Transcript A Study on Quality of Service Issues

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
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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
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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
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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. 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