Advanced VoIP Information

If you are new to VoIP and want to know more details regarding the technical aspects of VoIP then you have come to the right spot. Below you will find a list of the common Technical points that interest most people.

How much bandwidth does VoIP need?

The following is a simple formula to calculate your bandwidth requirements:

VoIP Bandwidth + Normal Usage + 25% Buffer = Internet Bandwidth Needed

When making your bandwidth calculations remember that only 80% to 85% of the bandwidth is available for data when using ADSL. The remaining portion is used in "overhead". For example a 128kbs broadband connection will usually only have 100kb/s of sustainable bandwidth available. Therefore we recommend that when calculate your bandwidth requirements it is better to over-estimate.

A single VoIP call consumes approximately 33kbs of internet bandwidth. Therefore most standard internet connections of 512/128kbs can handle 3 simultaneous calls. Also, you must remember if you are sharing your internet connection with VoIP and everyday internet usage you will need to have adequate bandwidth.

ADSL Connection SIP Voice Channels
256/64kb/s 1
512k/128kb/s 3
1500/256kb/s 6
512/512kb/s 12

* The above estimates are for "VoIP only" traffic on a quality internet connection. Also note, these benchmarks are made using the standard G729 Codec.

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What factors affect voice quality

Factors that affect voice quality include:

Factor Description
Loudness Ratings This is a measure of the speech volume
Sidetone The user’s ability to hear their own voice (so the service doesn’t sound “dead”).
(Absolute) Delay Transmission and processing time between a speaker’s mouth and a listener’s ear. Note: Delay is increased for long-distance calls, slow data rates, and when the available data rate is exceeded.
Echo Loudness The fraction of speech echoed back to a sender by a receiver.
Echo Delay The delay before speech is echoed back to a speaker.
Transcoding The change from one coding method to another e.g. fixed and mobile networks use different codecs.
Noise levels In a room and on a line.
Packet Loss Frequency of packet loss, and immunity of the codec to packet loss.
Use of QoS If a VoIP call is not given priority over other, less time sensitive services (e.g. web browsing) then the voice quality may fluctuate with the amount of other traffic on the data link.
Available Data Rate This affects the number of calls that can be maintained on a connection.

Traditional telephony requires a certain data rate per call to operate and if the limit is reached then no more concurrent calls are possible. With VoIP it might be possible to squeeze an extra call in but that means reducing the voice quality for all users.

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What factors Affect Voice Service Reliability

Factors that affect voice service reliability include:

  • Power Supply: Devices used for VoIP services (e.g. headset and/or adapter) almost always require connection to a plug pack or a PC, unlike traditional phones that get power from the line.
  • Network Performance: VoIP services over networks with low delay and low packet loss generally have better performance.
  • Use of a Wireless Access Network: A wireless network can give a user freedom to move around but it may be subject to more interference than a fixed network.
  • Presence of a Firewall: Use of a firewall (e.g. for data or network security) can make it hard for a VoIP call to take place between parties on either side of the firewall.
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VoIP Related Standards and Publications

* Some of the following information was sourced from IHS Electronics.

The Internet Engineering Task Force (IETF)
Alliance for Telecommunications Industry Solutions (ATIS)
Consumer Electronics Association (CEA)
European Telecommunications Standards Institute (ETSI)
International Telecommunications Union (ITU)
International Organization for Standardization (ISO)
Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Telecommunications Industry Association (TIA)

 

The Internet Engineering Task Force (IETF)
SIP Standards
Standard Description
RFC 2076 Common Internet Message Headers
RFC 2396 Uniform Resource Identifiers (URI): Generic Syntax
RFC 2543 SIP: Session Initiation Protocol (Obsoleted by RFC3261, RFC3262, RFC3263, RFC3264, RFC3265)
RFC 3261 SIP: Session Initiation Protocol. (Updated by RFC3265, RFC3853, RFC4320)
RFC 3262 Reliability of Provisional Responses in Session Initiation Protocol (SIP).
RFC 3263 Session Initiation Protocol (SIP): Locating SIP Servers.
RFC 3264 An Offer/Answer Model with Session Description Protocol (SDP).
RFC 3265 Session Initiation Protocol (SIP)-Specific Event Notification. A. B. Roach. June 2002. (Obsoletes RFC2543) (Updates RFC3261) (Status: PROPOSED STANDARD)
RFC 4028 Session Timers in the Session Initiation Protocol (SIP) S. Donovan, J. Rosenberg. April 2005. (Format: TXT=65363 bytes) (Status: PROPOSED STANDARD)

RPT Standards
Standard Description
RFC 1889 RTP: A Transport Protocol for Real-Time Applications

ENUM Standards
Standard Description
RFC 2915 The Naming Authority Pointer (NAPTR) DNS Resource Record. Also covers a std for Dialplan substitution
RFC 2916 ENUM Protocol (obsolete)
RFC 3761 ENUM Protocol (obsoletes RFC 2916)
RFC 3762 Usage of H323 addresses in ENUM Protocol
RFC 3764 Usage of SIP addresses in ENUM Protocol
RFC 3026 Berlin Liaison Statement
RFC 2870 Root Name Server Operational Requirements

QoS Standards
Standard Description
RFC 3246 An Expedited Forwarding PHB (Per-Hop Behavior). This is related to QoS.

 

VOIP Related IETF Working Groups
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Alliance for Telecommunications Industry Solutions (ATIS)
Standard Description
ATIS T1.312 Voice Packetization - Packetized Voice Protocol
ATIS T1.508 Telecommunications - Loss Plan for Evolving Digital Networks
ATIS T1.521 A Packet Loss Concealment Technique for use with ITU-T Recommendation G.711
ATIS T1.607 Integrated Services Digital Network (ISDN) - Layer 3 Signaling Specification for Circuit-Switched Bearer Service for Digital Subscriber Signaling System Number 1 (DSS1)
ATIS T1.TR.61 Transmission Performance Planning Issues Regarding the Introduction of Voice Over ADSL Technology into Networks Supporting Voiceband Services
ATIS T1.TR.80 Descriptors for User-Perceived Impairments in Speech over Voice-Over-Internet-Protocol (VOIP) Networks
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Consumer Electronics Association (CEA)
Standard Description
CEA-851.1 IP-Based Digital Telephony for the Versatile Home Network
CEA-2002 Test Procedure for Powerline Carrier Technology
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European Telecommunications Standards Institute (ETSI)
Standard Description
ETSI TR 102 024-6 Telecommunications and Internet Protocol Harmonization over Networks (TIPHON) Release 4; End-To-End Quality of Service in TIPHON Systems; Part 6: Actual Measurements of Network and Terminal Characteristics and Performance Parameters in TIPHON
ETSI TS 102 024-9 Telecommunications and Internet Protocol Harmonization over Networks (TIPHON) Release 4; End-to-End Quality of Service in TIPHON Systems; Part 9: Call Performance Classification (Voice)
ETSI TR 101 329-1 Telecommunications and Internet Protocol Harmonization over Networks (TIPHON) Release 3; End-to-End Quality of Service in TIPHON Systems; Part 1: General Aspects of Quality of Service (QoS)
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International Telecommunications Union (ITU)
Standard Description
ITU-T G.711 Pulse Code Modulation (PCM) of Voice Frequencies
ITU-T G.823 The Control of Jitter and Wander within Digital Networks which are Based on the 2048 kbit/s Hierarchy
ITU-T H.248.1 Series H: Audiovisual and Multimedia Systems - Infrastructure of Audiovisual Services - Communication Procedures - Gateway Control Protocol: Version 2
ITU-T H.323 Packet-Based Multimedia Communications Systems
ITU-T H.450.1 Generic Functional Protocol for the Support of Supplementary Services in H.323
ITU-T Y.1540 Internet Protocol Data Communication Service - IP Packet Transfer and Availability Performance Parameters Amendment 1: New Appendix VIII: Background on IP Service Availability
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International Organization for Standardization (ISO)
Standard Description
ISO/IEC 10646-1 Information Technology - Universal Multiple-Octet Coded Character Set (UCS) - Part 1: Architecture and Basic Multilingual Plane
ISO/IEC 11571 Information Technology - Telecommunications and Information Exchange Between Systems - Numbering and Sub-Addressing in Private Integrated Service Networks
ISO/IEC 11801 Information Technology - Generic Cabling for Customer Premises
ISO/IEC 17343 Information Technology - Telecommunications and Information Exchange Between Systems - Corporate Telecommunication Networks - Signalling Interworking Between QSIG and SIP - Basic Services
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Institute of Electrical and Electronics Engineers, Inc. (IEEE)
Standard Description
IEEE 1329 Standard Method for Measuring Transmission Performance of Handsfree Telephone Sets
IEEE 269 Methods for Measuring Transmission Performance of Analog and Digital Telephone Sets, Handsets, and Headsets
IEEE 802.1D Information Technology - Telecommunications and Information Exchange Between Systems - Local and Metropolitan Area Networks - Common Specifications - Part 3: Media Access Control (MAC) Bridges
IEEE 802.1Q Virtual Bridged Local Area Networks
IEEE 802.3 Information Technology - Telecommunications and Information Exchange Between Systems - Local and Metropolitan Area Networks - Specific Requirements Part 3: Carrier Sense Multiple Access with Collision Detection (CSMA/CD) Access Method and Physical Layer
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Telecommunications Industry Association (TIA)
Standard Description
TIA TSB 146 Telecommunications - IP Telephony Infrastructures - IP Telephony Support for Emergency Calling Service
TIA/EIA TSB 32 Overall Transmission Plan Aspects for Telephony in a Private Network
TIA/EIA/IS-811 Telecommunications - Telephone Terminal Equipment – Performance and Interoperability Requirements for Voice-over-IP (VoIP) Feature Telephone
TIA/EIA-810 Transmission Requirements for Narrowband Voice over IP and Voice over PCM Digital Wireline Telephones
TIA-912 Telecommunications IP Telephony Equipment Voice Gateway Transmission Requirements
TIA-920 Telecommunications Telephone Terminal Equipment Transmission Requirements for Wideband Digital Wireline Telephones
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SIP Response Codes

SIP response codes, class 1: Provisional messages
These are sent within a SIP dialogue
  • 100 Trying
  • 180 Ringing
  • 181 Call Is Being Forwarded
  • 182 Queued
  • 183 Session Progress
SIP Response codes: 2xx class
The 2xx class of responses indicates a success
  • 200 OK
  • 202 accepted: Used for referrals
SIP response codes, class 3xx
The 3xx class of responses indicates a redirection of the call
  • 300 Multiple Choices
  • 301 Moved Permanently
  • 302 Moved Temporarily
  • 305 Use Proxy
  • 380 Alternative Service
SIP responses, class 4: Request failures
  • 400 Bad Request
  • 401 Unauthorized: Used only by registrars. Proxys should use proxy authorization 407
  • 402 Payment Required (Reserved for future use)
  • 403 Forbidden
  • 404 Not Found: User not found
  • 405 Method Not Allowed
  • 406 Not Acceptable
  • 407 Proxy Authentication Required
  • 408 Request Timeout: Couldn't find the user in time
  • 410 Gone: The user existed once, but is not available here any more.
  • 413 Request Entity Too Large
  • 414 Request-URI Too Long
  • 415 Unsupported Media Type
  • 416 Unsupported URI Scheme
  • 420 Bad Extension: Bad SIP Protocol Extension used, not understood by the server
  • 421 Extension Required
  • 423 Interval Too Brief
  • 480 Temporarily Unavailable
  • 481 Call/Transaction Does Not Exist
  • 482 Loop Detected
  • 483 Too Many Hops
  • 484 Address Incomplete
  • 485 Ambiguous
  • 486 Busy Here
  • 487 Request Terminated
  • 488 Not Acceptable Here
  • 491 Request Pending
  • 493 Undecipherable: Could not decrypt S/MIME body part
SIP responses, class 5: Server failures
  • 500 Server Internal Error
  • 501 Not Implemented: The SIP request method is not implemented here
  • 502 Bad Gateway
  • 503 Service Unavailable
  • 504 Server Time-out
  • 505 Version Not Supported: The server does not support this version of the SIP protocol
  • 513 Message Too Large
SIP response codes, class 6: Global failures
  • 600 Busy Everywhere
  • 603 Decline
  • 604 Does Not Exist Anywhere
  • 606 Not Acceptable
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What training in VoIP is available?

The following Cisco courses are available

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Top 10 Internet Security Tips

The following information is from the Australian Government's Department of Broadband, Communications and the Digital Economy website

Consider the following tips.  They will help you secure your computer and make sure your business is using the Internet safely.
1. Develop a 'culture of security'
Businesses need to have Internet security measures in place and make sure staff are aware of and follow Internet security practices.

2. Install anti-virus software and keep it updated
Anti-virus software scans and removes known viruses your computer may have contracted. It will help protect your computer against viruses, worms and trojans.

3. Install a firewall to stop unauthorised access to your computer
Firewalls work like a security guard to protect your computer from intruders.

4. Protect yourself from harmful emails
Be cautious about opening emails from unknown or questionable sources.

5. Minimise spam
While it is not possible to completely stop spam from entering your email box, it is possible to take steps to prevent a large amount of spam.

6. Back-up your data
Creating a copy or back-up of data is a sensible way to ensure that you can recover all of your business information from your computer or website quickly and easily.

7. Develop a system for secure passwords
Creating effective passwords can provide additional means of protecting the information on your computer.

8. Keep your software up-to-date
If your software is out of date, you are more vulnerable.

9. Make sure your online banking is secure
If you bank online you should follow security advice provided by your financial institution.

10. Develop and maintain a security policy
You need to monitor and test security policies.

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Where do I find More Information?

Below are some links to Additional Resources you will find helpful:

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