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Posts Tagged ‘VoIP telephony’

VoIP – Open Issues and Future Work

August 31st, 2009 admin No comments

The main open issue is the inability to properly handle Skype, to date probably the most widespread VoIP protocol. As explained before, this is due to the lack of documentation about the protocol, and the use of payload encryption. This is not only a limitation of ntop and nProbe but of any other VoIP analysis tool.

VoIP support is relatively new into ntop/nProbe hence several extensions can be added to their implementation. The current measurements focus mainly on high-level metrics such as jitter or packet loss, and are independent of the codecs being used. However as new codecs such as H.264 [h264] are becoming increasingly popular, a planned enhancement is the ability to decode some of these codec formats in order to also provide precise information about the RTP payload (e.g. voice quality), as well provide support for RTP XS reports.

The implementation of these voice analysis metrics has been delayed with respect to the original plan, as they are described in ITU documents (e.g. ITU E.411 recommendation) that are not freely available on the Internet, that is usually a problem for the open source community. Nevertheless in the next release two new common metrics such as MOS score and r-factor will be implemented thanks to bits and pieces found googling on the Internet.

Source: Luca Deri (ntop.org)

Handling VoIP Service Requirements – Minimizing Latency

August 10th, 2009 admin No comments

As with most real-time services, VoIP demands that the network provide predictable performance within a constrained boundary of transport parameters. This section surveys the key networking issues that an organization or service provider must carefully consider when deploying a VoIP solution.

Latency (also referred to as delay) is the time that it takes a packet to make its way through the network to the terminating device. In other words, latency is the time it takes the speaker’s voice to reach the listener’s ear. While large latency values do not necessarily degrade the sound quality of a phone call, they can disrupt the rhythm of conversation, making it difficult to interact.

Several factors contribute to latency in a multiservice network, including:

• The time it takes for the endpoints to create the packets used in voice services, known as packet creation latency
• The time it takes to serialize the digital data onto the physical links of the interconnecting equipment Read more…