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  • About VoIP
    • What is VoIP and what it can do for you
    • Introduction to VoIP (video)
    • Why should you switch to VoIP services?
    • Analog Telephony
    • Digital Telephony
    • What is SIP?
    • How to start with VoIP telephony
    • Web based VoIP
    • How to choose a right VoIP provider?
    • Wi-Fi network and VoIP
    • VoIP Codecs
    • Free sip account
    • Confidential calls
    • VPN: UDP or TCP?
    • Mobile VoIP
    • VoIP on your mobile
  • Asterisk IP-PBX
    • All about Asterisk
    • About Mark Spencer
    • Asterisk SIP Media NAT
    • VoIP Codecs
    • Cisco ATA186 + Asterisk Config
  • Who we are?
  • How to start
  • Free SIP account
  • Configs
    • Grandstream Budgetone
facetime_150x150

QoS For FaceTime, bandwidth requirements and Firewall config

By Nik On January 3, 2013 · In Apple, FaceTime, Prioritization and traffic shaping, QoS - Quality of Service, VoIP calls quality

Facetime – Calls Quality, QoS and firewall ports in use

Video calling will become an increasingly widespread form of communication in coming years, moving from
broad adoption in the consumer market to selective endorsement in the corporate world. The same executives
who video-call their families when on the road will also use FaceTime [...]

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VoIP Quick start guide

By Nik On November 9, 2011 · In Open source, QoS - Quality of Service, SIP protocol, VoIP calls quality, VoIP over VPN

Hi there. Today I found a quick start guide on VoIP written for Cisco 3600 series modular routers- click here to download PDF. I think it could be useful even if you just a newbie and what to learn how VoIP works, so I decided to share it. Please let me know if you think it’s [...]

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Voip via VPN SSL voice quality

Running VoIP via VPN (SSL) – voice quality

By Nik On October 18, 2011 · In QoS - Quality of Service, SIP protocol, VoIP calls quality, VoIP over SSL VPN, VoIP over VPN, VoIP via VPN

The protocol overhead caused by the encapsulation of VoIP protocol within VPN dramatically increases the bandwidth requirements for VoIP calls, thus making the VoIP over VPN protocols too “fat” to be used over a mobile data connections like GPRS, EDGE or UMTS. Although VoIP over VPN is not as usable in mobile environments, it is [...]

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G.711: u-law or a-law?

By Nik On August 23, 2011 · In Asterisk, Google Voice, Mobile VoIP, QoS - Quality of Service, VoIP calls quality

As you already know G.711 is a high quality voice codec that we have support for in Asterisk as well as in  many other open source and commercial VoIP platforms. G.711 uses logarithmic PCM (pulse code modulation), a standard as old as from 1972. G.711 is pretty much the norm for IP-telephony where there is enough [...]

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Wi-Fi access point/router optimization for VoIP and other real time apps

By Nik On August 23, 2011 · In Mobile VoIP, Prioritization and traffic shaping, QoS - Quality of Service, SIP protocol, VoIP calls quality, VoIP over VPN

In order to improve the bandwidth allocation for certain apps in your wireless network you need to enable QoS (quality of service) feature on your router/access point. Please make sure that you have enough bandwidth available (down and up) on your [...]

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Think on solutions: VoIP phone system

By Nik On June 21, 2011 · In Android, Asterisk, Mobile VoIP, Prioritization and traffic shaping, QoS - Quality of Service, SIP protocol, VoIP calls quality, VoIP over VPN

It is often taken for granted that installing VoIP solutions will save customers time and money, but in actual fact getting Return on Investment depends on a number of factors.

Firstly if you are planning to use VoIP phone system for business purposes, think about how it will integrate into your existing Public Switched [...]

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The QoS Dilemma

By Nik On June 15, 2011 · In Prioritization and traffic shaping, QoS - Quality of Service, SIP protocol, VoIP calls quality

The problem with IP is that, like Ethernet, it is a connectionless technology and does not guarantee bandwidth.  Specifically, the protocol will not, in itself, differentiate network traffic based on the type of flow to ensure that the proper amount of bandwidth and prioritization level are defined for a particular type of application. By contrast, [...]

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VoIP calls over satellite links

By Nik On June 14, 2011 · In Asterisk, Prioritization and traffic shaping, QoS - Quality of Service, SIP protocol, VoIP calls quality

VoIP services via high latency satellite links: overview

As we all know the VoIP market expanding rapidly over the past few years and SIP based services allows the voice communication over global Internet network, giving the customers great savings and more convenient service.  In the past with cooper wires you can do nothing about bad [...]

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VoIP client behind a VPN with DD-WRT

By Nik On June 14, 2011 · In Asterisk, Mobile VoIP, QoS - Quality of Service, SIP protocol, VoIP over VPN

We have two sites with local networks interconnected by VPN tunnel and we have a VoIP server at one of the sites. The server side based on a PC with Windows 2003 Server (also acting as VoIP and VPN server) installed. On the other side we have a DD-WRT as the VPN client.

As we [...]

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frantic-phone-call

Typical VoIP Problems: not right codec, packet loss, jitter, out‐of‐order packets

By Nik On June 9, 2011 · In Asterisk, Mobile VoIP, Prioritization and traffic shaping, QoS - Quality of Service, SIP protocol, VoIP calls quality

Due to human perception, VoIP is much more sensitive to certain network conditions that are considered well within spec for most applications.

Network issues such as packet loss, jitter, and packet sequence errors are inherent to IP networks, and are well corrected and tolerated by data transfer protocols. Voice transmissions are real‐time by the nature; [...]

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SIP based VoIP service is on path to become one of the most significant protocols since HTTP and SMTP

By Nik On June 9, 2011 · In QoS - Quality of Service, SIP protocol, Uncategorized, VoIP calls quality

Understanting SIP

The growing thirst among communications providers,their partners and subscribers for a new generation of IPbased services is now being quenched by SIP – the SessionInitiation Protocol.  An idea born in a computer sciencelaboratory less than a decade ago, SIP is the first protocolto enable multi-user sessions regardless of media contentand is [...]

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  • Recent articles

    • Plain explain: IP-phone
    • Plain explain: What is SIP termination and what is SIP origination?
    • US fixed VoIP market to see steady growth
    • Save money with VoIP and unified communications
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    • Ubuntu (Linux) on your phone? Yes! And now officially!
    • QoS For FaceTime, bandwidth requirements and Firewall config
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    • What is VoIP termination
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    • Hey!
    • How to configure Internet tel. and SIP settings on Nokia phone (E52)
    • TCP vs UDP – you must know this
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    • VoIP or IP Telephony?
    • Asterisk 10.0.0-rc1 Now Available!
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    • History: Kellogg Field Phone (World War I)
    • Running VoIP via VPN (SSL) – voice quality
    • Skype App directory
    • G.711: u-law or a-law?
    • Wi-Fi access point/router optimization for VoIP and other real time apps
    • Agri-Cube grows mass quantities of vegetables in a one-car parking spot
    • Quick Comparison of freeware IP-PBX platforms: Asterisk vs Open SER
    • Microsoft enabling the ability to eavesdrop on VoIP conversations
    • Nimbuzz growing even without Skype
    • Skype protocol hack
    • Call DSN number
    • How to make your VoIP calls private and confidential
    • Think on solutions: VoIP phone system
    • Asterisk and Google Voice
    • VoIP Codec: Payload size
    • Nokia SIP settings
    • The QoS Dilemma
    • VoIP calls over satellite links
    • VoIP for Facebook!
    • VoIP client behind a VPN with DD-WRT
    • Digital Telephony
    • Analog Telephony
    • Mobile VoIP – the future of mobile communications
    • Sip on Android
  • VoIP SIP IP telephony tags

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  • More to read on VoIP

    • About
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    • Asterisk SIP Media NAT
    • Browser-based VoIP: web page code to call over IP (to your VoIP account)
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    • How to start with VoIP telephony
    • Multiplexing RTP Data and Control Packets on a Single Port
    • On-line payments
    • VoIP Codecs
    • VPN: UDP or TCP?
    • What is VoIP and what it can do for you
    • Why should you switch to VoIP services?
  • Blogroll

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    • Ubuntu how-to www.ubuntuka.com Miscellaneous Ubuntu Tips, Tricks and Hints
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voip-sip.org – Internet VoIP calls, International SIP termination, VoIP hardware

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