1, the Education MAN VOIP Application
Education system for all telephone contacts are using China Telecom's telephone network, in its establishment at all levels have no other phone support system. The high cost of enterprise communications business who is troubled by many problems, but it seems that the cost can not be compressed, the internal high service costs if eager to be able to resolve through existing technology. The VoIP technology has matured to address this problem plagued the industry for many years. By the Education Commission and the VoIP network between schools could save a lot communication costs, support a variety of ways and almost do not change the call the call habits VoIP network users to provide users with a complete solution.
So how can the education system to reduce these costs, enhance competitive? In addition to management to find ways to save costs, VoIP technology was a business from a technical level, provide a possibility to reduce communication costs.
The so-called VoIP (Voice over Internet Protocol) is a IP-phone-based, and introduce the corresponding value-added services technology. It is through voice IP packets sent to achieve business is built on packet-based IP technology, digital transmission technology, the basic principle is: The voice compression algorithm to compress encrypted voice, and then the voice data by IP and related protocols to package, through IP network to transmit data packets to the destination, then the voice packets are linked up through the decoding decompression treatment, restored to the original speech signal, so as to achieve the purpose of IP networks to transmit voice.
Driven the rapid development of Internet voice service factor is it? The main factor is price. The same talk time, voice services through the Internet to spend the cost of traditional voice services only through a fraction of the cost of spending, or even a few one-tenth of the price difference so poor, Internet telephony also can imagine the speed of development the. Another factor is technology. In the past, Internet phone and give us the impression that the voice of poor quality, long delays, often broken. But with technology, these phenomena would be lost. On the one hand, a series of agreements with the launch of ITU H.323, making gradual improvement of Internet phone standard, the manufacturers of the equipment to communicate with each; on the other hand, Internet network and hardware devices to build the strong increase in performance as the smooth Internet phone provided a guarantee; Finally, the voice compression technology for Internet phone paved the way for the development.
Now experts believe that the VoIP phone technology will represent the future direction, and even such a word: everything is over IP (everything will be based on IP). Many people believe that the coming era of IP phones, according to the company of London Phillips, IP phones worldwide in 2003 accounted for 43.12% overall telephone business, while research firm Frost & Sullivan report estimated that by 2007, VoIP call capacity will account for 75% of total call volume.
Second, VoIP voice system, the advantages and disadvantages
VoIP's biggest advantage is the widespread use of Internet and the global IP interconnect environment, providing more than the traditional business, better service, the following is the IP Phone features:
To support flexible business approach
Support the users pay by card. General account and the fixed-caller, etc. for international and domestic long-distance business, the full support of telephone billing system - Telephone, phone-PC and smart card numbers and other business operations, as well as instructions and busy such as call transfer, and other new value-added services. And make full use of advantages of Internet technology to ensure call quality.
Excellent cost performance
IP phone of the biggest advantages is the price advantage, which is the fundamental reason for the rise of IP phones. System transmission costs include direct phone locally, IP network utilization and remote city of the bill, with the traditional long-distance telecommunications network communications costs compared to expensive, can greatly reduce the user communication costs.
Compatible with the existing telephone network
IP Phone Gateway support the "phone to phone" mode, with the traditional phone connected to an ordinary telephone line, fully capable of traditional telephone features, easy to use, and automatic identification and (PSTN and Internet users) network between conversion, sharing of profits, billing, voice mail and a series of professional services.
Improve the transmission efficiency
Telephone gateway adopts international standard algorithms (G.723.1, G.729a) compressed voice signal for low rate speech coding, such as G.723.1 can be conventional 64 kb / s voice signal compression to 5.3kb / s, while still maintaining good sound effects. When people call about 60% of quiet time, use of silence compression technology transfer calls only during the actual talk-time data, but will double the bandwidth of the release of the silent period to the other voice services, to improve bandwidth utilization, savings in communications resources.
Full use of resources
IP communications network using Internet by the packet switching and connectionless packet switching with integrated, therefore, IP protocol used for communications resources, is much higher than traditional communications networks.
Third, VOIP Technology Overview
Widespread use of VoIP and VoIP network technology is inseparable from development. VoIP uses the IP network based on statistical time division multiplexing-based voice services for transmission of data using advanced signal processing technology for voice coding, voice compression, silent monitoring, comfort noise generation and other technologies can provide the traditional circuit-switched mode network voice quality comparable to PSTN voice services. Technology development today, the current implementation of two main VoIP standards: ITU-T standards and IETF standards.
ITU-T standards: H.323 protocol
H.323 is the first VoIP standard to be adopted. H.323 protocol system provides a network based on points for two / multi-point real
When the media communication system logic components, message definitions and the communication process. H.323 protocol design ideas using ISDN, use the Q.931 protocol to complete the call establishment and release, significant areas are telecommunications networks and centralized manageability features. Currently, H.323 protocol has been widely used in the Internet, such as ITXC and international AT & T, China such as China Telecom, China Unicom all adopts the architecture of H.323 IP phone operator network formation. Compared with the SIP, H.323 is more mature.
1, the Education MAN VOIP Application
Telephone contact with the education system is to use all the phone networks of China Telecom, in its establishment at all levels have no other phone support system. The high cost of enterprise communications business who is troubled by many problems, but it seems that the cost can not be compressed, the internal high service costs if eager to be able to resolve through existing technology. The VoIP technology has matured to address this problem plagued the industry for many years. By the Education Commission and the VoIP network between schools could save a lot communication costs, support a variety of ways and almost do not change the call the call habits VoIP network users to provide users with a complete solution.
So how can the education system to reduce these costs, enhance competitive? In addition to management to find ways to save costs, VoIP technology was a business from a technical level, provide a possibility to reduce communication costs.
The so-called VoIP (Voice over Internet Protocol) is a IP-phone-based, and introduce the corresponding value-added services technology. It is through voice IP packets sent to achieve business is built on packet-based IP technology, digital transmission technology, the basic principle is: The voice compression algorithm to compress encrypted voice, and then the voice data by IP and related protocols to package, through IP network to transmit data packets to the destination, then the voice packets are linked up through the decoding decompression treatment, restored to the original speech signal, so as to achieve the purpose of IP networks to transmit voice.
Driven the rapid development of Internet voice service factor is it? The main factor is price. The same talk time, voice services through the Internet to spend the cost of traditional voice services only through a fraction of the cost of spending, or even a few one-tenth of the price difference so poor, Internet telephony also imagine the speed of development the. Another factor is technology. In the past, Internet phone and give us the impression that the voice of poor quality, long delays, often broken. But with technology, these phenomena will cease to exist. On the one hand, a series of agreements with the launch of ITU H.323, making gradual improvement of Internet phone standard, the manufacturers of the equipment to communicate with each; on the other hand, Internet network and hardware devices to build the strong increase in performance as the smooth Internet phone provided a guarantee; Finally, the voice compression technology for the development of Internet phone paved the way for.
Now experts believe that the VoIP phone technology will represent the future direction, and even such a word: everything is over IP (everything will be based on IP). Many people believe that the coming era of IP phones, according to the company of London Phillips, IP phones worldwide in 2003 accounted for 43.12% overall telephone business, while research firm Frost & Sullivan report estimated that by 2007, VoIP call capacity will account for 75% of total call volume.
Second, VoIP voice system, the advantages and disadvantages
VoIP's biggest advantage is the widespread use of Internet and the global IP interconnect environment, providing more than the traditional business, better service, the following is the IP Phone features:
To support flexible business approach
Support the users pay by card. General account and the fixed-caller, etc. for international and domestic long-distance business, the full support of telephone billing system - Telephone, phone-PC and smart card numbers and other business operations, as well as instructions and busy such as call transfer, and other new value-added services. And make full use of advantages of Internet technology to ensure call quality.
Excellent cost performance
IP phone of the biggest advantages is the price superiority, which is the rise of IP phones root cause. System, direct phone transmission cost, including local, IP network utilization and remote city of the bill, with the traditional long-distance telecommunications network communications costs compared to expensive, can greatly reduce the user communication costs.
Compatible with the existing telephone network
IP Phone Gateway support the "phone to phone" mode, with the traditional phone connected to an ordinary telephone line, fully capable of traditional telephone features, easy to use, and automatic identification and (PSTN and Internet users) network between conversion, sharing of profits, billing, voice mail and a series of professional services.
Improve the transmission efficiency
Telephone gateway adopts international standard algorithms (G.723.1, G.729a) compressed voice signal for low rate speech coding, such as G.723.1 can be conventional 64 kb / s voice signal compression to 5.3kb / s, while still maintaining good sound effects. When people call about 60% of quiet time, use of silence compression technology transfer calls only during the actual talk-time data, but will double the bandwidth of the release of the silent period to the other voice services, to improve bandwidth utilization, savings in communications resources.
Full use of resources
IP communications network using Internet by the packet switching and connectionless packet switching with integrated, therefore, IP protocol used for communications resources, is much higher than traditional communications networks.
Third, VOIP Technology Overview
Widespread use of VoIP and VoIP network technology is inseparable from development. VoIP uses the IP network based on statistical time division multiplexing-based voice services for transmission of data using advanced signal processing technology for voice coding, voice compression, silent monitoring, comfort noise generation and other technologies can provide the traditional circuit-switched mode network voice quality comparable to PSTN voice services. Technology development today, the current implementation of two main VoIP standards: ITU-T standards and IETF standards.
ITU-T standards: H.323 protocol
H.323 is the first VoIP standard to be adopted. H.323 protocol system provides a network based on points for two / multi-point real
When the media communication system logic components, message definitions and the communication process. H.323 protocol design ideas using ISDN, use the Q.931 protocol to complete the call establishment and release, significant areas are telecommunications networks and centralized manageability features. Currently, H.323 protocol has been widely used in the Internet, such as ITXC and international AT & T, China such as China Telecom, China Unicom all adopts the architecture of H.323 IP phone operator network formation. Compared with the SIP, H.323 is more mature.
H.323 protocol characteristics
Based on H.323 gatekeeper concept for centralized control network is to facilitate unity of maintenance management;
But this is a clear disadvantage is that a large call processing delay;
Network size and therefore severely limited;
IETF standards: SIP
The agreement system by Level3, Bellcore and Cisico initiated service providers / vendors proposed by the IETF community
(Internet Engineering Task Force: Internet Engineering Task Force) developed the formation. SIP protocol is simple, good flexibility and scalability, as well as Internet applications and close the existing features, many people believe that the agreement easier to implement, recently, in particular the rapid development in the United States, while SIP will be the third generation mobile communication core network and intelligent services to be widely applied. Currently, SIP protocol is still in the early stages of development, many of the appropriate standard is not completely uniform, perfect management, large-scale Web applications are not.
It should be said, H.323 and SIP protocol IP phones are for building a network of evolving, both in the early stages of development are mutually exclusive, but with the SIP protocol development, particularly in the application of soft exchange made great progress has been SIP protocol, the current H.323 and SIP interconnection agreement has also been in preparation.
In addition, whether the agreement is based on H.323 or SIP-based IP telephone network structure, completion of all necessary agreements related to the media gateway gatekeeper load control, resource control and management. At present, this interface uses Media Gateway Control Protocol (MGCP: Media Gateway Control Protocol) or H.248 (MeGaCo: Media Gateway Control) protocol.
H.248 protocol called the Media Gateway Control Protocol, is 16 by the ITU-T group put forward the application in the media gateway and media gateway controllers, media gateway controller and H.248/MeGaCo terminal between.
MGCP is defined in the earlier IETF Media Gateway Control Protocol, the main function of the angle from the definition of media gateway controllers and media gateways between the behavior of relatively simple to achieve, not H.248 packets and attributes, as detailed on the definition of events are relatively simple mechanism for interaction. MGCP with simple characteristics, but the interoperability and limit the ability to support the business.
H.248/MeGaCo its functional flexibility and ability to support the business strong and valued, but there are always new accessories to complement its capabilities, is the mainstream media gateway protocol, which addresses the shortcomings of the original MGCP, and feasible, However, the agreement released soon, the industry still lack mature products and application examples, need further development.
H.248 and MeGaCo the same in the text of the agreement, but the syntax in the protocol message transmission differ, H.248 uses ASN.1 syntax format (ITU-T X.680 1997), MeGaCo using ABNF grammar format (RFC2234).
VoIP's key technologies include:
Media Coding
Including the popular G.723.1, G.729, G.729A voice compression algorithms and MPEG-II multimedia compression technology.
Control signaling
Including ITU-T H.323 and IETF Session Initiation Protocol SIP [4] (Session Initation Protocol) two sets of standards, is also involved in real-time synchronization control continuous media streaming real-time streaming protocol TRSP.
Packet Transmission
Real-time Transport Protocol mainly RTP.
Business Quality Assurance Technology
Use of resource reservation protocol RSVP, Quality of Service Qos and for operational quality control of real-time transmission control protocol RTCP come to avoid network congestion, protect the call quality.
Network transmission technology
Mainly TCP and UDP.
In addition to the groups involved in reconstruction and delay jitter smoothing, dynamic routing balancing technology, and gateway interconnect technology (including media sharing and control signaling exchange), network management (SNMP), secure authentication and billing technology, IVR interactive voice response technologies.
VOIP technology in our key emphasis is to resolve the technical aspects of voice quality, voice quality as the basis for the realization of VOIP related to the success of VOIP network construction, believed that an intermittent voice calls that no one could bear. VoIP voice quality affecting the main factors: delay and delay jitter, voice coding, packet loss, echo, voice call equality.
Delay and delay jitter
End to end delay, including encoding and decoding delay, packing and unpacking time delay, and network transmission delay. Delay changes, real-time delay variation (jitter) caused mainly by the network, if the end of the transmission path through the intermediate nodes (routers, switches, etc.) the more the delay jitter caused by the greater.
Speech Coding
Speech coding technology in the efficient use of bandwidth while providing high-quality voice. Different coding techniques will bring a different voice quality, the following table lists several coding MOS (mean opinion score) values (test results). Reference in the following table results, G.729 and G.711 encoding scheme can meet the educational system of two IP telephone quality requirements.
Coding
Bit rate (K bps)
MOS value
Coding delay (ms)
G.711
64
4.4
0.75G.723.1 (5.3K)
5.3
3.6
30G.723.1 (6.3K)
6.3
3.4
30G.729
8
4.0
10MS GSM
13
3.1
20
Coding techniques and performance indicators
On the other hand, G.729 bandwidth requirements are far lower than G.711, line quality and bandwidth in the same circumstances, G.711 packet transmission time-line on the road longer than G.729, and this time is more than voice compression into the data filing. So from the perspective of hearing, use of G, 729 and G.711 compression algorithm is basically the same effect.
Taken into account, using G.729, both to meet the requirements of voice effects, but also saves bandwidth, saves the investment cost line, is the preferred encoding compression algorithm.
Packet loss rate
In the IP network, IP packets are lost there factors: the network transmission packet loss, network congestion and packet loss when the active gateway device. When the packet loss rate of over 10% will seriously affect the voice quality.
Echo
Echo due to impedance mismatch at both ends of the call caused. End to end delay in a large IP networks, especially the impact of echo interference.
Voice level
The right to send and receive voice level is to affect the call quality is another important factor, the voice gateway must have a voice level adjustment function.
Can be seen from the above analysis, to improve voice quality IP telephony should mainly twofold. First, appropriate choice of speech coding technology, use high-quality voice gateways and appropriate gatekeeper control; second IP network is to improve the quality of service (QoS), the network delay, delay jitter and packet loss rate control within certain limits.
The following analysis of the gateway, gatekeeper over the above factors made for voice quality assurance.
Gateway Voice Quality assurance measures
Delay and jitter are critical factors affecting voice quality, the gateway through the following techniques can be carried out in these two factors, better control:
Mute Suppression
Silence Suppression, also known as VAD (Voice Activity Detection), which is based on everyday conversation and silence the voice features, to detect silent (Silence) to contain the time, so that it does not occupy or rarely occupied channel bandwidth, test when unexpected events to be compressed speech coding and transmission. The results show that the use of VAD technology enables the effective utilization of channel bandwidth increases about twice.
Jitter Buffer Technology
Jitter buffer for the receiving end, the purpose is to smooth the delay jitter, and both decoding and compression operations.
Echo Cancellation
In the IP phone devices usually need to take measures to eliminate echo, usually in the gateway device to achieve the industry's G.165 echo suppression.
Gatekeeper to ensure voice quality measures
Gatekeeper has a bandwidth management function, use this function can be set on certain routes on the voice bandwidth to control the IP telephone call process to ensure call quality. For example:
On certain routes bandwidth 1M bps, in order to ensure the video, the bandwidth data services, we set the gatekeeper on the bandwidth of voice service 60K bps, according to each telephone channels calculated bandwidth 12K bps, 5 way when the following voice to ensure call quality, but if more than 5-way call quality can not be guaranteed.
Use Gatekeeper bandwidth management features, configure the gatekeeper call when the line peak bandwidth, and monitor each way call. When the Sixth Road a call, the gatekeeper to reject the connection based on configuration, to ensure that the other 5-way call quality, but also ensure video and data services to carry out. IP telephone network in the educational system design, we can talk in different situations to determine the number of road-line peak, through the gatekeeper to ensure voice quality.
IP QoS to ensure voice quality
To ensure voice quality, the network must have a good QoS assurance mechanism. Diff-Serv can be used for QoS guarantees or RSVP:
IP network support required Diff-Serv, RSVP mode QoS guarantee mechanisms;
Voice gateway supports packet classification on the allocation of high priority voice packets;
Network Equipment Support CAR (Committed Access Rate) function, to achieve a measure of packet and flow monitoring;
Network device support WFQ function, high-priority packets to priority.
4, VOIP design principles
According to the characteristics of VoIP applications on their network design into consideration the following factors:
QoS guarantee
Voice applications on the delay, jitter, distortion, etc. are very sensitive, because VoIP is based on the statistical time division multiplexing of the IP network as the basis for voice service delivery, thus requiring the full QoS guarantee, in the specific implementation method, you can use a variety of technical means such as RSVP bandwidth reservation, using a variety of queue algorithms such as CBWFQ, PQ, CQ, etc. to ensure VoIP voice data have higher priority, ToS parameter settings and so on.
Efficient speech coding and decoding technology
Ordinary analog voice through sampling, coding, usually 64K bandwidth, which is G.711 coding technology to achieve high efficiency of voice transmission, voice quality must not affect the premise of the voice data compression, based on the G.711 , has the following development of coding techniques: G.726/G.727 (16,24,32 Kbps), G.728 (16Kbps), G.729, G.729A, G.729B, G.729AB (8Kbps), G. 723.1 (5.3/6.4Kbps), etc., and efficient compression technology makes the voice less and less occupied bandwidth, and voice quality control can be received within, and promote the VoIP technology.
Smart Choice voice compression technology
Between different VoIP devices can auto-negotiate before establishing the call using voice compression technology to support dynamic selection of both standard voice compression codec to be able to establish call connection, to achieve VoIP communications.
Voice calls with local connections
VoIP IP voice network enables communication within the network to achieve with the local telecommunications network connection (that is commonly referred to as ground) can enhance the value of VoIP networks, to achieve a more rich call applications, provides users with more convenient services .
Call VoIP network manageability
VoIP applications for the different ways to provide flexible and convenient means of control, and configuration process as simple and easy to understand, and call for the landing approach, the need for certification and reminder. For the different needs of users, can support a variety of flexible connectivity options, the process of achieving a different call.
Network reliability
Network single point of failure will not lead to loss of local networks and the network connection, multi-point failure will not cause the whole network is divided into several disconnected parts.
Standardized and easy network scalability
Network structure, technology and product standardization, the structure of trade expansion, technology and products can be continuous, satisfying the user based on the current application, the user fully guaranteed investment.
The modular structure of the network
VoIP network, the physical structure, logical structure and the address space of modular, similar to and the current PSTN network to provide users with a good user interface, easy to network, build, maintain and extend.
Easy network management and maintenance
The whole network can be unified or distribution management, network maintenance is simple and effective;
Network availability
According to the present needs and anticipated growth in demand can be designed VoIP network, not the pursuit of the technical nature empty, and the latest technology to avoid the high cost of pursuing a great price. Full use of existing resources and tap the potential of existing equipment to achieve with the VoIP network.
1, the Education MAN VOIP Application
Education system for all telephone contacts are using China Telecom's telephone network, in its establishment at all levels have no other phone support system. The high cost of enterprise communications business who is troubled by many problems, but it seems that the cost can not be compressed, the internal high service costs if eager to be able to resolve through existing technology. The VoIP technology has matured to address this problem plagued the industry for many years. By the Education Commission and the VoIP network between schools could save a lot communication costs, support a variety of ways and almost do not change the call the call habits VoIP network users to provide users with a complete solution.
So how can the education system to reduce these costs, enhance competitive? In addition to management to find ways to save costs, VoIP technology was a business from a technical level, provide a possibility to reduce communication costs.
The so-called VoIP (Voice over Internet Protocol) is a IP-phone-based, and introduce the corresponding value-added services technology. It is through voice IP packets sent to achieve business is built on packet-based IP technology, digital transmission technology, the basic principle is: The voice compression algorithm to compress encrypted voice, and then the voice data by IP and related protocols to package, through IP network to transmit data packets to the destination, then the voice packets are linked up through the decoding decompression treatment, restored to the original speech signal, so as to achieve the purpose of IP networks to transmit voice.
Driven the rapid development of Internet voice service factor is it? The main factor is price. The same talk time, voice services through the Internet to spend the cost of traditional voice services only through a fraction of the cost of spending, or even a few one-tenth of the price difference so poor, Internet telephony also can imagine the speed of development the. Another factor was technology. In the past, Internet phone and give us the impression that the voice of poor quality, long delays, often broken. But with technology, these phenomena will cease to exist. On the one hand, a series of agreements with the launch of ITU H.323, making gradual improvement of Internet phone standard, the manufacturers of the equipment to communicate with each; on the other hand, Internet network and hardware devices to build the strong increase in performance as the smooth Internet phone provided a guarantee; Finally, the voice compression technology for the development of Internet phone paved the way for.
Now experts believe that the VoIP phone technology will represent the future direction, and even such a word: everything is over IP (everything will be based on IP). Many people believe that the coming era of IP phones, according to the company of London Phillips, IP phones worldwide in 2003 accounted for 43.12% overall telephone business, while research firm Frost & Sullivan report estimated that by 2007, VoIP call capacity will account for 75% of total call volume.
Second, VoIP voice system, the advantages and disadvantages
VoIP's biggest advantage is the widespread use of Internet and the global IP interconnect environment, providing more than the traditional business, better service, the following is the IP Phone features:
To support flexible business approach
Support the users pay by card. General account and the fixed-caller, etc. for international and domestic long-distance business, the full support of telephone billing system - Telephone, phone-PC and smart card numbers and other business operations, as well as instructions and busy such as call transfer, and other new value-added services. And make full use of advantages of Internet technology to ensure call quality.
Excellent cost performance
IP phone of the biggest advantages is the price advantage, which is the fundamental reason for the rise of IP phones. System, direct phone transmission cost, including local, IP network utilization and remote city of the bill, with the traditional long-distance telecommunications network communications costs compared to expensive, can greatly reduce the user communication costs.
Compatible with the existing telephone network
IP Phone Gateway support the "phone to phone" mode, with the traditional phone connected to an ordinary telephone line, fully capable of traditional telephone features, easy to use, and automatic identification and (PSTN and Internet users) network between conversion, sharing of profits, billing, voice mail and a series of professional services.
Improve the transmission efficiency
Telephone gateway adopts international standard algorithms (G.723.1, G.729a) compressed voice signal for low rate speech coding, such as G.723.1 can be conventional 64 kb / s voice signal compression to 5.3kb / s, while still maintaining good sound effects. When people call about 60% of quiet time, use of silence compression technology transfer calls only during the actual talk-time data, but will double the bandwidth of the release of the silent period to the other voice services, to improve bandwidth utilization, savings in communications resources.
Full use of resources
IP communications network using Internet by the packet switching and connectionless packet switching with integrated, therefore, IP protocol used for communications resources, is much higher than traditional communications networks.
Third, VOIP Technology Overview
Widespread use of VoIP and VoIP network technology is inseparable from development. VoIP uses the IP network based on statistical time division multiplexing-based voice services for transmission of data using advanced signal processing technology for voice coding, voice compression, silent monitoring, comfort noise generation and other technologies can provide the traditional circuit-switched mode network voice quality comparable to PSTN voice services. Technology development today, the current implementation of two main VoIP standards: ITU-T standards and IETF standards.
ITU-T standards: H.323 protocol
H.323 is the first VoIP standard to be adopted. H.323 protocol system provides a network based on points for two / multi-point real
When the media communication system logic components, message definitions and the communication process. H.323 protocol design ideas using ISDN, use the Q.931 protocol to complete the call establishment and release, significant areas are telecommunications networks and centralized manageability features. Currently, H.323 protocol has been widely used in the Internet, such as ITXC and international AT & T, China such as China Telecom, China Unicom all adopts the architecture of H.323 IP phone operator network formation. Compared with the SIP, H.323 is more mature.
H.323 protocol characteristics
Based on H.323 gatekeeper concept for centralized control network is to facilitate unity of maintenance management;
But this is a clear disadvantage is that a large call processing delay;
Network size and therefore severely limited;
IETF standards: SIP
The agreement system by Level3, Bellcore and Cisico initiated service providers / vendors proposed by the IETF community
(Internet Engineering Task Force: Internet Engineering Task Force) developed the formation. SIP protocol is simple, good flexibility and scalability, as well as Internet applications and close the existing features, many people believe that the agreement easier to implement, recently, especially in the United States, rapid development, while SIP will be the third generation mobile communication core network and intelligent services to be widely applied. Currently, SIP protocol is still in the early stages of development, many of the appropriate standard is not completely uniform, perfect management, large-scale network applications yet.
It should be said, H.323 and SIP protocol IP phones are for building a network of evolving, both in the early stages of development are mutually exclusive, but with the SIP protocol development, particularly in the application of soft exchange made great progress has been SIP protocol, the current H.323 and SIP interconnection agreement has also been in preparation.
In addition, whether the agreement is based on H.323 or SIP-based IP telephone network structure, completion of all necessary agreements related to the media gateway gatekeeper load control, resource control and management. At present, this interface uses Media Gateway Control Protocol (MGCP: Media Gateway Control Protocol) or H.248 (MeGaCo: Media Gateway Control) protocol.
H.248 protocol called the Media Gateway Control Protocol, is 16 by the ITU-T group put forward the application in the media gateway and media gateway controllers, media gateway controller and H.248/MeGaCo terminal between.
MGCP is defined in the earlier IETF Media Gateway Control Protocol, the main function of the angle from the definition of media gateway controllers and media gateways between the behavior of relatively simple to achieve, not H.248 packets and attributes, as detailed on the definition of events are relatively simple mechanism for interaction. MGCP with simple characteristics, but the interoperability and limit the ability to support the business.
H.248/MeGaCo its functional flexibility and ability to support the business strong and valued, and constantly have to add a new annex to its capacity, is the media gateway of the mainstream protocols, it solves Le MGCP Yuanyou the shortcomings, operational, However, the agreement released soon, the industry still lack mature products and application examples, need further development.
H.248 and MeGaCo the same in the text of the agreement, but the syntax in the protocol message transmission differ, H.248 uses ASN.1 syntax format (ITU-T X.680 1997), MeGaCo using ABNF grammar format (RFC2234).
VoIP's key technologies include:
Media Coding
Including the popular G.723.1, G.729, G.729A voice compression algorithms and MPEG-II multimedia compression technology.
Control signaling
Including ITU-T H.323 and IETF Session Initiation Protocol SIP [4] (Session Initation Protocol) two sets of standards, is also involved in real-time synchronization control continuous media streaming real-time streaming protocol TRSP.
Packet Transmission
Real-time Transport Protocol mainly RTP.
Business Quality Assurance Technology
Use of resource reservation protocol RSVP, Quality of Service Qos and for operational quality control of real-time transmission control protocol RTCP to avoid network congestion, protect the call quality.
Network transmission technology
Mainly TCP and UDP.
此外还涉及到分组重建技术和时延抖动平滑技术、动态路由平衡传输技术、网关互联技术（包括媒体互通和控制信令互通）、网络管理技术（SNMP）、安全认证和计费技术、IVR交互式语音响应技术等等。
VOIP关键技术中我们重点强调的是解决语音质量方面的技术，因为语音质量是实现VOIP的基础，关系到VOIP网络的建设成败，相信一次断断续续的语音通话是任何人都无法忍受的。影响VoIP语音质量的主要因素有：时延与时延抖动、语音编码技术、包丢失率、回声、语音电平等。
时延与时延抖动
端到端的时延包括编解码时延，打包与解包时延，以及网络传送时延。时延的变化，即时延抖动（jitter）主要由网络引起，如果端到端的传输路径中经过的中间节点（路由器、交换机等）越多，带来的时延抖动越大。
语音编码技术
语音编码技术在有效地利用带宽的同时，能提供高质量的语音。不同的编码技术将带来不同的语音质量，下表列出了几种编码技术MOS（mean opinion score）值（测试结果）。参考下表所列结果，G.729与G.711两种编码方案可以满足教育系统IP电话的质量要求。
编码技术
比特率（K bps）
MOS值
编码延时（ms）
G.711
64
4.4
0.75G.723.1(5.3K)
5.3
3.6
30G.723.1(6.3K)
6.3
3.4
30G.729
8
4.0
10MS GSM
13
3.1
20
编码技术与性能指标
另一方面，G.729的带宽要求远远低于G.711，在相同线路质量和带宽情况下，G.711数据报在线路上传输的时间要长于G.729，而且这个时间大于语音压缩成数据报的时间。因此从听觉角度分析，使用G,729和G.711压缩算法的效果基本相同。
综合考虑，采用G.729，既满足语音效果的要求，又节约了带宽，节省了线路的投资费用，是首选的编码压缩算法。
包丢失率
在IP网络中存在IP包被丢失的因素有：网络传输中丢包、网络拥塞时网关设备主动丢包。当包丢失率超过10%时将会严重影响语音质量。
回声
回声是由于通话两端的阻抗不匹配所引起。在端到端时延较大的IP网络中，回声干扰的影响尤为明显。
语音电平
合适的语音发送与接收电平是影响通话质量的又一重要因素，因此语音网关必须具有语 音电平调节的功能。
从以上的分析可以看出，要提高IP电话的语音质量主要应从两方面着手。一是选用合适的语音编码压缩技术，选用高质量的语音网关及适当地网守控制方式；二是提高IP网络的服务质量（QoS），使网络时延、时延抖动及包丢失率控制在一定限度以内。
下面分析一下网关、网守围绕上面因素对语音质量的所做的保证。
网关保证语音质量的措施
时延与抖动是影响语音质量至关重要的因素，网关通过下列技术可以在这两个因素上进行较好的控制：
静音抑制技术
静音抑制，又称活动语音检测（Voice Activity Detection），它是根据人们日常谈话的语音和静默特性，检测到静音（Silence）时加以抑制，使其不占用或极少占用信道带宽，检测到突发的活动语音时才将其进行压缩编码与传输。研究表明运用VAD技术能使信道带宽的有效利用率提高约一倍。
抖动抑制缓冲技术
抖动抑制缓冲区用于接收端，目的是平滑延迟抖动，并兼顾解码和压缩操作。
回声消除
在IP电话设备中通常都必须采取消除回声的措施，在网关设备中通常采用业界的G.165实现回波抑制。
网守保证语音质量的措施
网守具有带宽管理的功能，利用这个功能可以通过设定某条线路上语音的带宽来控制IP电话的呼叫过程，从而保证通话质量。举例如下：
某条线路带宽为1M bps，为了保证视频、数据业务的带宽，我们在网守上设定语音业务的带宽为60K bps，按照每条话路占用带宽12K bps计算，有5路以下语音时能保证通话质量，但要是超过5路通话质量就得不到保证。
利用网守的带宽管理功能，在网守上配置该线路峰值呼叫时需要的带宽，并对每一路呼叫进行监控。当第六路发起呼叫时，网守根据配置拒绝其建立连接，以确保其它5路的通话质量，同时也保证了视频和数据业务的开展。在教育系统IP电话网的设计中，我们可以根据不同情况在线路上确定通话数的峰值，通过网守来保证语音质量。
IP QoS对语音质量的保证
为了保证语音质量，网络必须具有很好的QoS保证机制。可以采用Diff-Serv或RSVP进行QoS的保证：
要求IP网络支持Diff-Serv、RSVP模式的QoS保证机制；
语音网关支持报文分类，对语音包分配高优先级别；
网络设备支持CAR（Committed Access Rate）功能，实现报文的度量和流量监控；
网络设备支持WFQ功能，对高优先级报文进行优先处理。
四、VOIP设计原则
根据VoIP应用的特点，对其网络设计主要考虑以下因素：
QoS保证
话音应用对延迟、抖动、失真等等非常敏感，因为VoIP是基于统计时分复用的IP网络为基础进行话音业务传送，因此要求充分QoS保证，在具体实现方法上，可以采用多种技术手段如RSVP带宽资源预留，使用各种队列算法如CBWFQ、PQ、CQ等等保证VoIP语音数据具有较高优先级别，ToS参数设置等等。
高效的语音编码和解码技术
普通模拟话音经过抽样、编码，通常为64K带宽，也就是G.711编码技术，要实现高效的语音传输，必须在不影响话音质量的前提下对话音数据进行压缩，在G.711基础之上，先后发展如下编码技术：G.726/G.727（16、24、32Kbps），G.728（16Kbps） ，G.729、G.729A、G.729B、G.729AB（8Kbps），G.723.1（5.3/6.4Kbps）等等，高效的压缩技术使得话音对带宽占有越来越少，而话音质量控制在可接收范围之内，促进了VoIP技术的应用。
智能的选择语音压缩技术
不同VoIP设备之间在建立呼叫前能够自动协商所使用的语音压缩技术，动态选择双方支持的语音压缩编解码标准，从而能够建立呼叫连接，实现VoIP通讯。
语音呼叫与本地网连接
VoIP语音网络能够实现IP网络内部的通讯，实现与本地电信网的连接（即通常所称的落地）可以提升VoIP网络的价值，能够实现更丰富的呼叫应用，为用户带来更多便利的服务。
VoIP网络的呼叫可管理性
对于VoIP的不同应用方式要提供灵活方便的控制手段，而且配置过程尽可能简单易懂，并且针对落地的呼叫方式，需要提供认证及提醒功能。针对用户的不同需求，可以支持多种灵活的连接方式，实现不同的呼叫过程。
网络的可靠性
网络中单点故障不会使局部网络失去与整个网络的连接，多点故障不会造成整个网络被分成几个互不相连的部分。
网络的标准化和易扩展性
网络的结构，技术和产品的标准化，结构的易扩展，技术和产品的可连续性，在满足用户当前应用的基础之上，充分保证用户投资。
网络的模块化结构
VoIP网络的的物理结构、逻辑结构和地址空间实现模块化，要和当前PSTN网络类似，为用户提供良好的使用界面，便于网络的构建、维护和扩展。
网络的易管理和维护性
全网可进行统一或分布管理，网络维护简单有效；
网络的实用性
根据现在的需求和可以预见的需求增长情况设计VoIP网络，不追求空洞的技术先进性，避免追求高档和最新技术花费的巨大代价。充分利用现有资源，挖掘原有设备的潜力，实现与VoIP网络的连接。
五、城域网VOIP解决方案
VOIP带宽分析
针对当前可用的语音编码方法的综合比较，一般选择G.729作为VoIP编解码协议标准。G.729是一种高效的语音压缩国际标准，速率为8K bps,加上包头等的开销单向语音的速率可达11.2K bps。IP电话为双工通信，但考虑到通信时每个方向至少有60%时间处于静音状态，可以采用VAD技术进行静音压缩，每路语音需12K bps带宽。G.729技术十分成熟，语音质量较好，得到了广泛的应用。
市教委VOIP连接
PBX提供E1或E&M中继线路，连接语音网关的对应中继接口。语音网关根据不同的设备形式，采用不同的连接方式：
语音网关采用地市广域网路由器＋语音卡方式：语音卡内置于语音网路由器。
纯语音网关方式：将语音网关以太网接入地市教委以太网交换机。
推荐地市教委的语音网关采用第１种连接方式，即语音卡内置于语音网路由器，具有更好的性价比。另外，如果设置地市级网守，将网守的以太网口接入地市教委以太网交换机。
区县教委VOIP连接
有PBX的区县教委，组网形式和连接方式同地市教委设计方案。无PBX或即将淘汰原有PBX的县教委，可采用IP电话终端方式组建区、县教委电话网络，这种情况下需配置本地呼叫管理系统，呼叫管理系统接受上级网守的管理，IP电话终端和呼叫管理系统直接挂接在县教委以太网交换机上；也可采用IP PBX的形式组建区、县教委电话网络，IP PBX的以太网接口直接接入县教委以太网交换机。
神州数码用于市教委和区县教委接入的DCR-3660、DCR-2600系列路由器全面支持H.323 VoIP功能，能够灵活提供E&M、FXO、FXS等多种语音接口方便与各类PBX和传统语音设备的连接，并提供等位拨号等针对教育网络定制开发的特性功能，是构建低成本、高性能教育系统IP语音网络和数据网络的可靠保证。
同时，DCR-3660、DCR-2600系列路由器为增强教育城域网VoIP管理，提供内置H.323网守功能，可以在区县教委提供免费本地网守，减轻市教委网守的压力。
中小学校VOIP连接
一般用户数量较少，可采用广域网路由器＋FXS语音卡的方式，直接由语音网关提供模拟电话接口，作为中小学电话网解决方案。
神州数码的蓝箱产品除了可以完成路由器的广域网接入功能、本地交换网的接入，同时还可以通过灵活的语音模块完成VOIP的功能。神州数码用于中小学接入的DCR-1700、DCR-2500V系列路由器提供低成本的H.323 VoIP功能，DCR-1700能够灵活提供E&M、FXO、FXS等多种语音接口方便与各类PBX和传统语音设备的连接，DCR-2500V内置2个FXS IP电话接口，完全满足一般中小学的IP语音需求。
神州数码路由器VoIP的特点
静音压缩
自动检测静音状态，取消静音在网络上的传输，从而减少网络上的无效负载，该技术同舒适噪音相结合，能够获得更好的通话效果。
舒适噪音
由于静音压缩带来通话间断现象，为满足人的听觉感官的需要，自动向本地用户发送舒适的背景声。
头部压缩
支持PPP、FR、HDLC上的CRTP协议；支持RFC2507及RFC1144定义的TCP/IP头部压缩。
语音调谐
可以通过命令调谐语音端口的输入输出增益，改善通话效果。
Quality of service
QoS一直是Internet上传送实时语音的关键技术，我们的QoS支持CQ、PQ、WFQ、WRED等先进的排队策略。我们的VoIP应用默认情况下使用最适用于实时应用的WFQ策略。
资源预留
支持RSVP协议，保证语音传输的带宽需求。
语音平滑
跟踪著名网络实验室的最新研究成果，按照权威的Jitter Buffer算法论文，并根据自身实践，设计了一套行之有效的Jitter Buffer控制策略，成功解决了由于网络传输拥塞和语音分片不均等原因造成的语音抖动，在多项应用中证明了其良好的效果。
Gatekeeper
实现了嵌入式的Gatekeeper软件，便于VoIP网络的集中管理维护，同时基于Windows的GUI界面的版本也在研发进行中，不久将投向市场。
FAXoIP
对传统的FAX的支持是神州数码语音产品的又一特色，支持FRF.11及T.38两种格式的FAX编码方式，同时还支持原有的直通模式的传真技术。
Value-added services
实现了基于标准的H.450系列协议的增值服务。
Safety Certification
使用AAA技术和Radius协议进行安全认证，保障信息的安全性。
记账功能
基于Radius协议的与Gatekeeper软件集成的记账系统，使用便捷。
IVR功能
利用交互式语音响应技术交互式完成认证、转接、计费等工作。