Operation Manual Voice Over (Voice Volume) Table of Contents Table of Contents Chapter 1 Voice Over... 1-1 1.1 Introduction to VoIP... 1-1 1.1.1 VoIP System... 1-1 1.1.2 Basic VoIP Call Flow... 1-2 1.1.3 VoIP Features... 1-2 1.2 Voice Function Configuration... 1-3 1.2.1 Configuration Procedure... 1-4 1.2.2 Voice Subscriber Line... 1-5 1.2.3 Voice Entity... 1-6 1.2.4 Voice Protocols... 1-6 1.2.5 Dial Plan... 1-8 1.2.6 Command View... 1-8 i
Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over Chapter 1 Voice Over 1.1 Introduction to VoIP VoIP makes it possible that voice services such as plain old telephone service (POTS) can be carried over the IP network. VoIP is implemented via voice packets. In VoIP, the voice gateway segments the voice signal into frames and stores them in voice packets to transmit. Currently, interworking between PSTN and Internet is implemented via VoIP gateways. Meanwhile, the PC-to-telephone, telephone-to-pc, and telephone-to-telephone technologies are mature and the call quality has been improved greatly. Therefore, VoIP can completely meet the commercial requirements. H.323 and session initiation protocol (SIP) are two common protocols used in VoIP. For details about H.323 and SIP, refer to section 1.2.4 Voice Protocols. 1.1.1 VoIP System For POTS, all functions from the call originator to the call receiver are implemented by the public switched telephone network (PSTN). VoIP is different from POTS. Gateway IP Gateway PSTN PSTN Telephone GK server /SIP server Telephone Figure 1-1 VoIP system In Figure 1-1, the VoIP gateway provides interfaces for communication between the IP network and PSTN/integrated services digital network (ISDN), users connect to the originating VoIP gateway through PSTN, the originating VoIP gateway converts analog signals into digital signals and compresses them into voice packets that can be transmitted over the IP network, and the IP network transmits the voice packets to the terminating VoIP gateway, which reduces the voice packets to recognizable analog signals and transmits them to the receiver. This is a complete telephone-to-telephone communication process. In practice, a gatekeeper (GK) server or SIP server may be applied in the VoIP system to implement the functions of routing and access control. 1-1
Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over 1.1.2 Basic VoIP Call Flow The following describes a basic VoIP call flow: 1) A user picks up a telephone and the modular voice card detects the user s off-hook action in real time 2) The modular voice card transmits the off-hook signal to the VoIP signal processing module on the VoIP gateway. 3) The VoIP signal processing module generates a dial tone. 4) The user hears the dial tone played by the session application and begins dialing before the dial tone expires. 5) The session application collects the digits dialed by the user. 6) The session application compares the collected digits with the match template in real time during digit collection. 7) After finding a match template for the called number, the originating VoIP gateway maps the number to the terminating VoIP gateway. 8) The originating VoIP gateway initiates a VoIP call to the terminating VoIP gateway over the IP network and establishes a logical channel for the call to send and receive voice data. 9) The terminating VoIP gateway receives the call from the IP network and seeks the destination telephone according to the match template. If the call is to be processed by a private branch exchange (PBX), the terminating VoIP gateway passes the call via PSTN signaling to the PBX for processing until the destination telephone is connected. When the calling party or the called party hangs up, the conversation ends. Note: During call connection, the calling party and called party negotiate the encoding/decoding method for the call and voice data is transferred through real time protocol (RTP) The RTP voice channel is used to transfer prompt signals during call connection and other signals suitable for in-band transmission across the IP network. When either party hangs up, the session application will end the conversation. 1.1.3 VoIP Features Silence compression The voice traffic to be transmitted can be reduced by automatically detecting the time ranges of silence in a conversation and stopping generating voice traffic within these time ranges. 1-2
Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over Comfortable noise Silent gaps during a call can be filled by comfortable background noise. QoS As the voice service is highly time-sensitive, the priority transmission of voice packets must be guaranteed. Some measures, such as priority queuing (PQ), custom queuing (CQ), weighted fair queuing (WFQ), class-based queuing (CBQ), and real time protocol (RTP) can be adopted on the sender side for this purpose. To ensure an adequate bandwidth for voice transmission, you can adopt the committed access rate (CAR) mechanism to implement traffic classification and policing. Fax over IP On basis of VoIP, the fax over IP (FoIP) system is responsible for setup of fax channels and the receiving and sending of fax data. FoIP implementation involves modulation and demodulation, fax protocol processing, and IP channel maintenance. One-stage dialing and two-stage dialing One-stage dialing and two-stage dialing can well fit in with the situation where there are differences when various PBXs transmit called numbers to the VoIP gateway. If a PBX sends the called number to the VoIP gateway, the VoIP adopts the one-stage dialing to connect the calling user. If the PBX does not send the called number to the VoIP gateway, the VoIP gateway adopts the two-stage dialing and plays the prompt tone, instructing the calling user to enter information such as called number. Automatic busy tone detection Different PBXs are likely to play different busy tones with different frequency spectra. Therefore, it is hard to recognize a busy tone feature according to a fixed threshold. With the smart busy tone identification technology, the VoIP gateway can sample, calculate, and analyze the busy tone played by the PBX to obtain a set of parameters reflecting a busy tone feature to the greatest extent. The busy tone detection can be implemented by configuring these parameters on interfaces. 1.2 Voice Function Configuration As the voice functions and objects vary, the voice function configuration includes four parts: voice subscriber line configuration, voice entity configuration, voice protocol configuration, and dial plan configuration, as shown in Figure 1-2. Each part implements a type of function. The functions implemented by all these parts are related. 1-3
Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over Voice function configuration Voice subscriber line Voice entity Voice protocol Dial plan Figure 1-2 Voice function configuration 1.2.1 Configuration Procedure Figure 1-3 shows the voice function configuration procedure. For details, see Table 1-1. Start a link connection Is the link available? No Yes Is number substitution necessary? No Yes number substitution for dial plans voice entity voice subscriber line number application for dial plans voice protocol Troubleshoot No Is the call established? Yes End Figure 1-3 Voice function configuration procedure 1-4
Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over Table 1-1 Description of the voice function configuration procedure Step Operation Reference 1 2 3 Connect the physical devices according to the network diagram. links and routes and ensure that the links and routes are available. Determine whether number substitution is necessary. If yes, configure number substitution for the dial plan. If no, proceed with the steps below. Access Volume and IP Routing Volume Dial Plan in Voice Volume. 4 POTS entity and VoIP entity VoIP in Voice Volume 5 6 7 8 related voice subscriber lines for voice entities The physical characteristics of voice subscriber lines are usually set to the defaults. number substitution for the dial plan adopted in the network diagram. the following voice protocols according to the service and networking environment. H.323 protocol SIP protocol Fax protocol Check whether the network requirements can be met. If yes, the configuration is completed. If no, check and remove the fault and perform re-configuration. VoIP and E1&T1 in Voice Volume Dial Plan in Voice Volume H.323, SIP, and FoIP in Voice Volume. 1.2.2 Voice Subscriber Line Voice subscriber line configuration is to implement the functions of the voice subscriber line. Voice subscriber lines, which are connected to telephone network devices such as analog telephone and PBX, implement all physical layer functions between VoIP gateways and PSTN devices. These functions include power supply to analog telephones, off-hook state detection, ringing signal generation, receiving & sending of analog or digital voice calls, and receiving & sending of dialed digits for call routing. For the voice subscriber line configuration, refer to VoIP and E1&T1 in Voice Volume. The router provides the following voice subscriber lines: 1-5
Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over Foreign exchange station (FXS) analog voice subscriber line, namely, FXS interface. FXS interfaces are usually connected foreign exchange office (FXO) subscriber line terminals, for example, an ordinary analog telephone, to provide ringing current, ringing voltage, and dial tone. FXO analog voice subscriber line, namely, FXO interface or 2-port loop trunk interface. FXO interfaces are usually connected to analog telephone interfaces of PSTN central offices (PBXs). Ear & mouth or receive & transmit (E&M) analog voice subscriber line, namely, E&M interface. E&M interfaces support analog E&M signaling and divide each voice connection into trunk circuit side and signaling unit side (similar to the relationship between DCE and DTE). PBXs send signals to routers via M lines and receive signals from routers via E lines. Digital E1/T1 voice subscriber line, namely, timeslot (TS) group created on an E1/T1 interface. After TS groups and signaling types (for example, R2 signaling, digital E&M signaling, or digital LGS signaling) are configured on E1 voice interface cards, the system will automatically generate the corresponding voice subscriber lines for the TS groups. E1/T1 interfaces support R2, DSS1, QSIG, and digital E&M signaling. 1.2.3 Voice Entity Voice entity configuration is to implement the functions of the voice entity. For the configuration of the voice entity, refer to VoIP in Voice Volume. There are two kinds of voice entities: plain old telephone service (POTS) entity and VoIP entity. The POTS entity corresponds to the local telephone or PSTN. POTS entity configuration is to associate a voice subscriber line on the VoIP gateway with a local telephone. The POTS entity configuration implements the binding between telephone numbers and voice subscriber lines. The VoIP entity relates a call entity with a routing policy. Compared with the POTS entity, the VoIP entity corresponds to the IP network. VoIP configuration implements the binding between telephone numbers and destination addresses (IP addresses or server addresses). 1.2.4 Voice Protocols The VoIP gateway can transfer voice or fax over the IP network by using different protocols. The basic voice protocols that the router supports are H.323 and session initiation protocol (SIP), and the fax protocol is T.38. 1) H.323 H.323 is a standard protocol established by ITU-T. The H.323 protocol stack, implemented at the application layer, mainly describes terminals, devices, and services used for multimedia communication without QoS guarantee over the IP network. An 1-6
Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over H.323 network usually consists of VoIP gateway, gatekeeper (optional), multipoint control unit (MCU), and terminals. According to the ITU-T specifications, the gatekeeper (GK) should provide H.323 terminals, gateway, or MCU in LANs or WANs with the following functions: Address translation Access permission Bandwidth control and management Area management and security check Call control signaling and call management Routing control and accounting The GK not only controls the call service, but also functions as the central control point within its management area. The GK implements the control function by exchanging information with the VoIP gateway. If there is any GK, the router will be under the control of the GK. To implement the control function of the GK, you need to perform related configurations on the router. For detailed configurations, refer to H.323 in Voice Volume. 2) SIP SIP is the core protocol of the IETF multimedia data and control architecture and is used for signaling control and communication with a softswitch platform in the IP network. A SIP network consists of user agent (namely, SIP endpoint), proxy server, registration server, location server, and redirect server. Here, the proxy server, registration server, location server, and redirect server are only functional entities. In practice, multiple functional entities may be integrated into one physical entity. In a complete SIP system, all SIP endpoints serve as user agents and should register with the registration server to inform of their locations, session capabilities, and call policies. The registration server sends the registration information to the location server for storage. SIP endpoints can use the proxy server to set up calls. SIP endpoints send signaling messages to the proxy server and then the proxy server forwards them to the next hop. In this process, multiple proxy servers may be involved. Eventually, channels are established to transfer the upper layer voice service. Unlike the proxy server, the SIP redirect server will not forward the received session request messages, but inform the originating SIP endpoints of the addresses of the terminating SIP endpoints by returning reply messages. The originating SIP endpoints directly re-originate a session request message to the terminating SIP endpoints. The terminating SIP endpoints also directly return a reply message to the originating SIP endpoints. As a SIP endpoint, the voice router needs to exchange information with the servers to accomplish the functions such as registration. For detailed configurations, refer to SIP in Voice Volume. 1-7
Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over 3) Fax protocol Fax over IP (FoIP) complies with ITU-T T.30 and T.4 on PSTN and T.38 on the IP network. T.30 defines the procedures necessary for document transmission between facsimile terminals on PSTN. It gives detailed descriptions and stipulations on the communication process, signal format, control signaling, and error correction of Group 3 facsimile terminals on the general switched telephone network. T.4 is a standard protocol used for document transmission between Group 3 facsimile terminals. It standardizes image coding, signaling modulation, rate, transmission time, error correction, and document transmission of Group 3 facsimile terminals. T.38 describes the technical features necessary to transfer facsimile document in realtime between Group 3 facsimile terminals over the Internet or other networks by using IP protocols. It gives descriptions and stipulations on communication mode, message format, error correction, and part of communication processes. Before applying the fax service, you need to configure the technical protocols and physical characteristics. For detailed configurations, refer to FoIP in Voice Volume. 1.2.5 Dial Plan Dial plan configuration is to provide diversified number management functions. Dial plan configuration involves number substitution and number application. Number substitution means establishing some substitution rules and applying them to calling and called numbers. Number substitution includes number substitution rules and binding of number substitution rules. Number application means matching numbers, controlling the sending of numbers, and selecting voice entities according to match templates. Number application includes number match policy, rules in the match order for voice entity selection, maximum-call-connection set, and number sending mode. The dial plan configuration directly affects the selection of voice entity and the final call connection. The dial plan configuration involves global configuration, voice subscriber line configuration, and voice entity configuration. You can select one or more configurations for a dial plan. The global configuration acts on calls of the whole VoIP gateway, the voice entity configuration on those of the voice entity, and the voice subscriber line configuration on those of the voice subscriber line. For detailed configurations, refer to Dial Plan in Voice Volume. 1.2.6 Command View The voice subscriber line configuration, voice entity configuration, voice protocol configuration, and dial plan configuration are implemented via command lines. Command s are the command line interfaces of the voice router. Most voice 1-8
Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over functions are implemented in corresponding. The command s of the voice router are arranged in a hierarchical structure. You can enter different function s under system, and sub-function (s) under function s. Figure 1-4 shows the command structure of the voice router. System Ethernet interface E1/T1 interface Loopback interface Dial interface...... Voice subscriber line Voice dial program CAS Voice entity Voice Voice GK client Voice number substitution Voice SIP client Figure 1-4 Hierarchical command structure of the voice router Table 1-2 Basic functions of voice command s View name Function Prompt Command to enter Command to System system parameters [system] Log in to the system logout. Disconnect telnet connection with the VoIP gateway. Ethernet interface Ethernet interface parameters [system-ethe rnet1/1] Key in interface ethernet 1/1 in any Return to system Loopback interface loopback interface parameters [system-loop Back1] Key in interface loopback 1 in any Return to system E1/T1 interface Create a TS group [system-e1 1/0] Key in controller e1 1/0 in system Return to system CAS signaling parameters [system-cas 1/0:5] Key in cas in E1/T1 interface Return to E1/T1 interface Voice global voice parameters e] Key in voice-setup in system Return to system 1-9
Operation Manual Voice Over (Voice Volume) Chapter 1 Voice Over View name Function Prompt Command to enter Command to Voice dial program dial plan e-dial] Key in dial-program in voice Return to voice Voice subscriber line voice subscriber line e-line3/0] Key in subscriber-line 3/0 in voice Return to voice Voice entity voice entity e-dial-entity1] Key in entity 1 pots or entity 1 voip in voice dial program Return to voice dial program Voice GK client H.323 e-gk] Key in gk-client in voice Return to voice Voice SIP client SIP e-sip] Key in sip in voice Return to voice 1-10