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Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).

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VoIP ISP
Why do some people recommend buying business VoIP from an ISP? What are the benefits to my company? What are the costs?
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Free Tool: Site Down Detector
LVL 11
Free Tool: Site Down Detector

Helpful to verify reports of your own downtime, or to double check a downed website you are trying to access.

One of a set of tools we are providing to everyone as a way of saying thank you for being a part of the community.

Sennheiser
This paper addresses the security of Sennheiser DECT Contact Center and Office (CC&O) headsets. It describes the DECT security chain comprised of “Pairing”, “Per Call Authentication” and “Encryption”, which are all part of the standard DECT protocol.
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Skype for Business
Skype is a P2P (Peer to Peer) instant messaging and VOIP (Voice over IP) service – as well as a whole lot more.
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Hey there

Heard about jingle, the add on for XMPP that enables point to point audio between two XMPP clients. No server config necessary. Actually quite a cool feature. However, how good is it if you can not use those voice capabilities to do a PSTN phone cal, or call some SIP device you have. Well that isn't the purpose of Jingle anyhow. So other tools can do that including Asterisk and Freeswitch and that is what we are going to do in the article below.


Prerequisites :

Good Linux Administration Knowlege
Basic Understanding Of XMPP and SIP
A XMPP server running on a public IP "Openfire used here"
Access to change DNS names for your domain
A SIP client such as XTEN
A XMPP client with jingle capabilities such as JABBIN



Setting up Jingle To SIP

Well first of you will have to install Freeswitch since it the one tool that worked for me in Jingle to SIP conversion.

To do so go and get the latest Freeswitch and compile.

Well if you do have all the dependencies that should be as easy 1 2 3

The dependencies are

http://wiki.freeswitch.org/wiki/Inst...#Prerequisites

One thing you have to pay attention too is to enable the jingle module before compiling or you will have to recompile later on.

To enable the jingle module do the following:

Go to the directory where you extracted the Freeswitch source code, then go to modules.conf and make sure that the following line is not commented.

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I recently purchased a Bluetooth headset called the Music Jogger (model BSH10). The control buttons on it look like this:

Music-Jogger-BSH10-buttons.jpg
One of my goals is to use it as the microphone and speakers for Skype calls. In that respect, it works well. However, I also want to be able to answer a Skype call with its Multi-Function Button (MFB), so that I don't have to be sitting at the computer when a call comes in. In that respect, the headset fails.

One possible solution is to configure Skype to answer incoming calls automatically, but I don't like this idea, for two reasons. First, most of the time I am at my computer. In those cases, I may not always want to answer a call – especially when I see CallerID . Second, I may not be at the computer and may not have the headset on, in which case I don't want Skype to answer the call. I could try to remember to enable/disable Skype's automatic answer feature depending on my whereabouts, but that is likely to be error-prone – and a nuisance to boot. The better solution is to configure the MFB to answer a call. Fortunately, there's a way to do this easily – and with free software.

The solution presented in this article should work on many Bluetooth headsets. For example, here's another one from Kinivo (model BTH220) with similar controls (excellent headset – I own this one, too):

Kinivo-BTH220-buttons.jpg
As long as your Bluetooth headset has a Play-Pause button, …
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How Sip Phone (User Agent) works and communicates with sip servers

1.  There is a sip server and a sip registrar.
 
The sip server and sip registrar can be one server or two different servers. The sip registrar is the server on which it is recorded that a phone number is on a certain ip address and port.

When someone wants to talk to that number he sends INVITE request to his SIP server with the phone number.

 Sip server through the registrar finds out what is the according ip, where to resend the INVITE request.

 According to RFC 3261 (http://tools.ietf.org/html/rfc3261 section 8.3): The "expires" parameter of a Contact header field value indicates how long the URI is valid.  The value of the parameter is a number indicating seconds.  If this parameter is not provided, the value of the Expires header field determines how long the URI is valid.

Then a 2xx response (200 OK) on REGISTER request is received.  It has Contact header expires or an Expires header, with the value in seconds when the Register is expiring. Before this value the device must send a new REGISTER request. Here there is an example of REGISTER and its response: http://wiki.snom.com/Networking/SIP/Registration.

2.  Next is the SDP(http://tools.ietf.org/html/rfc4566). SDP is sent with the INVITE request and a 200 OK response and ACK request. Through it the sip user agents are negotiating the media voice, video.

These are the codecs G711, G729, T38 and others.
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LVL 58

Expert Comment

by:tigermatt
Comment Utility
Hi user_n,

A very interesting and useful discussion here. Thanks very much for putting this article out there for us all. The use of SIP technology is growing as the world moves in leaps and bounds towards IP-based PBXs, so this article could probably not come at a better time! I have hit the "Yes" button to vote this one up.

It is really quite interesting that you say "every user agent must support G711 G711A, G711U..." - I wish my SIP provider supported both of them because the transcoding I have to do between codecs from my SIP device to the provider has caused its own slew of issues on more than one occasion!!

I thought it might be worth mentioning here and pointing out that there are also two types of network element in the SIP protocol - the traditional proxies as laid down by the RFCs but also the back-to-back user agents (B2BUAs) like Asterisk and all its various derivitives.

In my experience, the commercial user with an IP PBX in their office will typically be using a B2BUA of some description while the providers are typically running a complex network of SIP proxies to route any calls they send/receive. Useful information to know. The B2BUAs sit in the middle of the call path, receiving the SIP invitations from the caller and then initiating a new call leg to the callee (which matches the usual definition of a "PBX"). Because the B2BUA is interpreting the SIP session and appears to both endpoints as the place the packets originate from, it's possible for additional features to be introduced to a call which may only be supported at the server-side or by one party. The PBX can just filter out SIP traffic which is not supported on both ends, or it can handle the server-side commands locally. Plus, the actual media traffic which carries the voice can still be the subject of re-invites, so one phone can still send RTP media packets direct to another phone and take the PBX out of the routing path of the voice itself.

That's different to a SIP proxy, which is roughly the same role as routers perform in an IP network. They don't actually do anything with the SIP packets they are given beyond finding the next SIP proxy which gets it a hop closer to the packet's destination, so if an endpoint wants to use a particular feature in the SIP session, both endpoints need to support it.

But, as always, a very nice article and just thought I would point the above out as I thought it was a useful addition to the discussion :-)

-Matt
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Author Comment

by:user_n
Comment Utility
You are right. The article is from user agent point of view. But I have not enough experience with pbx to write for them.
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Every year the snow affects people and businesses. According to the Federation of Small Businesses (FSB), in 2009, UK businesses lost an estimated £1.2bn because of bad weather.

This article was created to show how businesses can use relatively new technology to stay open during bad weather.

Remote Desktop and Cloud Computing

Although cloud computing is quite new, the concept has been around for a long time. Two example of cloud computing are the Hotmail and Gmail email services. Emails are stored on large computers and accessible from anywhere in the world via the internet. So, in the event of bad weather, people are still able to access their emails from home.

Businesses can use this to their advantage. By storing information in a format that’s accessible online, during snowy weather, businesses can allow staff to work from home and avoid the need to drive to work.

Remote Desktop is a technology that’s part of the Microsoft Windows system. Small companies where each member of staff has a designated computer can use Remote Desktop to give staff access during bad weather. It can be used to give people fully access to their work computer from home. Microsoft Small Business Server builds on Remote Desktop, and allows users to simply select their computer from a list, and start work without even leaving the house. If a company does not have Small Business Server, or the technical knowledge to set it up, they can sign up for a free LogMeIn account. LogMeIn Free
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Although VoiceOver IP has been around for a while, internet connections have only recently become fast enough to provide good call quality. Now, VoIP has become a real option for businesses looking at ways to improve their business model. In this article, we will explain how your company can make use of Hosted PBX service and employ staff who work from home, while cutting costs.

What is Hosted PBX?

Hosted PBX is a service which gives you most of the features of a regular office-based phone system. For example, you can transfer calls, put calls on hold and divert calls. You also have access to advanced features which are often not included in office-based PBX systems, such as call queuing, call recording, and IVR Menu Systems. Rather than buying expensive PBX Phone Systems, you pay a monthly fee to rent a place on a Hosted PBX system. This allows for substantial money saving over buying a phone system.

Cut Costs

When you look at employing a home/remote worker (for example, to answer calls and make appointments), you need to be able to transfer incoming calls to them without incurring divert charges which can mount up over time. You also need to give the caller confidence that their call is being handled professionally. With a Hosted PBX system, calls are transferred using VoIP and the internet, which means there are no call transfer charges. And because the call control never leaves the Hosted PBX system, the caller still receives a professional …
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Expert Comment

by:DanRollins
Comment Utility
Good article.
My company uses OneBox (http://www.new.onebox.com) and we find it to be very flexible and convenient.  The feature I find most useful is that voicemails are turned into emails -- so I have just one "inbox queue" of things to get done.  There is some discussion of this as a key component of a "virtual office" in my article:
     Create and Run a Virtual Office
     http://www.experts-exchange.com/ITPro/Consulting/A_3826.html
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In my office we had 10 Cisco 7940G IP phones that were useless as they were showing
PROTOCOL APPLICATION INVALID
when started. I searched through Google and worked for a week continuously on those phones, and finally got them working. This is a difficult problem for those who are new to IP phones, so I would like to share a step by step procedure to get them working again.

In this explanation I have set this Cisco IP phone to SIP mode, much as you can set this phone to other modes. As VOIP is more preferred I think SIP mode is more useful than others.

1. Step 1

Get TFTP server software.I prefer the Solarwinds TFTP server as it has worked for me. Install this on a workstation in your network.

2. Step 2

Get a configuration file editor (ASCII editor)Download the free hex editor Neo and install it.

3. Step 3

Create the configuration filesNow the important part comes: creating configuration files. I have searched a lot in Google for configuration files, but have not seen them together in one place, so I am putting the formats of those configuration files.

These are the files required.

1.CTLSEP001121F11A5A.tlv (001121F11A5A is MAC address of a Cisco IP phone)
2.OS79XX.TXT
3.P0S3-8-12-00.sb2 (name of ISO image)
4.P0S3-8-12-00.loads
5.P003-8-12-00.bin
6.P003-8-12-00.sbn
7.SEP001121F11A5A.cnf.xml (MAC address of the Cisco phone after SEP)
8.SIP001121F11A5A.cnf (MAC address of the Cisco IP phone after SIP)
9.SIPDefault.cnf

1. CTLSEP001121F11A5A.tlv
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Author Comment

by:lentinsun
Comment Utility
sorry for the delay in response....i was a bit busy with some other activities.
What is the status of these phones? If it is not working still , Could you please post the TFTP server log
during upgradation.

Regards,
Lentin  
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Expert Comment

by:vikrantambhore
Comment Utility
Hi Lentin,

It's has been fixed thanks for response
0

Voice Over IP

Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony, Internet telephony, broadband telephony, and broadband phone service. The term specifically refers to the provisioning of communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone network (PSTN).  Examples of the VoIP protocols are H.323, Media Gateway Control Protocol (MGCP), Session Initiation Protocol (SIP), H.248 (also known as Media Gateway Control (Megaco)), Real-time Transport Protocol (RTP), Real-time Transport Control Protocol (RTCP), Secure Real-time Transport Protocol (SRTP), Session Description Protocol (SDP), and Inter-Asterisk eXchange (IAX).