Download miễn phí Đề tài Developping Service VoIP in Viet Nam



Index
Glossary
Generel of the thesis
Chapter I: Voice over Internet Protocol (VoIP) Technology
1. Fundamental of channel switching network and Internet
Fundamental features of channel switching network
Fundamental features of Internet
Advantages of VoIP against PSTN
Outlook of VoIP technology
+ Some technical features of IP telephone
- Terminal equipment and gateway
- Tranmission equipment
+ Special feature of VoIP
- Adjustable quality
- Security
- User interface
- Connecting telephone and computer
1.5. Conclusion
2. Problems relating to VoIP technology and talk quality on VoIP
Coding techniques and talk signal compression
Voice Activity Detector (VAD)
Number and address
+ Numbering on SCN network
+ Numbering on IP
Fee
Signal cooperation
Confidence
Troubles relating to calls quality
+ Delay
+ Echo suppression
+ Jitter changeable delay
+ Package loss
+ Bandwidth
3. Transfer modes
3.1 Real Time Mode
Real Time Post
Real Time Control Mode
RSVP
Conlusion
4. Introduction of standards
4.1 Introduction of standards
4.2 Standard H323
4.2.1. Introduction in H323
4.2.2. H323 Elements
+ Main functions of gateway
4.2.3. H323 Structure
4.2.4. Signal and control system in H323
4.2.5 Establishing the call in H323
5. The Session Initiation Protocol (SIP)
The SIP Network Architecture
SIP Call Establishment
Information in SIP Messages
The Resource Reservation Protocol (RRP)
Chapter II: Voice Communication
2.1 . Grabbing and reconstruction
2.1.1. Sampling and quantisation
2.1.2. Reconstruction
2.1.3 Mixing audio siganals
2.2. Communication requirements
2.2.1. Error tolerance
2.2.2. Delay requirements
2.2.3. Tolerance for jitter
2.3. Communication patterns
2.4. Impact on VoIP
2.4.1. Sampling rate and quantisation
2.4.2. Packet length
2.4.3. Buffering
2.4.4. Delay
2.5.5. Silence suppression
2.5. Summary
Chapter III: Voice Communication
1. Quick Concept
1.1 How traditional long distance works
1.2 How long distance works with VoIP
2 Overview
Chapter IV. Compression Techniques
4.1.Preliminaries
4.2.General compression techniques
4.2.1. Lempel-Ziv compression
4.2.2 .Huffman coding
4.3. Waveform coding
4.3.1. Differential coding
4.3.1.1 Differential PCM (DPCM)
4.3.1.2 Adaptive DPCM (ADPCM)
4.3.1.3 Delta modulation (DM)
4.3.2 Vector quantisation
4.3.3 Transform coding
4.4 Vocoding
4.4.1 Speech production
4.4.2 Vocoding basics
4.4.3 Linear Predictive Coding (LPC)
4.5 Hybrid coding
4.5.1 Residual Excited Linear Prediction (RELP)
4.5.2 Codebook Exciter Linear Prediction (CELP)
4.5.3 Multipulse and Regular Pulse Excited coding (MPE and RPE)
4.6 Other compression techniques
4.7 Dalay by compression
4.8 Voice compression standards
4.9 Summary
 
 
 
 


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est in both the ARQ and the ACF is the call Model parameter, which is optional in the ARQ and mandatory in the ACF. In the ARQ call Model indicates whether the endpoint wants to send call signaling directly to the other party, or prefers that call signaling be passed via in the gatekeeper.In the ACF, it represents the gatekeeper’s decision as to whether call signaling is to pass via the gatekeeper or directly between the terminals. In the example of figures 11, the calling gatekeeper has choosen not to be in the path of tha call signaling.
The Setup message is the first call signaling message sent from one-terminal to the other to establish the call. The message must contain the Q.931 Protocol Discriminator, a Call Reference Setup, a Bearer Capability , and the User-User information element. Although the Bearer Capability information element is mandatory, the concept of a bearer, as used in the circuit switched world , does not map very well to an IP network. For example, no B-channel exists in IP and the actual agreement between endpoints regarding the bandwidth requirements is done as part of H.245 signaling, where RTP information such as the payload type is exchanged. Consequently, many of the fields in the Bearer Capability information element, as defined in Q.931, are not, used in H.225.0. Of those fiejds that are used in H.225.0, many are used only when the call has originated from outside the H.323 network and has been received at a gateway, where the gateway performs a mapping from the signaling received to the appropriate H.225.0 messages.
A nember of parameters are include within the mandatory. User-to-User information element. Those include the call identifier, the call type, a conference identifier, and information about the originating endpoint. Among the optional parameters, we may find a source alias, a destination alias, an H.225.0 address. The User-to-User information element is included in all H.225.0 call signaling messages. It is the inclusion of this information element that enables Q.931 messages, originallydesigned for ISDN, to be adapted for use with H.323.
The Call Proceeding message may optionally be sent by the recipient of a Seup message to indicate that the Setup message has been received and that call establishment procedures are underway. When sent, it ususlly precedes the Alerting message, which indicates that the called device is “ringing” Strietly speaking, the Alerting message is optional .
In addition to Call Proceeding and Alert, we may also find the optional Progress message(not shown). Ultimately, when the called party answersthe called terminal returns a connect message. Although some of the message from the called party to the calling party, such as Call Proceeding and Alerting, are optional, the connect message must be sent if the call is to be completed. The User-to -User information elementcontains the same set of parameters as defined for the Call Proceeding, Progress, and Alert message , with the addition of the Conference Identifier. These parameters are also used in a Setup message and their use in the Connect message is to correlate this conference with that indicated in a Setup. Any H.245 address sent in a Connect message should match that sent in any earlier. Call Proceeding, Alerting, or Progress message. In fact, the called terminal must include at least an H.245 signaling address to which H.245 message must be sent because H.245 message are used to establish the media (that is voice) flow between the parties.
In the example of figure 11, H.245 message exchange begins after the Connect message is returned. This message exchange could. In fact, occur earlier than the Connect message. It is important to note that H.245 is not responsible for carrying the actual media. For example, there is no such thing as an H.245 packet containing asample of coded voice. That is the fob of RTP. Instead, H.245 is a control protocol that message the establishment and release of media sessions. H.245 does this through messaging that enables the establiment of logical channels, where a logical channel is a unidirectional RTP stream from one party to the other.
A logical channel is opened by sending an Open Logical Channel (OLC) request message. This message contains a mandatory parameter called forward Logical Channel Parameters, which relates to the media to be sent in the forward drection, that is, from the endpointissuing this command. It contains information such as the type of data to be sent, an RTP session ID, an RTP payload type, and an indication as to whether silince suppression is to be used. If the recipient of the message wants to accept the media to be sent, then it will return an Open Logical Channel Ack message containing the same logical channel number as received in the request and a transport address to which the media stream should be sent.
Strictly speaking, a logical channel is unidirectional. Therefore, in order to establish a two-way conversation, two logical channel must be opened-one in each direction. According to the description just presented, this requires four messages, which is rather cumbersome. Consequently, H323 defines a bidirectional logical channel. This is means of establishing two logical channel, one in each direction, in a slightly more efficient manner. Basically, a bidirectional logical channel really means two logical channels that are associated with each other. The establishment of these two channels can be achieved with just three H.245 message rather than four. In order to do so, the initial OLC message not only contains information regarding the media that the calling endpoint wants to send, but it also contains reverse logical channel parameters . These indicate the type of media that the endpoint is willing to receive and to where that media should be sent.
Upon receipt of the request, the far endpoint may send an Opne Logical Channel Ack message containing the same logical channel number for the forward logical chanel, a logical channel number for the reverse logical channel, and descriptions related to the media formats that it iswilling to send. These media formats should be chosen from the options originallyreceived in the request, thereby ensuring that the called and will only send media that the calling end supports.
Upon receipt of the Open Logical Channel Ack, the originating endppoint responds with an Open Logical Channel Confirm message to indicate that all is well.RTP stream and RTCP message can now flow in each direction
5. The Session Initiation Protocol (SIP)
The Session Initiation protocol (SIP) is considered by many to be a powerful alternative to H.323. It is considered to be a more flexible solution, simpler than H.323, easier to implement, better suited to the support of intelligent user devices, and better suited to the implementation of advanced features. Although H.323 may still have a larger installed base than SIP, most people in the VoIP community believe that the future of VoIP revolves around SIP. In fact 3GPP has endorsed SIP as the session management protocol of choice for 3GPP. Release 5 albeit with some enhancements.
Like H.323, SIP is simply a signaling protocol and does not earry the voicce packets itself. Rather, it makes use of the services of RTP for the transport of the voice packets (the media stream).
5.1 The SIP Network Architecture
SIP defines two basic classes of network entities- clients and servers. Stricetly speaking, a client, also known as a user agent client, is an application program that sends SIP requests. A server is an entity that responds to those requests. Thus, SIP is a client-server protocol. VoIP calls using SIP are originated by a clien t and terminaled at a servers. A client may be found within a user’s device, which could be, for example, a SIP phone. Clients may also be found within the same platfoem as a server. For example, SIP enables the use of proxies, which act as both clients and servers.
Four different types of servers are available- proxy servers, redirect servers, user agent srevers, and registrars. Proxy server acts similarly to a proxy server used for Web access from a corporate local area network (LAN). Clients send requests to the proxy, which either handles those requests itself or forwards them on to ether servers, perhaps after performing some translation. To those other servers, it appears as though the message is coming from the proxy rather than some entity hiden behind it. Given that a proxy both receives requests and sends requests, it incorporates both server and client functionality. Figure 12 shows an example of the operation of a proxy servers . It does not take much imagination to realize how this type of functionality can be used for call forwarding/ follow-me services.
A redirect server is a srevers that accepts SIP req...

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