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WUSTL CSE 574S - VoIP/Multimedia over WiMAX

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Maneesh BakshiThe paper discusses the implementation of Voice over IP (VoIP) and IP Multimedia Subsystem services (IMS)over the much sought after wireless standard WiMAX (802.16). The key issues addressed are : Introductionto the working of VoIP and multimedia transmission over wireless. The softswitching technique forcompatibility with WiMAX. The challenges being faced that need to be overcome for successful deploymentand their solutions. The pre-standard installations and developments happening around the world at this time.The various products offered by the vendors compliant to the technology, and the impact of thesedevelopments on the existing Telecom and Satellite/CableTV industry.Keywords - WiMAX, VoIP, Multimedia, IMS, 802.16, SIP, H.323, IPTV, SoftswitchSee also: TCP Optimizations over Wireless | QoS for WiMAX |Other Reports on Recent Advances in NetworkingBack To Raj Jain's Home PageIntroduction to VoIP1.1 How does VoIP work?1.2 Signaling Protocols1.2.1 H.3231.2.2 Session Initiation Protocol (SIP)1.3 Softswitch1.4 Main Challenges1.4.1 Voice Quality1.4.2 Security1.4.3 E911/CALEA Support1.Introduction to IMS2.1 WiMAX IPTV2.2 How does it work?2.3 Services offered2.3.1 Personal Video Recorder2.3.2 Interactive TV2.3.3 PC-TV2.WiMAX - Quality of Service (QoS)3.VoIP/Multimedia over WiMAX file:///F:/www/cse574-06/ftp/wimax_voip/index.html1 of 13 11/27/2013 2:15 AMImpact on the Telecom and Satellite/Cable TV Industry4.Recently Launched Products5.Pre WiMAX Happenings and Industry Forecasts6.Summary7.References8.List of Acronyms9.1.0 Introduction to VoIPEver since its advent VoIP has opened new doors for telephony bringing forward immense possibilities. Thebasic reason for the popularity of VoIP is the cost which is very low as compared to the conventionaltelephony services. The concept of transmission of voice over data stream makes it possible to have VoIPtransmitted and received using anything that uses IP - laptops, PC's, WiFi enabled handsets etc..1.1 How does VoIP work?VoIP uses Internet Protocol for transmission of voice as packets over IP networks. The process involvesdigitization of voice, the isolation of unwanted noise signals and then the compression of the voice signalusing compression algorithms/codecs.After the compression the voice is packetized to send over an IPnetwork, each packet needs a destination address and sequence number and data for error checking. Thesignaling protocols are added at this stage to achieve these requirements along with the other call managementrequirements. When a voice packet arrives at the destination, the sequence number enables the packets to beplace in order and then the decompression algorithms are applied to recover the data from the packets. Herethe synchronization and delay management needs to be taken care of to make sure that there is properspacing. Jitter buffer is used to store the packets arriving out of order through different routes, to wait for thepackets arriving late.1.2 Signaling ProtocolsVoIP/Multimedia over WiMAX file:///F:/www/cse574-06/ftp/wimax_voip/index.html2 of 13 11/27/2013 2:15 AMThe signaling protocols H.323, SIP are used to setup the route for the transmission over the IP network, theGateway protocols like the Media Gateway Control Protocol are used to establish control and status in themedia and signaling gateways. Routing (UDP, TCP) and transport protocols (RTP) are used once the route isestablished for the transport of the data stream as shown in Figure 11.2.1 H.323H.323 is the ITU-T standard for packet based multimedia communication, though originally developed formultimedia conferencing over LAN's it was later modified for VoIP as well. With versions coming out in 1996and 1998, the standard has faced stiff competition from the other protocol SIP which was specificallydesigned for VoIP, but is more used because of its wide existence in the already installed networks. Thestandard is interoperable and has both point to point and multipoint capabilities. H.323 uses a number of othersub protocols for the various functions.H.255.0 - Registration, Admission, Status, Call Signaling, ControlH.245 - Terminal Capability Exchange, Media Description, Control of Logical ChannelAlso H.323 offers specifications for call control, channel setup, codecs for the transmission of Real time videoand voice over the networks where the QoS and guaranteed services are not available. For the transport RTPis used for real time audio and video streaming.1.2.2 Session Initiation Protocol (SIP)SIP is the IETF standard for VoIP signaling. It is based on the existing protocols like SMTP and HTTP, anduses a text based syntax that is comparable to HTTP uses in web addresses. A web address is comparable to atelephone number in a SIP network, also the PSTN phone numbers are also compatible in a SIP networkensuring interfacing with PSTN systems. SIP also provides a mobility function to the users. SIP also supportsmultiple media sessions during a single call hence users can - share a game, use instant message (IM), and talkat the same time. SIP works with most protocols like RTP, Session Description Protocol (SDP), SessionAnnouncement Protocol (SAP). A lot of other protocols are also needed when it comes to the transport andsignaling with the PSTN networks - RSVP, LDAP, RADIUS. SIP works on a client server architecture, wherethe clients are referred to as User Agents (UA). UA interact with the server mainly through a PC with atelephony agent or IP phone. The servers are of four types :RegistrationRedirect ServerProxy ServerVoIP/Multimedia over WiMAX file:///F:/www/cse574-06/ftp/wimax_voip/index.html3 of 13 11/27/2013 2:15 AMUA serverThe SIP messages used for communication between the client-server are INVITE, ACK, OPTIONS,REGISTER, CANCEL, BYE1.3 SoftswitchThe new innovation in switching - Softswitch is far less expensive both in terms of purchase and maintenanceas compared to the conventional switches used in PSTN networks. Interoperability is primarily the biggestcutting edge advantage that the service provider gets out of using a Softswitch. Softswitch is an enhancementover the existing gatekeeper technology which supported H.323, since H.323 was only restricted to LAN's theactivities and capabilities of the gatekeeper were restricted to a few gateways which were managed by asingle gatekeeper. As the networks became larger, there was a need for more efficient and smart solutions formanaging all these services, which were answered by the


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WUSTL CSE 574S - VoIP/Multimedia over WiMAX

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