What Is Voice over IP – Part 2

Other VoIP related problems and solutions are:

• Some routers can be configured to distinguish from other VoIP packets, and therefore assign them a higher priority.
• You can store in buffer packets, to make more asynchronous transmission, but this can result in an increase in latency, similar to satellite broadcasts.
• The system operator should provide a bandwidth wide enough to reduce the latency and data loss. However, that, while it is relatively easy in private networks, is much more difficult when using the Internet as a medium.
• With lower speeds of 256 kbit / s jitter problems can occur (literally nervousness, technical error in the time base when a digital sample is converted to analog). Indeed, on slow networks, the transmission delay becomes significant, and therefore the VoIP protocols use smaller packets of maximum size (usually 1500 bytes). Other protocols typically based on TCP, normally used instead of maximum size packets. If a packet arrives on a VoIP connection to the switch while it is transmitting a packet of maximum size belonging to another connection, will be delayed important, and certainly not constant.
• Privacy issues: the listing of call or conversation may be recorded by your VoIP, rather than attacks by third man in the middle. In particular, if the program is closed source and proprietary, it is more difficult to establish that there is some code spyware installed on your computer directly to the sender and receiver to record the calls, similar risk occurs if the communication is directly between sender and recipient, and all packets passing through a central server maintained by your VoIP.
• The digitization of the signal VoIP allows you to implement encryption and other forms of data protection in transit on the network. The most common call to PSTN line travels in the clear, at least two of deliveries, with one node and another starting with the receiver (the common telephone may not have installed a protocol for data protection or communication by analog signal could support it).
• Another guard is the fact that voice packets are routed on different nodes, which vary over time, whereas in communication with the analog signal the entire contents of the call follows the same path and the same goes for network equipment.

VoIP protocols

VoIP requires two types of communication protocols in parallel, one for the transport of data (voice packets over IP), and one for the “alert” of the conversation (reconstruction of audio frames, synchronization, caller ID, etc.). For data transport, the vast majority of VoIP deployments, is adopted RTP (Real-time Transport Protocol). For the second type of protocols needed for Internet telephony, the standardization process has not yet concluded.

Currently, three entities are involved in international standardization: ITU (International Telecommunications Union), the IETF (Internet Engineering Task Force) and ETSO (European Telecommunication Standards Institute) with some associations (e.g., Softswitch, H.323ORG, Vivid etc.) The management of voice calls over the IP network is at present directed towards two different proposals, developed within the ITU and IETF, which are respectively H.323 and SIP (Session Initiation Protocol).

Supporters of the proposal argue that the ITU H.323, historically came first, has now obtained the support of all providers of VoIP equipment, while supporters of SIP doubt H.323 interoperability of products from different manufacturers and, at same time, demonstrate the advantages of SIP in particular with regard to reduced signaling during activation of the call. Wanting to compare H.323 and SIP must however be noted that the purpose of the two standards is quite different. SIP was created as a protocol for real-time voice communication over IP and presides over all the basic functions of a call control: introduction and Logging operations, signaling, dial tone, call waiting, transfer, and identification of calling and so on.

While SIP is a protocol for reporting and control of multimedia sessions, H.323 defines a complete architecture for the development of multimedia conferencing, including the definition of application-level encoding formats, the definition of protocols for signaling and control the transport of audio, video and data and to manage security aspects, all with reference to local network architectures.

The H.323 protocol suite was initially the only standards adopted by device manufacturers for IP telephony and multimedia applications in general, and is supported by both PC applications from network devices (routers) and user terminals (IP phones). The proposed SIP, however, is finding increasing favor, thanks to its excellent integration with other protocols of TCP / IP suite (whereas H.323 is designed for a generic network packet) and its simplicity. H.323, in fact, born in the area where phone specifications are extremely accurate and complete but also very complicated, also includes, within it, other components previously defined by the ITU, greatly increasing the complexity.

Despite the spread of the H.323 standard, the market for devices for end users, software and gateways to connect to the PSTN is now oriented towards solutions based on SIP. Almost all products on the market are compatible with both standards and numerous consortia and standards bodies, including 3GPP for UMTS, SIP included in their specifications.

From the perspective of end users, and then market solutions such as VoIP phones, adapters, cordless phones and dual mode (VoIP and traditional) but the producers seem to move towards a “standard” is not technically designed to be this: Skype. With 100 million users the company by Niklas Zennstrom and Janus Friis has managed to take advantage of the first-comer and, thanks to its closed network (Skype users can only call other Skype users on the PSTN or mobile) and the signaling protocol for the owner using its software, was able to expand its activities to the management of partnerships with hardware manufacturers and software developers.

Other protocols used to encode the signal of the conversation (reconstruction of audio frames, synchronization, etc.) are:

• Skinny Client Control Protocol, proprietary protocol of Cisco
• Megaco (aka H.248) and MGCP
• MINET, Mitel’s proprietary protocol
• Inter Asterisk
• Xchange (supplanted by IAX2) used by the Asterisk open source PBX server and client software for
• XMPP used by Google Talk. Initially designed for the IM now extended to functions with VoIP module Jingle

Continued…

Related Posts:

• What Is Voice over IP | VoIP | Part 3
• What Is Voice over IP | VoIP | Part 1