Jumat, 14 Januari 2011
VoIP voice quality is influenced by several parameters
Voice over Internet Protocol (also called VoIP, IP Telephony, Internet telephony or Digital Phone) is a technology that allows voice conversations remotely via the Internet. Voice data is converted into digital code and transmitted over the network which sends packets of data, rather than through traditional telephone analog circuits.
VoIP voice quality is influenced by several parameters: capacity bandwidth, packet loss and delay has occurred in the network. Bandwidth capacity is the availability of network resources in the form of bandwidth used to transmit data packets. Packet loss is a parameter that states the rate of errors that occurred during packet data transmission path from sender to receiver. Time delay is a parameter stating the timeframe required to send packages from sender to receiver.
One of the available VoIP applications is Skype. Skype is a [[software]] IP-based voice communication application via the Internet among users of Skype. When using Skype, the Skype users who are online will find other Skype users and began building a network to find other users. Skype has a variety of features that can facilitate the users. Skype is also equipped with SkypeOut and SkypeIn which allows Skype users to communicate with conventional telephone users and mobile phones.
Skype uses the HTTP protocol to communicate with the Skype server to authenticate the username / password and registration with the Skype directory server. Modified version of the HTTP protocol used to communicate with other Skype clients. The advantage of this application is owned by the availability of security services in the form of voice data transmission. Security services provided are as follows:
Privacy
Skype uses AES (Advanced Encryption Standard) 256-bit encryption key with a total probability experiment (brute-force attack) by 1.1 x E-77 times, while for the key exchange (key exchange) simetrik using 1024-bit RSA . User's public key to be certified by Skype server at login using 1536 or 2048 RSA certificate-bit. Skype automatically encrypts all data before it is transmitted over the Internet.
Authentication
Every Skype user has a username and a password. And each username has an e-mail address registered. To enter the Skype system, users must include a username and password pair. If users forget their password then Skype will change it and send the new password to your e-mail users who've registered. This approach is known as E-mail Based Identification and Authentication. Because Skype is a voice communication system, each user can directly identify his interlocutors through the vote.
VoIP Benefits
Lower costs for direct dial long distance. The main emphasis of VoIP is cost. With two locations that are connected to the internet then the conversation becomes very low cost.
Infrastructure utilizing existing data networks for voice. Useful if the company already has a network. If possible the existing network with a VoIP network can be constructed easily. Not required additional monthly fee for the addition of voice communication.
The use of smaller bandwidth than a regular phone. With the rapid advancement of technology, the use of bandwidth for voice had now become very small. Data compression technique allows voice only need about 8kbps bandwidth.
Allows combined with local phone network that already exists. With this form of VoIP network gateway can be connected with existing office PBX. Communication between the office can use a regular telephone
Various forms of VoIP networks can be merged into a larger network. The example is VoIP Rakyat Indonesia.
The variation of the use of existing equipment, eg from a PC connects to a regular phone, IP phone handsets
Weakness of VoIP
The sound quality is not as clear as the PSTN network. Represents the effects of voice compression with a small bandwidth then there will be reduction in sound quality compared to the conventional PSTN network. But if the internet connection used is broadband internet connection / broadband such as Telkom Speedy, it will be crystal clear sound quality - even more clearly than Telkom connections and not to falter.
There was a pause in communication. The process of changing data into sound, network lag, making the gap in communication by using VoIP. Except if you use a broadband connection (see point above).
Regulation of the Government of Indonesia to limit the use to be connected to a network owned by Telkom.
If not connected to the internet 24 hours to promise to be interconnected.
If using the internet and computers behind NAT (Network Address Translation), it takes a special configuration to make it run VoIP
There is never a guarantee of quality when VoIP through the internet.
Equipment is relatively expensive. VoIP equipment that connects between a VoIP PBX (IP telephony gateway) are relatively expensive. It is expected that with the popularity of VoIP is the price of the equipment will also start down the price.
Potentially cause tissue obstructed / Stuck. If the use of VoIP is increasing, then there is the potential of the existing data network to be full if not managed properly. Bandwidth settings are necessary for the company's network does not become saturated due to the use of VoIP.
Surviving without a coordinated network properly will cause chaos in the numbering system
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Community VoIP
Community of users / developers of VoIP in the community, developed in 2000. Community early users / developers of VoIP is the "VoIP Merdeka". "VoIP Merdeka" (VM) is triggered by Onno W. Purbo. The technology used by the "VoIP Merdeka" (VM) is H.323 which is the beginning of VoIP technology. Central VoIP Merdeka in hosting in Indonesia Internet Exchange (IIX) for the support of some ISPs and Asossiasi Internet Network Operator (APJII). Area code "VoIP Merdeka" at that time unanimously determined to be 6288, of course without obtaining approval from the government.
In 2005, Anton Raharja et al of the ICT Center Jakarta began to develop new types of VoIP-based Session Initiation Protocol (SIP). SIP technology is a replacement technology is difficult to penetrate H.323 proxy server. In 2006, SIP VoIP infrastructure, known as VoIP Rakyat
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