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A Study on Quality of Service Issues in Internet Telephony. IP Telephony – Applications and Services Advantages and benefits of Voice over IP Technical Challenges – QoS issues Proposed Solutions. VOIP- Applications and Services. Integration of Data, Voice and Fax Sound Grading
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A Study on Quality of Service Issues in Internet Telephony IP Telephony – Applications and Services Advantages and benefits of Voice over IP Technical Challenges – QoS issues Proposed Solutions
VOIP- Applications and Services • Integration of Data, Voice and Fax • Sound Grading • Unified messaging • Video telephony • Web-based call centers • Low-cost voice calls • Remote teleworking
Advantages and Benefits Benefits put in three categories as • Cost Reduction • Simplification • Consolidation • Efficient usage of already existing network resources • Reduced number of access links • No per-minute distance sensitive charges. • No bandwidth limitation.
Technical Challenges- QoS Issues • Packet Loss • Packet Delay • Network Jitter
Packet Loss • Packet loss in IP networks affect time sensitivity of voice transmission. • Possible solutions • Noise Substitution • Packet repetition • Packet Interpolation • Frame interleaving • Network upgrade • Forward Error correction
Packet Delay • Codec delay • Encoding delay ( frame processing delay + lookahead delay) • Decoding delay is half the encoding delay • Higher compression achieved at the price of longer delays • Serialization delay • Longer frames result in higher delay in transmitting the packet • Higher speed lines reduce serialization delays • Queuing delay • Occurs at the switching and transmission points of the network • Can be reduced using mechanisms such as differentiated services and Resource Reservation Protocol ( RSVP) • Other sources of delay • Delays caused by modems in dial up networks, delays due to inefficient operating systems and sound card delays • Can be avoided by using digital lines and using gateway cards with specialized Digital signal processors
Network Jitter • Variance in the inter-frame arrival time at the receiver is called jitter • Jitter occurs due to variability of queuing delays in the network • Can be reduced by using Jitter buffers • To allow for variable packet arrival times and still achieve steady stream of packets, the receiver holds the first packet in a jitter buffer , before playing it out. • Selection of Jitter buffer is crucial to IP telephony systems. • Cisco, Hypercom and Netrix offer intelligent buffers that adjust automatically according to network availability.
Network Support for QoS • Providing controlled networking environment • Using management tools to configure network nodes , monitor performance and manage capacity and flow on a dynamic basis • Traffic prioritized by protocol, location and application type • Queuing mechanisms manipulated to reduce delays • Adding control protocols such as RTP, RTCP, RSVP to provide greater assurance of controlled QoS within the network • Other Networking tools to provide QoS include • Congestion Management ( Weighted fair queuing) • Qos Signaling (IP precedence and RSVP) • Packet Residency • RTP header compression • Generic traffic shaping • Weighted Random Early detection
Existing Service models and mechanisms • Two keymodels: Intserv and Diffserv. • THE INTEGRATED SERVICE MODEL • Guaranteed service for applications requiring a fixed delay bound • Controlled-load service for application requiring reliable and enhanced best-effort service • THE IETF DIFFERENTIATED SERVICES FRAMEWORK • The first approach specifies the QoS in deterministically or statistically quantitative terms of throughput, delay, jitter, and/or loss. Such approach is called quantitative Diffserv. • The second approach specifies the services in terms of some relative priority of access to network resources and is called prioritybased Diffserv. • Existing Solutions • The CISCO Solution : Enterprise IP Telephony • LUCENT Gateway Solution for Service Provider networks
References 1. G. A. Thom, “H.323: The Multimedia Communications Standard for Local Area Networks,” IEEE Commun. Mag., Dec. 1996. 2. ITU Rec. H.323, “Visual Telephone Systems and Equipment for Local Area Networks which Provide a Non-Guaranteed Quality of Service,” Nov. 1996. 3. Samir Mohamed, Francisco Cervantes-pérez, Hossam Afifi, "Integrating networks measurements and speech quality subjective scores for control purposes", IEEE INFOCOM 2001 - The Conference on Computer Communications, no. 1, April 2001 pp. 641-649 4. Goodman, O. Lockart, and W. Wong, “Waveform Substitution Techniques for Recovering Missing Speech Segments in Packet Voice Communications,” IEEE Trans. Acoustics, Speech and Sig. Processing, Dec.1986, vol. ASSP-34, no.6, pp. 1440–48. 5. IEEE Communication Society Library. http://dl.comsoc.org/cocoon/comsoc/servlets/GetPublication;jsessionid=01C893DE28A4EBF38E5B9DFFBD461325?id=12149