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internet telephony


Voip Gateways

Saturday, December 30, 2006

Analog VoIP Gateways
SmartLink™ M-ATA Micro-Analog Telephone Adapter
SmartNode™ 4110 Series Analog VoIP Gateway with up to 8 FXS/FXO ports
Analog VoIP Routers
SmartLink™ 4020 Series Analog VoIP SoHo Router
SmartNode™ 4520 Series Analog VoIP Router with up to 8 FXS/FXO ports
SmartNode™ 4830 Series Analog VoIP IAD with up to 8 FXS/FXO ports and integrated WAN connectivity
SmartNode™ 4900 Series IpChannelBank™ Multi-Port FXS/FXO Analog VoIP Gateway Router with up to 32 ports
Digital VoIP Gateways
SmartNode™ S-DTA ISDN BRI VoIP Gateway
Digital VoIP Routers
SmartNode™ 1200 ISDN/BRI VoIP Gateway (2 channels)
SmartNode™ 1400 ISDN/BRI VoIP Gateway (4 channels)
SmartNode™ 4552 SoHo VoIP Router for ISDN/BRI (2 channels)
SmartNode™ 4562 Secure ISDN VoIP and VPN
SmartNode™ 4630 Series Multiport BRI So Gateway Router (up to 8 channels)
SmartNode™ 4650 G.SHDSL Multiport ISDN VoIP IAD with G.SHDSL.bis Interface
SmartNode™ 4960 T1/E1/PRI VoIP IAD
SmartNode™ 4960 G.SHDSL Multiport PRI VoIP IAD with G.SHDSL.bis Interface
SIPxNano Small to Medium Business IP-PBX
Modular VoIP Routers
SmartNode™ 2300 Modular VoIP Gateway Router
SmartNode™ 2400 Modular VoIP Gateway Router
SmartNode™ Interface Card IC-4BRV Interface Card with 4 ISDN BRI Interfaces
SmartNode™ Interface Card IC-4FXS Interface Card with 4 analog FXS ports
SmartNode™ Interface Card IC-E1V Interface Card for ISDN PRI Interface
SmartNode™ Interface Card IC-T1V Interface Card for ISDN PRI Interface
VoIP Telephones
SmartLink™ 4050 Series SIP Telephones

VoIP 101: Voice over IP Explained

Thursday, December 28, 2006

VoIP 101: Voice over IP Explained

For those who have never heard about the potential of VoIP, be prepared to radically change the way you think about your current long-distance calling plan. VoIP (Voice over Internet Protocol) is very simply, a method for taking ordinary analog audio signals and turning them into digital signals that can be sent over the Internet.

So what? Well, for those of you who are already paying a monthly fee for an Internet connection, this means that you can use that same connection to place free long distance phone calls. This process works by using already available VoIP software to make phone calls over the Internet, essentially circumventing phone companies and their service charges.

Interestingly, VoIP is not an entirely new thing. In fact, a number of providing companies have been around for some time. But it has only been with the more recent explosion of high-speed internet access usage, that VoIP has gotten any attention. Now the major telephone carriers are setting up their own VoIP calling plans throughout the US, another testament to the potential of the technology.

How VoIP Is Used

While there are a number of ways that VoIP is currently being used, most individual callers fall into one of three categories: ATA, IP Phones, and Computer-to-Computer.

ATA or Analog Telephone Adaptor, is the most common way of using VoIP. This adaptor actually allows you to hook up the phone that is already in your house, to your computer, and then your Internet connection. What the ATA does, is turn the analog signals your phone sends out into digital signals that can be sent over the Internet. Setting up this system is quite simple. It simply requires that you order an ATA (its an adaptor remember), plug the cable from your phone which would normally go into the wall socket into the ATA, and then the ATA gets plugged into your computer, which is connected to the internet. Some ATAs include software that has to be installed on your computer before its ready, but basically it's quite a simple process. Then you are ready to make some calls.

The next type of VoIP usage utilizes IP Phones instead of your home phone. The IP Phone looks just like a normal phone, with all the same buttons and cradle, the only difference is that instead of having a normal wall jack connector, it has an Ethernet connector. This means, that instead of plugging in your IP phone to the wall jack like you would with a regular analog phone, it gets plugged directly into your router. This option allows you to circumvent your personal computer, and it also means that you will not have to install any software, because its all built in to the handset. In addition, the fact that Wi-Fi IP phones will soon be available, which will allow subscribing callers to make VoIP calls from any Wi-Fi hot spot, make this option an exciting possibility.

The simplest and cheapest way to use VoIP is through computer-to-computer calls. These calls are entirely free, meaning no calling plan whatsoever. The only thing you need, is the software which can be found for free on the internet, a good internet connection, a microphone, speakers, and a sound card. Except for your monthly internet service fee, there is literally no cost for making these calls, no matter how many you make.

For large companies, VoIP also offers some very unique possibilities. Some larger companies are already utilizing the technology by conducting all intra-office calls through a VoIP network. Because the quality of sound is comparable to and in some cases surpasses that of analog service, some international companies are using VoIP to route international calls through the branch of their company nearest the call's destination and then completing it on an analog system. This allows them to pay local rates internationally and still utilize the same intra-office VoIP network that they would if they were calling someone in the next cubicle over.

Other Advantages of VoIP

While your current long-distance plan covers you for only one location, say calls made from your office, with VoIP, you can make a call anywhere that you can get a broadband connection. That is because all three methods above, unlike analog calls, send the call information via the Internet. This means you can make calls from home, on vacation, on business trips, and almost anywhere else. Anywhere you go, with VoIP you can bring your home phone along with you. In the same way, computer-to-computer connections mean that as long as you have your laptop and a connection, you're ready to go.

There are also some nifty benefits to having your calls transmitted over the Internet. For example, some VoIP service providers allow you to check your voicemail via your e-mail, while others allow you to attach voice messages to your e-mails.

How VoIP Works

The current phone system relies on a reliable but largely inefficient method for connecting calls known as circuit switching. This technique, which has been used for over 100 years, means that when a call is made between two people a connection is maintained in both directions between callers for the duration of the call. This dual directional characteristic gives the system the name circuit.

If, for example, you made a 30-minute call the circuit would be continuously open, and thus used, between the two phones. Up until about 1960, this meant that every call had to have an actual dedicated wire connecting the two phones. Thus a long distance call cost so much, because you were paying for pieces of copper wire to be connected all the way from your phone to the destination phone, and for that connection to remain constant throughout the call. Today, however, your analog call is converted after leaving your house to a digital signal, where your call can be combined with many others on a single fiber optic cable. While this system is certainly an improvement over the past copper wire system, it is still quite inefficient. This inefficiency is due in part to the fact that the telephone line can't distinguish between useful talking and unneeded silences. For example, in a typical conversation while one person is talking the other person is listening. Thus the current analog system uses roughly half its space sending useless messages like this silence. But there is also more information, even down to pauses in speech, which under a more efficient system can be effectively cut out rather than wasting the circuit space. This idea of only transmitting the noisy bits of a telephone call and saving a great deal on circuit space, is the basis of Packet-Switching, the alternative method to circuit switching that the VoIP phone system uses.

Packet-Switching is the same method that you use when you view a website. For example, as you read this website, your computer is not maintaining a constant connection to the site, but rather making connections to send and receive information only on an as needed basis (such as when you click on a link). Just as this system allows the transfer of information over the Internet to work so quickly, so also does it work in the VoIP system. While circuit switching maintains a constant and open connection, packet switching opens connections just long enough to send bits of data called packets from one computer to another. This allows the network to send your call (in packets) along the least congested and cheapest lines available, while also keeping your computer or IP phone, free to send and receive messages and calls with other computers. This way of sending information, not to mention data compression, makes the amount of information which must be transmitted for every call at least 3-4 times less for VoIP than the exact same call in a conventional telephone system. For this reason, VoIP is so much cheaper than conventional calling plans.

The Future of VoIP

While most analysts believe it will be at least a decade before companies and telephone providers make the full switch to VoIP, the potential for the technology's use today is already quite astounding. A report by the Forrester Research Group predicts that by the end of 2006, nearly 5 million U.S. households will be using VoIP phone service. With the savings and flexibility that the technology already offers, and new advances just ahead on the horizon, we can expect those numbers will only increase in the future.

Voip Technology

Sunday, December 10, 2006

Overview on a VoIP connection

To setup a VoIP communication we need:

  1. First the ADC to convert analog voice to digital signals (bits)
  2. Now the bits have to be compressed in a good format for transmission: there is a number of protocols we'll see after.
  3. Here we have to insert our voice packets in data packets using a real-time protocol (typically RTP over UDP over IP)
  4. We need a signaling protocol to call users: ITU-T H323 does that.
  5. At RX we have to disassemble packets, extract datas, then convert them to analog voice signals and send them to sound card (or phone)
  6. All that must be done in a real time fashion cause we cannot waiting for too long for a vocal answer! (see QoS section)

Base architecture

Voice )) ADC - Compression Algo - Assembling RTP in TCP/IP ----
----> |
<---- | Voice (( DAC - Decompress. Algo - Disass. RTP from TCP/IP ----

Analog to Digital Conversion

This is made by hardware, typically by card integrated ADC.

Today every sound card allows you convert with 16 bit a band of 22050 Hz (for sampling it you need a freq of 44100 Hz for Nyquist Principle) obtaining a throughput of 2 bytes * 44100 (samples per second) = 88200 Bytes/s, 176.4 kBytes/s for stereo stream.

For VoIP we needn't such a throughput (176kBytes/s) to send voice packet: next we'll see other coding used for it.

Compression Algorithms

Now that we have digital data we may convert it to a standard format that could be quickly transmitted.

PCM, Pulse Code Modulation, Standard ITU-T G.711

  • Voice bandwidth is 4 kHz, so sampling bandwidth has to be 8 kHz (for Nyquist).
  • We represent each sample with 8 bit (having 256 possible values).
  • Throughput is 8000 Hz *8 bit = 64 kbit/s, as a typical digital phone line.
  • In real application mu-law (North America) and a-law (Europe) variants are used which code analog signal a logarithmic scale using 12 or 13 bits instead of 8 bits (see Standard ITU-T G.711).

ADPCM, Adaptive differential PCM, Standard ITU-T G.726

It converts only the difference between the actual and the previous voice packet requiring 32 kbps (see Standard ITU-T G.726).

LD-CELP, Standard ITU-T G.728
CS-ACELP, Standard ITU-T G.729 and G.729a
MP-MLQ, Standard ITU-T G.723.1, 6.3kbps, Truespeech
ACELP, Standard ITU-T G.723.1, 5.3kbps, Truespeech
LPC-10, able to reach 2.5 kbps!!

This last protocols are the most important cause can guarantee a very low minimal band using source coding; also G.723.1 codecs have a very high MOS (Mean Opinion Score, used to measure voice fidelity) but attention to elaboration performance required by them, up to 26 MIPS!

RTP Real Time Transport Protocol

Now we have the raw data and we want to encapsulate it into TCP/IP stack. We follow the structure:

VoIP data packets
I,II layers

VoIP data packets live in RTP (Real-Time Transport Protocol) packets which are inside UDP-IP packets.

Firstly, VoIP doesn't use TCP because it is too heavy for real time applications, so instead a UDP (datagram) is used.

Secondly, UDP has no control over the order in which packets arrive at the destination or how long it takes them to get there (datagram concept). Both of these are very important to overall voice quality (how well you can understand what the other person is saying) and conversation quality (how easy it is to carry out a conversation). RTP solves the problem enabling the receiver to put the packets back into the correct order and not wait too long for packets that have either lost their way or are taking too long to arrive (we don't need every single voice packet, but we need a continuous flow of many of them and ordered).

                    Real Time Transport Protocol

0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|V=2|P|X| CC |M| PT | sequence number |
| timestamp |
| synchronization source (SSRC) identifier |
| contributing source (CSRC) identifiers |
| .... |


  • V indicates the version of RTP used
  • P indicates the padding, a byte not used at bottom packet to reach the parity packet dimension
  • X is the presence of the header extension
  • CC field is the number of CSRC identifiers following the fixed header. CSRC field are used, for example, in conference case.
  • M is a marker bit
  • PT payload type

For a complete description of RTP protocol and all its applications see relative RFCs 1889 and 1890.


There are also other protocols used in VoIP, like RSVP, that can manage Quality of Service (QoS).

RSVP is a signaling protocol that requests a certain amount of bandwidth and latency in every network hop that supports it.

For detailed info about RSVP see the RFC 2205

Quality of Service (QoS)

We said many times that VoIP applications require a real-time data streaming cause we expect an interactive data voice exchange.

Unfortunately, TCP/IP cannot guarantee this kind of purpose, it just make a "best effort" to do it. So we need to introduce tricks and policies that could manage the packet flow in EVERY router we cross.

So here are:

  1. TOS field in IP protocol to describe type of service: high values indicate low urgency while more and more low values bring us more and more real-time urgency
  2. Queuing packets methods:
    1. FIFO (First in First Out), the more stupid method that allows passing packets in arrive order.
    2. WFQ (Weighted Fair Queuing), consisting in a fair passing of packets (for example, FTP cannot consume all available bandwidth), depending on kind of data flow, typically one packet for UDP and one for TCP in a fair fashion.
    3. CQ (Custom Queuing), users can decide priority.
    4. PQ (Priority Queuing), there is a number (typically 4) of queues with a priority level each one: first, packets in the first queue are sent, then (when first queue is empty) starts sending from the second one and so on.
    5. CB-WFQ (Class Based Weighted Fair Queuing), like WFQ but, in addition, we have classes concept (up to 64) and the bandwidth value associated for each one.
  3. Shaping capability, that allows to limit the source to a fixed bandwidth in:
    1. download
    2. upload
  4. Congestion Avoidance, like RED (Random Early Detection).

For an exhaustive information about QoS see Differentiated Services at IETF.

H323 Signaling Protocol

H323 protocol is used, for example, by Microsoft Netmeeting to make VoIP calls.

This protocol allow a variety of elements talking each other:

  1. Terminals, clients that initialize VoIP connection. Although terminals could talk together without anyone else, we need some additional elements for a scalable vision.
  2. Gatekeepers, that essentially operate:
    1. address translation service, to use names instead IP addresses
    2. admission control, to allow or deny some hosts or some users
    3. bandwidth management
  3. Gateways, points of reference for conversion TCP/IP - PSTN.
  4. Multipoint Control Units (MCUs) to provide conference.
  5. Proxies Server also are used.

h323 allows not only VoIP but also video and data communications.

Concerning VoIP, h323 can carry audio codecs G.711, G.722, G.723, G.728 and G.729 while for video it supports h261 and h263.

More info about h323 is available at Openh323 Standards, at this h323 web site and at its standard description: ITU H-series Recommendations.

You can find it implemented in various application software like Microsoft Netmeeting, Net2Phone, DialPad, ... and also in freeware products you can find at Openh323 Web Site.


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