VoIP Using SwitchFree as PBX Alicia Castro University of Colorado at Colorado Springs ABSTRACT Evaluating VoIP for enterprise use or for your home phone setup means a lot of experimentation and you will need to build a test server with which to hone your VoIP skills. That test server should be something you can get a lot out of without spending a bundle or committing to a specific vendor’s commercial VoIP platform before you have done your homework. Free telephony software lets you do that homework. [3] 1. INTRODUCTION VoIP (voice over IP) technology is a rapidly expanding field. More and more VoIP components are being developed, while existing VoIP technology is being deployed at a rapid pace. This growth is fueled by two goals: decreasing costs and increasing revenues. That is one of the reasons for free open source software like asterisk and freeSwitch. Unlike Asterisk, freeswitch support the windows platform and it was “the” reason I chose freeswitch over asterisk [2]. Voice over IP allows users to make phone calls over the internet or any other IP network, using the packet switched network as a transmission medium. The maturity of VoIP standards such as Sip and quality of service on IP networks maximize network efficiency, streamlines the network architecture, reduces capital and operational costs and opens up new services opportunities such as web enable multimedia conferencing and unified messaging.. VoIP offers compelling advantages, but it also presents a security paradox [3[. Security over VoIP will not be covered in this paper. The objective of this project is to set up my own VoIP network using an open source PBX implementation like freeSwitch. 2. VoIP Background Voice over Internet Protocol is a method for taking analog audio signals, like the kind you hear when you talk on the phone and turning them into digital data that can be transmitted over the internet. 2.1 What is it good for? Many engaging VoIP services are already available, and services providers are planning even more exciting services. Continued deployment of IP networks and IP endpoint devices will 1
Transcript
VoIP Using SwitchFree as PBX Alicia Castro
University of Colorado at Colorado Springs ABSTRACT Evaluating VoIP for enterprise use or for your home phone setup means a lot of experimentation
and you will need to build a test server with which to hone your VoIP skills. That test server
should be something you can get a lot out of without spending a bundle or committing to a
specific vendor’s commercial VoIP platform before you have done your homework. Free
telephony software lets you do that homework. [3]
1. INTRODUCTION VoIP (voice over IP) technology is a rapidly expanding field. More and more VoIP components
are being developed, while existing VoIP technology is being deployed at a rapid pace. This
growth is fueled by two goals: decreasing costs and increasing revenues.
That is one of the reasons for free open source software like asterisk and freeSwitch. Unlike
Asterisk, freeswitch support the windows platform and it was “the” reason I chose freeswitch over
asterisk [2].
Voice over IP allows users to make phone calls over the internet or any other IP network, using
the packet switched network as a transmission medium. The maturity of VoIP standards such as
Sip and quality of service on IP networks maximize network efficiency, streamlines the network
architecture, reduces capital and operational costs and opens up new services opportunities such
as web enable multimedia conferencing and unified messaging.. VoIP offers compelling
advantages, but it also presents a security paradox [3[. Security over VoIP will not be covered in
this paper. The objective of this project is to set up my own VoIP network using an open source
PBX implementation like freeSwitch.
2. VoIP Background Voice over Internet Protocol is a method for taking analog audio signals, like the kind you hear
when you talk on the phone and turning them into digital data that can be transmitted over the
internet.
2.1 What is it good for? Many engaging VoIP services are already available, and services providers are planning even
more exciting services. Continued deployment of IP networks and IP endpoint devices will
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enable further development of new services. Also, as the processing capacity of IP endpoints
increases – allowing them to deal directly with network access controls, multiple data formats,
and transformations- more innovative and convenient services will become possible [4].
VoIP technology provides a richer, more flexible foundation for building communication
services. IP networks support independent connections for signaling and media traffic. This
decoupling of signal and bearer traffic eliminates interference between the information flows; in
band signaling is not required. Thus, communication with application servers is simplified.
In addition, IP network topology allows any node to act as server. Therefore, multiple
application servers and user endpoints – located in one or several services provider domains-
can communicate via IP to participate in service support [5].
Finally, IP transport is provided by various underlying networks and different network
technologies can support different sets of services. For example, DSL and cable networks
provide broadband IP connections that support real time voice, data and video services. Hence,
these network providers can offer “triple-play” services to their customers.
2.3 Why Use VoIP? There are two major reasons to use VoIP: lower cost and increased functionality.
• Lower cost: In general phone service via VoIP costs less then equivalent service from
traditional sources. This is largely a function of traditional phone services either being
monopolies or government entities. There are also some cost savings due to using a
single network to carry voice and data. This is especially true when users have existing
under utilized network capacity that they can use for VoIP without any additional costs. In
the most extreme case, users see VoIP phone calls (even international) as FREE. While
there is a cost for their internet service, using VoIP over this service may not involve any
extra charges, so the users view the calls as free. There are a number of VoIP service
providers [2].
• Increased functionality: VoIP makes easy some things that are difficult to impossible with
traditional phone networks [3].
o Incoming phone calls are automatically routed to your VoIP phone where ever
you plug into the network. Take your VoIP phone with you on a trip and
anywhere you connect it to the internet, you can receive your incoming calls.
o Call center agents using VoIP phones can easily work from anywhere with a
good internet connection.
2.4 Advantage of using VoIP VoIP technology uses the internet’s packet-switching capabilities to provide phone services. VoIP
has several advantages over circuit switching. For example, packet switching allows several
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telephone calls to occupy the amount of space occupied by only one in a circuit switched
network. Using PSTN, that 10 minute phone call we talked about earlier consumed 10 full
minutes of transmission time at a cost of 128 Kbps With VoIP, that same call may have occupied
only 3.5 minutes of transmission time at a cost of 64 Kbps, leaving another 64 Kbps free for that
3.5 minutes, plus an additional 128 Kbps for the remaining 6.5 minutes. Based on this simple
estimate, another three or four calls could easily fit into the space used by a single call under the
conventional system. And this example does not even factor in the use of data compression,
which further reduces the size of each call [3].
2.5 Disadvantages of Using VoIP The current Public Switched Telephone Network is a robust and fairly bulletproof system for
delivering phone calls. Phones just work, and we've all come to depend on that. On the other
hand, computers, e-mail and other related devices are still kind of flaky. Let's face it -- few people
really panic when their e-mail goes down for 30 minutes. It's expected from time to time. On the
other hand, a half hour of no dial tone can easily send people into a panic. So what the PSTN
may lack in efficiency it more than makes up for in reliability. But the network that makes up the
Internet is far more complex and therefore functions within a far greater margin of error. What this
all adds up to be is one of the major flaws in VoIP: reliability [5].
• First of all, VoIP is dependant on wall power. Your current phone runs on phantom
power that is provided over the line from the central office. Even if your power goes
out, your phone (unless it is a cordless) still works. With VoIP, no power means no
phone. A stable power source must be created for VoIP.
• Another consideration is that many other systems in your home may be integrated into
the phone line. Digital video recorders, digital subscription TV services and home
security systems all use a standard phone line to do their thing. There's currently no
way to integrate these products with VoIP. The related industries are going to have
to get together to make this work.
• Emergency 911 calls also become a challenge with VoIP. As stated before, VoIP uses
IP-addressed phone numbers, not NANP phone numbers. There's no way to
associate a geographic location with an IP address. So if the caller can't tell the 911
operator where he is located, then there's no way to know which call center to route
the emergency call to and which EMS should respond. To fix this, perhaps
geographical information could somehow be integrated into the packets [3].
• Because VoIP uses an Internet connection, it's susceptible to all the hiccups normally
associated with home broadband services. All of these factors affect call quality:
• Latency
• Jitter
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• Packet loss
Phone conversations can become distorted, garbled or lost because of transmission
errors. Some kind of stability in Internet data transfer needs to be guaranteed before
VoIP could truly replace traditional phones.
• VoIP is susceptible to worms, viruses and hacking, although this is very rare and VoIP
developers are working on VoIP encryption to counter this [6].
• Another issue associated with VoIP is having a phone system dependant on individual
PCs of varying specifications and power. A call can be affected by processor drain.
Let's say you are chatting away on your softphone, and you decide to open a
program that saps your processor. Quality loss will become immediately evident. In
a worst case scenario, your system could crash in the middle of an important call. In
VoIP, all phone calls are subject to the limitations of normal computer issues.
One of the hurdles that were overcome some time ago was the conversion of the analog audio
signal your phone receives into packets of data. How it is that analog audio is turned into packets
for VoIP transmission? The answer is codecs.
3 Evaluating VoIP
3.1 FreeSwitch as a PBX
FreeSwitch is free PBX software and easy to work once you understand what you are doing.
FreeSwitch is very powerful, supports several Voice over IP communication protocols: H.323,
SIP, IA and others. Using these protocols, it can support just about any IP telephone, as well as
traditional analog and digital telephones [4]. FreeSwtich has some industrial strength features like
call queuing, conference calling, voice mail and caller ID. Using freeSwitch you can build
something as simple as an answering machine that sends its recorded messages to your email
address or something as sophisticated as a thousand subscriber corporate communications
system with least cost call routing and advance call accounting. The included freeSwitch gateway
interface allows you to develop computer aided telephony tools using PHP, Perl, Java or C and
the management API allows you to build socket-based monitoring and automation applications for
your PBX [2].
Freeswitch has been built on the following platforms [3]:
• Linux (x86 & x86_64)
• Windows (MSVC 2005)
• Mac OS X (intel & ppc )
• FreeBSD 6
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Additionally, the experimental external modules make use of several external modules:
The simplest and most common way is through the use of a device called an ATA (analog
telephone adaptor). The ATA allows you to connect a standard phone to your computer or your
Internet connection for use with VoIP. The ATA is an analog-to-digital converter. It takes the
analog signal from your traditional phone and converts it into digital data for transmission over the
Internet. You simply crack the ATA out of the box, plug the cable from your phone that would
normally go in the wall socket into the ATA, and you're ready to make VoIP calls. Some ATAs
may ship with additional software that is loaded onto the host computer to configure it; but in any
case, for a basic setup it's a very straightforward setup [7].
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Linksys PAP2T-NA Quick Info [6]
• Enables feature-rich telephone service over your broadband Internet connection • Two standard telephone ports for analog phones or use one of the ports for a fax
machine, each with an independent phone number • High quality, clear sounding voice service simultaneous with Internet use • Compatible with all common telephone features: Caller ID, Call Waiting, Voicemail, etc.
Linksys PAP2T Features [6]:
• Two voice ports (RJ-11) for analog phones or Fax machines • Impedance Agnostics - 8 Configurable Settings • Call Waiting, Cancel Call Waiting, Call Waiting Caller ID • Caller ID with Name/Number (Multi-national Variants) • Caller ID Blocking • Call Forwarding: No answer, Busy, All • Do Not Disturb • Call Transfer • Three-way Conference Calling with Local Mixing • Message Waiting Indication - Visual and Tone Based • Call Return • Call Back on Busy • Call Blocking with Toll Restriction • Delayed Disconnect • Distinctive Ringing - Calling and Called Number • Off-hook Warning Tone • Selective/Anonymous Call Rejection • Hot line and Warm Line Calling • Speed Dialing of 8 Numbers/Addresses • Music on Hold
Equipment used on experiment: • Lynksys PAP2T (ATA) • FreeSwitch (PBX) • Regular cord phone (GE) • Home network router with four ports
Connection sample from VoIP:
Let's say that you and your friend both have service through a VoIP provider. You both have your
analog phones hooked up to the service-provided ATAs. Let's take another look at that typical
telephone call, but this time using VoIP over a packet-switched network [6]:
1. You pick up the receiver, which sends a signal to the ATA. 2. The ATA receives the signal and sends a dial tone. This lets you know that you
have a connection to the Internet. 3. You dial the phone number of the party you wish to talk to. The tones are converted
by the ATA into digital data and temporarily stored. 4. The phone number data is sent in the form of a request to your VoIP company's call
processor. The call processor checks it to ensure that it's in a valid format. 5. The call processor determines to whom to map the phone number. In mapping, the
phone number is translated to an IP address (more on this later). The freeSwitch
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switch connects the two devices on either end of the call. On the other end, a signal is sent to your friend's ATA, telling it to ask the connected phone to ring.
6. Once your friend picks up the phone, a session is established between your computer and your friend's computer. This means that each system knows to expect packets of data from the other system. In the middle, the normal Internet infrastructure handles the call as if it were e-mail or a Web page. Each system must use the same protocol to communicate. The systems implement two channels, one for each direction, as part of the session.
7. You talk for a period of time. During the conversation, your system and your friend's system transmit packets back and forth when there is data to be sent. The ATAs at each end translate these packets as they are received and convert them to the analog audio signal that you hear. Your ATA also keeps the circuit open between itself and your analog phone while it forwards packets to and from the IP host at the other end.
8. You finish talking and hang up the receiver. 9. When you hang up, the circuit is closed between your phone and the ATA. 10. The ATA sends a signal to the soft switch connecting the call, terminating the
session. [6] 3.3.1 Some Programming samples:
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3.4 VOIP Using IP Phone
These specialized phones look just like normal phones with a handset, cradle and buttons. But
instead of having the standard RJ-11 phone connectors, IP phones have an RJ-45 Ethernet
connector. IP phones connect directly to your router and have all the hardware and software
necessary right onboard to handle the IP call. Wi-Fi Phones allow subscribing callers to make
VoIP calls from any Wi-Fi hot spot [7].
When started for the first time it will prompt for an IP address. When you choose some non-
existing one from your network you can administer the telephone from the local site computer
with a web browser. Every single setting you can make to the phone can be made also from the
site [7]. Here you can see what it looks like. It has three submenus – ‘status’, ‘basic settings’ and
advance settings
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You may configure the IP address as dynamic or static. I prefer a static IP so you can see my
network settings on the screenshot above. You can also adjust your time zone. The last
submenu Advances settings are the one where you register any user who can use the telephone.
Make sure the user is also registered on freeSwitch that will manage the calls [6]. In Advanced
Options you can choose your preferred voice coder. You can also choose the rate of the audio
codec. Initially I used G729 but the sound was robotic and had from 1 to 2 sec. delay. Changing it
to G723 made a big difference; it delivered a better sound. G.729 should not be used if your
network is all running on Ethernet [7]. The length of each frame was set to 30ms and check yes
for SIP registration.
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SIP port 5060 continuously “looks” for IP addresses to connect to the IP phone, ATA and
softphone.
Extensions use:
• 1000 for the softphone
• 1001 for the IP Phone
• 1002 for the ATA phone.
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Equipment use during experiment:
4. CONCLUSION The experiment was done with all the different types of VoIP. The reason for using the router with
four ports was because of the ability to test the softphone, the regular phone with an ATA
connector and an IP phone. Each phone acted like an extension and communicated with each
other. I encounter some bugs with the software, being that it is software that has not been
released to the public, this is understandable. Also a codec audio was used with the ATA and IP
phone; the best sound was obtained using the G.723 and the worse sound was obtained when I
used G.729. G.729 should not be use when the VoIP network is all running on Ethernet, and this
project was running on Ethernet. The windows firewall was a big problem at the beginning of the
project, because I forget to disable it. Also my laptop was connected to a wireless internet
connection and the router provided me with the local internet connection. It seems that the
freeSwitch got confused and did not find the IP address.
This was a very interesting project, there is still a lot of work to prepare freeSwitch to be
integrated with Windows; but there are so many nice things that can be done.
