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

Delays Encountered in IP Telephony

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

The Cisco data and IP Telephony Network Architecture

Lucent IP and PSTN Architecture

References 1. G. A. Thom, “H.323: The Multimedia Communications Standard for Local Area Networks,” IEEE Commun. Mag., Dec ITU Rec. H.323, “Visual Telephone Systems and Equipment for Local Area Networks which Provide a Non-Guaranteed Quality of Service,” Nov Samir Mohamed, Francisco Cervantes-pérez, Hossam Afifi, "Integrating networks measurements and speech quality subjective scores for control purposes", IEEE INFOCOM The Conference on Computer Communications, no. 1, April 2001 pp 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– IEEE Communication Society Library. =01C893DE28A4EBF38E5B9DFFBD461325?id= =01C893DE28A4EBF38E5B9DFFBD461325?id= =01C893DE28A4EBF38E5B9DFFBD461325?id=12149