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Major Telecommunications Part 1: Voice Over IP

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Major Telecommunications Part 1: Voice Over IP Cairo University Faculty of Engineering Electronics and Electrical Communications Department Professional Masters Program – Major Telecommunications Part 1: Voice Over IP (VoIP) Dr. Wagdy Anis Aziz, Adjunct Doctor Senior Manager, Core Network Support, Mobinil [email protected] +201222201073 Mobile Networks Evolution Mobile Networks Evolution EVOLUTION OF TELECOMMUNICATION PLATFORMS TOWARD SMART COMMUNICATIONS Mobile Architecture Networks Evolution Mobile Networks Evolution – 3GPP R99 , R4 and R5 Mobile Networks Evolution – 3GPP R7 and R8 EPS – 3GPP Architecture Domains • From an operator’s perspective, in order to provide a data-only mobile broadband service, the infrastructure must be upgraded to EPS. • EPS provides components that allow existing 2G/ G access networks to utilize EPC components. • For those incumbent 2G/ G operators the existing CS network can provide access to voice calls in the short term, but the deployment of IMS in conjunction with EPS would provide an All-IP network with access to speech services. • Our focus here is how EPS is supporting voice services. • There are two fundamentally different ways that voice services can be realized for LTE users; using circuit-switched or IP Multimedia Subsystem (IMS) technologies. • Voice services based on circuit-switched technology Circuit-switched fallback CSFB • Voice services with IMS technology Single-Radio Voice Call Continuity SRVCC Moving to Full IP network Why IP Networks? 1. Cost reduction - There can be a real savings in long distance telephone costs which is extremely important to most companies, particularly those with international markets. 2. Simplification - An integrated voice/data network allows more standardization and reduces total equipment needs. 3. Consolidation -The ability to eliminate points of failure, consolidate accounting systems and combine operations is obviously more efficient. 4. Advanced Applications -The long run benefits of IP include support for multimedia and multileveled applications, something which today's telephone system can't compete with. Migration of all 2G/3G network interfaces to IP Interface Vendor Status Main Benefits Huawei Validated * important for the implementation of Iu-Flex feature IuCs which is needed for the completion of the MSC pool (RNC>>CS core) NSN Validated in lab, live trial in May project ALU trial planned in June Huawei Validated and deployed *Saving on transport side due to better utilization of IuPS pooled BW. NSN Validated and deployed (RNC>>PS core) *Saving on PaCo side due to better SGSN resources utilization. ALU Validated and deployed Huawei Validated and deployed Iur * Saving on TX side due to better utilization of pooled NSN Validated and deployed BW. G (RNC>>RNC) 3 ALU No Need Huawei Validated and deployed * Saving on TX side due to higher transmission efficiency Iub Dual Stack NSN Validated and deployed * Allows reaching higher throughput per site by using (nodeB>>RNC) the IP BW offered by MBH. ALU Validated Huawei trial planned in May * More savings on TX side Iub Full IP NSN trial planned in June * Voice & Data sharing the same transmission resources (nodeB>>RNC) pool allows for higher throughput during low voice traffic period. ALU trial planned in June Huawei live trial ongoing * Saving (CAPEX&OPEX) from BSS side due to removal of TC *Better voice quality after removing transcoding and A NSN trials planned in May enabling TrFO calls. (BSC>>CS core) *Important for the implementation of A-Flex feature which is needed for the completion of the MSC pool project. ALU trial planned after B12 upgrade G 2 Huawei live trial ongoing * Offer higher data throughput by better utilization of Gb the pooled resources of packet core and transport (BSC>>PS core) NSN trial planned in May networks ALU Validated Huawei No plans yet Abis * Saving on TX side due to statistical multiplexing (but (BTS>>BSC) NSN No plans yet needs introducing of new Synchronization methods) ALU No plans yet Voice Over Internet Protocol (VoIP) Topics • What is VoIP? • Why is VoIP Attractive? • How Does it Work? • Voice Coding • VoIP Signaling Standards • QoS Impairment Factors • QoS Measurement Methods • Design of VoIP Network Using SIP VoIP Main Factors . VoIP Codecs – G Series , AMR. vocoder (VOice enCODER),) . VoIP Signaling Protocols – H.323 , SIP , H.248 , Megaco , SIGTRAN, BICC . VoIP QoS Impairment Parameters – Delay , Packet Loss , Jitter. VoIP QoS Measurement Methods – Subjective Methods (MOS) and Objective Methods ( PESQ , E-Model) What is VoIP? • “VoIP” stands for Voice over Internet Protocol. • VoIP is a technique that encodes voice to the low rates and route the relatively low bandwidth signals as packetized "data", over dedicated transmission facilities or the "Internet" using the Internet Protocol. • VoIP is the ability to make telephone calls and send faxes over IP-based data networks with a suitable quality of service (QoS) and superior cost/benefit. Why is VoIP Attractive? 1. Cost reduction - As described, there can be a real savings in long distance telephone costs which is extremely important to most companies, particularly those with international markets. 2. Simplification - An integrated voice/data network allows more standardization and reduces total equipment needs. 3. Consolidation -The ability to eliminate points of failure, consolidate accounting systems and combine operations is obviously more efficient. 4. Advanced Applications -The long run benefits of VoIP include support for multimedia and multileveled applications, something which today's telephone system can't compete with. Introduction to VoIP Codecs • In VoIP applications, voice call is the mandatory service even when a video session is enabled. A VoIP tool (e.g., Skype,….) normally provides many voice codecs which can be selected or updated manually or automatically. • Typical voice codecs used in VoIP include ITU-T standards such as 64 kb/s G.711 PCM, 8 kb/s G.729 and 5.3/6.3 kb/s G.723.1; 3GPP standards such as AMR; opensource codecs such as iLBC and proprietary codecs such as Skype’s SILK codec which has variable bit rates in the range of 6 to 40 kb/s. • Some codecs can only operate at a fixed bit rate, whereas many advanced codecs can have variable bit rates which may be used for adaptive VoIP applications to improve voice quality. • Voice codecs or speech codecs are based on different speech compression techniques which aim to remove redundancy from the speech signal to achieve compression and to reduce transmission and storage costs. • In practice, speech compression codecs are normally compared with the 64 kb/s PCM codec which is regarded as the reference for all speech codecs. Speech codecs with the lowest data rates (e.g., 2.4 or 1.2 kb/s Vocoder) are used mainly in secure communications. • In general, the higher the speech bit rate, the higher the speech quality and the greater the bandwidth and storage requirements. • In practice, it is always a trade-off between bandwidth utilization and speech quality. Speech Compression and Coding Techniques • Speech compression aims to remove redundancy in speech representation to reduce transmission bandwidth and storage space (and further to reduce cost). There are in general three basic speech compression techniques, which are: 1- Waveform-Based Coding 2- Parametric-Based Coding 3- Hybrid Coding Techniques. Speech Compression and Coding Techniques (1) 1- Waveform -Based Coding • As the name implied, waveform based speech compression is mainly to remove redundancy in the speech waveform and to reconstruct the speech waveform at the decoder side as closely as possible to the original speech waveform. • Waveform-based speech compression techniques are simple and normally low in implementation complexity, whereas their compression ratios are also low. The typical bit rate range for waveform-based speech compression coding is from 64 kb/s to 16 kb/s. At bit rate lower than 16 kb/s, the quantization error for waveform-based speech compression coding is too high, and this results in lower speech quality. Typical waveform-based speech compression codecs are PCM and ADPCM (Adaptive Differential PCM) Speech Compression and Coding Techniques (1) 1- Waveform-Based Coding • Typical ones are Pulse Code Modulation (PCM) and Adaptive Differential PCM (ADPCM) • For PCM, it uses non-uniform quantization to have more fine quantization steps for small speech signal and coarse quantization steps for large speech signal (logarithmic compression). Statistics have shown that small speech signal has higher percentage in overall speech representations. Smaller quantization steps will have lower quantization error, thus better Signal-to-Noise Ratio (SNR) for PCM coding. • There are two PCM codecs, namely PCM μ-law which is standardized for use in North America and Japan, and PCM A-law for use in Europe and the rest of the world. ITU-T G.711 was standardized by ITU-T for PCM codecs in 1988. • For both PCM A-law and μ-law, each sample is coded using 8 bits (compressed from 16-bit linear PCM data per sample), this yields the PCM transmission rate of 64 kb/s when 8 kHz sample rate is applied (8000 samples/s × 8 bits/sample = 64 kb/s). 64 kb/s PCM is normally used as a reference point for all other speech compression codecs. • ADPCM, proposed by Jayant in 1974 at Bell Labs, was developed to further compress PCM codec based on correlation between adjacent speech samples. Consisting of adaptive quantiser and adaptive predictor, a block diagram for ADPCM encoder and decoder (codec). Speech Compression and Coding Techniques (1) 1- Waveform-Based Coding • If an ADPCM sample is coded into 4 bits, the produced ADPCM bit rate is 4 × 8 = 32 kb/s. This means that one PCM channel (at 64 kb/s) can transmit two ADPCM channels at 32 kb/s each. If an ADPCM sample is coded into 2 bits, then ADPCM bit rate is 2 × 8 = 16 kb/s. One PCM channel can transmit four ADPCM at 16 kb/s each. ITU-T G.726 defines ADPCM bit rate at 40, 32, 24 and 16 kb/s which corresponds to 5, 4, 3, 2 bits of coding for each ADPCM sample.
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