Delay and Jitter

Delay and Jitter:  If it takes too long for the voice to be sampled, packetized, sent across the network, de-packetized, and replayed to the receiving end, the natural characteristics of a conversation can be affected – resulting in confusion and frustration by both parties.  Delay can be caused by the distance and speed of the network.  For toll quality voice, delay cannot exceed 100 ms and 150 ms for acceptable voice.

Jitter is the variation of delay across the network.  The uneven pattern of latency across the network is detected as poor quality voice.  Jitter can be caused by network congestion, queuing methods used in network equipment, and routing options used in Wide Area Networks.  The IP phone system should be able to continuously measure jitter and compensate by changing the size of the send and receive jitter buffers.

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Voice over IP – A Simple Explanation

We’ll use the example of a VoIP business phone system. With a VoIP phone system, the phones are essentially computing devices that are plugged into your local network. Some companies plug these “IP Phones” into a separate network specifically used for VoIP; others connect the IP phones directly to the network that the rest of their computers are connected to.
When a user talks into the microphone of an IP phone, their voice gets convert to electrical signals called sine waves. These sine waves then get sampled. The sampling process essentially converts the sine wave to a binary form (1’s and 0’s) that represents the characteristics of the original voice (the highs, lows, frequency, etc). Then these 1’s and 0’s get “encapsulated” or put into Internet Protocol (IP) packets. IP is the industry standard for sending ALL forms of data across networks (web, email, video, voice, etc). These IP packets then traverse the network to the end destination. The end destination could be an IP phone sitting right next to the original phone, an IP phones on the other side of the world, or perhaps a phone line that connects the phone system to the outside world. Once the IP packets carrying this voice information reaches its destination, the 1’s and 0’s get converted back to the analog sine wave, the analog sine wave get’s amplified and then ran through a speaker in the receiving phone that recreates the audio speech of the original speaker.

Unified communications food for thought: Legacy phone system integration

We often have larger, multi-site customers who have the following challenges:

1. They realize their antiquated phone system is / has become a liability. Failure is probable and securing replacement parts and service is becoming more difficult.
2. They want to benefit from the latest in Unified Communications tools (mobility, presence management, single system image with web-based admin, etc).
3. They don’t have the budget to migrate their technology.
4. They don’t want to invest any more money in legacy technology (expansion, replacement, maintenance contracts, etc).

Ways a migration plan can meet these challenges head on:

1. Integrate a ShoreTel VOIP system with the legacy phone system using VOIP to TDM (or digital) gateway technology to facilitate dial tone routing and station-to-station dialing between systems.  This also allows you to replace a subset of your critical handsets with IP phones at one location.

Replace the existing legacy voicemail system with a ShoreTel business phone system.  Using protocols like AMIS, SMDI, and QSIG – provide traditional integration services to the legacy digital sets (message waiting indicator).  Even better, provide these “legacy” users the ability to take advantage of other ShoreTel Unified Communications features like find-me / follow-me, voicemail-to-email, presence management, etc.