VoIP Australia - FAQ Voice Over Internet Protocol

 
VoIP Information
  Home / Technical Questions - Australian VoIP Providers
   

VoIP stands for "Voice over Internet Protocol", and like "Internet Telephony" or "IP Telephony" , is used as a general term to describe a range of technologies which allow telephone-grade voice conversations over the Internet. IP telephony is the technology of transmitting voice and fax over data networks using the Internet Protocol (IP).

The technology to convert analogue signals to digital and back has been around for decades, and is the same as used to play music and movies on your PC. If your PC has a microphone, you can digitise your speech and record it - or send it digitally to another PC which can play it through attached speakers. About 10 years ago several visionaries realised that the Internet could be used to carry the digitised packets of voice data, but the technology of the time had a couple of limitations due to the dial-in modems in common use (56Kbps modem speed, and occasional Internet connections). Today’s broadband is 'always-on' (so callers can always ring you) and broadband's bandwidth, combined with improvements in codecs, provide several simultaneous toll-quality voice conversations with a minimum of interruptions and drop-outs.

VoIP technology offers the same voice quality which customers expect from Australian telcos; but utilises the public Internet as the transmission medium. It doesn't matter whether you call your neighbour or the other side of the world; each call is simply a bit more data transferred through your existing Internet account, and so calls to other VoIP subscribers are effectively free for most broadband users.

Long distance phone card companies and even telcos have been quick to take advantage of the new capabilities and enormous cost savings resulting from more efficient use of the Internet and their private networks. This in turn creates the potential for new opportunities and services.

How much data of my ISP's data allowance will be used by VoIP calls?

This depends on the codec used (negotiated separately for each individual call), but

  • G.711 = 64Kb/sec + overhead * 60 /8 = 480K bytes/min, or approximately 1MB every 2 minutes
  • G.729 = 8Kb/sec + overhead * 60 /8 = 94.5K bytes/min, or approximately 1MB every 10 minutes

What is SIP and RTP, and how do they work?

SIP uses two protocols –

The Session Initiation Protocol (SIP) is an industry-standard application-layer control protocol that can establish, modify and terminate multimedia sessions or calls. Examples of multimedia sessions include multimedia conferences, distance learning, and Internet telephony. SIP is used for the control messages – register with SIP server, initiate a phone call, negotiate the codec to use, etc. - and consists of text messages in a predefined but flexible format. SIP is fully specified in RFC 2543.

With SIP, you join a SIP Registrar and configure your phone with the Registrars name and your username. Your SIP phone then keeps the SIP Registrar informed of your presence (whether you are online, and which IP address you can be contacted at).

The Real-time Transport Protocol (RTP) provides end-to-end network transport functions suitable for applications transmitting real-time data such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of-service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP is fully specified in RFC 1889.

Note that the once the phone call is initiated, the two VoIP devices sending SIP and RTP packets directly to each other.

What is the difference between FXS and FXO?

FXS (Foreign Exchange Service).
This interface allows the VoIP device to be connected directly to phones, faxes and modems. The FXS ports can also be connected to FXO ports (trunk lines) on your office PBX. The FXS interface generates the Dial Tone, and listens for the dialling digits.

FXO (Foreign Exchange Office)
This interface allows the VoIP device to be connected to your telephone company's exchange or to FXS ports (extensions) on an analog PBX extension or PSTN line. The FXO interface listens for Dial Tone, and sends the dialling digits.

POTS and PSTN?

POTS is an official abbreviation for "Plain Old Telephone System" (as opposed to new technologies such as VoIP).

PSTN is the "Public Switched Telephone System" as run by the Telcos (Telephone Companies).

What is a PSTN Gateway?

Many VoIP Providers provide a gateway to the PSTN, which allows you to dial regular (PSTN) phone numbers as well as SIP addresses on your VoIP device.

The call goes to your VoIP Provider by SIP; your VoIP Provider checks whether the SIP Address/number dialled is on its SIP network; otherwise it directs the call through the gateway onto PSTN and dials the phone number.

What about Firewalls?

These days most routers include Firewall and NAT functions, though they can be perfomed by dedicated devices.

Firewalls block unauthorized port activities, and can affect proper working of VoIP.

NAT (Network Address Translation) is used to share a single Internet connection among multiple computers on your LAN.  

There are several NAT types, with different characteristics and capabilities. The most common being symmetric NAT, which does not support STUN.

SIP is a simple protocol and can be routed through firewalls by opening specific port numbers in advance. STUN has also been developed as a method for traversing NAT without having to open ports, but is not available on all router/firewalls.

STUN (Simple Traversal of UDP Through NATs) server support is required if you will be using more than one VoIP device on your LAN. If your NAT supports STUN, then you don't need to Open/forward ports.

Some VoIP providers use Outbound Proxy as a method for achieving the same objective.

Opening of certain ports is the correct way to allow VoIP working while protecting your own LAN.

Your NAT router / firewall needs to allow voice calls to pass through, and redirect them to your VoIP device. This is achieved by Opening or Forwarding ports (sometimes referred to as Virtual server). Port TCP 5060 is used to initiate a VoIP call, and a range of UDP ports is used to carry the voice conversation (UDP port numbers depend on your VoIP device).

The ports to be opened are commonly: Open Ports for SIP (TCP port 5060) and RTP (UDP ports 10050-15000)

Quality of Service (QoS) support is desirable, because this can give your VoIP packets priority over other data packets.

How can QoS (Quality of Service) improve quality?

Consider the situation where you are having a VoIP (Voice over Internet Protocol) conversation with a friend when your workmate sends an email with all his holiday photos attached - during which your friend cannot hear you properly because your voice is choppy.

VoIP is achieved as two 'streams' of data - your voice is digitised and sent as a stream of packets to the other party; at the same time as their voice is sent to you. As long as the packets arrive quickly and in the correct order, your friend hears your voice at excellent quality.

Since your ADSL or cable connection has less uplink bandwidth than downlink bandwidth, the email and its attachments could take a minute to send during which other types of Internet traffic (including the outgoing VoIP data) have to fight to get through. The result is that some of your voice packets time out before reaching your friend, and what your friend hears is very choppy.

QoS allows you to instruct your router to send VoIP at high priority (to maintain good voice quality) and email at low priority (since e-mail isn't time critical). So when your router becomes congested it will prioritise the traffic and send VoIP data as it is received, fitting the email around the VoIP packets.

Does a VoIP call transmit continuously across the IP network ?

No, bandwidth is only used when someone is speaking, periods of silence (no speech) are automatically suppressed - this can account for up to 60% of a typical voice call. To compensate for these 'quiet' periods, 'comfort' or 'background' noise is automatically generated, so that the receiving end knows that the line is still active.

Codecs?

VoIP devices use programs called codecs to co mpress/ dec ompress the data and break it into packets for sending across the network. The devices used at each end negotiate to determine which codec they both support to give the best quality and lowest bandwidth. Use of higher bandwidth codecs is more susceptible to congestion on the Internet.

In general, if you have sufficient transmit rate between the end points, then the higher codec brings better quality; however, modern day codecs including 723.1A works well even at 6.3/5.3 kbps.
Bandwidth is related to the Codec being used:

  • The G.711 Codec uses 64kbps of bandwidth, similar to an ISDN/PCM call.
  • The G.729 Codec compresses voice to 8Kbps.
  • The G723.1 Codec has the ability to compress voice at two separate rates i.e. 6.3kbit/s and 5.3kbit/s. It should be noted that packet overhead (typically 46 octets) will increase these respective bandwidths, so it is best to choose a compressed voice Codec to minimize this effect. For example, the total required bandwidth including the packet header is about 9.33kbit/s for G723.1 (5.3kbit/s) and about 10.40kbit/s for G723.1 (6.3kbit/s).

What is the difference between G.711 A-law and u-law?

G.711 is the international standard for encoding voice at 64 kbps; and is typically used by Telcos for carrying phone calls between telephone exchanges. G.711 operating at 64 kbps is generally considered as the uncompressed reference against which the speech quality of lower bit rate algorithms are measured.

G.711 is a pulse code modulation (PCM) scheme operating at an 8 kHz sample rate, with 8 bits per sample (hence 64kbps). You can select between two different variants of G.711: A-law and law. A-law is the standard for international circuits including Europe and much of APAC, while law is used in the United States.

What is H.323, and how does it work?

H.323 is an earlier internet protocol for multimedia (including VoIP) which was very common in the 1990's.

Should I use SIP or H.323?

SIP and H.323 are two standard VoIP protocols. H.323 is the "legacy" protocol with a big installed base, but it was not originally designed for IP networks, existing VoIP networks may require H.323 operation for interworking and compatibility reasons.

SIP (Session Initiation Protocol) is the new VoIP protocol optimized for IP networks. This has lower overhead than H.323 and a simpler architecture. This is the preferred protocol for greenfield sites and new VoIP users.

What is a H.323 Gatekeeper?

A Gatekeeper is used on an H323 network to provide user registration and call routing capabilities analogous to the SIP Server on a SIP network. It also lets network managers set policies and control network resources such as bandwidth utilization. An H323 Gatekeeper is necessary in order to provide routing on an H323 VoIPTalk network.

What is MGCP?

The Media Gateway Control Protocol, or MGCP, is a recent protocol designed to address the requirements of 'carrier-based' IP telephony networks. MGCP is a protocol used by external call control devices called Media Gateway Controllers (MGCs) for controlling Media Gateways e.g. VoIPTalk units. With MGCP, the gateway device (e.g. VoIPTalk unit) is 'dumb', the MGCP protocol is used by the MGC to activate ringing/dialling/busy tones etc. on the 2600V unit. The 'intelligence' of the gateway moves to the centre of the network, to the MGC.

 

Copyright © 2013 VoIPAustralia.net. All Rights Reserved.