VoIP monitoring explained

Although voice over Internet protocol is key to the next generation communication technology, it might have varied types of problems. These problems need to be solved, and Ozeki Phone System solves it for you, but it never hurts knowing basic features of a protocol we are using or planning to use.

Voice communication over Internet Protocol networks and transmitting multimedia sessions are done by Voice over Internet Protocol or VoIP. VoIP phones can digitalise analog signals into data packets to be sent over the Internet, but other phones do it with the help of an ATA (Analog Telephone Adapter, which is used to convert analog and digital data).

When the packet reaches its destination it is decoded back into a voice signal. If the packets are compressed for faster transmission then they will be called codecs in their compressed format. These codecs vary in bandwidth requirements, voice quality, bitrate and ways of compression.

VoIP solutions are often used at corporate level as a cost effective solution to telephone communications, whereas proprietary VoIP applications are used for letting people talk either computer-to-computer or computer-to-telephone using a PC equipped with a special application and a headset.

Slight problems in connectivity can easily interfere with the VoIP related network traffic. Sound quality is the most sensitive to these problems: when the packets do not arrive in time, or they get lost, the sound quality will dramatically drop.

VoIP monitoring is used to analyze quality of VoIP calls. It is based on delay variation and packet loss. It has two types: proprietary and standard voip protocol monitoring. This article will explain how VoIP traffic is monitored in two different ways: Proprietary VoIP traffic Monitoring and Standard VoIP Traffic monitoring (Figure 1).

voip monitoring explained
Figure 1 - VoIP Monitoring explained

1. Proprietary VoIP Traffic Monitoring Skype typically falls into this category. It is possible to detect not only Skype in general, but also the conversation type (skype2skype or skype-in/out call). The main differences between a P2P and Skype conversation are:

  • Traffic is bidirectional in a Skype conversation and the frequency of the packets is high and the size of the packets is limited.
  • A P2P session, however, is mostly unidirectional, packet rate is varied and size of the packets is larger.

2. Standard VoIP Traffic Monitoring VoIP traffic analysis has two parts:

  • Analyzing signaling protocols
  • Analyzing voice traffic.

You can either get a traffic analysis overview that is simple to use and understand, or precise traffic metrics that is probably understood by professionals only. Those metrics satisfy basic traffic measurements, like:

  • SIP (Session Initiation Protocol is used to establish, maintain and terminate sessions over the Internet)
  • Time of call events, like the beginning and and end of the call, to identify performance issues on the SIP gateway.
  • RTP (Real-Time Transport Protocol specifies the packet format required for multimedia transmission over the Internet)
  • RTP ports where the call will take place to associate a signaling flow with the phone call.
  • Time-stamp and source identifiers of the first and last RTP flow packet.
  • RTP payload type identifier.
  • Called and caller party.
  • Unique call identifier for tracking problems.
  • Codecs to identify voice quality issues.
  • Number of lost packets and maximum packet time delta.
  • Number of jitters in both directions.

All of the above mentioned features can be monitored through Ozeki Phone System. To be more exact, it monitors all these, and the results are given to the customer at request because Ozeki Phone System is one of the most flexible, advanced and user-friendly next generation technologies.

Read the following pages for further information:

More information