Huawei AR G3 Series Enterprise Routers V200R002C01. Voice Feature White Paper. Issue 01. Date HUAWEI TECHNOLOGIES CO., LTD.

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1 V200R002C01 Issue 01 Date HUAWEI TECHNOLOGIES CO., LTD.

2 2012. All rights reserved. No part of this document may be reproduced or transmitted in any form or by any means without prior written consent of Huawei Technologies Co., Ltd. Trademarks and Permissions and other Huawei trademarks are trademarks of Huawei Technologies Co., Ltd. All other trademarks and trade names mentioned in this document are the property of their respective holders. Notice The purchased products, services and features are stipulated by the contract made between Huawei and the customer. All or part of the products, services and features described in this document may not be within the purchase scope or the usage scope. Unless otherwise specified in the contract, all statements, information, and recommendations in this document are provided "AS IS" without warranties, guarantees or representations of any kind, either express or implied. The information in this document is subject to change without notice. Every effort has been made in the preparation of this document to ensure accuracy of the contents, but all statements, information, and recommendations in this document do not constitute a warranty of any kind, express or implied. Huawei Technologies Co., Ltd. Address: Website: Huawei Industrial Base Bantian, Longgang Shenzhen People's Republic of China i

3 Keywords Abstract Acronyms AR IP PBX, VoIP, SIP This document describes voice features supported by the AR G3 series enterprise routers. Acronym AR IMS VoIP SIP IP PBX AG FXO FXS SIPUE POTS CDR Full Name Access Router IP Multimedia Subsystem Voice over Internet Protocol Session Initiation Protocol IP Private Branch exchange access gateway Foreign Exchange Office Foreign Exchange Station Sip user agent Plain Old Telephone Service Call Detail Record ii

4 Contents Contents 1 SIP AG Overview IP PBX Overview SIP SIP Structure SIP Messages User Registration Process VoIP (SIP) MO Process VoIP (SIP) MT Process Call Release Process FoIP (FAX over IP) FoIP Overview FoIP Transmission Mode Low-Speed Fax and High-Speed Fax MoIP (Modem over Internet Protocol) MoIP Connection Type Basic IP PBX Services FXS Access FXO Access FXO as the Calling Party FXO as the Called Party E1/PRI Access ISDN Signaling Q.931 Call Instances IP PBX Access to the PSTN SIP UE Access SIP UE Registration Process of a Call Between SIP UEs Access to the IMS Through SIP Access to the IMS Through SIP with Registration Calling and Called Parties Access to the IMS Through SIP PBX Communication Through SIP Fax/Modem iii

5 Contents 4.8 Number Change Intelligent Routing CDR IVR Service Dialing an Extension Number Triggering a Simultaneous Ringing Service Triggering a Sequential Ringing Service Triggering the Line Selection Service Triggering Call Queuing BEST Function Description Overview Description Power Outage Survival Call Manager System Advantages Deployment Other Services Supported by AR G3 Series Routers SIP NAT Traversal Overview SIP NAT Traversal Principles AR SIP NAT Traversal Solution (SBC Solution) AR Voice Solution AR Inter-Branch Voice Communication Solution Centralized Call Control Model Distributed Call Control Model Hybrid Call Control Model AR Connecting to an IMS/NGN Network as AG Market Positioning and Intended Customers Network Topology and Solution iv

6 1 SIP AG Overview 1 SIP AG Overview Definition SIP AG is a voice access gateway (AG) device based on the Session Initiation Protocol (SIP). It is configured between the public switched telephone network (PSTN) and IP multimedia subsystem (IMS), and is mainly used to convert signals between analog and digital forms. Purpose The emergence of the packet-switched network leads to revolutionary changes to the telephony system. Many new technologies are also developed for this new bearer network. The Voice over IP (VoIP) service enables IP networks to carry voice services (such as traditional telephone services). In addition, the new IMS provides powerful support for VoIP application. An IMS network is a standard next-generation carrier network that provides mobile or fixed-line multimedia services. It supports traditional packet switched and circuit switched telephony systems. Compared with the traditional PSTN, the IP bearer network features higher resource utilization and shared lines for calls. Currently, the VoIP technology has been put into commercial use. Traditional circuit switched telephone networks have been developing for years and a large number of devices are still in service now. Replacement of existing telephone networks with IP bearer networks can cost too much. SIP AGs can be used to connect the voice network and data network cost-effectively. Huawei AR G3 series routers can function as SIP AGs to connect the PSTN network and IP data network. As shown in Figure 1-1, ARs serve as SIP AGs to integrate the voice network and the IP data network based on the SIP protocol. 1

7 1 SIP AG Overview Figure 1-1 Typical networking where the SIP AG functions as the voice gateway IMS SIP IP Network SIP SIPAG SIPAG POTS Modem FAX POTS Benefits Although the VoIP service shares bandwidth with other services on the Ethernet, proper network planning and quality of service (QoS) configuration ensure high quality of enterprise voice services. The use of AR G3 series routers as the SIP AGs to provide VoIP services brings the following benefits to enterprises: Low costs: Traditional calls and fax services use circuit switched mode and occupy communications lines exclusively. Long distance call and fax services are expensive. The VoIP service with SIP AGs serving as voice gateways can reduce the communication costs for enterprises. High call quality: SIP AGs ensure call completion rate, voice quality, and service types by configuring QoS. Smooth upgrade/capacity expansion: A VoIP system is compatible with the existing telephony systems and office platforms, and the service capacity can be increased when the enterprise scale expands. 2

8 2 IP PBX Overview 2 IP PBX Overview Definition A private branch exchange (PBX) is a telephone exchange that serves a particular business or office. An IP-based PBX (IP PBX) is the server used on the internal IP telephone network of an enterprise for call control and configuration management. Purpose VoIP technology converts analog voice signals to digital signals, encapsulates digital signals in IP data packets, and transmits IP data packets on the IP data network in real time. By using the Internet, VoIP provides more and better services than the traditional PBX. For example, VoIP can transmit voice, fax, video, and data services on the IP network with low costs. VoIP provides unified messaging, virtual phone, virtual voice/fax , number query, Internet call center, Internet call management, video conference, ecommerce, fax S/F, and store and forward of other information. Traditional PBXs exchange calls inside an enterprise and between the enterprise network and the PSTN. One PBX integrates the telephone, fax, and modem functions. PBXs are widely used in enterprise offices and greatly enhance enterprise efficiency. However, traditional PBXs cannot meet the requirements for computer telephony integration (CTI) and VoIP. In addition, these PBXs are expensive and do not use standard and open platforms, making the interconnection between PBXs of different vendors difficult. IP PBXs provide local exchange and IP user access functions. AR G3 series routers can function as IP PBXs to integrate voice communications into enterprise data networks so that an integrated voice and data network is established to connect offices and employees around the world. AR G3 series routers can also connect to traditional POTS phones through voice gateways, making voice networks scalable. Benefits Compared with traditional PBXs, AR G3 series routers provide the following benefits to enterprises when functioning as IP PBXs: Low construction costs: IP PBXs can be deployed on the existing IP network of an enterprise, saving the costs on constructing and maintaining multiple networks. Low management costs: IP PBXs simplify the process to add, replace, or remove a terminal. For example, an IP phone can be moved by simply connecting the phone to another network interface. Unlike a traditional PBX, an IP PBX does not require additional configuration for the moved IP phone. 3

9 2 IP PBX Overview High work efficiency: IP PBXs can rapidly integrate multiple related systems so that enterprises do not need to deploy single-function systems. Highly reliable communication: IP PBXs ensure normal provisioning of internal services when egress transmission channels of an enterprise fail. Flexible solution: IP PBXs can be deployed in distributed networking to meet requirements of IP-based voice and data communication. This distributed networking allows enterprises to construct enterprise networks cross the cities, provinces, and even countries. Self-service maintenance: IP PBXs provide an individual service management system for enterprises and helps reduce carriers' maintenance costs. For example, an IP PBX provides extension number selection, short number self-planning, toll call right modification, number portability, and internal line addition. Customized development: To improve work and communication efficiency, IP PBXs can integrate the enterprise OA process, enterprise address book, and the click-to-dial function based on enterprises' needs. Featured solution: IP PBXs provide featured application solutions such as hotel telephone service and voice record. Abundant ICT applications: IP PBXs can be integrated with the UC system to enrich the ICT applications of enterprises. High resource utilization efficiency: An IP PBX on a local area network (LAN) manages the computer network and telephone network effectively based on actual conditions and implements resource sharing. 4

10 2 IP PBX Overview Figure 2-1 shows a typical IP PBX networking. Figure 2-1 Typical IP PBX networking SIP Headquarters Newly built area FAX POTS SIP SIP SIP FAX POTS IAD Branch IP PBX POTS POTS IAD (Distributed call control) VOICE FAX IAD IP PBX POTS New built&migrated area VOICE POTS FAX SIP SIP AG VOICE TDM PBX Migrated area Branch (Centralized call control) POTS SIP AG VOICE SIP FAX IAD SIP POTS POTS FAX FAX POTS SIP Access switch Aggregation switch FXS (RJ11 telephone line) E1 Ethernet 5

11 3 SIP 3 SIP The Session Initiation Protocol (SIP) is an application-layer protocol used to create, modify, and terminate multimedia sessions. Multimedia sessions are used for applications such as multimedia conferences, remote education, and Internet calls. SIP can be used to initiate a session and to invite members to the session established in other ways (for example, multi-party conference). SIP transparently supports name mapping and redirection services to implement ISDN, intelligent network (IN), and personal mobility services. Once a session is set up, media streams are directly transmitted at the bearer layer using the Real-Time Transport Protocol (RTP). SIP, proposed by the Internet Engineering Task Force (IETF) in 1999, is a signaling protocol implementing real-time communication on an IP network. SIP supports the following functions for establishing and terminating multimedia communications: 1. User location: determines the end system used for communication. 2. User availability: determines the media and media parameters to be used in communication. 3. User availability: determines the willingness of the called party to engage in communication. 4. Session setup: establishes session parameters for the called and calling parties. 5. Session management: includes transfer and termination of sessions. 6

12 3 SIP SIP is designed as part of the overall IETF multimedia data and control architecture, as shown in Figure 3-1. Figure 3-1 IETF multimedia data and control architecture H.323 SIP RTSP RSVP RTCP H.263 etc. RTP TCP UDP IP PPP AAL3/4 AAL5 PPP Sonet ATM Ethernet V.34 SIP is used with other protocols. For example, the Resource Reservation Protocol (RSVP) reserves network resources, the Real-Time Transport Protocol (RTP) transports real-time data and provides QoS feedback, the Real-time Stream Protocol (RTSP) controls delivery of streaming media, the Session Announcement Protocol (SAP) advertises multimedia sessions in multicast mode, and the Session Description Protocol (SDP) described multimedia sessions. However, the functionality and operation of SIP do not depend on any of these protocols. SIP can also be used with other session setup and signaling protocols. In that mode, an end system uses SIP to determine an appropriate end system address and protocol from a given address that is protocol-independent. For example, SIP can be used to determine whether the local end can communicate with the peer through H.323. If so, SIP obtains the H.245 gateway address and user address, and then uses H to establish the call. In another example, SIP can be used to determine whether the called party is connected through the PSTN and specify the called number. It is recommended that the Internet-to-PSTN gateway be used to establish the call. SIP fundamentally changes the communications service provisioning mode and the consumption habits of communications users. Services such as video and audio calls, messaging, web, , synchronous browse, and conference services are integrated, bringing innovations to the telecommunication industry. SIP has the following advantages as a control layer protocol: 1. Based on open Internet standards, SIP is suitable for integration of voice and data services, and can implement call control across media and devices. In addition, SIP supports various media formats and can dynamically add or delete media streams, so that various services can be deployed easily. 2. Extends the intelligent network to service side and end systems, reducing the network burden and facilitating service development 3. Supports application-layer portability functions, including dynamic registration, location management, and redirection management. 7

13 3 SIP 4. Provides presence/fork/subscription features, which facilitate new service development 5. It is simple and scalable. 3.1 SIP Structure The SIP protocol logically consists of the following elements: User agent: also called the SIP terminal. It is the end user of the SIP system and is defined as an application in RFC3261. Based on roles in a session, user agents can be classified into the user agent client (UAC) and user agent server (UAS). The UAC initiates a call request, and the UAS responds to the call request. SIP proxy server: an intermediate device. It can function as a server to parse user names and function as a client agent to initiate a call request to the next-hop server, which then determines the next hop address. SIP register server: an important part in the SIP system. It receives user registration information and maintains the information into the address database. Location server: stores and returns user address information. It obtains address information from the register server or other databases, and then uploads the address registration information to the location server. Redirect server: determines paths of call. After obtaining the next hop address, this server requests the previous-hop user to initiate a request directly to the next hop. At the same time, this server stops controlling the call. For example, if Bob wants to call Lara and this request is sent to the redirect server. The redirect server obtains the address of Lara and returns the address to Bob. Then, Bob can resend the session invitation to the address. Actually, functions of the preceding SIP servers are provided by one server. They are only identified logically. The following figures show interactions between the SIP components. Interaction between the UA, register server, and location server: registration This is I am at is at 2 UA 4 Register Server OK. The registration completes. 3 I have made a record. Location Server 8

14 3 SIP Interaction between the UA, proxy server, and location server: call routing Location Server UA1 I want to chat with UA Hi. Proxy Server 2 Where is UA2? 3 Address of UA2 UA1 is asking for you. 4 Is it Tom? This is Jerry. 5 Hello. 7 UA2 Interaction between the UA, redirect server, and location server: call redirection Redirect Server Where is UA2? 2 Location Server I want to chat with UA This is the address of UA2. 3 Address of UA2 Hello. Is it Tom? This is Jerry. 5 UA1 6 Hi, Jerry. 7 RTP UA2 9

15 3 SIP 3.2 SIP Messages SIP messages are encoded in text format. There are two types of SIP messages: request and response. RFC 3261 defines the following SIP request messages: INVITE: invites a user to a call. ACK: acknowledges a response message. OPTIONS: negotiates communication capabilities with the peer. BYE: terminates a session. CANCEL: cancels a session establishment. REGISTER: registers user location information with a registrar server. SIP response messages are sent in response to request messages, informing calling parties of call or registration results. Status codes identify the types of response messages. A status code is a 3-digit integer. The leftmost digit indicates the response message type, and the other two digits provide additional information, such as how a received request message is processed. RFC 3261 defines the following status codes: 100 to 199: provisional. A request has been received and is being processed. 200 to 299: success. A request has been successfully processed. 300 to 399: redirection. Further action needs to be taken to complete the request. 400 to 499: client error. A request contains incorrect syntax or cannot be processed by the server. 500 to 599: server error. The server failed to process a valid request. 600 to 699: global failure. A request cannot be processed by any servers. 3.3 User Registration Process Before a SIP user initiates a call, the user must register user information (for example, mapping between the domain name and the IP address) on the home network. The registration process can be implemented in non-authentication mode or authentication mode. After the system is powered on or a user is added, the user registration process starts. 10

16 3 SIP Registration Process in Non-authentication Mode Figure 3-2 Registration process in non-authentication mode SIP AG IMS Core Register Response 200 As shown in Figure 3-2, the SIP AG sends a Register message to the IMS Core for each user. The Register message contains information such as the user identity. When receiving the Register message, the IMS Core checks whether the user is configured in the IMS. If the user is configured, the IMS Core returns a Response-200 message to the SIP AG. If the user is not configured, the IMS Core returns a Response-403 message to reject the registration. The AR supports individual registration and group registration. In individual registration mode, users register on the IMS core individually through SIP AT0 trunks. In group registration mode, multiple users can register on the IMS core together, which reduces the number of register messages and avoids registration storms. Registration Process in Authentication Mode Figure 3-3 Registration process in authentication mode SIP AG IMS Core Register Response 401/407 Register Response

17 3 SIP As shown in Figure 3-3, the SIP AG sends a Register message to the IMS Core for each user. The Register message contains information such as the user identity. When receiving the Register message, the IMS Core queries and learns that this SIP AG registration requires authentication. Then, the IMS Core returns Response-401/407, which contains information such as the key and encryption method. The SIP AG encrypts the user name and password with the key, and sends them in a Register message to the IMS Core. The IMS Core decrypts the Register message and checks whether the user name and password are correct. If they are correct, the IMS Core returns Response-200. The AR supports the DIGEST MD5, DIGEST MD5-SESS, and AkAv1-MD5 algorithms for authentication and encryption. 3.4 VoIP (SIP) MO Process Figure 3-4 shows the VoIP (SIP) call process on the calling party side. Figure 3-4 VoIP MO process IMS Network USER1 AG P-CSCF-O Caller offhook dialtone 1st digit Dialtone stopped 2st digit 3st digit P1 P2 P3 P4 P5 P6 D1:1NVITE(SDP) D2:100 Trying D3:180 Ringing D4:200 OK D5:ACK conversation 200(callee offhook) 12

18 3 SIP P1: The AG receives the pick-up message from the calling party and plays the dial tone for the calling party. P2: When receiving the first dial number, the AG stops the dial tone and matches the number with the digitmaps. P3: After receiving N numbers, the AG detects that the numbers match a digitmap. Then the AG constructs an Invite message and sends it to the P-CSCF. P4: When receiving 100 Trying, the AG learns that the peer has received the Invite message. Then the AG stops the process of retransmitting the Invite message. P5: The AG receives 180 Ringing, indicating that the phone of the called party rings. The AG plays the RBT for the calling party. P6: The AG receives 200 OK message, indicating that the called party has picked up the phone. Then the AG stops playing the RBT and changes the flow mode to bidirectional. The AG constructs an ACK message to the P-CSCF. Besides normal calls, there are other scenarios. When the calling party initiates a call, the P-CSCF performs either of the following operations: If the data about the calling party exists but is not registered, the P-CSCF rejects the call from the calling party and returns message 403. If there is no data about the calling party, the P-CSCF rejects the call from the calling party and returns message VoIP (SIP) MT Process Figure 3-5 shows the VoIP (SIP) call process on the called party side. Figure 3-5 VoIP (SIP) MT process IMS Network USER1 AG P-CSCF-T ring Callee offhook P1 P2 P3 D1:INVITE(SDP) D2:100 Trying D3:180 Ringing D4:200 OK D5:ACK conversation 13

19 3 SIP P1: After receiving an INVITE message from the P-CSCF, the AG constructs a 100 Trying message and sends it to the P-CSCF. The AG locates the called party according to the P-Called-Party-ID header field, RequestURI, and TO header field carried in the INVITE message. If the TEL-URI field is used, the header fields can be not used. The AG can locate the called party according to the phone number in the TEL-URI field. Then the AG plays the ring tone to the called party. The AG constructs a 180 Ringing message and sends it to the P-CSCF, notifying that the phone of the called party is ringing. P2: After receiving the off-hook message from the called party, the AG stops ringing. In addition, the AG constructs a 200 OK message and sends it to the P-CSCF, notifying the called party has picked up the phone. P3: The AG receives an ACK message and the calling party and the called party talk with each other. Besides normal calls, there are other scenarios. When receiving the Invite message, the AG performs either of the following operations: If the data about the called party exists but is not registered, the AG rejects the call from the calling party and returns message 403 to P-CSCF. If there is no data about the called party, the AG rejects the call from the calling party and returns message 404 to P-CSCF. 3.6 Call Release Process Figure 3-6 shows the VoIP (SIP) call release process. Figure 3-6 Call release process IMS Network USER1 AG P-CSCF-O conversation onhook P1 P2 D1:BYE D2:200 OK 14

20 3 SIP P1: After receiving the onhook message of the user, the AG constructs a BYE message and sends it to the P-CSCF to release the DSP resource allocated to the user. P2: After receiving 200 OK from the P-CSCF, the AG releases the call. 3.7 FoIP (FAX over IP) FoIP Overview Traditional fax is sent and received through the PSTN. Fax services are widely used because various types of information can be easily transmitted at a high speed. The International Telegraph and Telephone Consultative Committee (CCITT) defines four fax machine standards, namely, G1, G2, G3, and G4 fax machines. G1: low-speed analog fax machines using analog frequency shift keying signals, and in black and white G2: medium-speed analog fax machines using analog phase shift keying signals in black and white; compressed frequency band at a transmission speed double that of G1 G3: high-speed digital fax machines using modulating signals in black and white at a transmission speed four times that of G1 G4: high-speed digital fax machines for the ISDN network at a speed of 64 kbit/s, using hybrid fax and telegraph terminals Due to the limitation of speeds or cables, G1, G2, and G4 fax machines are not widely used. Only G3 fax machines are commonly used for fax communication. G3 fax machines use a digital signal processing technology. Image signals are digitalized and compressed in a fax machine, converted to analog signals by a modem, and finally transmitted to a PSTN switch through common subscriber lines. Fax over IP (FoIP) sends and receives fax over the Internet. Compared with traditional fax, FoIP has the following benefits: Low fee: FoIP fully use the worldwide deployment and low communication fees of the Internet, and significantly reduces fax fees for enterprises. High security and QoS: FoIP uses advanced transmission and encryption technologies to improve the content definition and confidentiality, which are better than those of the traditional fax and IP telephone fax. High intelligence: FoIP automatically resends fax in a specified period and returns success or failure information to the user's box FoIP Transmission Mode FoIP supports two transmission modes (pass-through and T.38) and two switching modes (auto-switch and initiated negotiation switch). That is, four fax modes are available: auto-switch pass-through, auto-switch T.38, negotiation pass-through, and negotiation T.38. Auto-switch: The AG detects fax signals and selects the transparent or T.38 mode based on the configuration. In this case, the AG does not need to send any signal to the peer end. Initiated negotiation: The AG detects fax signals, and then sends a REINVITE message carrying negotiation parameters to negotiate the codec mode with the peer based on the configuration. 15

21 3 SIP In pass-through fax mode, fax data from the PSTN is modulated and then forwarded over an end-to-end (E2E) voice tunnel on the IP network. The AG functions as the gateway between the PSTN and the IP network and does not participate in modulation or demodulation. The AG is used as the gateway and fax machine to forward voice flows. Fax can be transmitted using pre-configured voice codes. Alternatively, the gateway automatically switches to the high speed coding mode of G.711. Compression loss of fax signals is relatively large when the G.729 protocol is used, and fax signals may not be demodulated correctly at the peer end. Therefore, the G.711 protocol, which causes less compression loss, is usually used for fax pass-through. Figure 3-7 shows the data forwarding process of fax pass-through. Figure 3-7 Data forwarding process of pass-through Fax analog data Gateway Analog data passes a VoIP tunnel at a rate of 64 kbit/s. Fax analog data Gateway FAX IP network FAX G.711 coding 64 kbit/s T.30 signaling G.711 coding 64 kbit/s During a T.38 fax call, the sending gateway demodulates a T.30 fax sent from the PSTN. The demodulated fax data is encapsulated in datagrams and sent to the receiver across the IP network. The receiver gateway modulates the datagrams into T.30 fax data and sends the fax data to the receiver. Figure 3-8 shows the data forwarding process. Figure 3-8 Data forwarding process of T.38 fax relay Data packet transmission Fax analog data Gateway FAX DSP demodulation IP network Fax analog data Gateway FAX DSP demodulation T.30 signaling T.38 signaling T.30 signaling 16

22 3 SIP Low-Speed Fax and High-Speed Fax Differences between low-speed fax and high-speed fax are as follows: Standard: High-speed fax uses the V.8 data transmission process. Low-speed fax uses the fax process defined by T.30. In addition, some low-speed fax terminals may use earlier standards. Rate range: Rate supported by high-speed fax ranges from 2.4 kbit/s to 33.6 kbit/s and that supported by low-speed fax ranges from 2.4 kbit/s to 14.4 kbit/s. Uplink transmission mode: High-speed fax uses only the pass-through mode. That is, fax is transmitted at a high rate from a modem to a gateway. Low-speed fax uses pass-through or T.38 mode. (T.38 mode does not support the rate of high-speed fax.) Error correction mode (ECM) requirement: High speed-fax must use the ECM mode, which is optional for low-speed fax. DSP EC requirement: High-speed fax requires DSP EC to be disabled (because it has an echo processing mechanism). Low-speed fax requires EC to be enabled (because it has no echo processing mechanism). 3.8 MoIP (Modem over Internet Protocol) A modulator demodulator (modem) is a device that is installed between a personal computer (PC) and a telephone to convert signals exchanged between them. A PC transmits digital signals to the modem port. The modem receives the signals and coverts (modulates) them into analog signals. Then, the signals are processed as normal voice signals in the telephony system. Signals sent from a telephone to a PC are processed reversely: Analog signals are transmitted over telephone lines to a modem, which converts the analog signals to digital signals, and sends the digital signals to a PC through the modem port. Modems are used for signal format conversion, including analog to digital conversion and digital to analog conversion. Other functions of a modem are as follows: Coverts signal frequency domain, such as the conversion from low-frequency signals to high-frequency signals and the modulation from digital baseband transmission to analog channel transmission. Extracts low-frequency signals from high-frequency signals. Demodulates digital baseband signals. Compresses network transmission data. Controls coding and error correction. MoIP provides modem services on the IP network or between the IP network and traditional PSTN network. 17

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