Monday 1 August 2011


Introduction to Voice over Internet Protocol (VoIP)


Voice over Internet Protocol (VoIP) is a technology that allows voice (and possibly video) communication over IP networks. These IP networks could be public, such as Internet, or private such as Enterprise networks, or a combination of both. This technology offers some advantages over the legacy telephone networks; however it also has some disadvantages. Engineers around the world are working hard to solve, and fortunately they are making lots of progress.  It is expected that legacy telephone systems shall disappear over the years.

In order to understand the advantages of VoIP systems over the legacy telephone networks, let us assume a scenario of a company that has an office with few hundred employees. Each employee has a LAN connection (usually Ethernet RJ45) as well as a telephone connection (usually RJ11). The IT room has all these LAN cables connected to a switch and the telephone cables connected to a PBX system. The PBX is also connected to external telephone lines (Known as FXO) providing public service telephone network (PSTN) service via a local carrier.

The PBX could have N lines for M users to share. Examples 60 lines for 300 users. This is based on the fact that not all employees are communicating to external numbers simultaneously and they probably need to communicate with each other more than externally. For an
employee to make a call he dials “9,desintation phone number”.

The main disadvantage of this setup, it contains extra cabling to each user. If you are in IT, you know that is an extra headache to lay and maintain these extra cables across the building. Also to add a new batch of users to the PBX extra hardware upgrade may be required (known as FXS cards).

If the same company has one or more extra branches in a remote location or even overseas, the local carrier shall charge the company long distance or overseas fees and the company cannot get around that unless they rent their own circuit from branch A to B.


In the VoIP world, the phone at the user-end is one of 3 types:
  • IP Phone
  • ATA device
  • Soft phone

IP phone is a piece of hardware, it looks like normal phones with the exception it connects via RJ45. Internally these phone sets have a different design than traditional phone set. I will illustrate the internals of these phones in my next article in a bit of details. Examples of these phones are:
  • Cisco
  • Aastra
  • Polycom
  • Grandstream


An ATA device is a device that can be connected to any legacy telephone set and it converts the output into VoIP and has a RJ45 jack on it. An example of ATA devices is Cisco PAP2T-NA, Internet Phone Adapter with 2 VoIP Ports

Softphones are piece of software that runs on your desktop or laptop and uses the microphone, speaker as well as the Ethernet card on your PC to emulate a phone set that hooks. An example of softphone is Xlite.

IP-PBX

In the IT room, there is a IP-PBX. IP-PBX Is the backend system that facliates phone calls between extensions or to the outside world. It also handles incoming calls, voicemail, call transfer, conferencing, Interactive Voice Response, Automatic All Distribution (ACD), Busy Light Field (BLF) …etc.

If the PSTN FXO lines are coming from the local carrier in the traditional telephone service form, a hardware system is required to convert the signal on the FXO lines to their VoIP equivalent.

One of the most famous open-source IP-PBX is Asterisk. It is free software that works on general purpose Intel-based PC or servers. There’s also a packaged flavor of Asterisk that has a Web User Interface for easy administration, known as Trixbox. Asterisk and Trixbox offer all small-medium enterprise PBX features: Extension-2-Extension, Call Transfer, Call Conference, Dial-plans, Voicemail, SIP Trunks …etc. They work with a number of hardware cards to connect to the telephone service providers (FXO lines). Cards are made by Digium (the creator of Asterisk), Sangoma, and probably many others. If the lines are coming in a bundles such as T1`s links (24 telephone lines) or E1`s (30 Telephone lines), a more sophisticated hardware is required, known as a Media Gateway. To learn more about how to setup Asterisk free PBX, refer to this book: Asterisk: The Definitive Guide

A Media gateway can do also other functions in a VoIP network. It can multiplex audio and/or video streams, allowing conference calls. The job of multiplexing audio / video streams in real-time requires mathematical computational power, may exceed the capabilities of normal general purpose hardware servers. That is especially true when the number of conference participants grows high. Media gateways rely on different technology of processors, known as digital signal processors (DSP) that has the computational power to perform such a job.

Networking Protocols

VoIP, nowadays, relies on bunch of networking protocols combined together to provide the voice communication service.

VoIP Advantages

  • Low rates for long distance and international calls
  • You can make and receive phone calls wherever there is a broadband connection
  • Free Caller ID, voicemail, call forward, conferencing are some of the many services included with most VoIP service providers.
  • Less Cabling for Enterprises
  • Most new IP-Phones can run XML Applications. Enterprises can use that to provide services to their employees.

VoIP Challenges

  • Emergency calls: When an emergency call is made from a legacy PSTN, the location of the call is associated with the callers due to the nature of the fixed land line. Because VoIP is portable to any place in the world, there is no guarantte that a person calling emergency is in the actual location he is claiming or registered at the service provider records.
  • Sound Quality And Reliability: VoIP may suffer from low sound quality and low reliability when public Internet is involved in delivering the voice traffic. High delays, jitter and packet loss due to congestion in the network will cause noticeable degradation the service quality.
  • Security: Since VoIP uses IP networks, it inherits all the security weakness of an IP network.
  • Power Failures: Legacy PSTN switches have batteries to overcome power outage for few hours, so that the service continues uninterrupted during power outages. Since VoIP calls relies on many systems: IP-Phone or ATA adapters, routers, switches, modems, servers more systems have to be power failure resilient. If all these systems are under one administration, it could be achievable, but that’s not always the case.

2 comments:

  1. I have an interest in the silence suppression topic. For work I have a Bell line that suppresses the line when I'm on conference calls so I loose the comments from people who aren't sitting near the conference phone. Bell says there's nothing they can do about their noise suppression algorithm.

    There should be a way to set the noise floor but that would probably also force vendors to provide more bandwidth for VOIP calls.

    ReplyDelete
  2. Hi DoTel, the silence suppression feature to save bandwidth works along with the voice activity detection. VAD turns on silence suppression when voice falls below certain level (threshold). Although most ip-phone vendors may not expose the VAD parameters to be configured by the user, most IP Phones allows the user to configure silence suppression on/off. This shows up during SIP call negotiation, in the SDP body of the messages, in order to inform the remote system whether your phone supports silence suppression or not:
    ...
    a=silenceSupp:off
    ...

    For example: Check this Aastra SIP Phone/Asterisk configuration example: search for “silsupp=off“ in this page http://www.voip-info.org/wiki/view/Asterisk+and+Aastra+Phones

    Note that in this particular version of the phone firmware, it was not a direct on/off switch on the phone admin web interface, it was actually a full text "payload=18;ptime=10;silsupp=off" the user need to write inside the "Customized Codec Preference List" text box. I don't now what phone/ATA adapter are you using, but you may need to scan the manual to find out how to turn on/off silence suppression.

    To overcome your problem:

    You can try to disable silence suppression on your own ipphone/ATA/PBX, see if the remote systems respect that during call setup (they should).

    ReplyDelete