1. 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

2. 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

3. 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.

4. Technical Challenges- QoS Issues  Packet Loss  Packet Delay  Network Jitter

5. 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

6. 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

7. Delays Encountered in IP Telephony

8. 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.

9. 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

10. 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

11. The Cisco data and IP Telephony Network Architecture

12. Lucent IP and PSTN Architecture

13. 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 http://dl.comsoc.org/cocoon/comsoc/servlets/GetPublication;jsessionid =01C893DE28A4EBF38E5B9DFFBD461325?id=12149 http://dl.comsoc.org/cocoon/comsoc/servlets/GetPublication;jsessionid =01C893DE28A4EBF38E5B9DFFBD461325?id=12149