voip-sip.org - Internet VoIP calls, International SIP termination, VoIP hardware
  • 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

Skype App directory

By Nik On August 31, 2011 · In VoIP calls quality

Many people using Facebook, Apple and Android applications. Skype has now introduced their applications from third party vendors that has to  enhance usability.

Looks like after Microsoft purchased  Skype, it’s on a roll. They bought several related tech companies in the last few months and they are ready to enhance their VoIP offering by allowing [...]

Continue Reading →

Microsoft enabling the ability to eavesdrop on VoIP conversations

By Nik On July 13, 2011 · In Asterisk, Mobile VoIP, SIP protocol

I’ve reading an interesting article on eweek.com that describes the problems that large companies face as they try to diversify – specifically the move of Microsoft from software towards IP telephony.

News Analysis: Now that Microsoft is on its way to becoming a telephone company, it is finding itself subject to a lot of things that a [...]

Continue Reading →

Skype protocol hack

By Nik On June 24, 2011 · In SIP protocol, VoIP calls quality

Skype protocol hack could have been prevented claims StarForce

StarForce’s comments come in the wake of blog postings by security researcher Efim Buchmanov who, earlier this month, claimed to have reverse engineered the Skype protocol.

“My aim is to make skype open source. [...]

Continue Reading →
  • 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
    • IP telephony can help level the playing field for small businesses
    • The loss of a landline means big change in communications
    • Ubuntu (Linux) on your phone? Yes! And now officially!
    • QoS For FaceTime, bandwidth requirements and Firewall config
    • Merry Christmas and Happy New Year!
    • How to Configure Axvoice Equipment?
    • Cloud VoIP vs. on-premise VoIP: Choosing the right one for your business
    • 26 terabits per second data transmission achieved
    • A fully functional VoIP Client (SIP) finally released (free download)
    • Fundamentals of SIP from Cisco :-)
    • WebRTC from Google: making real-time communication free to implement
    • What is VoIP termination
    • Introduction to Voice over IP (VoIP)
    • Linux: how to check OS version installed
    • Hey!
    • How to configure Internet tel. and SIP settings on Nokia phone (E52)
    • TCP vs UDP – you must know this
    • Google voice / Google talk and Asterisk configuration
    • VoIP or IP Telephony?
    • Asterisk 10.0.0-rc1 Now Available!
    • SIP based VoIP behind NAT
    • VoIP Quick start guide
    • 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

    android Asterisk bandwidth bandwidth requirements best effort cellular codec delay encryption options facebook free calls g273 g726 g729 google gsm high latency How it works ilbc IPsec issues jitter listen to voip mobile Nimbuzz nokia order packet payload. g711 privacy protocol analyzer pstn QoS - Quality of Service QoS protocols real-time applications record voip calls satellite link sipdroid SIP protocol Skype speex TLS voice quality voip voip becomes social
  • Quick navigation

    • Android (6)
    • Apple (1)
    • Asterisk (22)
    • Cloud VoIP (1)
    • FaceTime (1)
    • Google Voice (6)
    • History (2)
    • IT news (2)
    • Mobile VoIP (16)
      • Symbian (1)
    • Non VoIP news (2)
    • Open source (10)
    • Prioritization and traffic shaping (9)
    • QoS – Quality of Service (11)
    • SIP protocol (32)
      • Cisco (2)
    • SIP termination (2)
    • SMS to email (1)
    • Softphone VoIP (1)
    • Ubuntu (1)
    • Uncategorized (1)
    • VoIP calls quality (24)
    • VoIP industry news (4)
    • VoIP over SSL VPN (1)
    • VoIP over VPN (8)
    • VoIP service (2)
    • VoIP via VPN (1)
  • More to read on VoIP

    • About
    • About Mark Spencer
    • Asterisk SIP Media NAT
    • Browser-based VoIP: web page code to call over IP (to your VoIP account)
    • Choosing the right provider
    • Cisco ATA186 notes
    • Free sip account
    • Grandstream Budgetone configuration manual
    • 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

    • Asterisk™: The Definitive Guide (new window) “Asterisk has been emblematic of the way that open source software has changed business—and changed the world”
    • Blog Jon FreeSWITCH VOIP SIP Asterisk Linux Open Source
    • Business.com Business.com is one of the Web’s largest directories for business products and services
    • Ubuntu how-to www.ubuntuka.com Miscellaneous Ubuntu Tips, Tricks and Hints
  • Our Twitter – latest

    • Introduction to Voice over IP (VoIP) - One important step into adopting VoIP is... voip-sip.org/introduction-t… 5 months ago
    • How to configure Internet tel. and SIP... voip-sip.org/how-to-configu… 5 months ago
    • Hey! - To all people who has an experience with Asterisk and Linux - quote of the day: "I have not failed.... voip-sip.org/hey/ 5 months ago
    • What is VoIP termination - Many people keep asking me - what is VoIP Termination?... voip-sip.org/what-is-voip-t… 5 months ago
    • How to configure Internet tel. and SIP... voip-sip.org/how-to-configu… 5 months ago
    Follow @sipcalls
  • Protected by Copyscape DMCA Copyright Detector
"Introduction to Voice over IP (VoIP) - One important step into adopting VoIP is... http://t.co/fYdnQ7Jc" — sipcalls

voip-sip.org – Internet VoIP calls, International SIP termination, VoIP hardware

Pages

  • About VoIP
  • Asterisk IP-PBX
  • Who we are?
  • How to start
  • Free SIP account
  • Configs

The Latest

  • Plain explain: IP-phone
    We all strive to get a quality telephony, but when we […]

More

Thanks for dropping by! Feel free to join the discussion by leaving comments, and stay updated by following ourVoIP-SIP.org Twitter.
© 2013 voip-sip.org
Platform by PageLines