IP phone is usually called Internet phone or network phone. As the name suggests, it is to make calls through the Internet. In a broad sense, it should be called Internet telecommunications because it includes voice, fax, video transmission and other telecommunications services.
2. Basic principles of IP phone
Voice of IP phone is transmitted by using IP (Internet/Intranet) data network based on router/packet switching. Since the Internet adopts the "store-and-forward" method to transmit data packets, it does not monopolize the circuit, and the voice signal is greatly compressed, so the bandwidth occupied by IP phone is only 8kbit/s-10kbit/s. In addition, the billing method of packet switching has nothing to do with the distance, which naturally greatly saves long-distance communication costs. The Internet is composed of many different computer networks connected all over the world. The emergence and popularization of the Internet has greatly changed the way people communicate and communicate. The Internet uses the standard TCP/IP protocol to realize mutual communication and data exchange between computers. The TCP/IP protocol is responsible for queuing the IP data packets to be transmitted and sending them to the network. Each packet contains address and data reorganization information to ensure data security and correct data packet exchange. IP Telephony uses the Internet as the main transmission medium for voice transmission. First, the voice signal is transmitted to the IP Telephony gateway through the public telephone network; then the gateway converts and compresses the voice signal into a digital signal and transmits it to the Internet; these digital signals are transmitted to the gateway at the other party's location through a low-cost network that spans the globe, and the gateway then restores the digital signal to an analog signal and inputs it into the local public telephone network, ultimately transmitting the voice signal to the recipient.
3. Key equipment of IP phone system - gateway The
gateway located in various places is represented by a unique IP address. It is a bridge between two communication transmission modes and is the "exchange office" on the Internet to realize the interconnection and communication between remote telephones. On one side, the gateway connects to the traditional circuit-switched network (Circuit-switched Network) such as the public switched telephone network (PSTN) and can communicate with any external telephone. On the other side, the gateway connects to the packet-switched network (Packet-Switched Network) such as the Internet, Intranet, etc., and can communicate with any computer connected to the network. In the entire Internet Phone system, gateways are distributed all over the world to handle the access and conversion between the local PSTN network and the Internet. The gateway receives standard telephone signals, digitizes and greatly compresses them, and uses the IP protocol to group them and send them to the Internet, find the transmission route, and send them to the destination through the Internet. Conversely, it receives data packets transmitted from the Internet and transfers them to the telephone network system. Accessing and transferring to the telephone network system can be carried out simultaneously to achieve full-duplex (two-way) calls. For example, if you make a long-distance call from Beijing to San Francisco, in Beijing, an ordinary public telephone is connected to the local gateway through the PSTN. The local gateway processes the data with a specific compression algorithm, organizes it into IP packets containing the calling and called numbers, time, call information and other data, and analyzes the called number. According to the routing table, it maps it to an IP address, and sends it to the remote gateway corresponding to the IP address (such as San Francisco) through routing selection. In San Francisco, the called party, the remote gateway receives the IP data packet transmitted by the local gateway in Beijing, decompresses it in the opposite process, and then sends it to the PSTN network at its local end. In this way, real-time communication between the two places is realized. The communication cost included is only the local ordinary telephone fee in Beijing + Internet communication fee + local telephone fee in San Francisco. Since the communication fee of the Internet is low, the cost of long-distance calls is greatly reduced.
4. Voice quality of IP phone Voice quality basically depends on two factors: one is the speed of the Internet communication line; the other is whether the Internet itself is busy. Compared with the voice quality of ordinary phones, the voice quality of IP phones
mainly differs in two aspects: First, the voice is delayed, and second, there is a slight distortion sometimes. Anyone who has used IP phones generally believes that the voice quality is better than expected, generally between ordinary phones and mobile phones. In order to improve the voice quality, the most direct way is to increase the Internet access rate and use a good Internet access line.
5. Several key technologies and standards in IP phone systems (1)
* Basic standards for IP phones The standard for Internet phones adopts the ITU-T H.323 standard. H.323 is one of the ITU's multimedia communication protocol series H.32x. H.323 provides basic standards for transmitting voice, video and data based on IP networks (including the Internet). It is a framework protocol. The related transmission, control and voice and video compression standards are shown in the table below (the table also includes a series of protocols for multimedia in other networks (ISDN, PSTN)). H.323 defines four basic components in network transmission: terminals, gateways, gatekeepers and multipoint controllers (MCUs). * Network protocol standards Generally speaking, the call establishment and control of Internet telephones are mostly based on TCP, while the transmission of audio streams is based on UDP. In order to ensure the real-time transmission, IETF has added several important protocols: RSVP (Resource Reservation Protocol): Generally speaking, it is very difficult to reserve enough bandwidth for multimedia transmission on IP networks. For this reason, IETF has defined the Resource Reservation Protocol (RSVP). RSVP allows the receiver to apply for a specific amount of bandwidth for data transmission. With RSVP, the traditional IP network without QoS (Quality of Service) guarantee has obtained QoS guarantee. To be able to use RSVP, H.323 terminals, gateways, MCUs, etc. must support it, and routers on IP networks must also support it. RSVP is defined in RFC2205-RFC2209. RTP/RTCP (Real-Time Protocol/Real-Time Control Protocol): RTP is a protocol defined by IETF for transmitting audio and video streams. RTP is based on UDP. In the header of RTP, a timestamp is defined to ensure the real-time transmission and synchronization of audio and video. RTCP is a protocol for controlling and monitoring RTP and its QoS. H.323 is based on RTP. The RTP/RCt protocol can be found in RFC1889 and C1890.
6. Several key technologies and standards in IP telephone systems (2)
* The speech coding standard H.323 defines the transmission of multiple voices. IETF has established the AVT (Audio/Video Transport) working group to study speech transmission. At present, the commonly used speech coding bit stream rates in Internet telephony are as follows: G.711 64Kbit/s, G722 48-64kbit/s, G.728 16kbit/s, G.723 and G.723.1 5.3kbit/s or 6.3kbit/s, G.729 and G.729A8 or 13kbit/s. Not transmitting voice data when the two parties are not talking can effectively save bandwidth. However, in order to prevent the sound of intermittent calls during silence compression, it is recommended to add background noise during silence. The VoIP Forum of IMTC has proposed a background noise transmission method with variable parameters. * The system control of the control module H.323 includes: H.245 control, Q.931 call signal control and RAS control. The H.245 control channel is a trusted channel used to carry control information for the operation of H.323 entities. These controls include: performance exchange, opening or closing logical channels, priority requests, process control information, and basic command instructions. The call signaling channel uses Q.931 to establish a connection between two terminals. The RAS signaling channel completes registration, access rights, bandwidth changes, and status updates. The RAS signaling channel is generally established between the terminal and the gatekeeper. If the gatekeeper does not exist, there will be no RAS channel.
Evolution of IP Telephone Technology - IP telephony has developed rapidly since 1995 with its economic, high efficiency and advanced technology development. It has become one of the most competitive technologies in data voice communication. Many countries around the world have launched IP telephony services. my country's IP telephony trial operation has been more than half a year. IP technology is showing vigorous vitality and will surely promote the further development of the information industry. The development of IP telephony has gone through two primary stages and is currently evolving rapidly to the third stage - unified integration. 1 Technology accumulation stage - In the technology accumulation stage, experts in the CTI field proposed the idea of packet transmission of voice: all packet voice systems follow a common model, and the packet voice transmission network can use IP, frame relay or ATM. Devices or components called "voice agents" are set at the edge of these networks. Their task is to convert voice information from traditional voice formats to formats suitable for packet transmission, and then send packet data to the voice agent device at the destination through the above network. - Voice agent connection mode Two problems need to be solved in the packet voice network transmission system to make the packet voice service meet the needs of users. The first is the conversion of voice coding, that is, how to convert voice information into digital signals; another issue is signaling conversion, which mainly identifies the object called by the caller and the location of the caller in the network. ——Human language is expressed in the form of analog signals. Early telephone analog signals can be described as smooth "sine waves". Although analog communication technology has been quite developed, the transmission efficiency is not high. When the transmission attenuation causes the analog signal to weaken, it is difficult to distinguish complex analog voice information from transmission noise. ——Digital signals have only two states, "1" and "0", which are easy to distinguish from noise and are not easy to distort. Therefore, the global communication system has been converted to a digital transmission format called pulse code modulation (PCM). PCM converts analog voice into digital format. Standard telephone PCM uses 8-bit code and 8000/second sampling frequency, so each phone line occupies 64kb/s channel bandwidth. Another telephone voice standard called adaptive differential PCM (ADPCM) converts voice into 4-bit code, so it only occupies 32kb/s. ADPCM is usually used for long-distance lines. ——Based on this technology, people successfully developed the first generation of IP telephone equipment. Using the voice collection principle of the sound card on the computer, the 64kb/s analog voice is converted into ADPCM digital signal, and the primary IP telephone function of computer to computer is realized on the Internet. Since this system mainly uses computers to complete voice compression and control, it can generally only realize real-time communication of one voice. For example, on the PII233 computer system, only 4 voice channels can be completed at most. In this gray system, there are many more practical IP telephone systems, such as Vocaltec's IPhone and Microsoft's Netmeeting system. The successful development of the first generation of IP system has aroused people's great interest in IP telephone system, thus promoting the application research of IP telephone technology. People hope to use IP telephone system like general telephone system. 2 Practical stage——The second development stage of IP telephone is a leap forward based on the first stage. It can not only realize the use of IP telephone system for communication like PSTN system, but also realize calls with large traffic volume. The stage of IP telephone communication using the current PSTN switching system is called "practical stage". The IP phone in the practical stage is mainly a network access device, which completes the data network transmission and PSTN transfer functions. A practical IP phone access terminal system (we call it Gateway) generally includes five parts: ——. Establishing and controlling the connection, conversation and disconnection of the phone ——. Voice compression and data encoding processing ——. Data network transmission and control: ——. System maintenance part ——. User information management——This type of system is still built on the computer system, but it is not a terminal user device. Therefore, for ordinary users, only one telephone is needed to realize IP communication. Let's study the functions and implementation methods of each part. 2.1 Establishing and controlling the connection, conversation and disconnection of the phone——Establishing and controlling the connection, conversation and disconnection of the phone is the information exchange interface between the IP phone system and the PSTN, and is also the entry and exit gateway for the current general telephone system to convert to the Internet/Intranet. This part of the work is mainly achieved through programming control of telephone cards (such as E1 cards). ——Since the E1 card can receive PSTN information, remove the relevant signaling, and record it as a pure digital voice signal, the signaling conversion work is basically completed by the E1 card. However, in a complete IP phone gateway, each component must exchange information and coordinate work. Sufficient information exchange is required between the E1 card and the voice compression card, between the voice compression card and the network card (NIC), and between each component and the user interface. The exchange of this information can be controlled by the behavior of the state machine. 2.2 Phone connection and disconnection work——First, the calling user A on the PSIN picks up the phone. After receiving the off-hook signal of the calling user, the sending Capitel sends a dial tone or IVR (interactive voice response) prompt to the calling user. The calling user hears the dial tone, starts dialing, and sends the called number to the A-end switch Capitel. ——The A-end Capitel selects the IP address and the best path according to the called number, and sends a channel occupation signal to the B-end Capitel on the selected path, that is, the outgoing signal of the A-end Capitel occupies the incoming signal of the B-end Capitel. Then the Capitel at end A sends the called number to the Capitel at end B. (Note: This system takes the Capitel IP phone system of Beijing Post and Telecommunications Equipment Factory as an example). --The Capitel at end B converts the pure digital signal into a PCM signal according to the called number and sends it to the PSTN at end B to connect the called user. The called user picks up the phone to answer and sends the off-hook signal to the Capitel at end B, which is then forwarded to the Capitel at end A, and the two parties begin to talk. When the call ends, if the user at end A hangs up first, the calling user sends a recovery or disconnection signal to the Capitel, and the Capitel at end B sends this signal to the PSTN at end B; if the user at end B hangs up first, B sends a recovery or disconnection signal to the Capitel at end A, and everything is restored. ——2.3 Data Processing of Voice Compression——Voice compression mainly compresses voice signals. Commonly used voice processing methods include: G.711, G.722, G.729 and G.723. These compression algorithms must be processed on hardware, otherwise, it is impossible to achieve the call task of large traffic volume. This part can use the program to control the language compression card so that it can process the voice signal in real time according to our needs. When the voice data is collected, it must be placed in the memory. In the process of collection. The first step is to collect the uncompressed digital signal, and then send it to the specified memory according to the required structure after compression processing, and under the control of the CPU, use the algorithm in the DSP to perform the corresponding data compression processing. The voice signal after compression processing is then grouped and encoded to form a standard data packet, and then the data of these packets are sent to the network in the form of a stream for transmission. ——2.4 Data Coding Processing——Data coding processing is the main task of the H.323 module. It is the key to whether the voice data transmission format can be received by different systems. The protocol was announced by ITU on May 28, 1996. It has been widely used in multimedia data communication. It is a multimedia communication protocol used in the Integrated Services Digital Network (ISDN). ——Specific protocol standards include: H.255.0 (call processing protocol), H.245 (control processing protocol), H.261 and H.263 (video processing protocol), T.12O (data processing protocol). In the IP telephone system, this part of the work mainly completes the following tasks: ——. Real-time audio coding processing ——. Control protocol ——. Data transmission protocol ——2.5—Data exchange between gateways—Data exchange between gateways is a very important and difficult technology in the development of IP telephone systems. Although IP telephone manufacturers claim that their equipment meets the basic requirements of the H.323 standard protocol, each manufacturer has its own processing methods in the specific processing of H.225, H.245 and Q.931. In the comparison between Vocaltec, the founding manufacturer of IP telephones, and the Capitel IP switch system of Beijing Post and Telecommunications Equipment Factory, both products meet the H.323 specification, but the processing of G sets in the H.323 protocol is completely different, because H.323 does not clearly state the processing method of G sets. Vocaltec uses a three-step encoding method to encapsulate H.323 packets, while the Capitel IP switch system uses the Chinese standard eight-step encoding method to encapsulate H.323 packets. In this way, when the two products communicate with each other, due to the different encapsulation methods of H.323 packets and different interpretations of the received H.323 packets, incompatibility occurs. Serial number requirements 1 The gateway supports G.729A and G.723 multimedia digital signal codec protocols. G.729A is supported first, followed by G.723.1 2 The gateway supports DTMF and MF decoding and encoding (when making an outgoing call), and the IVR system can be used for final call 3 The gateway supports intercommunication with the gateway of the switching center 4 The gateway supports the "Quick Setup" in the protocol H.323 V2 5 The end-to-end information record code can be transmitted between the gateway and the gateway 6 The gateway can use the settlement system and the call confirmation from the settlement system operator for authentication 7 The call detail record can be generated and transmitted to the settlement center in real time 8 To sell gateways and gateways that comply with iNOW! 2.0 version, first of all, you must use the iNOW! authoritative organization and successfully complete the iNOW! organization's authentication procedures 9 For calls to the settlement center, through the settlement center and the terminal signal, iNOW! The platform provides compatibility capabilities10. It provides intercommunication functions between gateways and settlement systems11. It can transmit gateway routing call signaling and terminal routing call signaling12. The local gateway clock can be synchronized with an accurate and reliable time source for at least 24 hours13. The terminal source code can be generated in the settlement system CDR using the following algorithm: H.323+1000 Q.931 14. The gateway supports the T.38 fax protocol. Mandatory support for TCP/UDP/IP and V.21, V.27 V.17 15 For settlement system calls, the gateway will ensure the integrity of the information. In view of the current lack of universal international standards, Lucent, Itexc and Vocaltec jointly developed the IP phone industry standard iNow! protocol in January 1999. The protocol mainly includes five aspects: ——. Gateway to Gateway intercommunication requirements ——. Gatekeeper to Gatekeeper intercommunication requirements ——. Gatekeeper to settlement center intercommunication requirements ——. Phone to Phone service requirements ——. FAX to FAX service requirements —— While meeting the above requirements, the information exchange process must be completed under the control of the settlement center. IP phone operators in different regions can complete various authentication and exchange tasks through the settlement center. In the iNow! protocol, the connection and disconnection process is also strictly defined, thus ensuring that the products of different manufacturers can be compatible with each other in the connection and disconnection process. While stipulating the algorithm and information exchange specifications, the iNow! protocol also stipulates various detailed message formats. In this way, when different manufacturers apply the protocol, there will be no objection, so that the products of various IP phone manufacturers can be compatible with each other. ——However, good wishes are not equal to reality. Since its birth, the INOW! protocol has had many problems. First of all, it is a supplement to the H.323 protocol. It does not define a new protocol and is still limited to the scope of the H.323 protocol. The incompleteness of the H.323 protocol in the network layer and the lack of guarantee for transmission cannot be solved by the iNOW! protocol. Secondly, the iNOW! protocol is an industry standard and has not yet been supported by the ITU. Therefore, although many manufacturers support the iNOW! protocol for more than a year after its launch, the products of manufacturers claiming to support the protocol are not compatible with each other. ——After more than half a year of trial operation, the Chinese IP phone system, under the organization of the Ministry of Information Industry, combined with my country's network conditions and the problems reflected by users, has formulated China's IP phone compatibility standards and performance requirements in response to the problems existing in the current IP phone system. With the cooperation of relevant units, the network access test and certification of IP phone equipment have been carried out, and good results have been achieved. 3 Technology Convergence - Network development is evolving towards broadband and intelligence. The current convergence of circuit switching and packet switching is the inevitable result of this trend. Due to its high transmission efficiency and low cost, packet switching will gradually replace the current circuit switching network. Various access networks (wireless, xDSL, Cable, optical access, etc.) will become a unified packet switching backbone network. In the future network architecture, the No. 7 signaling system will coexist with the IP network for a certain period of time, and it will play an important role in the IP network. In the trend of various network convergence, an obvious change is that the powerful functions of the circuit switch in the past are being continuously decomposed, and the interface is being standardized. The MGCP (Media Gateway Control Protocol) protocol has unified the interface between the IP network and the PSTN network, and the IPST (Internet Protocol Standard Transmit) protocol has a unified implementation method for circuit switching signaling in the IN network. This makes the distributed call processing structure in the IP telephone field possible, laying a solid network foundation for the current and future applications of the IP telephone system. The IP telephone system developed in this stage is called the "unified stage" IP telephone system. ——The most notable feature of IP telephony in the unified stage is that various IP telephony devices are compatible with each other, extending the circuit switching concept to the entire network, and operators can exchange without obstacles on the entire IP network. Protocols represented by MGCP and IPST unify the specifications of H.323 and iNOW! protocols, and standardize the interface signaling between IP networks and PSTN networks (IPST protocol). We know that IP telephony systems are generally divided into three layers, namely: connection layer, control layer and service management layer. ——The connection layer is responsible for establishing and implementing the connection of the physical layer. While completing the information exchange between the IP network and the PSTN network, it is responsible for transmitting the encoded voice signal to the control system. The control layer completes the call request connection. The relevant protocols of this layer are: H.323, H.GCP and SIP, etc. The main task of these protocols is to complete the encapsulation of voice signals and establish appropriate bearer connections. The service management layer mainly completes the service control of operators, such as user management, billing, settlement and user authorization. This layer must support the A interface (intelligent network interface), so this layer is also closely related to the H.GCP (MGCP) protocol. ——MGCP is a protocol standard defined for all gateways between PSTN and IP networks. The most typical applications are IP phone gateways and dial-up access servers. Therefore, the future structures of IP phone gateways and dial-up access servers are very similar. The difference is that the IP phone gateway completes the bundling of PSTN voice resources and RTP session resources, while the dial-up access server completes the bundling of PSTN voice resources and IP sessions. Therefore, future dial-up access servers will be able to automatically identify IP access and IP phone (or fax) access, and dynamically implement channel allocation and resource bundling on demand.323 specification, but the processing of G set in H.323 protocol is completely different, because H.323 does not clearly explain the processing method of G set. Vocaltec uses three-step encoding method to encapsulate H.323 packets, while Capitel IP switch system uses eight-step encoding method of Chinese standard to encapsulate H.323 packets. Therefore, when the two products communicate with each other, due to the different encapsulation methods of H.323 packets, the interpretation of received H.323 packets is different, resulting in incompatibility. Sequence number requirement 1 The gateway supports G.729A and G.723 multimedia digital signal codec protocols. G.729A is supported first, followed by G.723.1 2 The gateway supports DTMF and MF decoding and encoding (when calling out), and finally the IVR system can be used for mourning 3 The gateway supports intercommunication with the gateway of the switching center 4 The gateway supports the "Quick Setup" in the protocol H.323 V2 5 The end-to-end information record code can be transmitted between the gateway and the gateway 6 The gateway can use the settlement system and the call confirmation from the settlement system operator for authentication 7 The call detail record can be generated and transmitted to the settlement center in real time 8 To sell gateways and gateways that comply with iNOW! 2.0 version, first of all, it is necessary to use the iNOW! authority and successfully complete the certification procedures of the iNOW! organization 9 For the call to the settlement center, through the signal of the settlement center and the terminal, iNOW! The platform provides compatibility capabilities10. It provides intercommunication functions between gateways and settlement systems11. It can transmit gateway routing call signaling and terminal routing call signaling12. The local gateway clock can be synchronized with an accurate and reliable time source for at least 24 hours13. The terminal source code can be generated in the settlement system CDR using the following algorithm: H.323+1000 Q.931 14. The gateway supports the T.38 fax protocol. Mandatory support for TCP/UDP/IP and V.21, V.27 V.17 15 For settlement system calls, the gateway will ensure the integrity of the information. In view of the current lack of universal international standards, Lucent, Itexc and Vocaltec jointly developed the IP phone industry standard iNow! protocol in January 1999. The protocol mainly includes five aspects: ——. Gateway to Gateway intercommunication requirements ——. Gatekeeper to Gatekeeper intercommunication requirements ——. Gatekeeper to settlement center intercommunication requirements ——. Phone to Phone service requirements ——. FAX to FAX service requirements —— While meeting the above requirements, the information exchange process must be completed under the control of the settlement center. IP phone operators in different regions can complete various authentication and exchange tasks through the settlement center. In the iNow! protocol, the connection and disconnection process is also strictly defined, thus ensuring that the products of different manufacturers can be compatible with each other in the connection and disconnection process. While stipulating the algorithm and information exchange specifications, the iNow! protocol also stipulates various detailed message formats. In this way, when different manufacturers apply the protocol, there will be no objection, so that the products of various IP phone manufacturers can be compatible with each other. ——However, good wishes are not equal to reality. Since its birth, the INOW! protocol has had many problems. First of all, it is a supplement to the H.323 protocol. It does not define a new protocol and is still limited to the scope of the H.323 protocol. The incompleteness of the H.323 protocol in the network layer and the lack of guarantee for transmission cannot be solved by the iNOW! protocol. Secondly, the iNOW! protocol is an industry standard and has not yet been supported by the ITU. Therefore, although many manufacturers support the iNOW! protocol for more than a year after its launch, the products of manufacturers claiming to support the protocol are not compatible with each other. ——After more than half a year of trial operation, the Chinese IP phone system, under the organization of the Ministry of Information Industry, combined with my country's network conditions and the problems reflected by users, has formulated China's IP phone compatibility standards and performance requirements in response to the problems existing in the current IP phone system. With the cooperation of relevant units, the network access test and certification of IP phone equipment have been carried out, and good results have been achieved. 3 Technology Convergence - Network development is evolving towards broadband and intelligence. The current convergence of circuit switching and packet switching is the inevitable result of this trend. Due to its high transmission efficiency and low cost, packet switching will gradually replace the current circuit switching network. Various access networks (wireless, xDSL, Cable, optical access, etc.) will become a unified packet switching backbone network. In the future network architecture, the No. 7 signaling system will coexist with the IP network for a certain period of time, and it will play an important role in the IP network. In the trend of various network convergence, an obvious change is that the powerful functions of the circuit switch in the past are being continuously decomposed, and the interface is being standardized. The MGCP (Media Gateway Control Protocol) protocol has unified the interface between the IP network and the PSTN network, and the IPST (Internet Protocol Standard Transmit) protocol has a unified implementation method for circuit switching signaling in the IN network. This makes the distributed call processing structure in the IP telephone field possible, laying a solid network foundation for the current and future applications of the IP telephone system. The IP telephone system developed in this stage is called the "unified stage" IP telephone system. ——The most notable feature of IP telephony in the unified stage is that various IP telephony devices are compatible with each other, extending the circuit switching concept to the entire network, and operators can exchange without obstacles on the entire IP network. Protocols represented by MGCP and IPST unify the specifications of H.323 and iNOW! protocols, and standardize the interface signaling between IP networks and PSTN networks (IPST protocol). We know that IP telephony systems are generally divided into three layers, namely: connection layer, control layer and service management layer. ——The connection layer is responsible for establishing and implementing the connection of the physical layer. While completing the information exchange between the IP network and the PSTN network, it is responsible for transmitting the encoded voice signal to the control system. The control layer completes the call request connection. The relevant protocols of this layer are: H.323, H.GCP and SIP, etc. The main task of these protocols is to complete the encapsulation of voice signals and establish appropriate bearer connections. The service management layer mainly completes the service control of operators, such as user management, billing, settlement and user authorization. This layer must support the A interface (intelligent network interface), so this layer is also closely related to the H.GCP (MGCP) protocol. ——MGCP is a protocol standard defined for all gateways between PSTN and IP networks. The most typical applications are IP phone gateways and dial-up access servers. Therefore, the future structures of IP phone gateways and dial-up access servers are very similar. The difference is that the IP phone gateway completes the bundling of PSTN voice resources and RTP session resources, while the dial-up access server completes the bundling of PSTN voice resources and IP sessions. Therefore, future dial-up access servers will be able to automatically identify IP access and IP phone (or fax) access, and dynamically implement channel allocation and resource bundling on demand.323 specification, but the processing of G set in H.323 protocol is completely different, because H.323 does not clearly explain the processing method of G set. Vocaltec uses three-step encoding method to encapsulate H.323 packets, while Capitel IP switch system uses eight-step encoding method of Chinese standard to encapsulate H.323 packets. Therefore, when the two products communicate with each other, due to the different encapsulation methods of H.323 packets, the interpretation of received H.323 packets is different, resulting in incompatibility. Sequence number requirement 1 The gateway supports G.729A and G.723 multimedia digital signal codec protocols. G.729A is supported first, followed by G.723.1 2 The gateway supports DTMF and MF decoding and encoding (when calling out), and finally the IVR system can be used for mourning 3 The gateway supports intercommunication with the gateway of the switching center 4 The gateway supports the "Quick Setup" in the protocol H.323 V2 5 The end-to-end information record code can be transmitted between the gateway and the gateway 6 The gateway can use the settlement system and the call confirmation from the settlement system operator for authentication 7 The call detail record can be generated and transmitted to the settlement center in real time 8 To sell gateways and gateways that comply with iNOW! 2.0 version, first of all, it is necessary to use the iNOW! authority and successfully complete the certification procedures of the iNOW! organization 9 For the call to the settlement center, through the signal of the settlement center and the terminal, iNOW! The platform provides compatibility capabilities10. It provides intercommunication functions between gateways and settlement systems11. It can transmit gateway routing call signaling and terminal routing call signaling12. The local gateway clock can be synchronized with an accurate and reliable time source for at least 24 hours13. The terminal source code can be generated in the settlement system CDR using the following algorithm: H.323+1000 Q.931 14. The gateway supports the T.38 fax protocol. Mandatory support for TCP/UDP/IP and V.21, V.27 V.17 15 For settlement system calls, the gateway will ensure the integrity of the information. In view of the current lack of universal international standards, Lucent, Itexc and Vocaltec jointly developed the IP phone industry standard iNow! protocol in January 1999. The protocol mainly includes five aspects: ——. Gateway to Gateway intercommunication requirements ——. Gatekeeper to Gatekeeper intercommunication requirements ——. Gatekeeper to settlement center intercommunication requirements ——. Phone to Phone service requirements ——. FAX to FAX service requirements —— While meeting the above requirements, the information exchange process must be completed under the control of the settlement center. IP phone operators in different regions can complete various authentication and exchange tasks through the settlement center. In the iNow! protocol, the connection and disconnection process is also strictly defined, thus ensuring that the products of different manufacturers can be compatible with each other in the connection and disconnection process. While stipulating the algorithm and information exchange specifications, the iNow! protocol also stipulates various detailed message formats. In this way, when different manufacturers apply the protocol, there will be no objection, so that the products of various IP phone manufacturers can be compatible with each other. ——However, good wishes are not equal to reality. Since its birth, the INOW! protocol has had many problems. First of all, it is a supplement to the H.323 protocol. It does not define a new protocol and is still limited to the scope of the H.323 protocol. The incompleteness of the H.323 protocol in the network layer and the lack of guarantee for transmission cannot be solved by the iNOW! protocol. Secondly, the iNOW! protocol is an industry standard and has not yet been supported by the ITU. Therefore, although many manufacturers support the iNOW! protocol for more than a year after its launch, the products of manufacturers claiming to support the protocol are not compatible with each other. ——After more than half a year of trial operation, the Chinese IP phone system, under the organization of the Ministry of Information Industry, combined with my country's network conditions and the problems reflected by users, has formulated China's IP phone compatibility standards and performance requirements in response to the problems existing in the current IP phone system. With the cooperation of relevant units, the network access test and certification of IP phone equipment have been carried out, and good results have been achieved. 3 Technology Convergence - Network development is evolving towards broadband and intelligence. The current convergence of circuit switching and packet switching is the inevitable result of this trend. Due to its high transmission efficiency and low cost, packet switching will gradually replace the current circuit switching network. Various access networks (wireless, xDSL, Cable, optical access, etc.) will become a unified packet switching backbone network. In the future network architecture, the No. 7 signaling system will coexist with the IP network for a certain period of time, and it will play an important role in the IP network. In the trend of various network convergence, an obvious change is that the powerful functions of the circuit switch in the past are being continuously decomposed, and the interface is being standardized. The MGCP (Media Gateway Control Protocol) protocol has unified the interface between the IP network and the PSTN network, and the IPST (Internet Protocol Standard Transmit) protocol has a unified implementation method for circuit switching signaling in the IN network. This makes the distributed call processing structure in the IP telephone field possible, laying a solid network foundation for the current and future applications of the IP telephone system. The IP telephone system developed in this stage is called the "unified stage" IP telephone system. ——The most notable feature of IP telephony in the unified stage is that various IP telephony devices are compatible with each other, extending the circuit switching concept to the entire network, and operators can exchange without obstacles on the entire IP network. Protocols represented by MGCP and IPST unify the specifications of H.323 and iNOW! protocols, and standardize the interface signaling between IP networks and PSTN networks (IPST protocol). We know that IP telephony systems are generally divided into three layers, namely: connection layer, control layer and service management layer. ——The connection layer is responsible for establishing and implementing the connection of the physical layer. While completing the information exchange between the IP network and the PSTN network, it is responsible for transmitting the encoded voice signal to the control system. The control layer completes the call request connection. The relevant protocols of this layer are: H.323, H.GCP and SIP, etc. The main task of these protocols is to complete the encapsulation of voice signals and establish appropriate bearer connections. The service management layer mainly completes the service control of operators, such as user management, billing, settlement and user authorization. This layer must support the A interface (intelligent network interface), so this layer is also closely related to the H.GCP (MGCP) protocol. ——MGCP is a protocol standard defined for all kinds of gateways between PSTN and IP networks. The most typical applications are IP phone gateways and dial-up access servers. Therefore, the future structures of IP phone gateways and dial-up access servers are very similar. The difference is that the IP phone gateway completes the bundling of PSTN voice resources and RTP session resources, while the dial-up access server completes the bundling of PSTN voice resources and IP sessions. Therefore, future dial-up access servers will be able to automatically identify IP access and IP phone (or fax) access, and dynamically implement channel allocation and resource bundling on demand.323 V2 "Quick Setup" 5 End-to-end message record code can be passed between gateways and gateways 6 Gateways can use the settlement system and the call confirmation from the settlement system operator for authentication 7 Call detail records can be generated and transmitted to the settlement center in real time 8 To sell gateways and gateways that meet iNOW! 2.0 version, you must first use the iNOW! authority and successfully complete the iNOW! organization's certification process 9 For calls to the settlement center, the iNOW! platform provides compatibility through the settlement center and terminal signals 10 Provides intercommunication between gateways and settlement systems 11 Can transmit gateway routing call signaling and terminal routing call signaling 12 For at least 24 hours, the local gateway clock can be synchronized with an accurate and reliable time source 13 The terminal source code can be generated in the settlement system CDR using the following algorithm: H.323+1000 Q.931 14 The gateway supports T.38 fax protocol. Mandatory support for TCP/UDP/IP and V.21, V.27 V.17 15 For settlement system calls, the gateway will ensure the integrity of the information. In view of the current lack of universal international standards, Lucent, Itexc and Vocaltec jointly developed the IP phone industry standard iNow! protocol in January 1999. The protocol mainly includes five aspects: ——. Gateway to Gateway intercommunication requirements ——. Gatekeeper to Gatekeeper intercommunication requirements ——. Gatekeeper to settlement center intercommunication requirements ——. Phone to Phone service requirements ——. FAX to FAX service requirements —— While meeting the above requirements, the information exchange process must be completed under the control of the settlement center. IP phone operators in different regions can complete various authentication and exchange tasks through the settlement center. In the iNow! protocol, the connection and disconnection process is also strictly defined, thus ensuring that the products of different manufacturers can be compatible with each other in the connection and disconnection process. While stipulating the algorithm and information exchange specifications, the iNow! protocol also stipulates various detailed message formats. In this way, when different manufacturers apply the protocol, there will be no objection, so that the products of various IP phone manufacturers can be compatible with each other. ——However, good wishes are not equal to reality. Since its birth, the INOW! protocol has had many problems. First of all, it is a supplement to the H.323 protocol. It does not define a new protocol and is still limited to the scope of the H.323 protocol. The incompleteness of the H.323 protocol in the network layer and the lack of guarantee for transmission cannot be solved by the iNOW! protocol. Secondly, the iNOW! protocol is an industry standard and has not yet been supported by the ITU. Therefore, although many manufacturers support the iNOW! protocol for more than a year after its launch, the products of manufacturers claiming to support the protocol are not compatible with each other. ——After more than half a year of trial operation, the Chinese IP phone system, under the organization of the Ministry of Information Industry, combined with my country's network conditions and the problems reflected by users, has formulated China's IP phone compatibility standards and performance requirements in response to the problems existing in the current IP phone system. With the cooperation of relevant units, the network access test and certification of IP phone equipment have been carried out, and good results have been achieved. 3 Technology Convergence - Network development is evolving towards broadband and intelligence. The current convergence of circuit switching and packet switching is the inevitable result of this trend. Due to its high transmission efficiency and low cost, packet switching will gradually replace the current circuit switching network. Various access networks (wireless, xDSL, Cable, optical access, etc.) will become a unified packet switching backbone network. In the future network architecture, the No. 7 signaling system will coexist with the IP network for a certain period of time, and it will play an important role in the IP network. In the trend of various network convergence, an obvious change is that the powerful functions of the circuit switch in the past are being continuously decomposed, and the interface is being standardized. The MGCP (Media Gateway Control Protocol) protocol has unified the interface between the IP network and the PSTN network, and the IPST (Internet Protocol Standard Transmit) protocol has a unified implementation method for circuit switching signaling in the IN network. This makes the distributed call processing structure in the IP telephone field possible, laying a solid network foundation for the current and future applications of the IP telephone system. The IP telephone system developed in this stage is called the "unified stage" IP telephone system. ——The most notable feature of IP telephony in the unified stage is that various IP telephony devices are compatible with each other, extending the circuit switching concept to the entire network, and operators can exchange without obstacles on the entire IP network. Protocols represented by MGCP and IPST unify the specifications of H.323 and iNOW! protocols, and standardize the interface signaling between IP networks and PSTN networks (IPST protocol). We know that IP telephony systems are generally divided into three layers, namely: connection layer, control layer and service management layer. ——The connection layer is responsible for establishing and implementing the connection of the physical layer. While completing the information exchange between the IP network and the PSTN network, it is responsible for transmitting the encoded voice signal to the control system. The control layer completes the call request connection. The relevant protocols of this layer are: H.323, H.GCP and SIP, etc. The main task of these protocols is to complete the encapsulation of voice signals and establish appropriate bearer connections. The service management layer mainly completes the service control of operators, such as user management, billing, settlement and user authorization. This layer must support the A interface (intelligent network interface), so this layer is also closely related to the H.GCP (MGCP) protocol. ——MGCP is a protocol standard defined for all gateways between PSTN and IP networks. The most typical applications are IP phone gateways and dial-up access servers. Therefore, the future structures of IP phone gateways and dial-up access servers are very similar. The difference is that the IP phone gateway completes the bundling of PSTN voice resources and RTP session resources, while the dial-up access server completes the bundling of PSTN voice resources and IP sessions. Therefore, future dial-up access servers will be able to automatically identify IP access and IP phone (or fax) access, and dynamically implement channel allocation and resource bundling on demand.323 V2 "Quick Setup" 5 End-to-end message record code can be passed between gateways and gateways 6 Gateways can use the settlement system and the call confirmation from the settlement system operator for authentication 7 Call detail records can be generated and transmitted to the settlement center in real time 8 To sell gateways and gateways that meet iNOW! 2.0 version, you must first use the iNOW! authority and successfully complete the iNOW! organization's certification process 9 For calls to the settlement center, the iNOW! platform provides compatibility through the settlement center and terminal signals 10 Provides intercommunication between gateways and settlement systems 11 Can transmit gateway routing call signaling and terminal routing call signaling 12 For at least 24 hours, the local gateway clock can be synchronized with an accurate and reliable time source 13 The terminal source code can be generated in the settlement system CDR using the following algorithm: H.323+1000 Q.931 14 The gateway supports T.38 fax protocol. Mandatory support for TCP/UDP/IP and V.21, V.27 V.17 15 For settlement system calls, the gateway will ensure the integrity of the information. In view of the current lack of universal international standards, Lucent, Itexc and Vocaltec jointly developed the IP phone industry standard iNow! protocol in January 1999. The protocol mainly includes five aspects: ——. Gateway to Gateway intercommunication requirements ——. Gatekeeper to Gatekeeper intercommunication requirements ——. Gatekeeper to settlement center intercommunication requirements ——. Phone to Phone service requirements ——. FAX to FAX service requirements —— While meeting the above requirements, the information exchange process must be completed under the control of the settlement center. IP phone operators in different regions can complete various authentication and exchange tasks through the settlement center. In the iNow! protocol, the connection and disconnection process is also strictly defined, thus ensuring that the products of different manufacturers can be compatible with each other in the connection and disconnection process. While stipulating the algorithm and information exchange specifications, the iNow! protocol also stipulates various detailed message formats. In this way, when different manufacturers apply the protocol, there will be no objection, so that the products of various IP phone manufacturers can be compatible with each other. ——However, good wishes are not equal to reality. Since its birth, the INOW! protocol has had many problems. First of all, it is a supplement to the H.323 protocol. It does not define a new protocol and is still limited to the scope of the H.323 protocol. The incompleteness of the H.323 protocol in the network layer and the lack of guarantee for transmission cannot be solved by the iNOW! protocol. Secondly, the iNOW! protocol is an industry standard and has not yet been supported by the ITU. Therefore, although many manufacturers support the iNOW! protocol for more than a year after its launch, the products of manufacturers claiming to support the protocol are not compatible with each other. ——After more than half a year of trial operation, the Chinese IP phone system, under the organization of the Ministry of Information Industry, combined with my country's network conditions and the problems reflected by users, has formulated China's IP phone compatibility standards and performance requirements in response to the problems existing in the current IP phone system. With the cooperation of relevant units, the network access test and certification of IP phone equipment have been carried out, and good results have been achieved. 3 Technology Convergence - Network development is evolving towards broadband and intelligence. The current convergence of circuit switching and packet switching is the inevitable result of this trend. Due to its high transmission efficiency and low cost, packet switching will gradually replace the current circuit switching network. Various access networks (wireless, xDSL, Cable, optical access, etc.) will become a unified packet switching backbone network. In the future network architecture, the No. 7 signaling system will coexist with the IP network for a certain period of time, and it will play an important role in the IP network. In the trend of various network convergence, an obvious change is that the powerful functions of the circuit switch in the past are being continuously decomposed, and the interface is being standardized. The MGCP (Media Gateway Control Protocol) protocol has unified the interface between the IP network and the PSTN network, and the IPST (Internet Protocol Standard Transmit) protocol has a unified implementation method for circuit switching signaling in the IN network. This makes the distributed call processing structure in the IP telephone field possible, laying a solid network foundation for the current and future applications of the IP telephone system. The IP telephone system developed in this stage is called the "unified stage" IP telephone system. ——The most notable feature of IP telephony in the unified stage is that various IP telephony devices are compatible with each other, extending the circuit switching concept to the entire network, and operators can exchange without obstacles on the entire IP network. Protocols represented by MGCP and IPST unify the specifications of H.323 and iNOW! protocols, and standardize the interface signaling between IP networks and PSTN networks (IPST protocol). We know that IP telephony systems are generally divided into three layers, namely: connection layer, control layer and service management layer. ——The connection layer is responsible for establishing and implementing the connection of the physical layer. While completing the information exchange between the IP network and the PSTN network, it is responsible for transmitting the encoded voice signal to the control system. The control layer completes the call request connection. The relevant protocols of this layer are: H.323, H.GCP and SIP, etc. The main task of these protocols is to complete the encapsulation of voice signals and establish appropriate bearer connections. The service management layer mainly completes the service control of operators, such as user management, billing, settlement and user authorization. This layer must support the A interface (intelligent network interface), so this layer is also closely related to the H.GCP (MGCP) protocol. ——MGCP is a protocol standard defined for all gateways between PSTN and IP networks. The most typical applications are IP phone gateways and dial-up access servers. Therefore, the future structures of IP phone gateways and dial-up access servers are very similar. The difference is that the IP phone gateway completes the bundling of PSTN voice resources and RTP session resources, while the dial-up access server completes the bundling of PSTN voice resources and IP sessions. Therefore, future dial-up access servers will be able to automatically identify IP access and IP phone (or fax) access, and dynamically implement channel allocation and resource bundling on demand.FAX to FAX service requirements - While meeting the above requirements, the information exchange process must be completed under the control of the settlement center. IP phone operators in different regions can complete various authentication and exchange tasks through the settlement center. In the iNow! protocol, the connection and disconnection process is also strictly defined, thus ensuring that the products of different manufacturers can be compatible with each other in the connection and disconnection process. The iNow! protocol specifies various detailed message formats while specifying the algorithm and information exchange specifications. In this way, when different manufacturers apply the protocol, there will be no objection, so that the products of various IP phone manufacturers can be compatible with each other. - However, good wishes are not equal to reality. Since its birth, the INOW! protocol has had many problems. First, it is a supplement to the H.323 protocol. It does not define a new protocol and is still limited to the scope of the H.323 protocol. The incompleteness of the H.323 protocol in the network layer and the lack of guarantee for transmission cannot be solved by the iNOW! protocol. Secondly, the iNOW! protocol is an industry standard and has not yet been supported by the I TU. Therefore, although the iNOW! protocol has been launched for more than a year and many manufacturers support it, the products of manufacturers claiming to support it are not compatible with each other. ——After more than half a year of trial operation, the Chinese IP telephone system, under the organization of the Ministry of Information Industry, combined with my country's network conditions and the problems reflected by users, formulated China's IP telephone compatibility standards and performance requirements in response to the problems existing in the current IP telephone system. With the cooperation of relevant units, the network access test and certification of IP telephone equipment were carried out, and good results were achieved. 3 Technology Convergence——The development of the network is evolving towards broadband and intelligentization. The current mutual integration of circuit switching and packet switching is the inevitable result of this trend. Due to the high transmission efficiency and low cost of packet switching, it will gradually replace the current circuit switching network. Multiple access networks (wireless, xDSL, Cable, optical access, etc.) will become a unified packet switching backbone network. In the future network architecture, the No. 7 signaling system will coexist with the IP network for a certain period of time, and it will play an important role in the IP network. In the trend of various network convergence, an obvious change is that the powerful functions of the circuit switch in the past are constantly being decomposed, and the interface is being standardized. The MGCP (Media Gateway Control Protocol) protocol makes the interface between the IP network and the PSTN network have a unified specification, and the IPST (Internet Protocol Standard Transmit) protocol makes the circuit switching signaling in the IN network have a unified implementation method. This makes the distributed call processing structure in the field of IP telephony possible, laying a solid network foundation for the current and future applications of IP telephony systems. The IP telephony system developed in this stage is called the "unified stage" IP telephony system. ——The most notable feature of the unified stage IP telephony is that various IP telephony devices are compatible with each other, the circuit switching concept is extended to the entire network, and operators can exchange without obstacles on the entire IP network. The protocols represented by MGCP and IPST unify the specifications of the H.323 and iNOW! protocols, and standardize the interface signaling between the IP network and the PSTN network (IPST protocol). We know that the IP telephony system is generally divided into three layers, namely: connection layer, control layer and service management layer. ——The connection layer is responsible for establishing and implementing the connection of the physical layer. While completing the information exchange between the IP network and the PSTN network, it is responsible for transmitting the encoded voice signal to the control system. The control layer completes the call request connection. The related protocols of this layer are: H.323, H.GCP and SIP, etc. The main task of these protocols is to complete the encapsulation of voice signals and establish appropriate bearer connections. The service management layer mainly completes the service control of operators, such as user management, billing, settlement and user authorization. This layer must support the A interface (intelligent network interface), so this layer is also closely related to the H.GCP (MGCP) protocol. ——MGCP is a protocol standard defined for all kinds of gateways between PSTN and IP networks. The most typical applications are IP phone gateways and dial-up access servers. Therefore, the future structure of IP phone gateways and dial-up access servers has great similarities. The difference is that the IP phone gateway completes the bundling of PSTN voice resources and RTP session resources, while the dial-up access server completes the bundling of PSTN voice resources and IP sessions. Therefore, the future dial-up access server will be able to automatically identify IP access and IP phone (or fax) access, and dynamically realize channel allocation and resource bundling on demand.FAX to FAX service requirements - While meeting the above requirements, the information exchange process must be completed under the control of the settlement center. IP phone operators in different regions can complete various authentication and exchange tasks through the settlement center. In the iNow! protocol, the connection and disconnection process is also strictly defined, thus ensuring that the products of different manufacturers can be compatible with each other in the connection and disconnection process. The iNow! protocol specifies various detailed message formats while specifying the algorithm and information exchange specifications. In this way, when different manufacturers apply the protocol, there will be no objection, so that the products of various IP phone manufacturers can be compatible with each other. - However, good wishes are not equal to reality. Since its birth, the INOW! protocol has had many problems. First, it is a supplement to the H.323 protocol. It does not define a new protocol and is still limited to the scope of the H.323 protocol. The incompleteness of the H.323 protocol in the network layer and the lack of guarantee for transmission cannot be solved by the iNOW! protocol. Secondly, the iNOW! protocol is an industry standard and has not yet been supported by the I TU. Therefore, although the iNOW! protocol has been launched for more than a year and many manufacturers support it, the products of manufacturers claiming to support it are not compatible with each other. ——After more than half a year of trial operation, the Chinese IP telephone system, under the organization of the Ministry of Information Industry, combined with my country's network conditions and the problems reflected by users, formulated China's IP telephone compatibility standards and performance requirements in response to the problems existing in the current IP telephone system. With the cooperation of relevant units, the network access test and certification of IP telephone equipment were carried out, and good results were achieved. 3 Technology Convergence——The development of the network is evolving towards broadband and intelligentization. The current mutual integration of circuit switching and packet switching is the inevitable result of this trend. Due to the high transmission efficiency and low cost of packet switching, it will gradually replace the current circuit switching network. Multiple access networks (wireless, xDSL, Cable, optical access, etc.) will become a unified packet switching backbone network. In the future network architecture, the No. 7 signaling system will coexist with the IP network for a certain period of time, and it will play an important role in the IP network. In the trend of various network convergence, an obvious change is that the powerful functions of the circuit switch in the past are constantly being decomposed, and the interface is being standardized. The MGCP (Media Gateway Control Protocol) protocol makes the interface between the IP network and the PSTN network have a unified specification, and the IPST (Internet Protocol Standard Transmit) protocol makes the circuit switching signaling in the IN network have a unified implementation method. This makes the distributed call processing structure in the field of IP telephony possible, laying a solid network foundation for the current and future applications of IP telephony systems. The IP telephony system developed in this stage is called the "unified stage" IP telephony system. ——The most notable feature of the unified stage IP telephony is that various IP telephony devices are compatible with each other, the circuit switching concept is extended to the entire network, and operators can exchange without obstacles on the entire IP network. The protocols represented by MGCP and IPST unify the specifications of the H.323 and iNOW! protocols, and standardize the interface signaling between the IP network and the PSTN network (IPST protocol). We know that the IP telephony system is generally divided into three layers, namely: connection layer, control layer and service management layer. ——The connection layer is responsible for establishing and implementing the connection of the physical layer. While completing the information exchange between the IP network and the PSTN network, it is responsible for transmitting the encoded voice signal to the control system. The control layer completes the call request connection. The related protocols of this layer are: H.323, H.GCP and SIP, etc. The main task of these protocols is to complete the encapsulation of voice signals and establish appropriate bearer connections. The service management layer mainly completes the service control of operators, such as user management, billing, settlement and user authorization. This layer must support the A interface (intelligent network interface), so this layer is also closely related to the H.GCP (MGCP) protocol. ——MGCP is a protocol standard defined for all kinds of gateways between PSTN and IP networks. The most typical applications are IP phone gateways and dial-up access servers. Therefore, the future structure of IP phone gateways and dial-up access servers has great similarities. The difference is that the IP phone gateway completes the bundling of PSTN voice resources and RTP session resources, while the dial-up access server completes the bundling of PSTN voice resources and IP sessions. Therefore, the future dial-up access server will be able to automatically identify IP access and IP phone (or fax) access, and dynamically realize channel allocation and resource bundling on demand.GCP and SIP, etc., the main task of these protocols is to complete the encapsulation of voice signals and establish appropriate bearer connections. The business management layer mainly completes the business control of operators, such as user management, billing, settlement and user authorization. This layer must support the A interface (intelligent network interface), so this layer is also closely related to the H.GCP (MGCP) protocol. ——MGCP is a protocol standard defined for all kinds of gateways between PSTN and IP networks. The most typical applications are IP phone gateways and dial-up access servers. Therefore, the future structure of IP phone gateways and dial-up access servers has great similarities. The difference is that the IP phone gateway completes the bundling of PSTN voice resources and RTP session resources, while the dial-up access server completes the bundling of PSTN voice resources and IP sessions. Therefore, the future dial-up access server will be able to automatically identify IP access and IP phone (or fax) access, and dynamically realize channel allocation and resource bundling on demand.GCP and SIP, etc., the main task of these protocols is to complete the encapsulation of voice signals and establish appropriate bearer connections. The business management layer mainly completes the business control of operators, such as user management, billing, settlement and user authorization. This layer must support the A interface (intelligent network interface), so this layer is also closely related to the H.GCP (MGCP) protocol. ——MGCP is a protocol standard defined for all kinds of gateways between PSTN and IP networks. The most typical applications are IP phone gateways and dial-up access servers. Therefore, the future structure of IP phone gateways and dial-up access servers has great similarities. The difference is that the IP phone gateway completes the bundling of PSTN voice resources and RTP session resources, while the dial-up access server completes the bundling of PSTN voice resources and IP sessions. Therefore, the future dial-up access server will be able to automatically identify IP access and IP phone (or fax) access, and dynamically realize channel allocation and resource bundling on demand.
提升卡
变色卡
千斤顶