Here's some ideas on how to manage VOIP for administrators.
As with most new technologies, Voice over Internet Protocol (VoIP) brings new challenges along with the benefits. The main challenge is VoIP’s extreme sensitivity to delay and packet loss compared with other network applications such as web and e-mail services. A basic understanding of VoIP traffic and of the quality metrics provided by VoIP monitoring tools will help you keep your VoIP network running smoothly.
How does VoIP work?
VoIP phones use codecs to translate analog sound streams into digital packets for transmission. On the receiving end, the codec translates the packets back to analog. For two people to converse normally, all of this must happen in as close to real time as possible.
For call setup, most enterprise VoIP solutions include one or more call managers, which are servers that set up calls between VoIP phones, and can also provide gateway connections to the Public Switched Telephone Network (PSTN) for calls outside the VoIP network. Typically, the call initiator contacts the call manager, which then rings the phone being called. Once the receiving party answers, the call manager provides a mechanism for the phones to negotiate codecs and connection parameters.
The connection itself is typically in the form of two full-duplex streams: a Real-time Transport Protocol (RTP) stream that carries the encoded audio, and an RTCP (Real-time Transport Control Protocol) stream to provide communications control. Once the call is set up, the call manager is no longer involved until the teardown phase, when the IP phones inform the call manager the call has been completed so the centralized call queue (a list of what phones are active) can be updated.
What can go wrong with VoIP?
Depending on what components of the network are compromised, users can experience audio quality problems with a call, or they might not be able to connect in the first place if a call manager is affected.
Audio quality problems
Problems with VoIP audio quality are almost always related to network delay, jitter, and packet loss, or some combination of the three. It is common to see them together because they are both related to a general deterioration of network conditions.
In any VoIP deployment, some delay is unavoidable. Codecs take time to encode/decode the audio stream, and even the fastest network medium is not instantaneous.
In addition to network delay, the VoIP equipment itself (IP phones, gateways, etc.) subtracts even more processing time from the overall delay budget. The delay budget for reasonable two-way conversations in real time is about 150 milliseconds (one way). When delay exceeds the budget, the callers can get confused about who should be speaking and who should be listening, and begin to talk over and past one another.
Network jitter and delay
Real-time voice communications are sensitive to delay and variation in packet arrival times. Codecs require a steady, dependable stream of packets to provide reasonable playback quality. Packets arriving too early, too late, or out of sequence result in jerky, jumbled playback. This phenomenon is called jitter.
Because no network can guarantee a perfectly steady stream of packets under real-world conditions, VoIP phones use jitter buffers to smooth out the kinks. A jitter buffer is simply a First-In, First Out (FIFO) memory cache that collects the packets as they arrive, forwarding them to the codec evenly spaced and in proper sequence for accurate playback.
While a jitter buffer can successfully mask mild delay and jitter problems, severe jitter can overwhelm the jitter buffer, which results in packet loss (see below). Increasing the size of the jitter buffer can help, but only to a point: A jitter buffer that increases overall round-trip delay to 300 ms will make normal conversation difficult.
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