Thursday, October 21, 2010

Why H.323 is an important Internet protocol?

The H.323 communication protocol is the first complete specification under which the products developed can be used with the transmission protocol widely disseminated (IP). There is much interest and excitement around the H.323 because it appears in the right time. Network administrators have extensive networks already installed and feel comfortable with the IP-based applications such as web access. Moreover, personal computers are becoming more powerful and therefore capable of handling real-time data such as voice and video.

Several independent consulting companies predicted a rapid adoption of H.323. The following chart explains the trend itself

H.323 Components


The H.323 specification defines a generic entity as any component that meets the standard.


One end H.323 is a network component that can send and receive calls. You can generate and / or receive data streams.


An H.323 terminal is one end of the network that provides real-time two-way communications with another H.323 terminal, gateway or multipoint control unit (MCU). This communication consists of control signals, signs, audio, moving color image and / or data between two terminals. According to the specification, an H.323 terminal can provide voice only, voice and data, voice and video, or voice, data and video.


The gatekeeper (GK) is an entity that provides address translation and control access to the network of H.323 terminals, gateways and MCUs. The GK may also offer other services to the terminals, gateways and MCUs such as bandwidth management and locating gateways or gateways.

The Gatekeeper performs two call control functions that preserve the integrity of corporate data network. The first is the translation of addresses of the terminals to the corresponding LAN IP or IPX, as described in the RAS specification. The second is bandwidth management, fixing the number of conferences that may be happening simultaneously in the LAN and rejecting new requests above the level set, so as to ensure sufficient bandwidth for data applications the LAN. The Gatekeeper provides all the above functions for terminals, gateways and MCUs, which are registered within the control zone known as H.323.


An H.323 gateway (GW) is an endpoint that provides real-time two-way communications between H.323 terminals on the IP network and other terminals or gateways in a switched network. In general, the purpose of the gateway is transparently reflect the characteristics of an endpoint in the IP network to another in a switched network and vice versa.

MCU (Multipoint Control Units)

Multipoint Control Unit is designed to hold the conference among three or more points under the H.323 standard, leading the negotiations between terminals to determine common capabilities for processing and controlling audio and video multicasting.

Communication under H.323 provides the audio and video signals. The audio signal is digitized and compressed under one of the supported algorithms such as G.711 or G.723, and the video signal (option) is treated with H.261 or H.263 standard. The data (optional) are handled under the T.120 standard that allows application sharing in conferences point to point and multipoint.

Benefits of H.323 Technology: An extension of H.320 Design

The H.323 is based on the H.320 specification. Many components of H.320 are included in H.323. In this regard, H.323 can be seen as an extension of H.320. The new standard was designed specifically with these ideas in mind:

     * Build on existing standards, including H.320, RTP and Q.931
     * Incorporate some of the advantages of the packet-switched networks to carry data available in real time.
     * Solving the problems posed by sending data in real-time packet switched networks.

Benefits of H.323 technology:

·    Reduced operating costs: You can use the wired campus, WAN connections based on IP routers and WAN services to send video. This is a potential source of significant operational savings. Support costs of infrastructure (eg SNMP) can be combined. H.320 technology typically requires you separate networks for video and data. This means double cabling and network infrastructure. This model increases the cost of system implantation.
·    Wider dissemination and greater portability: With H.323, each port over IP video can potentially support. This makes the technology accessible to a wider range of users. It is also easier to move a team in our environment, which will make the same computer can be used for more applications. With H.320, you must dedicate one line for each location. Most rooms or personal computers can not easily support video, which also limits the accessibility and portability of systems.

Client/Server features of H.323 network design

The design relies heavily on the H.323 network components. Its capabilities are distributed across the network. One example is the gatekeeper. A gatekeeper can reside on a server in a gateway or an MCU. Is responsible for registering users or customers (videoconferencing systems) and can potentially provide a set of communication functions.

As a rule, H.320 equipment is not connected to a server. The characteristics of the system lie in the same videoconferencing platform. This approach oriented communication terminal does not support supplementary services such as call routing, transfer or retention. These are services that are used by the technology of the telephone exchange.

What is H.323 Protocol?

The H.323 is a family of standards as defined by ITU for multimedia communications over LAN networks. It is defined specifically for LAN technologies that do not guarantee quality of service (QoS). Examples include TCP / IP and IPX over Ethernet, Fast Ethernet or Token Ring. The most common networking technology in which they are deploying H.323 is IP (Internet Protocol).

This standard defines a rich set of features and functions. Some are required and others optional. The H.323 defines much more than the terminals. The standard defines the following most important component:




Multipoint Control Unit

The H.323 uses the same compression algorithms for video and audio to the H.320 standard, although it introduces some new ones. It is used for T.120 data collaboration.

The H.323 in historical perspective:

Prior to H.323, the ITU is focused exclusively on the standardization of global telecommunications networks. For example, in 1985 he began work in the specification that defines the image and voice transmission on circuit-switched networks such as ISDN. The ratification of the standard (H.320) took place 5 years later (was approved by the CCITT in December 1990). Only 3 years after equipment was available that met the standard and allow inter-operability between them.

In January 1996, a group of vendors of computer networks and proposed the creation of a new ITU-T standard to incorporate video conferencing on the LAN. Initially, research focused on local area networks, as these are easier to control. However, with the expansion of the Internet, the group had to consider all IP networks within a single recommendation, which marked the beginning of H.323.

The supports H.323 real time video, audio and data over local area networks, metropolitan, regional or large area. Likewise supports Internet and intranets. In May 1997, the Group 15 of ITU H.323 redefined as the recommendation for "multimedia communications systems in those situations in which the means of transport is a packet switching network that can provide quality service guaranteed.

Note that also supports H.323 videoconference on point to point connections, and ISDN telephone. In these cases, there must be a transport protocol such as PPP packet.

Improving VoIP quality of service and packet loss

It is therefore necessary not only for QoS in the network, but also in the terminals, and in the processes that develop in them, hence it is necessary also to say that sensitivity to packet loss, delays and fluctuations, experienced by voice over IP services depend largely on the mechanisms implemented in the terminals.

The preparation of media in the terminals to be sent and transferred over an IP network involves several processes: scanning, compression and packaging at the sending end, and the inverse processes at the receiving end. All this is accomplished through a complex algorithm processing that is determined, which in turn develops in some time interval, ie delay involves delay processing and packaging:
o Processing delay: delay for implementing the encryption algorithm, which delivers a stream of bytes ready to be packaged;
o packetization delay is the time required to form a voice packet from the encoded bytes.

It should be noted that the outcome of this consolidation - packetization directly affects the QoS, and also how they perform. Thus, when encoding speed reduces the bandwidth requirements are also reduced, allowing the network towards being able to handle more concurrent connections, but increases the delay and distortion of voice signals. The opposite occurs when increasing the speed of encoding.

Another aspect to consider is the tradeoff between packetization delay and channel utilization (the ratio between bytes and bytes of header information in each voice packet), ie, the search for greater use of the channel leading to further delay of packetization for a coding standard. Of course, as the encoding standard used shall be the resulting delay in connection with the use of the channel, which widened when the use of the channel is above 50% with a growth delay in exponentially If low speed codecs such as G.723.1. The packetization delay can also be reduced by multiplexing several voice connections on the same IP packet.

A delay of processing and packaging also adds the delay introduced by the process of buffering in the terminals, and the delay of "glue" in the network. This gives an end-to-end delay perceived by the end user a greater or lesser extent. End to end delays below 400 milliseconds do not compromise the interactivity in the conversation, but above 150 milliseconds, echo control is required.

The delays discussed above are logical result of the characteristics and mode of operation of IP networks, as well as the nature of voice signals.

Wednesday, October 20, 2010

Operating VoIP from PC to PC and VoIP phone via Computer

Operating VoIP from PC to PC and VoIP phone via Computer

Keyspan VP-24A Cordless Voip Phone Skype Compatible for Mac Or Pc

Connecting LAN and Internet on Network

Problem: I have a 5 desktop PC & a VOIP router. I have Cable Internet Connection. This Internet cable is connected to switcher and all PCs also connected to switcher. My Internet Connection is configured in one system. Now I get Internet on all 5 systems, but I can't connect LAN when I configured IP in these systems and then LAN working but Internet cannot be connected. Please give me solution. I want to connect LAN as well as Internet on all these systems

Reply: Your internet connection is of single user connection and you need multiple user connections.

Problem:  That is true but when I removed IP, then I was able to connect Internet in all systems.

Reply:  When you are using internet sharing:
Step 1: Connect your router to single machine via cross cable.
Step 2: Browse your router in internet explorer.
Step 3: configure the IP address of your LAN series and add the gateway address your internet sharing machine.
Step 4: Add DNS server and then remove the router and connect to your switch and check it and reply me.
Step 5: You can install a proxy server in one machine and give the IP address of this machine to all the machine as proxy server in IE options. It should work.

Ideas for Networking Research Paper

When we talk about research in networking, let me tell you there are different domains in networking and each domain in itself is huge like switching, routing, security, VOIP, and network management. Which layer in OSI model interests you the most depends on you but anyways following ideas I can give:

1. (Routing) - Develop an algorithm on as to how you can transfer a packet from one wireless base station to another in a different network in minimum round trip time.

2. (Switching) - develop a s/w and h/w asic that would enable fast switching in a switch between ports.

3. VoIP - develop a unique algorithm that would compress voice packet from 64 kbps (G711) to maybe 2-3 Kbps per sec. Can we do it?

4. Network Security - enforce security policy in network and push it to OS level (kernel), so that network elements can take action right at the OS (maybe blocking it or shutting down the PC)

5. Or maybe take up the industry standard VOIP protocols and then find their drawbacks and work on improving them, for example MGCP, Megaco, SIP, H.323 protocols.

Finally if you want to add some spice in your networking research paper, then how about adaptive learning by network and then find a shorter path by doing a heuristic search, etc.

VOIP problem - Local ping rate is high

Problem: I am getting a problem since long time. My LAN works under 192.168.3.X and VOIP works under 192.168.2.X. In the night time, we use VOIP to make call to clients. Only in the night after 9 O' clock, I will get ping rate from local system to local router which is in 2.X series will be high. It goes up to 900 ms. Sometimes Request Timed Out also occurs. The whole day it used to be fine (i.e. 1 ms only). But night it gives a lots of trouble.

I tried to find out the traffic congestion system. But I could not be. My both VOIP & Data goes through same switch only.

How can I solve this VOIP problem? Thanks in advance!

Solution: According to me it has to be a network congestion from the client side because as you say the traffic at your end remains the same during day and night. See ping is the round trip time taken so the latency can be at other end as well. You can ask your client to put a packet sniffer to monitor if there is an increase in traffic at their end or not.

Voice over IP, TAPI, SIP protocols

Voice over IP is a suite of related protocols working together to carry media (voice and others as well -- video) over a packet-switched network such as the Internet (based on IP protocol).

For example, you may write some code to sample data from a microphone and send over the net using UDP and that's VoIP. Or you may use more appropriate protocols such as RTP, RSTP, etc.

To learn VOIP, try to study TAPI (Telephonic Application Programming Interface).You can make SIP proxy server in any High Level language (one I know is VC++) and use it for VoIP features. see help for the following in MSDN

•    TAPI
•    SIP (session initiation protocol)
•    H.323 protocol.

VoIP - Understanding Voice over Internet Protocol

I hope you are familiar with the IP address thingy. VoIP is using latest generation IP addressing and it is converting our phone number to an IP address and then routing that information using the destination address and then again converting back to phone numbers. Likewise you use codes like 001-789-4566664. This is the format for a phone number, but when it is converted to an IP address, it is something of this sort, I hope you are getting a picture and all this conversation is being done using a gatekeeper which handles all the traffic and routing and conversions. You should try searching some material about it on the net or on my hubs.

You should get a hands on the rate at which compression of data is being done. For example if you phone lines are working at 64kb/sec, they are not efficient to send data at this size and normally what a good solution requires is some data compression. You can compress a data packet to a max of 8kb/sec and in technical terms it is called G.729. G.728 (16kb/sec_.G.727(32kb/sec) and so on... The present telephony structure is using H323 protocol. but what I am working on is a conversion of H323 protocol to H248 and get some knowledge about H321..H322.. H323.. H324..H325 ..H245.... There are some protocols like SIP (session initiation protocol). When you will start your search using a VoIP, you will come across all these terminologies and I have explained most of them in my articles.

Traditional phone calls set up a circuit between caller and callee and some points of concern are:
1) 50% of the line is continuously idle (because either you are talking/other person is talking, but NOT both at same time).
2) Lot of silence is present.
So VoIP is better as it sends packets only when there is a need to send voice. Therefore Resource Utilization is CUT by about 70%!
3) Ordinary Telephone sends 64Kbps. But VoIP does a lot of compression.
And many other such features.

1) Jitter
2) Delay
3) Lack of QoS in Public Networks

But these are fast fading and VoIP is emerging as the communication protocol of choice in long-distance voice communications. Actually VoIP is where Telecom Networks and Data Networks intersect.

Telecom companies want to do VoIP their way - H.*** protocols are that suite. (Intelligence in the network is what they believe). They say that we are around for over 100 years, so we have the most tested tech kinda stuff.

Data Networking (IETF etc) wants to use SIP suite of protocols. (Intelligence in the End‑Points is what they believe). They show the rapid development of Internet kind of stuff to support their views.
PS: Do NOT confuse "intelligence" above with the standard "intelligence" we use. In this context, intelligence is about who is taking the decisions, etc.

Now I think you are able to understand what is VOIP and its advantages and disadvantages.
VoIP Phones - Use of signaling protocol, call manager, and SIP clients