Goal We want to know. Introduction. What is VoIP? Carrier Grade VoIP. What is Meant by Carrier-Grade? What is Meant by VoIP? Why VoIP?

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1 Goal We want to know Introduction What is Meant by Carrier-Grade? What is Meant by VoIP? Why VoIP? VoIP Challenges 2 Carrier Grade VoIP Carrier grade Extremely high availability % reliability (high reliability) Fully redundant, self-healing AT&T carries about 300 million voice calls a day (high capacity). Highly scalable Short call setup time, high speech quality No perceptible echo, noticeable delay and annoying noises on the line Interoperability Carrier grade and VoIP mutually exclusive Phone A serious alternative for voice communications with enhanced features 3 What is VoIP? SIP phone Transport voice traffic using the Internet Protocol (IP) One of the greatest challenges to VoIP is voice quality. One of the keys to acceptable voice quality is bandwidth. Control and prioritize the access Internet: best-effort transfer VoIP!= Internet telephony The next generation Telecommunications Access and bandwidth are better managed. 4 1

2 IP and TCP IP A packet-based protocol Routing on a packet-by-packet base Packets transfer with no guarantees May not receive in order May be lost or severely delayed TCP Retransmission Assemble the packets in order Congestion control Useful for file-transfers and Data and Voice Data traffic Asynchronous - can be delayed Extremely error sensitive Voice traffic Synchronous - stringent delay requirements More tolerant for errors IP is not for voice delivery. VoIP must Meet all the requirements for traditional telephony Offer new and attractive capabilities at a lower cost 5 6 Why VoIP? Lower Equipment Cost Why carry voice? Internet supports instant access to anything However, voice services provide more revenues. Voice is big business. Why use IP for voice? Traditional telephony carriers use circuit switching for carrying voice traffic. Circuit-switching is not suitable for multimedia communications. IP: lower equipment cost, low operating expense, integration of voice and data applications, potentially lower bandwidth requirements, the PSTN switch Proprietary hardware, OS, applications High operation and management cost Training, support and feature development cost The IP world Modulized and mass-produced (hardware/software) Distributed client-server architecture Rapid feature development Customize Compared with IN Not as open and flexible as IP. A few highly successful services widespread availability of IP 7 8 2

3 Voice/Data Integration Lower Bandwidth Requirements A single network to support a wide range of applications Voice, data, video IP-based call centers E-commerce IP-based telephony Click-to-talk application Web collaboration Video conferencing IP-based PBX IP-based voice mail Welcome to customer service. How can we help you? IP-based Call center 9 PSTN Human speech frequency < 4K Hz The Nyquist Theorem: 2*4k samples per second G kbps=8k * 8 bits Sophisticated coders Save more bandwidth by silence-detection Compression 32kbps, 16kbps, 8kbps, 6.3kbps, 5.3kbps GSM 13kbps Traditional telephony never changes. VoIP two ends of the call negotiate the coding scheme 10 The Widespread Availability of IP VoIP Challenges IP LANs and WANs Dial-up Internet access The ubiquitous presence Packet-based solutions for commercial carriers Voice over IP (VoIP) Voice over Frame Relay (VoFR) Voice over Asynchronous Transfer Mode (VoATM) Only for the backbone of the carriers VoIP must offer the same reliability and voice quality as PSTN % and Toll quality Mean Opinion Score (MOS) 5 (Excellent), 4 (Good), 3 (Fair), 2 (Poor), 1 (Bad) International Telecommunication Union Telecommunications Standardization Sector (ITU- T) P.800 Toll quality means a MOS of 4.0 or better. How to do?

4 Speech Quality (1/3) Must be as good as PSTN Delay Coding/Decoding + Buffering Time + Tx. Time G.114: the round-trip time 300 ms Echo High Delay ==> Echo is Critical speak Speech Quality (2/3) Jitter Delay variation Due to different routes or queuing times Use of jitter buffer Jitter buffers add delay speak speak 120ms 120ms listen listen listen total delay: ms 13 round-trip time: ms constant delay variance =0 variable delay variance!= 0 14 Speech Quality (3/3) Packet Loss Traditional retransmission cannot meet the real-time requirements Call Set-up Time Address Translation Directory Access Percentage Packet Loss Impact of Packet Loss Packet Reconstruction Algorithms 35% 30% 25% 20% 15% 10% 5% 0% perfect excellent good acceptance annoying bad unstable

5 Impact of Latency Managing Access and Prioritizing Traffic milliseconds perfect excellent good acceptance annoying bad A single network for a wide range of applications Resource management: call is admitted if sufficient resources are available Prioritization: different types of traffic are handled in different ways If a network becomes heavily loaded, traffic should feel the effects before synchronous traffic (such as voice). QoS has required huge efforts Speech-coding Techniques In general, coding techniques are such that speech quality degrades as bandwidth reduces. The relationship is not linear. Speech coder Bit rate MOS G kbps 4.3 G kbps 4.0 G.723 (celp) 6.3kbps 3.8 G kbps 3.9 G kbps 4.0 GSM 13 kbps 3.7 Network Reliability and Scalability % reliability Today s VoIP solutions are ok. Redundancy and load sharing Reliability v.s. cost Scalable easy to start on a small scale and then expand as traffic demand increases Distributed architecture of IP

6 VoIP Implementations (1/2) VoIP Implementations (2/2) IP-based PBX solutions A single network Enhanced services IP voice mail One of the easiest applications IP call centers Use the caller ID Automatic call distribution (ACD) Load the customer s information on the agent s desktop Click to talk VoIP Evolution 23 6

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