VoIP has emerged rapidly in medium and big organizations as an inexpensive and efficient communication service within the organization and in handling communications outside of the organization in an efficient way.
Protocols are responsible for the activities of IP telephony signaling.
An essential part in enabling interoperability between the circuit-switched and packet-switched networks is the "VoIP gateway". The gateway provides fundamental functions for operating VoIP: interface to the IP network and the PSTN network, call processing functions, voice compression and decompression, packing and unpacking the compressed voice to/from packets. The gateway setups the call, takes telephone signals, digitizes them, compresses them, and organizes them in packets. Then these packets are routed using the Internet Protocol (IP). Meanwhile, a gateway at the other end reverses the whole procedure for packets coming in from the network and going into a telephone device.
As described in (LINK) VoIP History, implementing VoIP incorporates some problems: delays, out of order packets, lost packets and more. These problems have a destructive effect on the communication quality.
Some tools were developed to overcome the VoIP built in problems and sustain an acceptable communication quality. To reduce to minimum VoIP packets delay and loss due to network congestion, routers software was updated: IP packet precedence was developed, giving VoIP traffic the highest precedence. Another addition is Weighted Fair Queuing (WFQ): a buffering mechanism that buffers IP packets according to a number of criteria.
The gateways have an important role in the struggle for VoIP performance. Gateways manage buffers to cope with "Jitter": packets from the same conversation can arrive out of order because of different delays they suffer on the way. Those delays can cause an unnatural-sounding voice. The Jitter buffers collect packets and keep acceptable continuity by giving subsequent packets time to arrive and still fit into a natural voice flow. Gateways can also fill in short periods of missing packets (up to 200 ms) using buffers.
Another important feature handled in the gateways is Voice compression. Using voice compression methods significantly improves the utilization of the bandwidth without compromising voice quality. LINK à Voice Compression methods
The gateway must execute the various actions (decode, buffering and more) in an acceptable delay period to produce an intelligible connection, otherwise a considerable degradation in the quality of the sound will occur, together with delays and jitter.
A problem that may occur due to wire separation (two to four-wire in the last mile) is Echo- a portion of the incoming signal is fed back to the sender. Echo cancellation is implemented to overcome this problem.
As mentioned before, VoIP saves bandwidth also by sending only the conversation data and not sending the silence periods. This is a considerable saving because generally only one person talks at a time while the other is listening. By removing the VoIP packets containing silence from the overall VoIP traffic we can reach up to 50% saving!
Nevertheless, to obtain an intelligible conversation the gateway fills in the silence periods to reproduce the exact conversation.
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