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)

- Nigeria and Mauritius have legalised VoIP
- Algeria contemplating the same approach
- “Back door deals” arranged between SOTELMA (Mali) and four VoIP operators
- Kenya examining ways to have VoIP services through cybercafés and ISPs

VoIP is a tool that African governments and regulators could use in a number of ways to address a wide range of pressing issues and it is therefore something that needs to be understood rather than feared.

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
• The time it takes an electrical (or photonic) signal to travel the length of a conductor, known as propagation delay
• The time that a packet remains buffered in a network element while it awaits transmission, referred to as the queuing delay
• The time it takes a network device (router, switch, firewall, etc.) to buffer a packet and make the forwarding decision, known as packet forwarding delay.

When designing a multiservice network, the total delay that a signal or packet exhibits is the sum of all the latency contributors. Generally, it is accepted that the end-to-end latency should be less than 150 ms for toll quality phone calls. The remainder of this section describes some basic steps network managers can take to assess and mitigate the impact of each latency contributor in a multiservice network.

Source: Juniper Networks, Inc. White Paper

Lower cost and greater flexibility are among the promises of VOIP for the enterprise, but VOIP should not be installed without careful consideration of the security problems introduced. Administrators may mistakenly assume that since digitized voice travels in packets, they can simply plug VOIP components into their already-secured networks and remain secure. However, the process is not that simple. This publication explains the challenges of VOIP security for agency and commercial users of VOIP, and outlines steps needed to help secure an organization’s VOIP network. VOIP security considerations for the public switched telephone network (PSTN) are largely outside the scope of this document.

VOIP systems take a wide variety of forms, including traditional telephone handsets, conferencing units, and mobile units. In addition to end-user equipment, VOIP systems include a variety of other components, including call processors/call managers, gateways, routers, firewalls, and protocols. Most of these components have counterparts used in data networks, but the performance demands of VOIP mean that ordinary network software and hardware must be supplemented with special VOIP components.

Not only does VOIP require higher performance than most data systems, critical services, such as Emergency 911 must be accommodated. One of the main sources of confusion for those new to VOIP is the (natural) assumption that because digitized voice travels in packets just like other data, existing network architectures and tools can be used without change. However, VOIP adds a number of complications to existing network technology, and these problems are magnified by security considerations.

Source: National Institute of Standards and Technology