Celideo Limited Inject confidence into your network

By using redundant audio as defined by RFC 2198 it is possible to reduce the impact that lost packets have on VoIP audio quality. The idea is to send a primary encoding of normal quality along with a secondary encoding of lower quality within the same stream, but temporally delayed. An example might be to have G.711 as the primary encoding and G.723 as the secondary encoding.

The transient nature of network problems and complex network configuration contribute to make the monitoring VoIP quality across networks a daunting task. Jitter, packet loss and delay all have a significant impact on voice and video quality and the ability to automatically monitor these metrics in real-time and to receive alerts can prove invaluable in detection and response.

Network Probe

VoIP QoS is affected by several types of network errors and characteristics. Focusing on the RTP QoS, or voice quality, these result in either and increase in end-to-end delay or a loss of audio clarity.

Delay

Recently we've been asked about lip-sync analysis a problem that is becoming common in Video over IP applications. The audio and video are normally sent as individual RTP streams, combined with differing encode/decode times, dejitter buffers size and potentially different origins this quickly gives rise to different user arrival times that manifest as lip-sync problems. Measuring the degree of lip-sync, in frames or milliseconds, is possible in real-time using various characteristics of RTP and RTCP.