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VoIP Multimedia over WiMAX 1 of 13 file F www cse574 06 ftp wimax voip index html Maneesh Bakshi The 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 Introduction to the working of VoIP and multimedia transmission over wireless The softswitching technique for compatibility with WiMAX The challenges being faced that need to be overcome for successful deployment and 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 these developments on the existing Telecom and Satellite CableTV industry Keywords WiMAX VoIP Multimedia IMS 802 16 SIP H 323 IPTV Softswitch See also TCP Optimizations over Wireless QoS for WiMAX Other Reports on Recent Advances in Networking Back To Raj Jain s Home Page 1 Introduction to VoIP 1 1 How does VoIP work 1 2 Signaling Protocols 1 2 1 H 323 1 2 2 Session Initiation Protocol SIP 1 3 Softswitch 1 4 Main Challenges 1 4 1 Voice Quality 1 4 2 Security 1 4 3 E911 CALEA Support 2 Introduction to IMS 2 1 WiMAX IPTV 2 2 How does it work 2 3 Services offered 2 3 1 Personal Video Recorder 2 3 2 Interactive TV 2 3 3 PC TV 3 WiMAX Quality of Service QoS 11 27 2013 2 15 AM VoIP Multimedia over WiMAX 2 of 13 file F www cse574 06 ftp wimax voip index html 4 Impact on the Telecom and Satellite Cable TV Industry 5 Recently Launched Products 6 Pre WiMAX Happenings and Industry Forecasts 7 Summary 8 References 9 List of Acronyms 1 0 Introduction to VoIP Ever since its advent VoIP has opened new doors for telephony bringing forward immense possibilities The basic reason for the popularity of VoIP is the cost which is very low as compared to the conventional telephony services The concept of transmission of voice over data stream makes it possible to have VoIP transmitted 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 involves digitization of voice the isolation of unwanted noise signals and then the compression of the voice signal using compression algorithms codecs After the compression the voice is packetized to send over an IP network each packet needs a destination address and sequence number and data for error checking The signaling protocols are added at this stage to achieve these requirements along with the other call management requirements When a voice packet arrives at the destination the sequence number enables the packets to be place in order and then the decompression algorithms are applied to recover the data from the packets Here the synchronization and delay management needs to be taken care of to make sure that there is proper spacing Jitter buffer is used to store the packets arriving out of order through different routes to wait for the packets arriving late 1 2 Signaling Protocols 11 27 2013 2 15 AM VoIP Multimedia over WiMAX 3 of 13 file F www cse574 06 ftp wimax voip index html The signaling protocols H 323 SIP are used to setup the route for the transmission over the IP network the Gateway protocols like the Media Gateway Control Protocol are used to establish control and status in the media and signaling gateways Routing UDP TCP and transport protocols RTP are used once the route is established for the transport of the data stream as shown in Figure 1 1 2 1 H 323 H 323 is the ITU T standard for packet based multimedia communication though originally developed for multimedia conferencing over LAN s it was later modified for VoIP as well With versions coming out in 1996 and 1998 the standard has faced stiff competition from the other protocol SIP which was specifically designed for VoIP but is more used because of its wide existence in the already installed networks The standard is interoperable and has both point to point and multipoint capabilities H 323 uses a number of other sub protocols for the various functions H 255 0 Registration Admission Status Call Signaling Control H 245 Terminal Capability Exchange Media Description Control of Logical Channel Also H 323 offers specifications for call control channel setup codecs for the transmission of Real time video and voice over the networks where the QoS and guaranteed services are not available For the transport RTP is 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 and uses a text based syntax that is comparable to HTTP uses in web addresses A web address is comparable to a telephone number in a SIP network also the PSTN phone numbers are also compatible in a SIP network ensuring interfacing with PSTN systems SIP also provides a mobility function to the users SIP also supports multiple media sessions during a single call hence users can share a game use instant message IM and talk at the same time SIP works with most protocols like RTP Session Description Protocol SDP Session Announcement Protocol SAP A lot of other protocols are also needed when it comes to the transport and signaling with the PSTN networks RSVP LDAP RADIUS SIP works on a client server architecture where the clients are referred to as User Agents UA UA interact with the server mainly through a PC with a telephony agent or IP phone The servers are of four types Registration Redirect Server Proxy Server 11 27 2013 2 15 AM VoIP Multimedia over WiMAX 4 of 13 file F www cse574 06 ftp wimax voip index html UA server The SIP messages used for communication between the client server are INVITE ACK OPTIONS REGISTER CANCEL BYE 1 3 Softswitch The new innovation in switching Softswitch is far less expensive both in terms of purchase and maintenance as compared to the conventional switches used in PSTN networks Interoperability is primarily the biggest cutting edge advantage that the service provider gets out of using a Softswitch Softswitch is an enhancement over the existing gatekeeper technology which supported H 323 since H 323 was only restricted to LAN s the activities and capabilities of the gatekeeper were restricted to a few gateways which were managed by a single gatekeeper As the networks became larger there was a need for more efficient and smart solutions for managing all these


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WUSTL CSE 574S - VoIP/Multimedia over WiMAX

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