Apparatus, method and system for improved quality of voice calls over a packet based network

Publication: EP2120416A1
Published: 2009-11-18
Family Size: 1
Granted: No

Simple SummaryContent extracted from patent full text and abstract with AI.

The invention provides an apparatus, method, and system for maintaining and improving the quality of voice calls over packet-based networks (like VoIP) when network conditions change. It enables seamless switching between different voice codecs during an active call by using two codecs in parallel, each with its own buffer, managed by a coordination unit. This setup reduces audible interruptions, keeps conversations smooth, and handles network changes without noticeable quality loss.

Use CasesContent extracted from patent full text and abstract with AI.

  • Voice-over-IP (VoIP) telephony systems for consumer and enterprise communications.
  • Mobile VoIP applications that encounter varying network conditions (e.g., 4G to Wi-Fi handover).
  • Unified communication systems in businesses needing high-quality, uninterrupted voice calls.
  • Call centers or customer service systems requiring reliable voice quality.
  • Conference call services that need to adapt to participants joining via diverse networks and devices.
  • Emergency or mission-critical communication systems where uninterrupted calls are essential.

BenefitsContent extracted from patent full text and abstract with AI.

  • Provides seamless and uninterrupted voice quality during codec changes, minimizing audio artifacts and dropped speech.
  • Enables on-the-fly adaptation to changing network conditions (e.g., bandwidth fluctuation, handover between networks).
  • Improves user experience by masking disruptions that normally occur during codec switching.
  • No dependency on intermediate network devices; negotiation and adaptation occur directly between call endpoints.
  • Supports compatibility and interoperability with multiple codecs, enhancing system flexibility.
  • Reduces risk of lost or out-of-order voice packets by coordinated buffer management.
  • Works with standard protocols (like SIP/SDP), supporting ease of integration into existing systems.

Technical Classifications (CPCs)

Main Classifications

Electrical & Electronic Tech

Sub Classifications

Electric Communication Technique

CPC Codes

H04L65/1083H04L65/752H04L65/80

Inventors & Applicants

Applicants

Deutsche Telekom Ag

Univ Berlin Tech

Patent Abstract

The invention relates to an apparatus and a method for providing data communication over a packet based network comprising a network interface (450) which is connectable to the packet based network. The apparatus comprises at least two codec means (430, 470) being arranged in parallel and being connected to said network interface (450). A first codec means (470) comprises a first buffer (460) and a second codec means (430) comprises a second buffer (440). A management unit (420) is connected to both codec means (430, 470) and a sound device (410). When a changeover signal is issued, said first codec means (470) continuous to decode an input data packet being passed through by said first buffer (460), and said second codec means (430) initiates an encoding process of an output data packet being received from the sound device (410) and directly transmits the encoded output data packet to the network interface (450).

Key Information

Publication No.

EP2120416A1

Family ID

39865219

Publication Date

2009-11-18

Application No.

EP08156401A

Application Date

2008-05-16

Priority Date

2008-05-16

Granted

No

Possible Cooperation

For further information please contact the transfer office.