Posts tagged voice over ip

An Introduction to Voice Over IP ( VoIP ) with Biamp

Biamp Systems has had a VoIP solution for use in audio conferencing systems for over 2 years now – this technology isn’t new to us. In the past two years, AVI-SPL has installed our solution in systems that are integrated with Cisco Call Manager 5.0 & higher, as well as SIP systems by Avaya, Shoretel and others.

What is VoIP?

Internet Protocol (IP) is the way you can digitally access and send information from a network. You communicate using IP every time you access the internet. Now you can leverage that in new and cost-effective ways. From audio to control to video, IP enabled products are the future backbone for A/V systems. Lower costs, greater flexibility and better management are all advantages of leveraging existing IT infrastructures. With the introduction of the VoIP-2 card, a SIP compliant two-channel Voice over Internet Protocol interface, Biamp Systems follows our tradition of innovating to better serve our customers.

The VoIP-2 Card allows Biamp’s AudiaFLEX to connect directly to IP-based phone systems. Used in conjunction with AEC-2HD Acoustic Echo Cancellation Cards and TI-2 Telephone Interface Cards, the VoIP-2 Card makes AudiaFLEX the most powerful, flexible and affordable telephone conferencing product available. Up to six VoIP-2 Cards can be installed into a single AudiaFLEX unit.

VoIP Basics

Before diving into technical details, this simple overview diagram is a good start to summarize steps involved in a VoIP call.

1. The voice signal is first encoded into a known compressed audio format, packetized in a real-time protocol and then transmitted over the network.

2. A VoIP protocol takes care of managing the communication session.

3. On the receiving side, data is extracted from packets and the signal is decoded back to analog audio. Success of this process is obviously sensitive to delay and packet loss.

Into into VoIP

What Are Those Voice Over IP Acronyms?

Voice over Internet Protocol (VoIP): Protocol specifically designed for voice transmission over networks (LAN/WAN).

Protocol: Similar to how a language enables communication between people, a protocol defines a set of rules used to control connection, communication and data traffic between different network devices.

Codec: It refers to the software algorithm used to encode the voice signal into a compressed data format optimized for transmission over IP. On the receiving side, signal is decoded back to analog audio. Codec quality obviously affects audio performance.

Session Initiation Protocol (SIP): SIP is a widely used peer-to-peer protocol that allows the set up, modification and tear down of a VoIP communication session. Peers of a SIP session are the User Agent Client (Initiating the call) and User Agent Server (Answering the call). Note that SIP does not handle voice transmission, it only manages the communication.

SIP servers: They include the Proxy, Redirect and Registrar Servers. Their purpose is to provide name resolution, user location and pass on messages to other servers in the network.

SIP addresses: Users in a SIP network are identified by unique SIP addresses. A SIP address is similar to an e-mail address and may be of two types: a user name ( or an E.164 address (5036417287). VoIP-2 card only supports E.164 address.

Real-time Transport Protocol (RTP): An IP packet format that is used for delivering real time audio/video over the LAN/WAN. Once the VoIP call session initialized by SIP, RTP is the protocol used to transmit the voice data.

Quality of Service (QoS): Mechanism used to prioritize applications, users, data flow by guaranteeing a certain level of performance. QoS is very important in the case of RTP applications such as VoIP where it is used to insure quality of the audio signal.

Domain Name System (DNS): DNS procedures provide translation from human friendly hostnames into IP addresses. The SIP session mainly uses DNS to allow a client to resolve a SIP URI into the IP address, port and transport protocol.

SIP call flow process: During the registration process, SIP devices register to a registrar server their SIP addresses. The network is then aware of the location of a device upon request. When a user initiates a call, the SIP discovery process starts by sending a request to a SIP server (proxy or redirect server). The challenge for the proxy server is to obtain the IP address of the device such that voice data can be routed between them. Negotiating a compatible data format (sample rate, codec …etc) is the next step before voice data can be transmitted between parties. SIP terminates the call session with a BYE message at the end of the call.

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