- 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
- Who we are?
- How to start
- Free SIP account
- Configs
After Microsoft bought Skype for US$8.5 billion, Google has released a developer preview of WebRTC – an open framework enabling implementation of voice and video Real Time Communications in the browser with the use of HTML 5 and JavaScript APIs.
It’s possible that the Skype acquisition could mean restricting the client from technologies and devices [...]
How to configure Internet tel. and SIP settings on Nokia phone (E52)
By Nik On March 28, 2012 · In Google Voice, Mobile VoIP, Open source, SIP protocol, Symbian, VoIP calls quality
In this article we will talk about Nokia Internet telephony and SIP settings since many people asking me about that. I have check all configuration myself on my old Nokia E52 as well as E71 buddy and everything works fine. Therefore you can use the same setup procedure on any E-series with Symbian.
This article describes the configuration process of Asterisk with Google voice.
Asterisk communicates with Google Voice and Google Talk using the chan_gtalk Channel Driver and the res_jabber Resource module. Before proceeding, please ensure that both are compiled and part of your installation. Compilation of res_jabber and chan_gtalk for use with Google Talk / Voice are dependant [...]
For all the technology behind Voice over IP (VoIP), you’d expect that it would work on every network, but this unfortunately isn’t the case. Network Address Translation (NAT) is a common practice used in networks, and it doesn’t play well with VoIP. Solving this problem requires an understanding of NAT, VoIP [...]
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 [...]
Asterisk: How to make calls using Google Prerequisites
Asterisk can communicate with Google Voice and Google Talk using the chan_gtalk Channel Driver and the res_jabber Resource module. Please ensure that both are compiled and part of your installation prior following the configuration guide below. Compilation of res_jabber and chan_gtalk for use with Google [...]
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
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- How to Configure Axvoice Equipment?
- Cloud VoIP vs. on-premise VoIP: Choosing the right one for your business
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- 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
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- 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
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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
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