencing, streaming and scheduling into a single, affordable solution. Both scheduled and ad-hoc conferences are supported for a network of H.323 endpoints and SIP endpoints. VCB includes management, as well as basic features such as continuous presence and rich, intelligent conference layouts (up to 16 participants at a time). Additionally, the VCB includes all the latest audio, video, and data algorithms.

• Scalable from 12 to 48 ports per server
• Includes embedded H.323 Gatekeeper
• Continuous Presence - H.264 4CIF 30 frames per second
• Data rates up to 4.0Mbps per participant
• Wide-band G.722.1, G.723.1, G.728 and AAC-LD audio with transcoding
• Mixed H.323 and SIP endpoints in a multipoint conference
• Web-based configuration program with scheduler, moderating and monitoring
• Incorporates H.329 support for simultaneously sending and receiving video and data streams
Continuous presence allows users to see up to 25 participants at the same time. Conference managers may choose among a wide combination of rich and intelligent layout formations. Layouts change dynamically according to the number of participant.


The VCB is shipped as an integrated solution in a rack-mount appliance, scalable from 12 to 48 port. Includes embedded H.323 Gatekeeper (MXM without vPoint HD licenses) that supports 25 to 100 users depending upon model.
VCB is also available as an optional software module of MXM (Media Xchange Manager), but this is limited to 12 ports and not expandable.

Brochure MCU EMBLAZE (249Kb)

Web Conference vs Video Conference

Web Conference Vs Video Conference

Terdapat cukup banyak variasi teknologi yang dapat membantu user ketika ingin melakukan meeting baik dari kantor ataupun dari tempat di seluruh dunia. Ada dua teknologi yang cukup populer, yakni Web Conference dan Video Conference. Keduanya memiliki kelebihan dan kekurangan masing-masing dalam penggunaannya.

Web Conference mengijinkan user untuk menghadiri meeting ‘virtual’ dari seluruh dunia. Untuk itu, diperlukan koneksi Internet dan software sebagai instalasi jika ada client yang ingin menggunakannya. Sementara Video Conference akan memberikan kemudahan bagi user untuk mendapatkan 2 tipe, yakni audio dan video sehingga user pengguna Video Conference dapat melihat dan berbicara dengan orang lain secara real time. Bahkan untuk sekarang ini, Video Conference dianggap sebagai teknologi masa depan dari videophone, namun dalam skala yang lebih besar.

Video Conference lebih efektif daripada Web Conference karena memiliki kemampuan untuk menampilkan wajah dan body user yang berbicara, dapat melakukan pengamatan mengenai reaksi bahasa tubuh, ekpresi muka dan intonasi suara. Sementara Web Conference masih terbatas pada hal-hal di atas pada fiturnya.
Keduanya memiliki persamaan untuk dapat melakukan percakapan dari mana saja, namun untuk Web Conference lebih menonjolkan personal user dengan desain web sesuai selera user yang bersangkutan. Walaupun hal tersebut benar, tetapi Web Conference juga dapat menyebarkan informasi dan bukan tempat yang tepat untuk sosialisasi user dengan tujuan bisnis.
Untuk Video Conference memiliki standard internasional yang dapat memprmudah komunikasi. Biasanya terdapat satu lokasi standard untuk server Video Conference yang dapat mengambil semua informasi dan kemudian mendistribusikannya ke lokasi conference lainnya. Hal ini akan membuat biaya lebih besar, tergantung dari jumlah koneksinya. Selain itu, masih ditambah dengan adanya camera video atau yang biasa disebut web cam, mikrofon, dan speaker, dan hub server yang terkoneksi dengan PC. Sementara Web Conference hanya meminta koneksi Internet.
Web Conference dapat dijalankan oleh ribuan user, namun jumlah user dalam satu channel terbatas. Sedangkan Video Conference lebih terbatas pada kapasitas servernya. Web Conference lebih fleksibel karena dapat dijalankan di banyak tipe koneksi Internet tetapi Video Conference hanya bisa dijalankan di koneksi Internet broadband (jarak jauh).
Kedua tipe komunikasi diperlukan dalam komunitas bisnis. Perusahaan dapat memilih mana yang lebih tepat digunakan dan yang lebih efisien dalam proses komunikasi. Sebagai contoh, biaya Video Conference yang lebih besar daripada Web Conference menjadi suatu penghalang bila dibandingkan dengan hasilnya yang jauh lebih efisien. Namun, masih banyak perusahaan yang lebih memilih Video Conference daripada Web Conference, mengingat bahwa dengan Web Conference, walaupun ringan di ongkos, tetapi penggunaanya masih kurang efektif.

Ditulis oleh : adityawarman


Pendapat saya tentang perbedaan antara Wb Conference vs Video Conference sangatlah mendasar. Saya asumsikan Web Conference itu seperti : Yahoo Messanger, Skype, MSN dll. Sedangkan Video Conference seperti Polycom, Tandberg, Aetra, Sony, dan EMBLAZE VCON.
Perbedaan mendasar dari kedua jenis Tele Conference ini antara lain.
Web Conference:
1. Memiliki protokol yang propetiery karena tergantung akan web server sehingga tidak akan bisa saling berkomunikasi antar Web Conference (YM, Skype, MSN,dll). Mereka hanya bisa berhubungan dalam satu jenis/merk Web Conference.
2. Tidak adanya feature untuk mengirimkan/menerima presentasi (H.239.
3. Tidak adanya Bandwidth Mangement untuk berapa bandwidth yang akan kita pakai.
4. Frame rate yang kecil (max 15 fps) sehingga kualitas gambar tidak bisa seperti gambar real.
5. Tidak adanya Lip Shyncronise sehinga gerakan bibir dam suara tidak bisa match/cocok.
6. Tidak ada Echo Cancelation sehingga kualiatas suara kurang bagus karena akan ada noise dan feedback sound.

Video Conference:
1. Memilik Protokol yang standard H.323 ( IP ) sehingga sangat memngkinkan untuk saling berkomunikasi antar jenis perangkat dalam biarpun berbeda merk/vendor.
2. Adanya Feature untuk mengirim / meneriama presentasi H.239
3. Adanya Bandwidth Management sehinga kita bisa mengatur dengan banwidth berapa kita akan caling atau terkoneksi dengan yang lain.
4. Frame Rate yang lebih real karena bisa sampai 30 fps sehingga bisa menghasilkan kualiatas gambar seperti TV/CD/DVD.
5. Dilengkapi dengan technology Lip Shyncronise sehingga antara gerakan dan suara bisa sesuai.
6. Adanya Echo Canclelation sehingga kualitas Audio akan lebih baik dan lebih jelas.

Itu adalah beberapa perbedaan yang ada antara Web Conference dan Video Conference.
Sebenarnya masih banyak perbedaan akan tetapai akan sangat banyak kalau di tulis semua.
Pada Intinya, kedua fasilitas Conference ini sangatlah berbeda. biarpun secara keliatannya hampir sama.
Terima kasih

IP Addresses Explained

IP Address (Internet Protocol Address): This number is an exclusive number all information technology devices (printers, routers, modems, et al) use which identifies and allows them the ability to communicate with each other on a computer network. There is a standard of communication which is called an Internet Protocol standard (IP). In laymans terms it is the same as your home address. In order for you to receive snail mail at home the sending party must have your correct mailing address (IP address) in your town (network) or you do not receive bills, pizza coupons or your tax refund. The same is true for all equipment on the internet. Without this specific address, information cannot be received. IP addresses may either be assigned permanently for an Email server/Business server or a permanent home resident or temporarily, from a pool of available addresses (first come first serve) from your Internet Service Provider. A permanent number may not be available in all areas and may cost extra so be sure to ask your ISP.





Domain Name Server (DNS):

This allows the IP address to be translated to words. It is much easier for us to remember a word than a series of numbers. The same is true for email addresses.

For example, it is much easier for you to remember a web address name such as whatismyip.com than it is to remember 192.168.1.1 or in the case of email it is much easier to remember anonymous@whatismyip.com than anonymous@192.168.1.1

Dynamic IP Address:

An IP address that is not static and could change at any time. This IP address is issued to you from a pool of IP addresses allocated by your ISP or DHCP Server. This is for a large number of customers that do not require the same IP Address all the time for a variety of reasons. Your computer will automatically get this number as it logs on to the network and saves you the trouble of having to know details regarding the specific network configurations. This number can be assigned to anyone using a dial-up connection, Wireless and High Speed Internet connections. If you need to run your own email server or web server, it would be best to have a static IP.

Static IP Address:

An IP address that is fixed and never changes. This is in contrast to a dynamic IP address which may change at any time. Most ISP's a single static IP or a block of static IP's for a few extra bucks a month.

IP version 4: Currently used by most network devices. However, with more and more computers accessing the internet, IPv4 addresses are running out quickly. Just like in a city, addresses have to be created for new neighborhoods but, if your neighborhood gets too large, you will have to come up with an entire new pool of addresses. IPv4 is limited to 4,294,967,296 addresses.

IP version 5: This is an experimental protocol for UNIX based systems. In keeping with standard UNIX (a computer Operating System) release conventions, all odd-numbered versions are considered experimental. It was never intended to be used by the general public.

IP version 6: The replacement for the aging IPv4. The estimated number of unique addresses for IPv6 is 340,282,366,920,938,463,463,374,607,431,768,211,456 or 2^128.

The old and current standard of addresses was this: 192.168.100.100 the new way can be written different ways but means the same and are all valid:

* 1080:0000:0000:0000:0000:0034:0000:417A

* 1080:0:0:0:0:34:0:417A

* 1080::34:0:417A

Kamus tentang IT ( Video Conferencing )

A

Application Sharing
This is a feature that allows two or more people to work together when one the individuals does not have the same application, or same version of the application. In application sharing, one user launches the application and it runs simultaneously. All users can impute information and otherwise control the application using the keyboard and mouse. Files associated with the application can be easily transferred, so the results of the collaboration are available to all users immediately. The person who launched the application can lock out the other person from making changes, so the locked-out person sees the application running but cannot control it.


Application Viewing
In personal conferencing, the users sharing the application can see every keystroke or mouse movement made by the one user who is running the application. The other users have no control over the application.

ATM. Asynchronous Transfer Mode
High speed low-delay transport technology, integrating multiple data types (voice, video, and data). ITU has selected ATM as the basis for the future broadband network because of its flexibility and suitability for both transmission and switching. May be used in the phone and computer networks of the future.

Audio
Signals that carry sounds.

Audio Bridge
Equipment that mixes multiple audio inputs and feeds back composite audio to each station after removing the individual station's input.

Automatic Bandwidth Adjustment
Algorithm in H.323 endpoint for automatic increasing and decreasing video bit rate due to the network behavior.


B

B channel
The ISDN circuit-switched bearer channels, capable of transmitting 64kps of digitized information.

B-ISDN
Broadband ISDN. The ITU-T is developing the B-ISDN standard, incorporating the existing ISDN switching, signaling, multiplexing and transmission standards into a higher-speed specification that will support the need to move different types of information around the public switched network.

Bandwidth
A term that defines the information carrying capacity of a channel - its throughput. In analog systems, it is the difference between the highest frequency that a channel can carry and the lowest, measured in hertz. In digital systems the unit of measure of bandwidth is bits per second.

Bit. Binary Digit
The basic signaling unit in all digital transmission systems.

Bit rate
The number of bits of information transmitted over a channel in a given second. Typically expressed bps.

Bps
Bits per second, a unit of measurement of the speed of data transmission and thus of bandwidth.

BRI. Basic Rate Interface In ISDN there are two interfaces, the BRI and the PRI or Primary Rate Interface. The BRI offers two circuit-switched B (bearer) channels of 64 kbps each and one packet-switched 16 kbps D (delta) channel that is used for exchanging signals with the network.

Bridge
In videoconferencing vernacular, a bridge connects three or more conference sites so that they can simultaneously communicate. Bridges are often called MCUs - Multiple Conferencing Units. A bridge is also considered a device that interconnects LAN segments at the data-link layer of the OSI model to extend the LAN environment physically. They work with frames of data, forwarding them between networks. They learn station addresses and they resolve problems with loops in the topology by participating in the spanning tree algorithm. Finally, the term bridge can be used in audio conferencing to refer to a device that connects multiple (more than two) voice calls so that all participants can hear and be heard.

Broadcasting
In packet-switched networks, this means sending a packet to all users connected to the specific network.

C

Call
Multimedia communication between two or more H.323 endpoints.

Call Signaling Channel
Reliable channel used to convey call setup messages following Q.931

Centralized Multipoint Conference
A call in which all participating terminals communicate in a point-to-point fashion with an MCU.

Caller ID
An identification (number, name) of the party being called. This identification is of interest when you transfer or forward a call. For example, when an unanswered call is forwarded to a voice messaging system, the called-ID of the original call is used to locate the mailbox of the called party.

CCITT
Consultative Committee for International Telegraphy and Telephony. As of 1994 known as the International Telecommunications Union. See ITU.

CIF
Common Intermediate Format, an optional part of the ITU-T's H.261 and H.263 standards. CIF specifies 288 non-interlaced luminance lines, that contain 176 pixels. CIF is to be sent at frame rates of 7.5, 10, 15, or 30 per second. When operating with CIF, the number of bits that result can not exceed 256 K bits (where K equals 1024).

Circuit-switched
An ISDN bearer service that provides a 64 kbps (sometimes 56 kbps) path between two users for the duration of the call. The term is also used for the networks with behavior similar to ISDN.

CODEC
A sophisticated digital signal-processing unit that takes an analog input and converts it to digital on the sending end. At the receiving end, another codec reverses this by reconverting the digital signal back to analog. Codec is a contraction of code/decode (some experts in the video industry assert it also stands for compress/decompress). A codec takes the form of a set of hardware or software components, or a combination of both.

Compression
Reducing the representation of the information, but not the information itself. Reducing the bandwidth or number of bits needed to encode information or encode a signal, typically by eliminating long strings of identical bits or bits that do not change in successive sampling intervals (e.g., video frames). Compression saves transmission time or capacity. It also saves storage space on storage devices such as hard disks, tape drives, and floppy disks.

D

Decentralized Multipoint Conference
Conference in which the participating terminals multicast to all other participating terminals without an MCU.

Document Sharing
See Whiteboard

E

E.16
Address format for ISDN networks. See ITU Recommendation E. 164 (1991). Added as alias for H.323 terminals.

Endpoint
A Terminal, Gateway, or MCU.

Ethernet
A LAN running on coaxial or twisted pair wiring, at 10 or 100 mbps. In Ethernet, all terminals are connected to a single common highway or bus.

Ethernet switch
A device than connects local area networks (LAN). Ethernet switching is viewed as one solution to deliver 10 Base-T or 100 Base-T networks that are bandwidth-constrained because of a new requirement to carry multimedia messages and interactive videoconferencing communications. To qualify as an Ethernet Switch, a device must be capable of switching packets from one Ethernet segment to another "on the fly" and exhibit very low port-to-port latency.

F

Full-duplex
A communication protocol in which the communications channel can send and receive data at the same time. Compare to half-duplex, where information can only be sent or received in one direction at a time.

Full High Definition (Full HD)
Full HD refers to the resolution used for the display of the video images during video or television broadcast .Full definition refers to 1080p (1920 x 1080 pixels). "1080" stands for 1080 lines of vertical display resolution, while "p" stands for progressive, or non-interlaced, scan. 1080p assumes a widescreen aspect ratio of 16:9. 1080p models offer pictures with even finer detail and more subtle colors than 720p models.

G

G.711
An ITU-T Recommendation entitled, "Pulse Code Modulation (PCM) of Voice Frequencies". G.711 defines how a 3.1 kHz audio signal is encoded at 64 kbps using Pulse Code Modulation (PCM) and either mu-law (US and Japan) or A-law (Europe).

G.721
An ITU-T Recommendation that defines how a 3.1 kHz audio signal is encoded at 32 kbps using Adaptive Differential Pulse Code Modulation (ADPCM).

G.722
An ITU-T Recommendation that defines how a 7.5 kHz audio signal is encoded at a data rate of 64 kbps.

G.722.1
G.722.1 is an ITU-T standard audio codec meant for high quality, moderate bit rate (24 and 32 kbit/s) wideband (50 Hz - 7 kHz audio bandwidth, 16 ksps) audio coding

G.722.1 Annex C
G.722.1 Annex C delivers 14 kHz audio fidelity at 24, 32, or 48 kbps (kilobits per second).

G.723
An ITU-T Recommendation entitled, "Dual Rate Speech Coder for Multimedia Communication Transmitting at 5.3 and 6.4 kbps".

G.728
An ITU-T Recommendation for audio encoding using Low Delay Code Excited Linear Prediction (CELP). The bandwidth of the analog audio signal is 3.4 kHz whereas after coding and compression the digitized signal requires a bandwidth of 16 kbps.

G.729
G.729 is an audio data compression algorithm for voice that compresses voice audio in chunks of 10 milliseconds. G.729 is mostly used in Voice over IP (VoIP) applications for its low bandwidth requirement.

Gateway
The gateway allows H.323 systems to interoperated with other H.32x products. For instance, the gateway could link the H.323 session with an H.320 (ISDN-based) system; an H.321 (ATM-based) system; an H.322 (iso Ethernet-based) system; or an H.324 (POTS-based) system. At the present, most H.323 gateway implementations are concerned with linking H.323 and H.320/H.324 systems across a LAN/WAN connection.

GateKeeper
A gatekeeper is a utility that controls H.323 videoconference access on a packet-switched network. It requires that multimedia terminals register "at the gate", which is accomplished when the terminal provides its address. The gatekeeper translates network addresses and aliases to make connections. It can also deny access or limit the number of simultaneous connections to prevent congestion.

H

H.221
A framing portion of the ITU-T's H.320 Recommendation that is formally known as "Frame Structure for a 64 to 1920 kbps Channel in Audiovisual Teleservices". The Recommendations specifies synchronous operation in which the coder and decoder synchronize timing.

H.222
ITU-T Recommendation specifies generic coding of moving pictures and associated audio information.

H.223
Part of the ITU-T's H.324 standard specifying a control/multiplexing protocol, which is formally called "Multiplexing protocol for low bit rate multimedia communication".

H.231
A Recommendation added to the ITU-T's H.320 family specifying multipoint control unit used to bridge three or more H.320 compliant codecs together in a multipoint conference.

H.233
A multiplexing Recommendation that is part of the ITU-T family of video interoperability Recommendations. The Recommendation specifies how individual frames of audiovisual information are to be multiplexed onto a digital channel.

H.235
H.235 covers security and encryption for H.323 and other H.245 based terminals. The standard addresses authentication by means of several algorithms and privacy which allows for encryption also of the media streams.

H.239
The H.239 ITU-T recommendation is titled "Role management and additional media channels for H.3xx-series terminals". Practical importance of this recommendation is its setting forth a way to have multiple video channels (e.g., one for conferencing, another for presentation) within a single session (call).

H.242
Part of the ITU-T's H.320 family of video interoperability Recommendations. H.242 specifying the protocol for establishing an audio session and taking it down after the communication has terminated.

H.245
Part of the ITU-T's H.323 and H.324 families defining control of communications between multimedia terminals.

H.261
The ITU-T's Recommendations that allows dissimilar video codecs to interpret how a signal has been encoded and compressed, and to decode and decompress that signal. It also defines two picture formats: CIF and QCIF.

H.263
H.263 is an ITU-T videoconferencing Recommendation originally designed as a low bit-rate compressed format for videoconferencing.

H.264
H.264 is the latest ITU-T videoconferencing Recommendation that offers a new video compression scheme. The main benefits of H.264 are higher video quality at a given bit-rate, higher resolution and lower storage requirements.



H.320
An ITU-T standard including a number of individual recommendations for coding, framing, signaling and establishing connections (H.221, H.230, H.321, H.242, and H.261). It applies to point-to-point and multipoint videoconferencing sessions and includes three audio algorithms, G.711, G.722 and G.728.

H.323
The H.323 extends the H.320 to Intranet, Extranet or Internet over packet-switched networks: Ethernet, Token-Ring, and others that may not guarantee QoS. It also specifies procedures for videoconferencing over ATM including ATM QoS. It supports both point-to-point and multipoint operations.

H.323 Alias
User logical name used for remote party calling. Translated by Gatekeeper to the network address.

H.324
An ITU-T standard that provides point-to-point data, video, and audio conferencing over analog telephone lines (POTS). It can incorporate H.261 video encoding, but most implementations will probably use H.263, a scalable version of H.261 that adds a 128-by-96 Sub-QCIF (SQCIF) format. Because of H.263's efficient design, it may produce frame rates much like those of today's ISDN H.320 systems through inexpensive hardware-assisted modems. The H.324 family includes H.223, a multiplexing protocol. H.245, a control protocol, T.120, a suite of audiographics protocols and V.34, a modem specification.

High Definition (HD)
High definition refers to the resolution used for the display of the video images during video or television broadcast. HD refers to 720p (1280 x 720 pixels) or higher. "720" stands for 720 lines of vertical display resolution, while "p" stands for progressive, or non-interlaced, scan. 720p assumes a widescreen aspect ratio of 16:9. With high definition (HD) videoconferencing systems, the improvement in video quality over standard definition (SD) videoconferencing is dramatic since HD carries more than 9 times the pixel count of standard CIF videoconferencing images.

I

Interactive Multicast
This method of streaming is a hybrid of the IP multicast protocol and is a unique streaming method invented by Emblaze-VCON. Emblaze-VCON Interactive Multicast uses the bandwidth-efficient IP Multicast protocol, but adds a layer of interactivity, where the "chair" or host of the multicast can virtually move the podium to any other participant, allowing interaction between all on the call, while using only one stream of video and audio, thus preserving bandwidth. This technology is ideal for bandwidth conscious networks and satellite-based networks.

IP. Internet Protocol
The most popular network protocol in corporate and public networks. May be used by H.323 endpoints for audio, video, and data packets transfer.

Interoperability
The ability of electronic components produced by different manufacturers to communicate across product line. The trend toward embracing standards has greatly furthered the interoperability process.

ISDN
Integrated Services Digital Network. ISDN is an entirely digital telephone service that can be installed by the local telephone company to replace the old analog local loop (the connection to the telephone company's nearest central switching office) with a digital line. As long-distance lines are usually digital already, replacing the local loop with an ISDN line provides "end-to-end" digital service. Two types of ISDN are: BRI and PRI. ISDN BRI is the interface to connect the desktop to the digital long distance network. ISDN BRI provides two 64 kbps B ("bearer") channel to carry information content, the voice, video, and data substance of a transmission. A separate 16 kbps D ("data") channel is used for call setup and signaling. ISDN BRI is often called "2B+D" ISDN, for its combination of two B and one D channel.

ITU
International Telecommunications Union. One of the specialized agencies of the United Nations that is composed of the telecommunications administrations of 113 participating nations. Founded in 1865 before telephone were invented as a telegraphy standards body. It now develops international standards for interconnecting telecommunications equipment across networks.

K

kbps
Kilo-bytes per second - one thousand bits per second.


L

LAN
Local Area Network (LAN). A network of computer and other devices for communication within a restricted geographic area, such as a building or a campus.

M

mbps
Megabits per second or approximately one million bits per second.

MCU – Multipoint Control Unit
The MCU is a bridging device that enables multipoint videoconferencing (3 or more sites). Once a MCU bridge is set up it is possible to add multiple sites to a video call and simultaneously allow several additional locations to participate in the session. Basic Features include:

* Voice Activated Switching - Switches the camera to the speaker, so that they may be seen by all participants.
* Continuous Presence – All the participants appear on the screen simultaneously and the speaker is highlighted
* Lecture Mode - Allows the speaker to see a mix of all participants; all other participants see the lecturer in full screen mode.

Multicasting
Sending a packet that can be received by multiple recipients, all of whom are listening on a single multicast address.

Multiplex
A method of transmitting multiple signals onto a single circuit so that each can be recovered intact.

Multiplexer
Electronic equipment that allows multiple signal to share a single communications circuit.

Multiplexing
The process of combing multiple signals onto a single circuit using various means.

Multipoint
Communication configuration in which several terminals or stations are connected. Compare to point-to-point where communication is between two stations only.

Multipoint Control Unit (MCU)
A device that bridges together multiple inputs so that three parties or more can participate in a video conference.

Multipoint Processor (MP)
An entity which provides for the processing of audio, video, and/or data streams in a multipoint conference. The MP provides for the mixing, switching, transcoding, or other processing of media streams under the control of the MC.

N

Network
A group of stations (computers, telephones, or other devices) connected by communications facilities for exchanging information. Connection can be permanent, via cable, or temporary, through telephone or other communications links. The transmission medium can be physical (copper, wire, fiber optic cable etc.) or wireless, for example via satellite.

P


POTS
Plain Old Telephone Service. Conventional analog narrowband telephone line using twisted-pair copper wire for transmitting voice calls.

Q

Q.931
Call signaling protocol for setup and termination of calls.

Quality of Service (QoS)
Guarantees network resources for specific application requirements.

R

RAS Channel
An unreliable channel used to convey the Registration, Admissions and Status messages and bandwidth changes between two H.323 entities through a Gatekeeper.

Reliable Transmission
Connection-oriented data transmission which guarantees sequenced error-free, flow-controlled transmission of messages to the receiver.

Resource Reservation Protocol (RSVP)
IETF specification. Allows applications to request dedicated resources.

Real-Time Protocol/Real-Time Control Protocol (RTP/RTCP)
IETF specification for audio and video signal management. Allows applications to synchronize and spoil audio and video information.

Router
Equipment that facilitates the exchange of packets between autonomous networks (LANs and WANs) of similar architecture. Routers move packets over a specific path or paths based on the packet's destination, network congestion and the protocols implemented on the network.

S


Switch
A device that establishes, monitors, and terminates a connection between devices connected to a network.

Switching
The process of setting up a connection between an input and an output. It allows a subscriber to establish communications with multiple parties by sending their address to the switch, which will then attempt to make a connection.

Switched Circuit Network (SCN)
A public or private switched telecommunications network such as GSTN or ISDN.

Switch Type
The type of ISDN network you are connected to. This information is available from the ISDN provider and provided to the buyer when purchasing an ISDN line.

T


T.120
The ITU-T's "Transmission Protocols for Multimedia Data", a data sharing/data conferencing specification that lets users share documents during any H.32x videoconference. Like H.32x specifications, T.120 is an umbrella Recommendation that includes a number of other Recommendations. Data-only T.120 session can be held when no video communications are required, and the standard also allows multipoint meetings that include participants using different transmission media. The mandatory components of T.120 include recommendations for multipoint file transfer and shared-whiteboard implementation.

TCP. Transmission control protocol
A reliable transport layer on top of IP.

Teleconferencing
The use of telecommunications links to provide audio, video and graphics capabilities. These systems allow distant workgroups or individuals to meet. An endpoint which provides for real-time, two-way communications with ano.

U


UDP
User Datagram Protocol. An unreliable transport layer on top of IP.

Unicast
Application of conferencing, usually over packet-switched networks, where only one user receives data. In contrast to this, multicast application, where data is received by more than one user.

Unreliable Transmission
Connection-less transmission which provides best-effort delivery of data packets. Messages transmitted by the sender may be lost, duplicated, or received out of sequence.

V


Videoconferencing
A collection of technologies that integrate video with audio, data, or both to convey in real-time over distance for meeting between dispersed sites.

Video Server
A specialized file server with enormous hard disc capacities (often measured in terabytes or trillions of bytes). These servers store MPEG compressed audio and video images and provide service to end-users over high-speed LANs and WANs. Applications that require video servers include entertainment, training/education, and video-enabled databases.

W

WAN
Wide Area Network. A communications network that services a geographic area larger than that served by a local area network or metropolitan area network.

Whiteboarding
A term used to describe the placement of shared documents on an on-screen "shared notebook" or "whiteboard". Multiple users can simultaneously view and annotate a document.


Z

Zone
In H.323 specifications, a collection of all Terminals, Gateways and MCUs managed by a single Gatekeeper. A zone must include at least one Terminal and may include LAN segments connected using routers.