-This document explains about VoIP systems. Recent happenings like Internet - diffusion at low cost, new integration of dedicated voice compression processors, - have changed common user requirements allowing VoIP standards to diffuse. This - howto tries to define some basic lines of VoIP architecture. -
-
-Please send suggestions and critics to
-Copyright (C) 2000,2001 Roberto Arcomano. This document is free; you can - redistribute it and/or modify it under the terms of the GNU General Public - License as published by the Free Software Foundation; either version 2 of the - License, or (at your option) any later version. This document is distributed - in the hope that it will be useful, but -
-
-WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY
- or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for
- more details. You can get a copy of the GNU GPL
-If you want to translate this document you are free, you only have to: - -
-
-
-Warning! You don't have to translate TXT or HTML file, you have to modify
- LYX file, so that it is possible to convert it all other formats (TXT, HTML,
- RIFF, etc.): to do that you can use "LyX" application you download from
-No need to ask me to translate! You just have to let me know (if you want) - about your translation. -
--Thank you for your translation! -
-
-Thanks to
-Thanks to
-Thanks to
-More than 30 years ago Internet didn't exist. Interactive communications
+ This document explains about VoIP systems. Recent happenings
+ like Internet diffusion at low cost, new integration of dedicated
+ voice compression processors, have changed common user requirements
+ allowing VoIP standards to diffuse. This howto tries to define some
+ basic lines of VoIP architecture.
+ Please send suggestions and critics to Copyright (C) 2000,2001 Roberto Arcomano. This document is free;
+ you can redistribute it and/or modify it under the terms of the GNU
+ General Public License as published by the Free Software Foundation;
+ either version 2 of the License, or (at your option) any later version.
+ This document is distributed in the hope that it will be useful,
+ but
+ WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY
+ or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License
+ for more details. You can get a copy of the GNU GPL If you want to translate this document you are free, you only
+ have to:
+ Warning! You don't have to translate TXT or HTML file, you have
+ to modify LYX file, so that it is possible to convert it all other
+ formats (TXT, HTML, RIFF, etc.): to do that you can use "LyX" application
+ you download from No need to ask me to translate! You just have to let me know
+ (if you want) about your translation.
+ Thank you for your translation!
+ Thanks to Thanks to Thanks to More than 30 years ago Internet didn't exist. Interactive communications
were only made by telephone at PSTN line cost.
-
-Data exchange was expansive (for a long distance) and no one had been thinking
- to video interactions (there was only television that is not interactive, as
- known).
-
-Few years ago we saw appearing some interesting things: PCs to large masses,
- new technologies to communicate like cellular phones and finally the great
- net: Internet; people begun to communicate with new services like email, chat,
- etc. and business reborned with the web allowing people buy with a "click".
-
-Today we can see a real revolution in communication world: everybody begins
- to use PCs and Internet for job and free time to communicate each other, to
- exchange data (like images, sounds, documents) and, sometimes, to talk each
- other using applications like Netmeeting or Internet Phone. Particularly starts
- to diffusing a common idea that could be the future and that can allow real-time
- vocal communication: VoIP.
-
-We cannot know what is the future, but we can try to image it with many
- computers, Internet almost everywhere at high speed and people talking (audio
- and video) in a real time fashion. We only need to know what will be the means
- to do this: UMTS, VoIP (with video extension) or other? Anyway we can notice
- that Internet has grown very much in the last years, it is free (at least as
- international means) and could be the right communication media for future.
-
-VoIP stands for 'V'oice 'o'ver 'I'nternet 'P'rotocol. As the term says
- VoIP tries to let go voice (mainly human) through IP packets and, in definitive
- through Internet. VoIP can use accelerating hardware to achieve this purpose
- and can also be used in a PC environment.
-
-Many years ago we discovered that sending a signal to a remote destination
- could have be done also in a digital fashion: before sending it we have to
- digitalize it with an ADC (analog to digital converter), transmit it, and at
- the end transform it again in analog format with DAC (digital to analog converter)
- to use it.
-
-VoIP works like that, digitalizing voice in data packets, sending them
- and reconverting them in voice at destination.
-
-Digital format can be better controlled: we can compress it, route it,
- convert it to a new better format, and so on; also we saw that digital signal
- is more noise tolerant than the analog one (see GSM vs TACS).
-
-TCP/IP networks are made of IP packets containing a header (to control
- communication) and a payload to transport data: VoIP use it to go across the
- network and come to destination.
-
-
-When you are using PSTN line, you typically pay for time used to a PSTN
- line manager company: more time you stay at phone and more you'll pay. In addition
- you couldn't talk with other that one person at a time.
-
-In opposite with VoIP mechanism you can talk all the time with every person
- you want (the needed is that other person is also connected to Internet at
- the same time), as far as you want (money independent) and, in addition, you
- can talk with many people at the same time.
-
-If you're still not persuaded you can consider that, at the same time,
- you can exchange data with people are you talking with, sending images, graphs
- and videos.
-
-Unfortunately we have to report some problem with the integration between
- VoIP architecture and Internet. As you can easy imagine, voice data communication
- must be a real time stream (you couldn't speak, wait for many seconds, then
- hear other side answering): this is in contrast with the Internet heterogeneous
- architecture that can be made of many routers (machines that route packets),
- about 20-30 or more and can have a very high round trip time (RTT), so we need
- to modify something to get it properly working.
-
-In next sections we'll try to understand how to solve this great problem.
- In general we know that is very difficult to guarantee a bandwidth in Internet
- for VoIP application.
-
-Here we see some important info about VoIP, needed to understand it.
-
-To setup a VoIP communication we need:
-
-
- Data exchange was expansive (for a long distance) and no one
+ had been thinking to video interactions (there was only television
+ that is not interactive, as known).
+ Few years ago we saw appearing some interesting things: PCs to
+ large masses, new technologies to communicate like cellular phones
+ and finally the great net: Internet; people begun to communicate
+ with new services like email, chat, etc. and business reborned with
+ the web allowing people buy with a "click".
+ Today we can see a real revolution in communication world: everybody
+ begins to use PCs and Internet for job and free time to communicate
+ each other, to exchange data (like images, sounds, documents) and,
+ sometimes, to talk each other using applications like Netmeeting
+ or Internet Phone. Particularly starts to diffusing a common idea
+ that could be the future and that can allow real-time vocal communication:
+ VoIP.
+ We cannot know what is the future, but we can try to image it
+ with many computers, Internet almost everywhere at high speed and
+ people talking (audio and video) in a real time fashion. We only
+ need to know what will be the means to do this: UMTS, VoIP (with
+ video extension) or other? Anyway we can notice that Internet has
+ grown very much in the last years, it is free (at least as international
+ means) and could be the right communication media for future.
+ VoIP stands for 'V'oice 'o'ver 'I'nternet 'P'rotocol. As the
+ term says VoIP tries to let go voice (mainly human) through IP packets
+ and, in definitive through Internet. VoIP can use accelerating hardware
+ to achieve this purpose and can also be used in a PC environment.
+ Many years ago we discovered that sending a signal to a remote
+ destination could have be done also in a digital fashion: before
+ sending it we have to digitalize it with an ADC (analog to digital
+ converter), transmit it, and at the end transform it again in analog
+ format with DAC (digital to analog converter) to use it.
+ VoIP works like that, digitalizing voice in data packets, sending
+ them and reconverting them in voice at destination.
+ Digital format can be better controlled: we can compress it,
+ route it, convert it to a new better format, and so on; also we saw
+ that digital signal is more noise tolerant than the analog one (see
+ GSM vs TACS).
+ TCP/IP networks are made of IP packets containing a header (to
+ control communication) and a payload to transport data: VoIP use
+ it to go across the network and come to destination.
+ When you are using PSTN line, you typically pay for time used
+ to a PSTN line manager company: more time you stay at phone and more
+ you'll pay. In addition you couldn't talk with other that one person
+ at a time.
+ In opposite with VoIP mechanism you can talk all the time with
+ every person you want (the needed is that other person is also connected
+ to Internet at the same time), as far as you want (money independent)
+ and, in addition, you can talk with many people at the same time.
+
+ If you're still not persuaded you can consider that, at the same
+ time, you can exchange data with people are you talking with, sending
+ images, graphs and videos.
+ Unfortunately we have to report some problem with the integration
+ between VoIP architecture and Internet. As you can easy imagine,
+ voice data communication must be a real time stream (you couldn't
+ speak, wait for many seconds, then hear other side answering): this
+ is in contrast with the Internet heterogeneous architecture that
+ can be made of many routers (machines that route packets), about
+ 20-30 or more and can have a very high round trip time (RTT), so
+ we need to modify something to get it properly working.
+ In next sections we'll try to understand how to solve this great
+ problem. In general we know that is very difficult to guarantee a
+ bandwidth in Internet for VoIP application.
+ Here we see some important info about VoIP, needed to understand
+ it.
+ To setup a VoIP communication we need:
+
-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.
-
-Now that we have digital data we may convert it to a standard format that
- could be quickly transmitted.
-
-
-
-
-It converts only the difference between the actual and the previous voice
- packet requiring 32 kbps (see Standard ITU-T G.726).
-
- 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.
+ Now that we have digital data we may convert it to a standard
+ format that could be quickly transmitted.
+ It converts only the difference between the actual and the previous
+ voice packet requiring 32 kbps (see Standard ITU-T G.726).
+
-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!
-
-Now we have the raw data and we want to encapsulate it into TCP/IP stack.
- We follow the structure:
-
- 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!
+ Now we have the raw data and we want to encapsulate it into TCP/IP
+ stack. We follow the structure:
+
-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).
-
- 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).
+
-Where:
-
-
-For a complete description of RTP protocol and all its applications see
- relative RFCs
-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
-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:
-
-
-For an exhaustive information about QoS see
-H323 protocol is used, for example, by Microsoft Netmeeting to make VoIP
- calls.
-
-This protocol allow a variety of elements talking each other:
-
-
-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
-You can find it implemented in various application software like
-To create a little VoIP system you need the following hardware:
-
-
-All that has to be present twice to simulate a standard communication.
-
-The tool above are the minimal requirement for a VoIP connection: next
- we'll see that we should (and in Internet we must) use more hardware to do
- the same in a real situation.
-
-Sound card has be full duplex unless we couldn't hear anything while speaking!
-
-As additional you can use hardware cards (see next) able to manage data
- stream in a compressed format (see Par 4.3).
-
-We can use special cards with hardware accelerating capability. Two of
- them (and also the only ones directly managed by the Linux kernel at this moment)
- are the
-
-
-Quicknet PhoneJack is a sound card that can use standard algorithms to
- compress audio stream like G723.1 (section 4.3) down to 4.1 Kbps rate.
-
-It can be connected directly to a phone (POTS port) or a couple mic-speaker.
-
-It has a ISA or PCI connector bus.
-
-Quicknet LineJack works like PhoneJack with some addition features (see
- next).
-
-VoiceTronix V4PCI is a PCI card pretty like Quicknet LineJack but with
- 4 phone ports
-
-VoiceTronix VPB4 is a ISA card equivalent to V4PCI.
-
-VoiceTronix VPB8L is a logging card with 8 ports.
-
-For more info see
-Quicknet LineJack and VoiceTronix cards can be connected to a PSTN line
- allowing VoIP gateway feature.
-
-Then you'll need a software to manage it (see after).
-
-We can choose what O.S. to use:
-
-
-Under Win9x we have Microsoft Netmeeting, Internet Phone, DialPad or others
- or Internet Switchboard (from
-Warning!!: Latest Quicknet cards using Swithboard (older version too) NEED
- to be connected to Internet to get working for managing Microtelco account
- (not free of charge), so if you plan to remain isolated from Internet you need
- to install
-For VoiceTronix cards you can find software at
-Under Linux we have free software
-Attention: all Openh323 source code has to be compiled in a user directory
- (if not it is necessary to change some environment variable). You are warned
- that compiling time could be very high and you could need a lot of RAM to make
- it in a decent time.
-
-To manage gateway feature (join TCP/IP VoIP to PSTN lines) you need some
- kind of software like this:
-
-
-You can choose as gatekeeper:
-
-
-
-
-Here we see how to configure special hardware card in Linux and Windows
- environment.
-
-As we saw, Quicknet Phonejack is a sound card with VoIP accelerating capability.
- It supports:
-
-
-Quicknet PhoneJack is a ISA (or PCI) card to install into your Pc box.
- It can work without an IRQ.
-
-Under Windows you have to install:
-
-
-all downloadable from
-After Switchboard has been installed, you need to register to Quicknet
- to obtain full capability of your card.
-
-When you pick up the phone Internet Switchboard wakes up and waits for
- your calling number (directly entered from your phone), you can:
-
-
-Internet Swichboard is h323 compatible, so if you can use, for example,
- Microsoft Netmeeting at the other end to talk.
-
-Warning!! Internet Switchboard NEED to be connected to Internet when used
- with newer Quicknet cards
-
-In place of Internet Switchboard you can use openh323 application
-Under Linux you have to install:
-
-
-With Internet Switchboard (and with other application) you can:
-
-
-This card is very similar to the previous, it supports also gateway feature.
+ Where:
+ For a complete description of RTP protocol and all its applications
+ see relative RFCs 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 We said many times that VoIP applications require a real-time
+ data streaming cause we expect an interactive data voice exchange.
-
-We only notice that we have to
-
-Follow README file for more help.
-
-I personally haven't tested VoiceTronix products so please contact
-In this chapter we try to setup VoIP system, simple at first, then more
- and more complex.
-
- 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:
+ For an exhaustive information about QoS see H323 protocol is used, for example, by Microsoft Netmeeting to
+ make VoIP calls.
+ This protocol allow a variety of elements talking each other:
+ 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 You can find it implemented in various application software like
+ To create a little VoIP system you need the following hardware:
+ All that has to be present twice to simulate a standard communication.
+ The tool above are the minimal requirement for a VoIP connection:
+ next we'll see that we should (and in Internet we must) use more
+ hardware to do the same in a real situation.
+ Sound card has be full duplex unless we couldn't hear anything
+ while speaking!
+ As additional you can use hardware cards (see next) able to manage
+ data stream in a compressed format (see Par 4.3).
+ We can use special cards with hardware accelerating capability.
+ Two of them (and also the only ones directly managed by the Linux
+ kernel at this moment) are the
+ Quicknet PhoneJack is a sound card that can use standard algorithms
+ to compress audio stream like G723.1 (section 4.3) down to 4.1 Kbps
+ rate.
+ It can be connected directly to a phone (POTS port) or a couple
+ mic-speaker.
+ It has a ISA or PCI connector bus.
+ Quicknet LineJack works like PhoneJack with some addition features
+ (see next).
+ VoiceTronix V4PCI is a PCI card pretty like Quicknet LineJack
+ but with 4 phone ports
+ VoiceTronix VPB4 is a ISA card equivalent to V4PCI.
+ VoiceTronix VPB8L is a logging card with 8 ports.
+ For more info see Quicknet LineJack and VoiceTronix cards can be connected to a
+ PSTN line allowing VoIP gateway feature.
+ Then you'll need a software to manage it (see after).
+ We can choose what O.S. to use:
+ Under Win9x we have Microsoft Netmeeting, Internet Phone, DialPad
+ or others or Internet Switchboard (from Warning!!: Latest Quicknet cards using Swithboard (older version
+ too) NEED to be connected to Internet to get working for managing
+ Microtelco account (not free of charge), so if you plan to remain
+ isolated from Internet you need to install For VoiceTronix cards you can find software at Under Linux we have free software Attention: all Openh323 source code has to be compiled in a user
+ directory (if not it is necessary to change some environment variable).
+ You are warned that compiling time could be very high and you could
+ need a lot of RAM to make it in a decent time.
+ To manage gateway feature (join TCP/IP VoIP to PSTN lines) you
+ need some kind of software like this:
+ You can choose as gatekeeper:
+ Here we see how to configure special hardware card in Linux and
+ Windows environment.
+ As we saw, Quicknet Phonejack is a sound card with VoIP accelerating
+ capability. It supports:
+ Quicknet PhoneJack is a ISA (or PCI) card to install into your
+ Pc box. It can work without an IRQ.
+ Under Windows you have to install:
+ all downloadable from After Switchboard has been installed, you need to register to
+ Quicknet to obtain full capability of your card.
+ When you pick up the phone Internet Switchboard wakes up and
+ waits for your calling number (directly entered from your phone),
+ you can:
+ Internet Swichboard is h323 compatible, so if you can use, for
+ example, Microsoft Netmeeting at the other end to talk.
+ Warning!! Internet Switchboard NEED to be connected to Internet
+ when used with newer Quicknet cards
+ In place of Internet Switchboard you can use openh323 application
+ Under Linux you have to install:
+ With Internet Switchboard (and with other application) you can:
+ This card is very similar to the previous, it supports also gateway
+ feature.
+ We only notice that we have to Follow README file for more help.
+ I personally haven't tested VoiceTronix products so please contact
+ In this chapter we try to setup VoIP system, simple at first,
+ then more and more complex.
+
-A and B should:
-
-
-In this kind of view A can make a H323 call to B (if B has Netmeeting active)
- using B IP address. Then B can answer to it if it wants. After accepting call,
- VoIP data packets start to pass.
-
-If you use Microsoft Windows in a lan you can call the other side using
- NetBIOS name. NetBIOS is a protocol that can work (stand over) with NetBEUI
- low level protocol and also with TCP/IP. It is only need to call the "computer
- name" on the other side to make a connection.
-
- A and B should have
+ In this kind of view A can make a H323 call to B (if B has server
+ side application active) using B IP address. Then B can answer to
+ it if it wants. After accepting call, VoIP data packets start to
+ flow.
+ Under Microsoft Windows a NetBIOS name can be used instead of
+ an IP address.
+
-This is possible cause John call request to Alice is converted to IP calling
- by the NetBIOS protocol.
-
-The above 2 examples are very easy to implement but aren't scalable.
-
-In a more big view such as Internet it is impossible to use direct calling
- cause, usually, the callers don't know the destination IP address. Furthermore
- NetBIOS naming feature cannot work cause it uses broadcast messages, which
- typically don't pass ISP routers .
-
-The NetBIOS name calling idea can be implemented also in a Internet environment,
- using a WINS server: NetBIOS clients can be configured to use a WINS server
- to resolve names.
-
-PCs using the same WINS server will be able to make direct calling between
- them.
-
- This is possible cause John call request to Alice is converted
+ to IP calling by the NetBIOS protocol.
+ The above 2 examples are very easy to implement but aren't scalable.
+
+ In a more big view such as Internet it is impossible to use direct
+ calling cause, usually, the callers don't know the destination IP
+ address. Furthermore NetBIOS naming feature cannot work cause it
+ uses broadcast messages, which typically don't pass ISP routers .
+ You can also use DNS to solve name in IP address: for example
+ you can call ''box.domain.com''.
+ The NetBIOS name calling idea can be implemented also in a Internet
+ environment, using a WINS server: NetBIOS clients can be configured
+ to use a WINS server to resolve names.
+ PCs using the same WINS server will be able to make direct calling
+ between them.
+
-A, B, C and D are in different subnets, but they can call each other in
- a NetBIOS name calling fashion. The needed is that all are using S as WINS
- Server.
-
-Note: WINS server hasn't very high performance cause it use NetBIOS feature
- and should only be used for joining few subnets.
-
-A problem of few IPs is commonly solved using the so called masquering
- (also NAT, network address translation): there is only 1 IP public address
- (that Internet can directly "see"), the others machines are "masqueraded" using
- all this IP.
-
- A, B, C and D are in different subnets, but they can call each
+ other in a NetBIOS name calling fashion. The needed is that all are
+ using S as WINS Server.
+ Note: WINS server hasn't very high performance cause it use NetBIOS
+ feature and should only be used for joining few subnets.
+ ILS is a kind of server which allows you to solve your name during
+ an H323 calling: when you start VoIP application you first register
+ to ILS server using a name, then everyone will be able to see you
+ using that name (if he uses same Server ILS!).
+ A problem of few IPs is commonly solved using the so called masquering
+ (also NAT, network address translation): there is only 1 IP public
+ address (that Internet can directly "see"), the others machines are
+ "masqueraded" using all this IP.
+
-In the example A,B and C can navigate, pinging, using mail and news services
- with Internet people, but they CANNOT make a VoIP call. This because H323 protocol
- send IP address at application level, so the answer will never arrive to source
- (that is using a private IP address).
-
-Solutions:
-
-
- In the example A,B and C can navigate, pinging, using mail and
+ news services with Internet people, but they CANNOT make a VoIP call.
+ This because H323 protocol send IP address at application level,
+ so the answer will never arrive to source (that is using a private
+ IP address).
+ Solutions:
+
-
-
-With Linux (as an h323 terminal) you can experiment everything done before.
-
-Sintax is:
-
-"ohphone -l|--listen [options]"
-
-"ohphone [options]... address"
-
-
-Also, when you start ohphone, you can give command to the interpreter directly
- (like decrease AEC, Automatic Echo Cancellation).
-
-You can also experiment gatekeeper feature
-
- Sintax is:
+ "ohphone -l|--listen [options]"
+ "ohphone [options]... address"
+ Also, when you start ohphone, you can give command to the interpreter
+ directly (like decrease AEC, Automatic Echo Cancellation).
+ Gnomemeeting is an application using GUI interface to make call
+ using VoIP. It is very simple to use and allows you to use ILS server,
+ chat and other things.
+ You can also experiment gatekeeper feature
+
-
-We have to notice that the Gatekeeper is able only to solve name in IP
- address, it couldn't join hosts that aren't reachable each other (at IP level),
- in other words it couldn't act as a NAT router.
-
-You can find gatekeeper code
-Program has only to be launch with -d (as daemon) or -x (execute) parameter.
-
-
-In addition you can use a config file (.ini) you find
-As we said, gateway is an entity that can join VoIP to PSTN lines allowing
- us to made call from Internet to a classic telephone. So, in addition, we need
- a card that could manage PSTN lines: Quicknet LineJack does it.
-
-From
-
-If executable doesn't work you need to download source code and
-After that you only need to launch PSTNGw to start your H323 gateway.
-
-First Matrix refers to:
-
-
- We have to notice that the Gatekeeper is able only to solve name
+ in IP address, it couldn't join hosts that aren't reachable each
+ other (at IP level), in other words it couldn't act as a NAT router.
+ You can find gatekeeper code Program has only to be launch with -d (as daemon) or -x (execute)
+ parameter.
+ In addition you can use a config file (.ini) you find As we said, gateway is an entity that can join VoIP to PSTN lines
+ allowing us to made call from Internet to a classic telephone. So,
+ in addition, we need a card that could manage PSTN lines: Quicknet
+ LineJack does it.
+ From If executable doesn't work you need to download source code and
+ After that you only need to launch PSTNGw to start your H323
+ gateway.
+ First Matrix refers to:
+
-Second Matrix refers to Gateway softwares that manage LineJACK card.
-
- Second Matrix refers to Gateway softwares that manage LineJACK
+ card.
+
-Notation:
-
-
-From all we said before we noticed that we still have not solved problems
- about bandwidth, how to create a real time streaming of data.
-
-We know we couldn't find a solution unless we enable a right real-time
- manager protocol in each router we cross, so what do we can do?
-
-First we try to use a very (as more as possible) high rate compression
- algorithms (like LPC10 which only consumes a 2.5 kbps bandwidth, about 313
- bytes/s).
-
-Then we starts classify our packets, in TOS field, with the most high priority
- level, so every router help us having urgently.
-
-Important: all that is not sufficient to guarantee our conversation would
- always be ok, but without an great infrastructure managing shaping, bandwidth
- reservation and so on, it is not possible to do it, TCP/IP is not a real time
- protocol.
-
-A possible solution could be starts with little WAN at guaranteed bandwidth
- and get larger step by step.
-
-We finally have to notice a thing: also the so called guaranteed services
- like PSTN line could not manage all clients they have: for example a GSM call
- is not able to manage more that some hundred or some thousand of clients.
-
-Anyway for a starting service, limited to few users, VoIP can be a valid
- alternative to classic PSTN service.
-
-PSTN: Public Switched Telephone Network
-
-VoIP: Voice over Internet Protocol
-
-LAN: Local Area Network
-
-WAN: Wide Area Network
-
-TOS: Type Of Service
-
-ISP: Internet Service Provider
-
-RTP: Real Time Protocol
-
-RSVP: ReSerVation Protocol
-
-QoS: Quality of Service
-
-
- Notation:
+ VoIP becomes very interesting when you start to use PSTN lines
+ to call other people in the world, directly to their home telephone.
+ A typical application is like that:
+ So your decision will be taken considering PSTN line costs. In
+ fact what VoIP does is the convert this:
+ into this:
+ To save money you need that:
+ Typically "short distance calling" refers to a "city cal" while "great
+ distance calling" can be an "intercontinental call"!
+ From all we said before we noticed that we still have not solved
+ problems about bandwidth, how to create a real time streaming of
+ data.
+ We know we couldn't find a solution unless we enable a right
+ real-time manager protocol in each router we cross, so what do we
+ can do?
+ First we try to use a very (as more as possible) high rate compression
+ algorithms (like LPC10 which only consumes a 2.5 kbps bandwidth,
+ about 313 bytes/s).
+ Then we starts classify our packets, in TOS field, with the most
+ high priority level, so every router help us having urgently.
+ Important: all that is not sufficient to guarantee our conversation
+ would always be ok, but without an great infrastructure managing
+ shaping, bandwidth reservation and so on, it is not possible to do
+ it, TCP/IP is not a real time protocol.
+ A possible solution could be starts with little WAN at guaranteed
+ bandwidth and get larger step by step.
+ We finally have to notice a thing: also the so called guaranteed
+ services like PSTN line could not manage all clients they have: for
+ example a GSM call is not able to manage more that some hundred or
+ some thousand of clients.
+ Anyway for a starting service, limited to few users, VoIP can
+ be a valid alternative to classic PSTN service.
+ PSTN: Public Switched Telephone Network
+ VoIP: Voice over Internet Protocol
+ LAN: Local Area Network
+ WAN: Wide Area Network
+ TOS: Type Of Service
+ ISP: Internet Service Provider
+ RTP: Real Time Protocol
+ RSVP: ReSerVation Protocol
+ QoS: Quality of Service
+