Voice Over IP Telephony

By

Mukesh Verma

MS- CLIS (IIIT -Allahabad)

Introduction

Telecommunication network has emerged as a strategic component of the worldwide infrastructure to support economic development, scientific discovery, educational opportunities & social advancements. Telecommunication today is not simply a means to communicate, with anyone anywhere in the world & to access information on timely basis, but also it acts as a catalyst in overall development of country.

The telephone is the most common of the today's communication facilities. It is the one that has been here for the longest. It would therefore only be natural for us to assume that the familiar telephone would be around for all times to come. But a silent revolution is taking place on telephone networks the world over could make the telephone of today useless or extinct.

As BT's advertising slogan says, "It's good to talk" the good news is that it's now cheaper as well. Pressure from telecom Watchdog Oftel is leading to deregulation & the competition to provide us with telephone services has never been so intense. The scramble for telecom ownership is fuelled by Internet services. And with the right software, the Net could also start competing with the telecom network. It might sound too good to be true, but there are a variety of reasonably priced options on the market.

WHAT IS VOIP?

For the record, IP telephony can be divided into two major sub-groups - Internet Telephony and Voice over IP. While the former uses the public Internet network for voice traffic and is highly disorganized, with little QoS guarantee; the latter refers to voice traffic over a managed IP-based network.

How VOIP Works

Voice traffic flows over the network through a gateway that acts as a link between an IP network and a telephone network. The analog voice signal is digitized using PCM. These digital voice samples are then buffered on an IP gateway. This device converts the PCM data stream into a compressed IP packet stream using DSP's (Digital Signal Processors). DSP's are responsible for converting from analog to digital as well as compression.

The basic element in the technology of Voice over Internet Protocol (VoIP) is the packet. It uses the packet switching network to transfer voice across the network. What happens is this: The sending computer chops data into these small packets, with an address on each one telling the network where to send them. When the receiving computer gets the packets, it reassembles them into the original data. Packet Switching minimizes the time that a connection is maintained between two systems, which reduces the load on the network. It also frees up the two computers communicating with each other so that they can accept information from other computers as well.

Finally, an IP Header is attached to this compressed data, which is then sent out on the network as discrete data packets. Each packet is put into separate IP "envelopes" containing addressing information that tells the Net where to send the data. As the packets are sent across the Internet, routers along the way examine addresses on the IP envelopes. They determine the most efficient path for sending each packet to the next router closest to its final destination. T. Even though some packets from the same message are routed differently than others, they will be reassembled at the destination .The packets thus arrived are received by the second gateway, which decompresses it & converts the call back to a standard analog signal and routes it to the receiver's standard telephone. Thus this system enables the transmission of analog signals over a network by first digitalizing & compressing the voice data & then again converting back to voice signals making a Full Duplex conversation possible.

VoIP Components And Requirements

Using Internet Telephony requires at the very least a PC with a internet connection & a full duplex sound card with speakers & microphones.

A basic setup between two locations will consist of an IP gateway on either location. The gateway is essentially a device that translates protocols from the network onto the PSTN network. So, the one you choose will depend on the type of connectivity you're using, and the protocols you'll use.

In addition to gateways and terminals, there are devices called gatekeepers. Simply speaking, a gatekeeper acts as a routing device for voice calls They determine how much bandwidth is required for a voice call, and can perform other functions like call authorization, bandwidth management, call management, and directory control.

Finally, there's a device called Multipoint Control Unit, or MCU. This is meant to provide conferencing facilities between three or more H.323 terminals or gateways

For compression of the voice into packets PCM technique is used. PCM is primarily the encoding of analog voice data into digital form without much compression. This is where speech codecs, also called voice coders or Vocoders come into picture. These are speech compression algorithms that let you drastically reduce the amount of data that goes into the network while still preserving the voice quality.

Benefits Of VoIP

In the case of a large organization, the amount of telephone calls made everyday is considerable, & the cost is high. Internet Telephony, using the Internet connection to make call ensures that all calls -long distance or not-cost no more than a local call.

In addition to this, Internet telephony provides all the functions of the telephone & more. Furthermore, in case where it is necessary to have lengthy conversations we can handle the call using headphone with a attached microphone which makes the operation cheap painless & more efficient.

Another advantage of using PC internet phone -installed on a laptop- is that no matter where one is as long as he has access to the internet he can place a call.

Limitations Of VoIP And Solutions

The process described above seems to be very rosy, but that's not the case as VoIP has several limitations.

The first and foremost being that IP is a slightly unreliable protocol when transmitting data over large distances and involving the Internet. Some of the voice 'packets' may get lost or may reach out of order in which they were sent leading to Latency. With digital data, this does not pose much of a problem, since the receiving end can always request for the missing data to be sent again and wait for them to reach before putting all the packets together. With voice, which is real-time, this cannot be done, as the data no longer exists on the sending side either.

. With digital data, this does not pose much of a problem, since the receiving end can always request for the missing data to be sent again and wait for them to reach before putting all the packets together. With voice, which is real-time, this cannot be done, as the data no longer exists on the sending side either.

Another problem with VoIP is packet errors .If a voice packet encounters a bad Router; it might get corrupt or get lost all together. If it's the former the packet that reaches the destination is of no value. All these delays are classified as (de) packetization, access & network delays.

There are many ways to solve these issues. One method is to differentiate between the various services- in other words traffic prioritization. Routers can be configured to give preference to voice packets over data packets. Another method is weighted fair queuing. Here, a minimum amount of bandwidth is allocated to certain traffic, in this case, voice. Thus we can prevent certain type of traffic from passing through a Router all together.

Another problem faced with VoIP is network congestion. This is one area in which research is still going on.

Legality is another issue faced by VoIP. Presently in India, sending voice over VSNL's Internet service is prohibited.

Voice quality is the first thing to get affected in VoIP. Overall voice quality is a function of many factors including the compression algorithm, errors and frame loss, echo cancellation, and delay. The H.323 standard was originally a multimedia standard meant for transferring audio and video data over a network such as Ethernet or Token Ring, which doesn't provide guaranteed QoS (Quality of Service) the voice quality is affected primarily due to the type of Codec used and also because of latency, jitter and packet loss

Delay can create two potential problems.

The first is that long delays in conversation cause the talk from two parties to overlap.

The second problem is echo, or the reflection of the original signal back to the sender. . It becomes more noticeable when the delay becomes too large. Echo chancellors are necessary to remove echo from such conversations.

With a large number of bandwidth-hungry applications pouring in, 'congestion' is the watchword in e-com circles.

Even though paucity of bandwidth is widely acknowledged as the major cause, inefficient utilization of bandwidth is also a key factor. Moreover, Internet traffic increases in proportion to available bandwidth, as fast as it is added; so delays and choking of networks are inescapable.

Protocol Hierchies:

For telephones to communicate with each other and with other devices, such as computers, over a data network, they need to speak a common language called a protocol.

Packet switching networks based on the Internet protocol generally consist of different sub networks of various technologies. Since the packetizesd voice is transmitted through the network using Internet protocol it is a necessity to discuss this topic in detail.

If we restrict ourselves to IP's suitability as a networking layer for communication media rather then a sub network we can make the following observations.

IP imposes no bandwidth restrictions. Since IP is a connectionless network layer; packets may traverse various routes between the source & the destination, encountering greatly varying link & router delays.

TCP/IP (transmission control protocol/Internet protocol) is the basic communication protocol of the Internet. When you set up a direct access to the Internet, communications take place over TCP/IP. TCP/IP is a two-layered program. The higher layer, TCP, manages the assembling of a message or file into smaller packets that are transmitted over the Internet and received by a TCP layer that reassembles the packets into the original message. The lower layer, IP, handles the address part of each packet so that it gets to the right destination.

Two relevant proposals for the IP telephone are the ITU-T recommendation H.323 & the Session Initiation Protocol (SIP)

Most VoIP implementations follow the ITU H.323 standard. This consists of four basic components: the terminals, IP gateways, gatekeepers, and Multipoint Control Units (MCUs). The H.323 standard was originally a multimedia standard meant for transferring audio and video data over a network such as Ethernet or Token Ring, which doesn't provide guaranteed QoS (Quality of Service). Multipoint control units (MCU) provide support for conferences between three or more H.323 terminals.

QoS happens when you can guarantee the timely delivery of information on networks, control bandwidth, set priorities for selected traffic, and provide a good level of security. The H.323 standard consists of a set of audio and video compression algorithms and control protocols.

Terminals support two-way communication over IP. A terminal uses certain protocols for communication. These include the H.245, which negotiates channel usage and capability. It also uses Q.931 for signaling and control; and RTP/ RTCP (Real-time Protocol/Real-time Transport Control Protocol) for sequencing the audio packets. . RTP carries the actual media and RTCP carries status and control information. These packets are sent via UDP (User Datagram Protocol). And finally, there's the RAS protocol, which sends Registration/Admission/Status signals to the gatekeeper. H.235 provides security and authentication. All control protocols are sent using TCP.

The primary goal of H.323 is to interwork with other multimedia terminals. Because the basic service provided by an H.323 terminal is audio communications, a H.323 terminal plays a key role in IP-telephony services. VoIP gateway is a product that delivers data and voice over an IP network. Gateway takes traditional telephony traffic, compresses it, and then places the compressed information into an IP packet and "routes" this into the IP network. A gateway connects two dissimilar networks. An H.323 gateway provides connectivity between an H.323 network and a non-H.323 network.

Being delay sensitive audio data requires the transmitted voice packets to reach the destination with minimum delay. The TCP messages are broken into datagrams & transferred. These datagrams are re assembled & the error check is performed. Even if one packet is missing the whole message needs to be resent.

The TCP is less effective for audio communication because it tries to ensure error free transfer at the cost of delay, whereas we need less transmission overheads & less stringent error checks. Thus UDP is a better choice than TCP.

The following problems may still arise when UDP datagrams are used. The packets may arrive out of sequence& packets may be dropped or duplicated.

As you can see, full implementation of H.323 requires a lot of overhead. An alternative to H.323 emerged with the development of Session Initiation Protocol (SIP) .SIP is a much more streamlined protocol, developed specifically for IP telephony, smaller and more efficient than H.323.SIP is an ASCII-based, application-layer control protocol (defined in RFC 2543) that can be used to establish, maintain, and terminate calls between two or more end points. Like other VoIP protocols, SIP is designed to address the functions of signaling and session management within a packet telephony network

Now we discuss the basic differences between the H.323 & SIP protocols.

most of the implications in SIP use UDP while those in H.323 use TCP. In case of SIP network servers provide intelligence & services while in case of H.323 gatekeepers provide them. The encoding is textual in case of SIP but in H.323 it is in binary format.

Although SIP messages are not directly compatible with H.323, both protocols can coexist in the same packet telephony network if a device that supports the interoperability is available.