In terms of VoIP, bandwidth represents a measure of the number of bits per second that can flow through an IP channel at a given time. The term “available bandwidth” applies to the amount of bandwidth that is left for applications like VoIP after the networks main applications are up and running and all other needed services are accounted for.
The CODEC (CODER/DECODER) is a standard through which voice information can be encoded into data or decoded back to voice information. Both a Coder and Decoder are necessary on both sides of the telephone call since telephone calls occur simultaneously in both directions. Bandwidth is an extremely important factor in QOS (Quality of Service). MOS (Mean Opinion Score) is an attempt to make a quantifiable benchmark of voice quality. Below are examples of the CODEC, bit rate and mean opinion score:
CODEC Kbps (kilobits per second) MOS
G.711 (toll quality) 64 4.1
G.726 16 (32) 3.8
G.729AB (cell phone quality) 8 3.7
Digital Signal Processors (DSP) standardize the different states of a digital signal into an organized and understandable signal. DSP circuits can differentiate between digital signals and digital noise. Signal-to-Noise ratio is one of the most important factors in telephony voice communication because if there is too much noise in a signal, the DSP will be unable to find a signal standard and lose it. DSP circuits always adjust digital signal levels so the can maintain a standard without noise.
|FoIP (Fax Over Internet Protocol)|
FoIP (Fax Over Internet Protocol)
FoIP is the transmission of fax messages over IP networks. FoIP works
flawlessly and is latency tolerant even on poor to slow IP connections. A
company that deploys FoIP solutions can benefit from significant cost and
H.323 is a call control protocol that defines the standards for the use of VoIP over WAN’s, LAN’s, multimedia and network equipment, as well as digital communication systems. H.323 has the largest installed base of any call-control protocol because it was the first developed. It was initially designed to ensure that multimedia applications could run over unreliable networks. H.323 covers and works with many types of applications, voice being only one of these applications.
Centrex is a telephony service, primarily for voice, where the equipment that provides the service logic functions is owned and operated by another party. Traditional Centrex service has been offered by the local phone company, IP Centrex it usually supplied by a service provider. Both versions offer services where the capital expenditures for the equipment and ongoing support and services are borne by the provider and the customer only pays an ongoing monthly operating expense for the service and usage.
Media Gateway Control Protocol is a one of the standard protocols used in VoIP and has its strongest niche in IP Centrex environments. MGCP is a media gateway control protocol as defined by the standards organizations IETF and ITU-T, for use in distributed switching environments. Simply put, this means that the protocol is used for handling the communications between the media gateway (which converts data from the format required for a switched circuit network) to the media gateway controller (that is required for a packet-switched network).
In terms of VoIP, the hardware or software driven action referred to as PAD (Packet Assembly / Disassembly) takes voice (analog signals of varying amplitude and frequency over time) and converts it into raw data bits of 0’s and 1’s. These raw data bits are then framed and “Packetized” or encapsulated into network transferable units (so it can travel through IP networks) using the Internet Protocol. Voice packets follow the TCP/IP OSI (open systems interface) model as shown below:
Transport Layer: RTP (Real Time Transport)
Network Layer: IP (packet segmentation)
Data Layer: Raw data is framed
Physical Layer: Analog signals (voltages) are converted into raw data
In the making of a standard Circuit Switched call, analog signals are converted into digital data through a process called Pulse Code Modulation (PCM). PCM data flows in a continuous data stream of 64K, the standard bandwidth for toll quality voice calls. Generally, in a standard PSTN (public switched telephone network) setup, voice is converted from the analog signals from a phone-user’s voice and coded into streams of PCM data bits over copper or fiber wires. PCM is generally what it used over T1 and ISDN voice connections. The difference of PCM from VoIP is that the raw voice data that makes up the PCM data bits are Central Office guided (one pathway from phone to phone) while VoIP raw data (speech) is packetized and addressed with destination information then it’s thrown into the busy Internet highway with as many pathways as possible to get to the addressed destination.
RTP (Real Time Protocol) is specifically concerned with the dependable transmission of latency-sensitive traffic across the network and is involved in using time stamping to determine network jitter tolerance and makes sure that voice packets are arriving in order.
SIP (Session Initiation Protocol) is a signaling protocol for Internet conferencing, telephony, presence, events notification and instant messaging. SIP is a text-based protocol, similar to HTTP and SMTP, for initiating interactive communication sessions between users.
Skinny is a protocol that was developed by Cisco specifically for use with Call Manager.
TCP (Transport Control Protocol) is by its nature a more reliable protocol the UDP, however it’s much slower and less efficient in passing data across an IP network. This is because the TCP protocol acts similarly to a “Supervised Transfer” during a voice call; asking the destination site whether or not the info it’s passing was meant for it where as UDP acts like a “Blind Transfer” in terms of sending packetized voice data it; passing the information as fast as possible, and not waiting for any confirmation.
UDP (User Datagram Protocol) is a connectionless transport protocol that is part of a suite of protocols (the others being RTP and IP) that allow for the timely and efficient transfer of voice data across and IP network. UDP is the better transport protocol for VoIP data than TCP.
VoIP is based on the principal of transmitting digitized voice packets over networks.
Basically, VoIP consists of converting voice signals into streams of digital packets and sending those packets of data through an IP-constructed network environment. VoIP can work in both LAN (local area network) and WAN (wide area network) environments for intranetwork or internetwork communication between VoIP channel users. Routers and switches and other special compression protocols direct the packetized voice data to their destination IP address.