INDHU 2 | Session Initiation Protocol | Voice Over Ip

VOICE TRANSMISSION ON IP BASED NETWORK USING WIMAX A PROJECT REPORT

Submitted by

R. ABINAYA(42007106002) S. INDHUMATHI(42007106035) P. LOGESWARI(42007106046)
In partial fulfillment for the award of the degree of

DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING TAGORE ENGINEERING COLLEGE CHENNAI JANUARY-2011

such as one-way delay and packet loss. The Wimax technology is the way to avert the impending crisis of rural connectivity . using H. VoIP quality is measured according the quality perceived by the end users as well as through conventional network parameters. A WiMAX infrastructure will let service providers deliver VoIP to rural areas. The results put in evidence a good quality for VoIP services with 60 simultaneous users in a WiMAX link with resources pre-provisioned. its implications and applications and its wireless capabilities. the study of VoIP and Wimax systems. .323 and SIP (session initiation protocol) protocols. This project explains about the purpose of implementation of VoIP over Wimax. VoIP applications are being widely used in today's networks challenging their capabilities to provide a good quality of experience level to the users. currently employed in some parts of the world.ABSTRACT: The Worldwide becoming new era of communication. The work presented here is a unique contribution assessing the VoIP sessions quality on a real WiMAX test-bed. is Interoperability for Microwave Access (WIMAX).

and 3. .5-GHz spectrum is not available for use in the United States . they will deliver VoIP services over a less expensive wireless infrastructure. The 2. AT&T and Covad's announcement should help confirm that wireless VoIP can be made reliable and cost-effective. ‡ ‡ It allows 2-way voice transmission over broadband connection.certified equipment has yet been released²though some shipments are expected in 2005. instead of trenching fiber everywhere. which carries the risk of interference from other wireless devices.5-GHz spectrum is owned largely by Sprint and Nextel. It is also called IP telephony.5. broadband telephony. voice over broadband. VoIP over WiMAX is attractive for both enterprises and carriers. ‡ VoIP is packetisation and transport of classic public switched telephone system audio over an IP network. Second. no WiMAX .5-GHz spectrum ranges. REASON FOR CHOOSING VOIP ‡ Voice over Internet Protocol (VoIP) is a technology that enables one to make and receive phone calls through the Internet instead of using the traditional analog PSTN (Public Switched Telephone Network) lines. the initial WiMAX equipment will be certified in the 2. First. but Challenges remain. just in time for AT&T and Covad's plans. while the 3. AT&T and Covad have revealed their intention to use WiMAX technology to provide VoIP and data services to rural areas to avoid local access charges. The announcement earlier this month clears up the mystery surrounding the service providers' membership in the WiMAX Forum Apparently. as Verizon is doing. even if it's pre-WiMAX. Organizations seeking a quick way to provide connectivity to branch offices should consider using high-speed wireless. internet telephony. This means that any near-term WiMAX equipment deployment will likely use unlicensed spectrum.VOIP OVER WIMAX A WiMAX infrastructure will let service providers deliver VoIP to rural areas without a backhoe.

Circuit switching is a very basic concept that has been used by telephone networks for more than 100 years. the connection is maintained for the duration of the call. Then came voip over broad band which was packet Switching.1. Unfortunately.EXISTING METHOD: Fig. that promise has been imperfectly met in the past because of both the immaturity of the existingtechnologies and the relatively high cost of networking equipment . When a call is made between two parties. Because you're connecting two points in both directions. This is the foundation of the Public Switched Telephone Network (PSTN). Broadband wireless has long held the promise of delivering a wide range of data and information services to business and residential customers quickly and cost-effectively. the connection is called a circuit. Voice transmission from PC to PC Existing phone systems are driven by a very reliable but some what inefficient method for connecting calls called circuit switching.

11b 802.16 AssignedSpectrum 2.9GHz 2.ATM Modulation System Frequency Hopping Direct Sequence No Frequency Hopping Direct Sequence No OFDM OFDM QUAM.4GHz 5.11 802.PSK.Me sh No No No Yes Fig.4 6GHz 2.COMPARISON OF DIFFERENT STANDARDS: FEATURES 802.11a 802.4GHz 10GHz-66GHz Maximum Throughput 2Mbps 11Mbps 54Mbps 54Mbps 70Mbps Transport Protocol Supported Ethernet Ethernet Ethernet Ethernet TCP/IP.2.Tabulation .11g 802. OFDM Adaptive Modulation No No Yes Network Architecture Supported QOS Point to Multipoint PTMP No PTMP PTMP PTMP PTMP.PTCM.

Modulation and coding schemes may be adjusted individually for each subscriber and may be dynamically adjusted during the course of a transmission to cope with the changing radio frequency (RF) environment. Voip over wi-max The 802.3.The orthogonal frequency division multiplexing (OFDM) modulation scheme is specified for the lower band with a single carrier option being provided as well The 802.16 protocols are highly adaptive. In the higher frequencies. 16 quadrature amplitude modulation (QUAM) and 64 QUAM are automatically invoked by the protocol to match signal characteristics with network conditions.PROPOSED MEHOD R O U T E R WWIMAX X IM A VOIP GATEWAY BOX WIMAX CPE BB INTERNET Fig. but it uses what is essentially a connection-oriented protocol somewhat akin to those of ATM and frame relay. Polling on . and they enable subscriber terminals to signal their needs while at the same time allowing the base station to adjust operating parameters and power levels to meet subscriber needs.16 standard can accommodate both continuous and bursty traffic.

‡ VoIP Application Service Provider Network: The network where the VoIP equipment is located.Mobile User ± VoIP platform located on the same premises as the WiMAX Provider¶s network 3.Fixed User ± VoIP platform located at a separate location from the WiMAX Provider¶s network. 4.Roaming ± When a WiMAX user is roaming the VoIP platform used may be located either in the visited network or the home network. Provisions for privacy. as follows: 2. The purpose here is to present the WiMAX network solutions. avoiding the simple contention-based network access schemes utilized for WLANs. to provide best practice guidelines that aggregate existing standards and recommendations to WiMAX network operators on deploying VoIP. VOIP NETWORK DEPLOYMENT SCENARIO Several categories of VoIP Networks are to be considered based on the terminal and the location of the VoIP platform 1. The three logical networks that make up the VoIP Ecosystem include the following: ‡ PSTN Carrier Network: Made up of traditional Cell and POTS line providers interconnecting using a circuit switched network.Mobile User ± VoIP platform located at a separate location from the WiMAX Provider¶s network 5. and authentication of subscribers also exist. VOIP ECOSYSTEM An ecosystem is a unit of interdependent systems interacting as a functional whole. security. The purpose of the interdependent systems is to make and receive calls. this network will include the WiMAX base stations. ‡ WiMAX Provider Network: For the objective of this paper. .Fixed User End-User VoIP Termination± VoIP platform located on the same premises as the WiMAX Provider¶s network. Advanced network management capabilities extending to layer 2 and above are not included in the standard. but the network operator also has the option of assigning permanent virtual circuits to subscribers²essentially reservations of bandwidth.the part of the subscriber station is generally utilized to initiate a session. ASN and CSN.

729 codec can also generate speech frames every 10 ms or longer sample rate . It is developed for multimedia simultaneous voice and data applications. the gateway converts the media stream as well as the control protocol between packetswitched and circuit-switched methods. however. However.Audio codec.729 CODEC is widely used in wireless networks .Signaling (SIP.Data transport (RTP.711 and G. Its voice sampling rate is 8 kHz and each sample is encoded with 8 bits resulting in a constant 64 kbps bit rate and offers very good voice quality. G.Addressing. 4. The G. The G. Samples can be packed into frames every 10 ms or another longer sampling rate. 2. or another longer sampling rate. G.GATEWAY A Gateway is a device used to translate media streams between different technologies. With VoIP. it requires a 5 ms look-ahead delay before producing any new frame.729 are the most widely used codecs in existing networks. BASIC TECHNIQUES IN VOIP 1.Each 10 ms frame contains 80 voice samples (collected at a sampling rate of 8000 samples per second).323). G729 has a data rate of 8 kbps and a 10-30 ms sample period.711 codec is currently used in a wide range of applications. Average bandwidth usage is ~40 kbps per call. . RTCP) RTP (Real-Time Transport Protocol) RTCP (Real-Time Control Protocol) 3. AUDIO CODEC The choice of CODEC used is important because it determines the required bandwidth per call. H.

Information provided by this protocol include timestamps (for synchronization).Responsible for periodical transmission of control packets to all participants in the session.711 G. Simple format addressing: <user | phone no. video etc.4.. ADDRESSING Here phone no.4kb/s 120 64 50 60 70 71 Fig. sequence numbers (for packet loss detection) and the payload format which indicates the encoded format of the data.4kb/s 8kb/s 8kb/s 8kb/s 12. you¶ll be able to dial just as you always have. .729 G. The RTCP is used to specify Quality of Service (QOS) feedback and synchronization between the media streams. Effective efficiency of different types of codecs DATA TRANSPORT PROTOCOL RTP RTP stands for Real-Time Transport Protocol.SPEECH CODEC NO. Application layer protocol for transmitting real time audio.729 G. OF BYTES PER FRAME G.The data transfer protocol. is converted into IP address.>@<domain | hostname | IP address> If you make a call using a phone with an adaptor.729 GSM-EFR 80 24 10 10 10 31 BIT-RATE TOTAL NO OF BYTES EFFECTIVE EFFICIENCY 67% 37% 20% 33% 43% 44% 64kb/s 6.Supports for multi-point communication. deals with the transfer of real-time multimedia data.723. RTCP RTCP stands for Real-Time Control Protocol. If there is no adapter we have to use the IP address of the particular phone.1 G.

These protocols. Agreeing on coding/decoding procedures. chat. Their charter States that SIP is a text-based protocol. they cannot apply only to some specific set of session types ‡ Simplicity is key ‡ Existing IP protocols and architectures are re-used and integrated tightly . Significant efforts were undertaken in past decades to develop the signaling protocols in use in today¶s telephone network. This group has worked long and hard to help SIP mature. are defined in large detailed specifications developed by various standardization organizations Signaling in VoIP is needed for: 1. Agreeing on port numbers for RTP/RTCP sessions.H.323. has become foundation for VoIP and unified communications. It was one of many efforts and has been led by the IETF-SIP working group. SIP services and features are provided end-to-end ‡ Extensions and new features must be generally applicable. OVERVIEW OF SIP Intense work on SIP really began in earnest with the Internet Engineering Task Force (IETF) in1999. The basic model and architecture defined for SIP sets out some specific characteristics: ‡ Wherever possible. 2. including numerous extensions. Types of signaling protocols: 1 . such as Signaling System H. video. similar to HTTP and SMTP.323 and SIP.SIP (Session Initiation Protocol). What began as a series of proposed drafts and standards . 2. and virtual reality. for initiating interactive Communication sessions between users. Such sessions include voice. Locating partners. 3. interactive games.SIGNALLING Signaling is one of the most important functions in the telecommunications infrastructure because it enables various network components to communicate with each other to set up and tear down calls. also known as the public switched telephone network (PSTN).

as SIP is a text protocol. Terminal gets the request and arranges with IWF. It sends an INVITE request to IWF. so there has to be a way to resolve some of the commonconventions in the active and current IP address. can be imbedded in email messages or Web pages. a URL might point to an email-like address.SIP uses an addressing structure similar to email addresses.25 with no changes. 5. an H. SIP operates independently of the IP network layer. Frame Relay. so the addresses. IWF finishes the dialog. Thus.323 address or even a telephone number on the PSTN. Additionally. AAL5. using SIP. 6. Terminal stops the dialog. which are SIP URLs or URIs (Uniform Resource Locaters or Indicators). UA informs IWF to finish the dialog. 3. . SIP is text based. SIGNAL CONTROL FOR SIP: 1. ATM. Instead of UA. a media format for the following dialogs. SIP URLs and URIs are network-neutral. 7. It requires only unreliable packet delivery and provides its own reliability mechanism. SIP can run over IPX. 2. Users may log in any where and be dynamically assigned an IP address. or X. the IWF arranges with Terminal to get a translation address. Although it¶s widely used in IP networks today(usually over UDP to avoid the overhead of TCP). UA wants to make a collection with Terminal. 4.425 8. and informs Terminal to disconnect the link with H. IWF translates INVITE to SETUP and tells Terminal. Terminal and UA communicate with media channel.

We also define how services can be implemented within the H.5. the gateway. the gatekeeper. Signal Control ± Start from SIP UA OVERVIEW OF H.323. and the multipoint control unit.323 architecture.Fig. In this section we describe. We then define the various protocols that are part of the H.323 architecture by defining the main components of the architecture: the terminal. the H.323 family and are used by the components of the architecture for communicating with each other. . at a high level.

IWF translates it to be INVITE without media capability description. The only requirement is that each zone contain exactly one GK. Each zone consists of a single H.6. interconnected via a LAN. A typical H. a number of H.323 gateways (GWs). That is because the Terminal¶s media capability is not recognized by IWF.Signal Control ± Start from H.323 Terminal When a Terminal invites a SIP UA.The H. or just a single LAN.323 standard was initially targeted to multimedia conferencing over LANs that do not provide guaranteed QoS. SIP media adopt Offer/Answer model. A zone can span a number of LANs in different locations. firstly. which acts as the administrator Fig.323 network is composed of a number of zones interconnected via a WAN. and a number of multipoint control units (MCUs). it will send SETUP to IWF.323 terminal endpoints (TEs).323 gatekeeper (GK). It means UA¶s . a number of H.

which produces high overhead and therefore wastes network bandwidth. This paper proposed the use of a novel multiplexing technique. its applications are injecting a huge number of small packets in the network. Accordingly. overload. IWF arranges with Terminal about transport layer address and media format. the Delta. 7. So. The details are as follows: 1. IWF runs the dialog stop program which is defined in H. Establishing the dialog. thus improving the overall network performance. In the Delta-Multiplexing technique. the VoIP packets destined to the same destination gateway are aggregated in a single UDP/IP header.media capability can only be advanced in INVITE or the Response of INVITE. IWF records it and arranges with Terminal. therefore reducing the header overhead and saving network bandwidth. 5. 6.Multiplexing technique greatly saves bandwidth. Terminal sends a SETUP to IWF and makes a connection between Terminal and SIP UA. 2. Delta-Multiplexing. Moreover. .245 and finish the dialog. 4. therefore reducing network traffic. to save the wasted bandwidth. and congestion. IWF translates SETUP into INVITE and then informs SIP UA with this INVITE. SIP UA accepts the request and sends response to IWF. 3. Unfortunately. The result showed that Delta-Multiplexing is capable of saving between 68% and 72% as compared to conventional techniques (without multiplexing). Terminal and UA can communicate each other with media channels.323 and informs UA stop with SIP BYE. Terminal informs IWF to release control connection with H. UA tells IWF its media capability in the Response of INVITE. the Delta-Multiplexing technique reduces the size of the packets payload by transmitting the difference between the consecutive packets payloads. EFFECTIVE BANDWIDTH UTILIZATION Voice over Internet Protocol (VoIP) has been dominating the telecommunications world. We have simulated the DeltaMultiplexing technique using a 14-byte LPC codec. Moreover. The response which include UA¶s media capability is an OFFER. the Delta-Multiplexing technique reduces the number of VoIP packets running over the network.

D. IEEE Comm. RFC 3261.. Peterson. June 2002 4. ³Voice over Internet Protocol and Human Assisted E-Commerce´. Internet Engineering Task Force. Trecordi. 1998. J. Furnell. ³Packet based Multimedia Communications Systems´.J. C. SIP: Session Initiation Protocol.velammal. H.323. Geneiatakis. http://tifac.voip. Johnston. Camarillo. 2. Ziti Pubs. ITU-T Rec. 2. Kambourakis..Lambrinoudakis.pdf . G. v. S. S. Decina and V. Gritzalis.Ramakhrishnan. and W. H. 147-156. 3. K. Samos. Schulzrinne. S. R. P. J Gerard . A. A. G. pp.). Rosenberg. "Voice-over-IP: The Future of Communications". Greece. Waldron.. Rachel. INC'05International Network Conference.2002 6.com 5.323´.org/CoMPC/articles/30. Katsikas (Eds. ³Session Initiation Protocol Security Mechanisms: A state-of-the-art review´. Magazine. Sparks et al. Dagiouklas.REFERENCE 1. Kumar.S. www. July 2005 7. ³Performance Analysis of Different Codecs in VoIP Using H. Sept 1999. April 29. R.V. M.

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