A Study of the ITU-T G.729 Speech Coding Algorithm ...

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Open MASTER THESIS Figure 18: VoIP solution Datum - Date Rev Dokumentnr - Document no. 04-09-28 PA1 the end point’s willingness to enter into a conversation, SIP initiates a session which it later also terminates. 4.3.3 A Comparison Between H.323 and SIP H.323 and SIP are fierce competitors in the market for IP telephony signaling. Both, SIP and H.323, have been developed to address session control and signaling functions in a distributed call control environment. Like H.323, SIP works most efficiently with intelligent end-points. However, whereas in the case of SIP network intelligence and services are provided by servers, H.323 employs gatekeepers. Also, in contrast to H.323, which operates on a binary code basis, SIP is ASCII-based. 4.4 RTP and RTCP Figure 19: IP header Real-Time Transport Protocol (RTP) and Real-Time Control Protocol (RTCP) enable the transport of voice packets, following the voice signaling and encoding. RTP is an end-toend delivery device providing services for data with real-time attributes, examples of which are interactive audio and video. Payload type identification, sequence numbering, timestamping and delivery monitoring are some of the services provided. While the majority 42 (78)
Open MASTER THESIS Datum - Date Rev Dokumentnr - Document no. 04-09-28 PA1 of applications runs RTP on top of UDP to use the advantages of its multiplexing and checksum services, RTP can also be used with a number of other underlying network or transport protocols. Multicast distribution, provided by the respective underlying network, enables RTP to support the transfer of data to multiple destinations. Lower-layer service guarantees timely delivery and the quality of service. RTP does neither give a guarantee for delivery nor does it rely on the underlying network to deliver the packets in sequence. The sequence numbers included in RTP enable to determine the location of a packet, which can take place without the decoding of packets in sequence, as well as the reconstruction of the packet sequence of the sender. RTP has primarily been developed for multimedia conferences with multiple participants. Other applications of RTP include the storage of continuous data, interactive distributed simulation, active badge, and control and measurement. RTP contains two parts. While the Real-Time Transport Protocol (RTP) ensures the transport of data with real-time attributes, the RTP Control Protocol (RTCP) assumes responsibility for the monitoring of the service quality as well as for the provision of information about the session’s participants. The basis of the RTCP is hereby the periodic transmission of control packets to all participants, which requires the multiplexing of the data and control packets by the underlying protocol, for instance, by using separate port numbers with UDP. RTCP performs four functions: • RTCP provides feedback on the quality of the data distribution, which is closely related to the flow and congestion control functions of other transport protocols. • RTCP carries a persistent transport-layer identifier for an RTP source called the canonical name or CNAME. Due to the possibility that the SSRC identifier may change in case a conflict is discovered or in case of the restart of a program, receivers require the CNAME to keep track of each participant. CNAME may also be required by receivers to associate multiple data streams from a given participant in a set of related RTP sessions, such as in the case of an audio-video synchronization • Since the first two functions require that all participants send RTCP packets, there must be a control for the rate in order for RTP to scale up to a large number of participants. Through the process of having each participant send its control packets to all of the others, the number of participants can be determined. On the basis of this number, the rate at which the packets are sent, is calculated. • RTCP conveys minimal session control information, which is mostly used in sessions with no membership control or parameter negotiation. 4.5 Quality of Service The ability of networks to deliver good, predictable service over a variety of underlying technologies, such as IP-routed networks, is defined as Quality of Service (QoS). Due to the real-time attribute of voice applications the latter are not only extremely intolerant to delivery delays of packets, but also to jitter, echo, congestion, packet loss, and miss-ordered packets arrival. Methods to overcome the hostile environment of the IP net and to achieve an acceptable Quality of Service, are therefore essential. 43 (78)
Page 1 and 2: A Study of the ITU-T G.729 Speech C
Page 3 and 4: Open MASTER THESIS Uppgjord - Prepa
Page 11 and 12: 1 Introduction 1.1 Thesis Backgroun
Page 13 and 14: 1.4 List of Abbreviations Open MAST
Page 15 and 16: 2 Speech Coding Overview 2.1 Introd
Page 17 and 18: Open MASTER THESIS Table 2: English
Page 19 and 20: Open MASTER THESIS Datum - Date Rev
Page 21 and 22: 2.3 Speech Analysis Open MASTER THE
Page 27 and 28: 2.5.2 Voice Coders Open MASTER THES
Page 39 and 40: 4 Voice over IP Overview 4.1 Introd
Page 41: 4.3 Gateway Control Open MASTER THE
Page 49 and 50: 5.5 Instruction Set Open MASTER THE
Page 63 and 64: 8.8.5 Debug Method Open MASTER THES
Page 67 and 68: Table 9: continued Word Parmeter De
Page 71 and 72: 10 References Open MASTER THESIS Da
Page 73 and 74: 11 Appendix A Open MASTER THESIS Da
Page 75 and 76: Table 12: continued Open MASTER THE
Page 77 and 78: Table 13: continued Open MASTER THE

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