VoIP 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.
Contact Inflow – we want to be your Unified Communications and VOIP business phone system partner!
Your Strategic Advisor for Unified Communications & Contact Center Success
Inflow elevates your customer experience and makes you a disruptive force in your industry through your unified communications and contact center technologies.
Supporting Over 800 Customers Around the World
- SIP Virtualization and Why the Right Session Border Controller is Important
- What To Expect When Migrating to Mitel MiVoice Connect
- Inflow Continues To Redefine Customer Experience By Partnering With Teleopti
- The State of Unified Communications As A Service in 2019
- Using the Mitel Emergency Notification Application