VOICE OVER INTERNET PROTOCOL (VOIP)

Transcription

1 Chapter 1 VOICE OVER INTERNET PROTOCOL (VOIP) 1.1 Introduction Voice over Internet Protocol is a technology (or a group of technologies) that allows us to make voice communication using a broadband Internet connection instead of a regular phone served by the Public Switched Telephone Network (PSTN). It was first successfully demonstrated in 1974 on the ARPANET, the ancestor of the Internet. In addition to delivering the voice communication, the VoIP also delivers multimedia sessions. Other commonly used terms for VoIP are IP telephony, Internet telephony, Broadband telephony, and the Broadband phone service. VoIP services convert the analog voice signal into a digital signal that travels over the Internet. In more common terms it is a phone service over the Internet. With a broadband (high speed internet) connection one can get phone service delivered through internet connection instead of from the regular phone line (PSTN). The VoIP calls to telephones served by PSTN can be referred as off-net calls. To have VoIP service, one needs a broadband connection and a computer or special VoIP phone or a traditional phone connected to VoIP adaptor. Voice over Internet Protocol (VoIP) is a growing technology which is becoming more and more popular due to two major reasons: the lower cost and the increased functionality. It avoid the tolls charged by ordinary telephone services. Because of these advantages, the VoIP has now become a good alternative to the traditional voice communication system i.e., the Public Switched Telephone Network (PSTN). Now days VoIP is also available on some smart phones and on the Internet access devices. Calls and text messages can be sent over 3G or 4G internet connection (Robert M. Gray, 2005). The first voice transmission was introduced in 1876 through a ring down circuit having two devices connected by the wire. It had no dialing facility and the whole system was a one way system. The one way voice transmission, later transformed to a two way (bi-directional) voice transmission system which was based on wired networks. It was accomplished by establishing a circuit between transmitting and receiving end. This circuit switched technology was based on analog infrastructure which was neither robust nor efficient due to line noise and many other factors. The 1

2 digital era began in 1960 s and the human switch was replaced by electronic switch. It convert speech into digital signals and all the transmission facilities were converted from analog to digital ones. Digital transmission provides the benefits of low interference and easier restoration of signal. In digital transmission repeaters were used, which amplify the signal and clean it to its original condition. This chapter presents the state of the art of transmission technology, effects of various factors of VoIP, VoIP Protocols, Voice Coding, Compression Algorithms and their benefits as well as limitations. Since 1970 s there has been an increase in the use of packet networks for transmitting data. The most obvious use of this new technology is the Internet. It splits the data into small Packets that includes address information added to each packet. These packets are then sent over the network and reach the destination by taking possibly the different paths through network. These packets are then reassembled at destination node. The packet switched network has significant advantages over circuit switched network. By the early 90 s certain fundamental technologies such as signaling methods, RTP/UDP, voice coding algorithms were developed, which led to the development of this new technology, the VoIP (Sherburne & Fitzgerald, 2004). VoIP is a blend of hardware as well as software that uses the Internet as a transmission medium. IP networks are accomplished of processing all kinds of network traffic, which includes voice as well. The ability and quality of a VoIP communication for conversation is governed by various elements such as network settings, coding process, speech content, kind of error correction etc. In addition to voice calls the VoIP also offers services such as fax, send message service and voice-messaging applications. The process involved in transmitting these services over the packet switched network (Internet) are digitization and encoding of the analog voice signal followed by packetization, signaling and media channel set up. The analogous steps are employed excluding for decoding and digital-analog alteration to generate original signal at the receiving end. Communication using Voice over Internet Protocol has many advantages over the traditional Public Switched Telephone Network (PSTN) which works on circuit switching technique. In VoIP, it takes only 6-8 Kbps or less bandwidth whereas PSTN takes 64Kbps of bandwidth to make a call and thus it saves tremendous amount of 2

3 bandwidth. The high-quality VoIP services are, therefore, becoming an alternative to Public Switched Telephone Network (PSTN) (Brezocnik et al. (2001); Davidson et al. (2006)). 1.2 VoIP System In recent years, communication via Internet is the most extensively used network applications. The bandwidth requirements of speech transmission are relatively low but it is highly sensitive to delay, jitter (delay variation) and the packet loss. The important fact for becoming VoIP an interesting application, is that the minimum effort is required for co-existing the VoIP standards with the existing network infrastructure. VoIP system uses Internet protocol (IP) as a transmission medium, which is network layer protocol, according to the seven layer Open System Interconnected (OSI) model. Figure 1.1 shows the architecture of VoIP system. It provides the description of the end to end path of the voice signal. At the sender side the analog voice signal which is the output of some audio device such as microphone is converted into digital form by an analog-to-digital converter. This digitized signal is first compressed and then packetized using the Real Time Protocol (RTP) packetizer and sent over the network. At the receiving end, these voice packets are reassembled and converted back into original signal and played out using the output audio device usually, a speaker. (Tanaka et al. (1979); Cohen (1977); Gold (1977); Gruber (1979); Magill (1973); Cohen (1999); Hallock (2004); Gray (2005)). Figure 1.1 Basic VoIP Architecture The figure 1.2 shows the basic steps involved in the conversion of an analog signal into voice packets. First the analog signal after passing through a low pass filter (not shown in the figure) is sampled and quantized to convert into its digital form. Then this digitized signal has been processed for echo detection. The Voice activity detection algorithms were used to remove the echo, if any, present in the signal. Then the signal is compressed using various compression algorithms defined by the standards of the International Telecommunication Union (ITU-T) codecs such as G.711, G.729 and 3

4 G etc. The codec is selected on the basis of the various factors such as bandwidth available and required quality of signal. The most of the available codecs used today also include VAD, SID, CNG and PLC algorithms. In case, the codec used in the network does not have these features, then these algorithms are separately deployed in the network. The encoded speech is then packetized into packets of equal size. Each such packet includes the headers at the various protocol layers such as Real Time Transport Protocol (RTP) - 12 bytes, User Datagram Protocol (UDP) - 8 bytes, Internet Protocol (IP) - 20 bytes and the payload composed of the encoded speech for certain duration depending on the codec used. These packets encapsulating the voice data were then sent over the Internet for the transmission. The packets were routed over the shared network (Internet) and reach their destination IP address independently. At the receiver end, the voice packets are depacketized and decoded. Decoding process includes dejittering, error correction and packet loss concealment. The decoded speech signal was then converted into the analog signal again using the digital to analog converter and played out using the output audio device usually, a speaker (Cohen (1977); Gold (1977); Gruber (1979), Vlaovic & Brezocnik, (2001)). Analog Voice Signal Sampled and quantized Digital Voice Signal Echo removal, Silence suppression IP UDP RTP Packetization IP Network Figure 1.2 Voice packet formation (Wallingford, 2005) 4

5 1.2.1 VoIP Protocols The performance of VoIP network is related with the three communication aspects: call signaling protocols, network environment and its parameters. Figure 1.3 illustrates relation between the QoS of VoIP system and various aspects of communication. Figure 1.3 Framework of VoIP (Amin, (2005)) VoIP s networks are made up of combination of protocols and applications that best suits according to the requirement. Similar to the telephone call, VoIP offers call signaling, quality of service (QoS), and media transportation. Most of the call signaling process has been offered via either H.323 or SIP protocol. Further, QoS can be regulated by Transport layer protocols like RSVP (Resource Reservation Protocol) and Real-time Control Protocol (RTCP). The actual media transport is through CODEC and Real-time Transport Protocol (RTP). Figure 1.4 shows the basic structure for VoIP protocols. So the most important protocols for VoIP are: data transport, control and media protocols. To make technical formation of a VoIP call, RTP, along with RTCP is used in conjunction H.323, SIP or other call signaling protocols (Qinxia, 2007). In the following sections, the basic VoIP protocols are discussed in detail along with the basic transport protocols on which they work. 5

6 Figure 1.4 VoIP Protocols structure (Hersent et al. (2005)) Internet Protocol (IP) The Internet Protocol (IP) is responsible for the delivery of data/datagrams from one computer to another by using Internet which actually operates at network layer. Each computer (or host) on the Internet has at least one unique IP address that recognizes it from all other systems on the Internet. When a user sends or receives data, the data content is distributed into little portions called packets which contain both the sender's and receiver s Internet address. Every packet is firstly directed to a gateway host that realizes a small part of the Internet which reads the destination address and forwards it to an adjacent gateway which subsequently reads the destination address. This process continuous till one gateway recognizes it as belonging to a system within its instantaneous domain which forwards it straight to the computer with specific address. Since a message can be divided into several packets which can be processed to different routes across the internet. The role of IP is to deliver while the Transmission Control Protocol (TCP) puts them in the desired sequence. IP is a connectionless protocol and every packet can be processed through Internet independently relative to the other data. Presently, IPv4 is widely used version while IPv6 is also drawing attention due to the advantage that it offers much longer addresses and hence enhanced possibility of more Internet users (Fairview Industries Primer Series Voice over Internet Protocol, 2004). 6

7 Transmission Protocols Generally, there are two protocols: TCP (Transmission Control Protocol) and UDP (User Datagram Protocol), used at the transport layer for the transmission of data through an IP network. Both protocols are associated with unique port numbers (for example, the HTTP application is usually associated with port 80). The TCP is a connection oriented protocol that is responsible for reliable communication between two end processes. Transport layer protocol (TCP) is used for the data transmission. Internet use TCP/IP for most of its communication applications. The sending and receiving TCP entities exchange data in the form of segments. A segment consists of a fixed 20-byte header (plus an optional part) followed be zero or more data bytes. The source port and the destination port specify the end points of the connection. The sequence number field identifies the first byte of data in the segment and the acknowledgment number contains the value of the next sequence number, the sender of the segment is expecting to receive. The TCP header length field tells the TCP header length is 32-bit word. There are six 1-bit flags. The window size field tells how many bytes may be sent starting at the byte acknowledged. The checksum field checks a sum of the bytes in the header. With the information in the header, TCP provides reliable transmission between the two end points. Since, TCP is the reliable and robust protocol which provides the guaranteed delivery of the packets because it allows the retransmission of packets if any packet is lost during transmission. No doubt, TCP is robust protocol for transmission but it increases the traffic load with transmission of acknowledgment and delay with retransmission. But for voice like applications in which we require non-interrupted and continuous communication, TCP cannot be used. (Comer, (2005); Tittel & Chappel, (2006)). Voice communication requires one shoot transmission of speech packets. The UDP is connectionless and reliable having minimal overhead. Each packet on the network is composed of small header and user data, and is called a UDP datagram. UDP routes data to its correct destination port, but does not try to perform any sequencing, or to ensure data reliability in common with IP. A UDP segment consists of an 8-byte header followed by the data. The two ports serves the same function as they do in TCP: 7

8 to identify the end points within the source and destination matches. The UDP length field includes the 8-byte header and the data. With only this information in the header, UDP provides unreliable transmission between the two end points. Voice is a real-time application, and mechanism must be in place to ensure that information received has been in the correct sequence, reliably and with predictable delay characteristics. Although TCP would address these requirements to a certain extent, there are some functions which are reserved for the layer above TCP. The extra overhead of TCP and the possibility of increased latency makes it unsuitable for real-time applications. Since UDP does not use the retransmission mechanism in transmission, it is better for the real time application. Therefore, for transport layer, TCP is not used, and the alternative protocol, UDP, is commonly used for VoIP communications (Commer, (2005)). The drawback of the UDP is that it does not provide guaranteed service. On the top of the UDP, Real Time transport is used. This protocol, along with another protocol Real Time Control Protocol (RTCP) provides all necessary information. Since it contains timestamp, payload type, sequence number etc. The brief detail of the VoIP network is also given by (Simak & Koska, (2001); ITU-T, E.492, (1996); Fallman, (2008); Oneiss & Craiger, (2002); Minoly & Minoli, (2002); Sriram & Whitt, (1986); Filka et.al, (2004)) Media Protocols It is essential in the real time applications to place a mechanism which ensures that a stream of data can be reconstructed accurately. To reconstruct the datagrams in the correct order, network delays detection must be placed in that mechanism. The main media protocols available are Real-Time Transport (RTP), Real-Time Transport Control Protocol (RTCP) and Resource Reservation Protocol (RSVP). The Real-Time Transport Protocol (RTP) describes a modest approach of sending and receiving encoded media streams in connectionless assemblies. It offers headers that afford VoIP systems an easy mode of discriminating between multiple sessions on the same host. It can be mentioned that the codec just defines how the digitized sample can be encoded, compressed, and decoded. However, RTP is accountable for transmitting the encoded sound data within a UDP datagram. RTP is capable to mix different streams into a single one and supports different applications 8

9 like conference calling etc. Although sometimes it can-not offer suitable controls for describing the multiplexed voice pathways that are usually related with telephony such as trunks (Wallingford, (2005)). The RTP is a protocol to carry data that has the real-time properties. It is the main transport protocol used for IP telephony media streams (Schulzrinne et al. (2003)) and it defines a standardized packet format for delivering media over the internet. RTP provided end-to-end network transport functions of applications transmitting real-time data, such as interactive audio and video over multicast or unicast network services. The network services include payload type identification, sequence numbering, time stamping and delivery monitoring. Both RTP and UDP contribute parts of the transport protocol functionality. To make use of UDP s multiplexing and checksum services, RTP is run on top of UDP. However, RTP may be used with other suitable underlying network or transport protocols. If a multicast distribution is provided by the underlying network, RTP protocol can transfer the data to the multiple destinations. RTP does not have a standard UDP port on which it communicates. The only standard that it obeys is that UDP communications use an even port and RTP Control Protocol communication are done on the next higher odd port. The RTCP provides the control services for data stream that uses RTP. The RTCP provides feedback on the quality of the transmission link. The other function of RTCP includes carrying a persistent transport-level identifier for an RTP source and this identifier is used be receivers to synchronize audio and video and convey minimal session control information such as participant identification to be displayed in the user interface (Schulzrinne et al. (2003)). RTCP uses the same transport protocols as RTP to periodically transmit control packets tp participants in the streaming multimedia session. As mentioned earlier, every RTP channel using port number N has its own RTCP protocol channel with port number N+1. It gathers statistics on a media connection and information such as bytes sent, packet sent, lost packets, jitter, feedback and round trip delay. An application may use this information to increase the quality of service perhaps by using a low compression codec instead of a high compression codec. RTCP is also used for QoS reporting. 9

10 The RSVP is used to reserve resources for a session on the internet. This aspect of the Internet which is quite different to the underlying design intent of the system, was only established to support a best effort service, without any regard to predefined requirement for user application. RSVP is intended to provide guaranteed performance by reserving the necessary resources at each machine that participates in supporting the flow of traffic (such as a video or audio conference). IP does not set up paths for the traffic flow, whereas RSVP is designed to establish these paths, as well as to guarantee the bandwidth on the paths (Awduche et al. (2001)). RSVP does not provide routing operations, but utilizes IPv4 or IPv6 as the transport mechanism in the same fashion as the Internet Control Message Protocol (ICMP) and the Internet Group Message Protocol (IGMP). RSVP operates with unicast or multicast procedures and interworks with the current and planned multicast protocols. Like IP, it relies on routing tables to determine routes for its messages. It utilizes IGMP to first join a multicast group and then executes procedures to reserve resources for the multicast group. RSVP enables endpoints to signal the network with the kind of QoS needed for a particular application. The receiver host application must determine the QoS profile which is passed to the RSVP. After the analysis of the request for QoS, RSVP is used to send request messages to all the nodes that participate in the data flow (Barden et al. (1997)) Call Signaling (H.323 & SIP) As in case of telephonic applications, there is a signaling arrangement that tells the users for the incoming call or if the transmission line is busy. There are few VoIP protocols stacks which are derived from various internet standards bodies such as International Telecommunication Union (ITU) and Internet Engineering Task Force (IETF). H.323 and Session Initiation Protocol (SIP) are most commonly used signaling protocols and were designed by ITU and IETF in 1996 and 1999 respectively. For VoIP, the process of signaling and encoding along with the packet transmission is executed by either the H.323 or the Session Initiation (SIP) protocol. Although both protocols are applied in different ways yet they offer the identical services. The H.323 (1999) provides a foundation for multimedia communication over networks that do not provide a guaranteed quality of service. The H.323 standards uses 10

11 the IP/UDP/RTP protocols and provides an infrastructure for audio, video and data communications over packet based networks. This standard is a part of the H.32x protocol family that includes standards like H.324 (standard for multimedia transport over SCNs (Sustainable Communities Network) and H.320 (standard for ISDNs, Integrated Services Digital Network), among others. The H.323 protocol format is presented in Figure 1.5. Figure 1.5 H.323 protocol stack ( Davidsonx et.al, (2006) The H.323 standard describes four key components of an H.323 system, namely the terminals, gatekeepers and Multipoint Control Units (MCU). The terminal are used to originate and terminate the calls. It may provide speech only, speech and data or speech, data and video. Gatekeeper performs the function of a manager in the particular zone in H.323 communication and provides the services to the end point components. The gateways are used for the communications of H.323 to other networks such as PSTN, ISDN and for the transformation of voice signal according to network. The H.323 uses the MCU for placing the multimedia conference between the different users. H specifies the message for the call control signaling, registration and admission into call. H.245 specifies the messages for opening and closing the channel for media streams and these messages are transported on TCP. T.120 is used for defining the data conference part. H.225.0/ RAS (Registration, Admission and Status) messages use the 11

12 UDP for the communication between the end points and gatekeeper for registration and admission. H.26x (H.261. H.263) provides the way for the video data to transmit over the IP network (Thom, (1996); Liu & Mouchtaris, (2000)). The H.323 protocol offers a typical voice and multimedia conferencing product that can communicate over IP network. To launch real time voice/video over the IP networks, H.323 has different CODECs to convert analog to digital audio like G.711, G.722, G.723.1, G.728, and G.729. The audio/video signal to be transmitted is converted into digital form by using CODECs and it can be decoded later at the receiving end. After the signaling procedure is completed, a transport protocols look out for all the data that is yet to be transmitted via the network between the two parties. While RTP offers end-to-end delivery of audio/video signal, the Real-Time Transport Control Protocol (RTCP) offers feedback on the quality of the connection. The functionality of SIP is entirely different from the H.323 protocol. Alike H.323, SIP also offers the means of signaling, setting up, and tearing down a VoIP session. SIP is a peer-to-peer protocol, where the peers in the session are called User Agents (UAs). SIP servers contains proxy, redirect and registrar servers. A proxy server accepts the intermediate SIP requests and forwards it to the next SIP server. In SIP, when a call is made, a request is sent to a SIP server which finds the end user or pass it to other SIP server. Finally, when the end client is found, RTP communication takes place between the two parties. The SIP (Rosenberg et al., 2002) is an application-layer control (signaling) protocol for initiating, manipulating and tearing down sessions. These sessions includes Internet telephone calls, multimedia distribution and multimedia conferences. SIP s main purpose is to help session originators deliver invitations to potential session participants. It is alternative to H.323, but is a more lightweight and general-purpose, text-based protocol based on HTTP (Schulzrinne and Agboh, (2005)). SIP makes use of proxy servers to help route requests to the user s current location, authenticates and authorizes users for services, implements provider callrouting policies and provides features to the users. SIP also provides a registration 12

13 function that allows users to upload their current locations for use by proxy servers. SIP is a request-response protocol, dealing with requests from clients and responses from servers. It is designed to address the functions of signaling and session managements. Signaling allows call information to be carried across network boundaries. Session management provides the ability to control the attributes of an end-to-end call. Such attributes are very important in a packet telephony network. From an architecture standpoint, the physical components of a SIP network can be grouped into two categories: clients and servers. SIP client s includes phones, which act as either a UAS or UAC and the gateway that provides call control. The functions of the gateway include translation between transmission formats and between communication procedures. SIP servers include proxy server, redirect server and register server. The proxy server is an intermediate entity that receives SIP requests from a client and then forwards the request on the client s behalf. The redirect server is a server that accepts a SIP request, maps the SIP address and returns them to the client. The register server is a server which processes request from UACs for the sake of registering and looking up their current location (Zhang, (2002); Johnson, (2004)). In current SIP or H.323 based Internet telephony client-server architectures, a registration server is employed for every domain. The user agents in the domain register their IP addresses with the server so that the other users can reach them. Traditional redundancy and failover methods are used for scalability and reliability of such server-based systems. The majority of the system cost is in maintenance and configuration, so it is not easy to set up the system in a small environment quickly. 1.3 Codecs The most necessary and important component of the VoIP system is the speech codec. It is the part where digital signal processing techniques plays an important role using compression and decompression schemes. The speech codecs has been an important part of speech processing over IP. It is used to encode the speech samples into small number of bits and compressed these into different size of packet frames. For a VoIP system the end users should have list of codecs since this network is capable to handling various types of codecs. The Speech codecs differ primarily in bit rate (measured in bits per sample or bits per second), complexity (measured in operations 13

14 per second), delay (measured in milliseconds between recording and playback), and perceptual quality of the synthesized speech. Codecs are of two types; Narrowband codec and wideband codec. Narrowband (NB) coding refers to coding of speech signals with bandwidth less than 4 khz and sampling frequency of 8 khz. It has been standardized by ITU-T. For wideband (WB) coding speech bandwidth can be extended from 7- khz to 14 khz and the sampling frequency can be increased from 8 to 16 khz. NB coding is more common than WB coding mainly because of the narrowband nature of the wire line telephone channel ( Hz). More recently, however, there has been an increased effort in wideband speech coding because of several applications such as videoconferencing. Some of the popular codecs which are generally used in VoIP system, are described below:- G.711 G.711 (1988) codec was introduced in digital telephony by ITU in It digitizes analog signal using a semi-logarithmic scale, called the compounded pulse code modulation (PCM). It increases the resolution for small signals while large signals are treated proportionally. It can encode the speech signal at 64 kbps with sampling frequency of 8 khz. It fulfills all the necessary requirements for proper communication. It provides the best voice quality for VoIP (MOS value 4.1) because of low delay and less distortion. The downside is that it consumes more bandwidth than other codecs, up to 84 Kbps including all TCP/IP overhead. However, with increasing broadband bandwidth, this should not be a problem (G.711 ITU, (1988)) G.723.1, G.726, G.728 These codecs allow cost-effective usage of the network, permitting high-quality sound reproduction at a bit rate of 8-32 kbps. Unlike G.711, this group of codecs uses ADPCM or CELP algorithms to decrease bandwidth requirements. Adaptive differential pulse code modulation (ADPCM) preserves bandwidth by quantifying the deviation of each sample from a foretold point rather than zero, permitting fewer bits to represent the historically 8- bit PCM scales. CELP, or code excited linear prediction, uses a newer variation of this approach.g.726 (1990) is the codec developed by ITU. This codec uses the Adaptive differential pulse code modulation (ADPCM) coding scheme. It operates on 16, 24, 32 and 40 kb/s bit rates. This coding scheme conserve the bandwidth by 14

15 measuring the deviation of each sample from predicted point. The ITU standard G (1996) specifies the compression of coded speech code or audio at very low bit rates. It has dual coding rates of 5.3 and 6.3 kbps. the input and output of this coding algorithm is 16 bit linear samples and the total algorithmic delay is 37.5 ms. This provides the highest compression of the current ITU standards without compromising with the quality of speech. The ITU-T G.728 (2006) standard coding the speech using low-delay code excited linear prediction (LD-CELP) can operate at bitrates other than 16 kbps, bit rate is extended to 40 kbps to optimized it for voice band data (VBD). The algorithmic delay is five samples long (0.625 ms). This type of codecs can perform compression and decompression for various multimedia applications like telephony. G.729 and G.729A G.729 (1996) codec can model speech based on the qualities of the human vocal tract. G.729 is a data compression algorithm that compresses digital voice in packets of 10 milliseconds period. The speech is coded at 8 kbps using conjugate-structure algebraic-code-excited linear prediction (CS-ACELP). Because of its low bandwidth requirements, G.729 is frequently used in VoIP applications (like Skype) where bandwidth is preserved. Standard G.729 operates at a bit rate of 8 kbps, but there are other extensions which offer rates of 6.4 and 11.8 kbps for slightly better speech quality, respectively. G.729 is extended to the next generation commonly designated as G.729A with different features and half the consumption of instructions. Due to the low data rates and low complexity algorithm G.729A could be preferred for IPTel applications. It has good balance of coding efficiency, speech quality and better stability under extreme conditions. G.729/G.729A have built-in packet loss concealment to produce the speech with same spectral qualities for the missing speech, making the gaps in the speech signal less noticeable to the listeners. These codecs could use the frame of 10 ms which is helpful in decreasing the effect of delay. The G.729B described a Speech Activity Detector (SAD) which can be used with the G.729 or G.729A. The SAD can enable silence suppression and generates comfort noise. The silence suppression scheme detects the parts of speech where there is no signal/information and discontinued the codec output. G.729 and G.729A could be made more effective with the use of G.729B to reduce the bandwidth utilization by not transmitting the silence 15

16 periods. The effectiveness can be achieved by using the concept of inter-frame dependency between the consecutive frames of speech. Hence the packet history becomes much important in this coding method (G.729B, (1996)). G.722 ITU developed first wideband codec G.722 in 1988 which is a low complexity waveform codec and uses the Sub-Band Adaptive Differential Pulse Coding Modulation (SB-ADPCM). This codec is called a wideband codec because it uses double sampling rate (16 khz rather than 8). It operates on 48, 56 and 64 kbps bit rates. The effect is much higher sound quality than the other VoIP codecs. It is identical to G.711. The next version G was developed in 1999 that operates on 24 and 32 kbps bit rates and widely used for conferencing terminals and hands-free systems with low frame loss. AMR NB For handling variable bit rate over the channel, the Adaptive Multi-rate (AMR) speech codec was developed by European Telecommunication standards (ETSI) and adopted by 3 rd Generation Partnership Project (3GPP). The codec uses algebraic code excited linear prediction (ACELP) technique to compress the speech. AMR-NB codec was primarily developed for mobile telephony over GSM and UMTS networks in This codec can operate at eight bit-rates from 4.75 to 12.2 kbps. it can switch bit-rate every 20ms according to the channel and network conditions, using lower bit rates during network congestion or degradation while preserving audio quality. The codec provides voice activity detection (VAD) and comfort noise generation (CNG) algorithms for reduction in bit rate, and uses an essential packet loss concealment (PLC) algorithm for handling the missing frames. AMR provides toll quality speech starting at 7.4 kbps, with near-toll quality at lower rates and greater robustness and better reproduction of non-speech sounds at higher rates. As it consists of 8 different bit- rates, AMR can be considered to be the most widely deployed codec in the world today. Adaptive Multi-Rate (AMR-WB) The Adaptive Multi-Rate Wideband Codec (AMR-WB) is a speech coder standard introduced by the 3rd Generation Partnership Project (3GPP), which is a 16

17 partnership project of various standards organizations, for compressing the toll quality speech (16,000 samples/second). The AMR-WB Codec has been approved by the ITU- T standards body and is referred to as G This speech coder is mainly used for speech compression in the 3rd generation mobile telephony. This codec has nine basic bit rates, 23.85, 23.05, 19.85, 18.25, 15.85, 14.25, 12.65, 8.85 and 6.6 kbps. This codec works on the principle of Algebraic Code Excited Linear Prediction (ACELP) for all bit rates. To reduce average bit rate, this codec supports the discontinuous transmission (DTX), using Voice Activity Detection (VAD) and Comfort Noise Generation (CNG) algorithms. The coder functions on a frame of 320 speech samples (20 ms), and a look ahead of 5 ms is required. So the algorithmic delay for the coder is 25 ms. AMR-WB is based on Adaptive Multi-Rate encoding, similar to Algebraic Code Excited Linear Prediction (ACELP). AMR-WB with wide speech bandwidth ( Hz) offers outstanding speech quality compared to narrowband speech coders ( Hz). AMR-WB is codified as G.722.2, by ITU-T, also known as Wideband coding of speech at 16 kbps. G AMR-WB is the same codec as the 3GPP AMR-WB (Peters & Davidson, 2000; Amin, 2005; Oneiss, 2002). A common file extension for AMR-WB file format is.awb. The other storage format for AMR-WB. 3GP also permits use of AMR-WB bit streams for stereo sound. Speex The Speex (2007) is an open - source codec. This was designed to be flexible and to support a wide range of speech quality and bit-rates. The Speex codec provides sampling rates of 8 to 32 khz and a variable packet rate. Speex could encode wideband speech (16 khz sampling rate) in addition to narrowband speech (8 khz sampling rate). Speex reduces the packet loss but has less to do for the corrupted packets. It has the modest (adjustable) complexity and a small memory footmark. Speex ultra-wideband (32 khz) compression in the same bit stream. It operates on kbps bit-rates for narrowband applications and for wideband it uses the kbps bit-rates. The codec also provides the Packet loss concealment, Variable bitrate operation (VBR), Voice Activity Detection (VAD) and Noise suppression and continuous Transmission (DTX) techniques to improve the quality of the speech signal. (Valin, (2007)). 17

18 3GPP2 has developed a standard wideband codec for cdma 2000 systems and successfully standardized the Variable-Rate Multimode Wideband (VMR-WB) speech codec in (VMR-WB, 2004). The operation of codec can be controlled by characteristics of speech signal and conditions of the network using selection of mode of operation. The total algorithmic delay for wideband codecs is 33.75ms and 35.06ms for narrowband. The attractive features of AMR-WB can enable the use of this codec for VoIP, streaming compressing voice messages in attachments etc. (Jelink & Ahmadi, 2006). There have been several more wideband codecs such as Internet Low Bit Rate (ilbc) coder (Anderson et al., 2004). ITU codec G.729 has also been extended to wideband using embedded variable bit-rate extension. This codec was approved by ITU in 2006 as a G (G.729.1, 2006). It also has attractive features and applications as discussed in (Trilling & Ragot, (2007); Massaloux et al. (2007)). 1.4 Speech Quality Measurement Techniques For the traditional circuit switched telephones (PSTNs) speech quality was not a major problem, so the quality measurement methods was limited. Generally the electrical signal measurement, signal to noise ratio or total harmonic distortion was applied to measure the speech quality as the signal was strictly electrical. With development of digital technologies such as VoIP, new technologies were proposed to measure to the speech quality more accurately and the perception of human end-users. Most of these measurement techniques, are active in nature i.e. injecting a signal as input of a system and analyzing the output to determine the performance of the system. The International Telecommunication Union (ITU) is one of driving forces in the world of telecommunications. Many ITU-T Recommendations are concerned with standardizing the measurement of speech quality for voice services, many of these standards are considered in this section Subjective Speech quality Measurement The obvious way to measure the speech quality is to go right to the source and receiver and use human subjects to rate the quality of the telephone calls. This is termed as Subjective Method, standardized in ITU-T Recommendation P.800 (ITU-T, 1996b) and P.830 (ITU-T, 1996). As a result of the subjective testing, the speech quality in 18

19 ITU-T standards is expressed as Mean Opinion Score (MOS), which rates the quality of a telephone call from an end-user perspective. Mean opinion score is a score of subjective testing of voice quality as perceived by large number of subjects (people) listening to the speech over a communication system. To determine the MOS for a particular telephone call, a statistically valid group (with equal number of males and females) is required. The testing considers a number of degradation factors such as packet loss, network noise, talker echo, distortion, end-to-end delay and other transmission problems. These recommendations uses the MOS scale that ranges between 1 and 5 and the MOS of some voice transmission session is the average estimate of voice quality rates assigned by a group of people as shown in the Table 1.1. Table 1.1 Subjective speech quality measurements Quality of Speech Mean Opinion Score (MOS) User s Opinion Excellent 5 Very Satisfied Good 4 Satisfied Fair 3 Some are Satisfied Poor 2 Almost all dissatisfied Bad 1 Not Recommended The MOS score 1 corresponds to bad quality and 5 to excellent quality. The MOS score 4.0 can be considered toll quality within the telephone industry, this type of voice quality can be heard for a local wired landline telephone call. Anything below MOS 4.0 would then be below the toll quality. The subjective measurement considers both listening opinion and conversation opinion tests. This also includes different rating methods such as Absolute Category Rating (ACR), Degradation Rating Category (DCR) and Comparison Category Rating (CCR) Objective Speech Quality Measurement Other methods are known as objective that depend on comparison of the received signal with the original signal to measure the perceived quality of speech in the 19

20 terms of MOS. These methods are known as intrusive methods as they require the injection of the original signal to analyze the distortion of the received signal. Several signal-based models were used for the estimation of speech quality under different bandwidths as Telecommunication Objective Speech Quality Assessment (TOSQA) (ITU-T Contr 19, (2001)). Perceptual Speech Quality Measure (PSQM) (P.861, 1998), Perceptual Analysis Measurement System Measure (PAMS) (ITU-T Contr D.001, 2001) and the normally used method for measuring the speech quality intrusively is known as Perceptual Evaluation of Speech Quality (PESQ). It is standardized as ITU-T Recommendation P.862 (ITU-T, (2001)). The PESQ algorithm is a fusion of two perceptually motivated measures PAMS and PSQM99. PESQ has been based on a designed model to have the high prediction accuracy for different types of codecs and network conditions. PESQ can produce robust estimate of Speech quality in the presence of wide range of noises that includes white Gaussian noise, Traffic Noise, Office Noise, Babble Noise and Street Noise. The recent model Wideband Extension to PESQ (WB-PESQ) (P.862.2, 2005) could be used as the reference for estimation of wideband speech quality. WB-PESQ has been very similar to PESQ which is used for narrowband speech, except these two points: (1) For WB-PESQ the input-filter has flat pas-band characteristic up to 7 khz and the PESQ used the IRS-Receive type high pass characteristic. (2) The mapping function differs for wideband transmission. The E-Model defined by the ITU-T recommendation G.107 (G.107, 2005) is an analytical. The other objective measurement category is non-intrusive estimate of the quality that depends on either the received signal or the networking parameters without the need for the original signal. The two main methods in this category are ITU-T Recommendation P.563 (ITU-T, (2004)) and the E-model as defined in ITU-T Recommendation G.107 (ITU-T, (2009)). Many other standards and methods have been proposed by other organizations, other researchers, and the authors of this chapter independent of the ITU-T, these attempts will be discussed in detail later in the chapter. 20

21 1.5 Advantages of VoIP VoIP is the hottest research topic of telecommunications. It has many advantages over the traditional PSTNs, notably the lower cost and the increasing functionality. There are so many factors/ parameters which makes voice over IP better than traditional circuit switched network (PSTN). There are so many successes that VoIP has over traditional circuit network. This has attracted the communication industries and business community towards VoIP service. Low cost: The main advantage of VoIP service is the low cost, which makes VoIP service very suitable for long distance calls. Another key advantage is being able to combine phone calls with business data. For daily use of this technology one has to pay only for internet connection or internet service only, there are no extra chargers (phone calls) like cabling charges etc. Several industrial organizations are using it for voice calls. As high-speed Internet access is cost effective, VoIP is very popular for small businesses these days. These days some telecommunication companies and Internet Service Providers (ISPs) are offering Voice over IP deals to target the small business sector (Naylor and Leneord (1982); Davidson et al. (2006)). Integration of Services: Since the number of calls on the traditional network are more. VoIP network provides the integration with traditional PSTN system. The communication from VoIP system to the PSTN is very easy (Davidson et al. (2006)). Bandwidth Efficiency: VoIP systems consume less bandwidth as compare to PSTNs. A VoIP takes only 6-8 Kbps using G.711 codec, whereas PSTN takes 64 Kbps of bandwidth to make a call and thus it saves tremendous amount of bandwidth. Besides G.711coding scheme VoIP systems are capable of implementing more advanced voice coding schemes which takes lesser bandwidth. For instance the VoIP system can implement G.726, G.728, G.729 and G.723 coding schemes which uses 32 kbps, 16 kbps, 8 kbps and 6.3 kbps bandwidth respectively (Xiong et al. (2003)). Scalability: Traditionally available PBX phone systems consist of various ports for the telephones to plug in to. Since the VoIP communication is software based so to update the network is easy by the addition of more features. VoIP systems offer larger flexibility to run a number of Virtual Users over each network socket (Davidson et al. (2006)) 21

22 Advanced features: It has more features that it has adopted from Internet technologies. Voice-related services, such as caller-id, call forwarding and broadcast messaging, become simpler to maintain and can be updated as needed. Fax over IP: The ITU-T recommendation T.38 provides the feature of Fax over the data network. Integration of VoIP with PSTN is not limited to just voice calls but with traditional application too. Using this service one can also send fax over IP networks (Wallingford, (2005); Davidson et al. (2006)). 1.6 Disadvantages of VoIP The VoIP is becoming popular due its above mention advantages but still it has many disadvantages and the brief description of these disadvantages is presented in this section. (a) No service during a power outage: During a blackout a regular phone is kept in service by the current supplied through the phone line. This is not possible with IP phones, so when the power goes out, there is no VOIP phone service. One solution to this problem is to use battery backups or power generators to provide electricity. If one decide to continue subscribing to a regular phone line as an emergency backup, consider that monthly cost cuts into ones overall VOIP savings. However, VOIP would still make sense in this case if your home or business made significant long distance calls (Wallingford, (2005)). (b) Voice Quality and Reliability: The quality of voice of VoIP system is not good, Because VoIP depends upon an Internet connection, the VOIP service shall be effected by the quality of broadband Internet service and occasionally by the demerits of a computer system which may results into distorted voice quality, so sometimes users complain about distortions, delay and sometimes echo in the signal quality. The quality of the VoIP signal is degraded by various impairment factors like packet loss, delay, jitter, Bad and unreliable internet connection and hardware components (Wallingford, (2005)).. (c) Emergency Calls: VoIP system provides 911 emergency services which are different from the traditional PSTN. The PSTN phone service was associated with a particular phone number having fixed address, while the VoIP services could be 22

23 used from internet, so the location of the caller could not be determined automatically. So in case of change in any information the VoIP service provider should be provided accurate address and information immediately (Davidson et al. (2006)). (d) Security: It is the most important factor which limits liability of VoIP. Security is the most concerned area of any communication network for the security and secrecy of voice and data transmission (Wallingford, (2005); Brezocnik & Valovic, (2001)). Overall, the disadvantages of VoIP aren't significant enough to deter the average consumer from using the technology; especially with the fact that they know that the calls they are making are free. But one can rest assured that the technology will only get more reliable as time goes by. 1.7 Factors Affective Quality of Service (QoS) of VoIP Quality of service (QoS) refers to the ability of the network to provide better service over various underlying technologies and maintain an adequate standard of voice quality. The QoS for VoIP, is made through some specific parameters and the values of these parameters determine the degradation of the speech signal (Peters & Davidson, (2000); Amin, (2005); Simak & Koska, (2001)). In this work one are concerned with the Network QoS, which is based on the QoS framework, as suggested by Peters & Davidson (2000). One has also identified some QoS parameters for the implementation in our future Simulink study of VoIP. Some of these factors are described as below: Delay It is defined as the time taken by the speech to reach from the speaker s mouth to the listener s ear and round trip (total delay) delay is the sum of two one way delays in a user s call (Kostas et.al. (1998)). In voice communication the voice signal should reach to the listener in a very short time interval, as it should seem to be a face-to face communication. If signal is delay, it will impair the quality of speech in VoIP network. The degradation caused in the signal depends upon the amount of delay added by VoIP network components. There are certain factors which are responsible for causing delay. These factors are: 1.) VoIP network components; 2.) Various parameters of VoIP 23