An Efficient VoIP System to Promote Voice Access Business

Transcription

1 An Efficient VoIP System to Promote Voice Access Business Xicheng Liu Centre for Communication Systems Research University of Cambridge Timothy J. Li Motorola Electronics Co., Ltd. Abstract VoIP is a rapidly growing area with great market potential. To promote it for both commercial and research purposes, we are developing an efficient voice access system based on state-of-art Motorola communication techniques. It is composed of a communication subsystems, a voice interface, and an information agent. Implementation of the high performance communication system is described in this paper. It is a gateway system integrating a PBX and a VoIP module. All components that H.323 defined to support VoIP are implemented in the VoIP module, currently in a simplified manner. As an embedded system, it features realtimeness and task distributiveness. A number of additional techniques are used to improve the communication performance, including noise suppression, zero copy, and buffer structure optimization. When integrated and refined in interoperability, the system will readily serve as a product. Keywords VoIP, gateway, gatekeeper, communication common platform, embedded system, voice processing, H Introduction With rapid development of Internet throughout the world, IP-based network has dominated the global information infrastructure. To maximize the value of the infrastructure, a great number of multimedia services are being developed. Voice over IP (VoIP) is among the hottest ones deployed recently. Many companies, including both giants and small players, have provided VoIP systems [1]. Nortel, Lucent, Cisco, 3Com, Cabletron, Ascend, Ericsson, etc., all provide total solutions including VoIP gateway, gatekeeper, and management software. Cisco also announced a series of routers with standards-based voice interface cards to support voice I/O directly [2]. Many small players look upon VoIP as a good point to cut into the IT market. They sharp their reputation by providing excellent components or systems. Among them are VocalTec, RadVision, and Telogy Networks. Lists of hundreds or even thousands of VoIP product vendors can be found on Internet, such as in the web site [3]. However, voice access business model is by no means fully explored. Telephone has been the most customized, effective, intelligent, and human way to handle complicated business. Internet provides a opportunity to enhance the voice communication means with remarkable flexibility while keep all its previous advantages. In fact, voice can be developed as an interface to access and manipulate information in both cyberspace and human world, either by itself or combined with other media. To promote such a general voice access business model, we need develop efficient technologies to enable integration and interaction of voice, information, and communication. A powerful, service friendly, and cost-effective communication platform readily hosting such integration and interaction is very helpful. Meanwhile, technical issues affecting large-scale commercial deployment of VoIP, such as QoS, interoperation, management, and security, need be addressed with network level study [4]. To promote voice access services and address the technical problems, we are developing a VoIP system based on state-of-art Motorola communication technologies. The system provides a gateway solution to allow interaction of voice, information, and communication. It is composed of a communication subsystems, a voice interface, and an information agent. With it users can access remote phones or manipulate Internet information through local phones. However, they need not pay local phone fees because the system connects their phones directly to Internet. It emphasizes the smalland medium-sized business market. We believe that this is a key step to push the voice access services greatly. In this paper we will focus on the implementation of the communication subsystem. Full description of

2 voice interface and information agent needs separate papers. The communication system integrates a VoIP module with a PBX. The VoIP module implements important components defined by H.323 [5], such as gatekeeper and gateway, in an efficient way. State-of-art innovations in communication and DSP from Motorola are adopted to set the system on a high performance platform. A number of additional techniques are used to improve the voice performance. It is running well on realistic Internet environment. The communication system provides an efficient platform to backup the voice and information subsystems on it. To allow integration and interaction of voice and information on gateway has many advantages than to put them on the end system. It can greatly simplify the operation of users, and is thus more user friendly. It allows best system optimization to maximize the performance obtained by each user. In section 2 of this paper we will introduce the overall system architecture. Section 3-7 will describe structure and important modules and techniques of the communication system in details. On-going work will be described in section 8. Conclusions will be given in section 9. 2 System Architecture Architecture of the system is illustrated in figure 1. It consists of a communication subsystem, a voice interface, and an information agent. The communication subsystem integrates a powerful CPU-based PBX and a VoIP module. It works as a gateway interfacing the PBX and Internet. It can exchange phone calls from both local ends and network side. At the same time, it will forward requests to manipulate Internet information to the voice interface. The voice interface interprets voice command in a phone call and sends it to the information agent. Result information from the information agent will also be gracefully translated into voice data. This module is developed in an evolutionary manner. First, simple combinations of digits are interpreted as commands, then phrase commands, and then voice speaking of limited areas. Eventually we will work out a Voice- Access-Internet-Language (VAIL) and a translation protocol between it and HTML. The information agent carries out information access to Internet according to user commands laid down from the voice interface, and returns the result information. Voice Interface Information Agent Voice Data Communication Subsystem Fig. 1. System Architecture A typical running setting of the system is shown in figure 2. Local phones connect to the communication subsystem, which bridges them to Internet. A phone call channel can be set up between a local phone and remote phone through Internet. The remote phone is not necessarily of the same provider, if only interoperation with other providers is realized. Through information agent a phone request to access Internet information, for example, a directory book, a database, and an e-shop, will also be satisfied. Of course, it is also possible for remote users to access Internet with phones by first dialing into the system. However, to prevent the system from being blocked by the flood calls from all over the Internet, some admission control is necessary. 1

3 URL Database Phone Voice Interface Information Agent Internet Phone Communication Subsystem Fig. 2. Typical System Running Setting 3 Communication Subsystem Structure The communication subsystem is based on Motorola Communication Common Platform (CCP). CCP is a state-of-art communication reference system for Motorola PowerPC860 processor [6]. It connects PowerPC860 control board, interfaces cards, and DSP boards using high-speed buses. In near future, it will introduce multiple processors on the control board. ATM switching module and interfaces will also be added. Development of communication systems will be greatly facilitated using CCP, such as router, remote access server, and modem bank. Modules of the system and their relations are depicted in figure 3 [7]. All modules compose two functional subsystems, a PBX and a VoIP gateway. The PBX is composed of hardware platform, control message handler, switching module, call control module, and operation and maintenance (OAM) module. The VoIP gateway includes a gatekeeper and multiple gateway objects. Both PBX and VoIP gateway can be managed through user interface module when running. Interface to the voice interface subsystem is omitted in the figure. User Interface Gatekeeper Internet OAM Call Control Gateway Internet Control Message Handler Switching Hardware Platform Fig. 3. Communication Subsystem Structure Control message handler transfers control messages between hardware components and software modules. Other software modules can ask the handler to send a message by placing it into the message queue of the handler. At the same time, the handler can deliver a message from hardware to the appropriate software module. Switching module carries out voice switching for local calls. It initializes switch setting, connects two channels on request from call control module, and tears down the connection after the call terminates. 2

4 Call control module centralizes the PBX subsystem. It controls message flow in the process of phone hook-off, channel setup, voice exchange, and call termination. It branches local calls and remote calls. Local calls are directly switched with the switching module. Remote calls are transferred to VoIP gateway relay. OAM module is responsible for hardware monitoring and resetting, hardware failure notifications, part of card failure recovery (the call related functions are handled by call control), and tasks watchdog. Gatekeeper maintains a calling information table, which have entries for mapping a phone number to a network address. It receives phone call requests from call control module, makes address translation, initializes gateway objects for channel set-up. For call requests from Internet, it also creates dedicated gateway objects to handle them. Gateway objects directly bridge local PBX and Internet. In our implementation, they transmit signalling messages and voice data in a unified manner. This can reduce the delay of setting up multiple channels for a single call. To enable VoIP, gatekeeper and gateway are key components. They interact with each other, and both interact with call control module. In these processes, call signalling messages are sent to and received from remote phones to set up call channels. Voice data are exchanged via the channels to fulfill the function of VoIP. In following sections, we will describe gatekeeper and gateway, and other techniques in more details. 4 Gatekeeper Gatekeeper is one of the most important components H.323 defined to realize interoperation [5]. Especially in a large network setting like Internet, it is indispensable as a central body to provide network call control and management. In our simplified implementation of our system, the functional flow of it is as figure 4. BEGIN Message Acquisition Message Interpretation Address Translation Gateway Management END When started, it enters an endless loop. As a first step, it extracts a message from local or network message queue. Then an interpreter works to analyze the message. For a call request to a remote phone, address translation will be made to map the phone number into an IP address. This is realized by looking up a calling information table, which is also maintained by the gatekeeper. Then a gateway object will be kicked off to set up a network call channel. The object will be responsible for successive voice data transmission if the channel is set up successfully. Thus the gatekeeper is released from the details of channel maintenance, and can process other call requests. This message loop continues until the system gets down. Thus messages from local and network sides can be processed one by one. We see gatekeeper does not set up channel itself, but leave it to a gateway object. This way a number of call requests can be processed simultaneously. Because the PBX is a small setting with tens of phones, bandwidth management is not absolutely necessary. But it can be easily added into the flow shown in figure 4 [7]. 5 Gateway Fig. 4. Functional Flow of Gatekeeper Gateway is the component that directly exchanges messages between PBX and Internet. In our 3

5 implementation, a gateway object is responsible for a single call channel between a local phone and a remote phone. It is created at the call initialization time by the gatekeeper. So there may be many gateway objects active at the same time. They process a number of calls simultaneously. Main functional flow of each gateway object is shown in figure 5. Messages from both local PBX and network are processed. There are similar message processing modules for the two kinds of messages. The processing fulfills a simplified version of H.225 [8] based on Q.931 [9]. In our implementation, a message can wrap either call signalling or voice data. They are transmitted and processed in a unified channel rather than in separate channels. This can reduce the channel set-up overhead. Different types of messages are branched to different message processing routines. For signalling messages, new messages will be generated based on message interpretation, then sent to network or local PBX. For voice data, direct exchange will be carried out using right output interfaces. When channel disconnecting messages are captured, the channel will be torn town and channel resources will be cleared. BEGIN Local Message Processing Local Message Analysis Processing Routines Network Message Processing Network Message Analysis Processing Routines END Fig. 5. Functional Flow of Gateway 6 Task Distribution VoIP system is a real-time system processing many calls simultaneously. Both computation intensive processing and time-bounded processing are involved. When a number of calls are present, each should be assigned an equal computing power. Our solution lies in three aspects. First, a real time operating system (RTOS) is employed. We use the psos from Integrated System, Inc. [10], whose performance has been proved by many industrial applications. Second, computation intensive processing is isolated and processed by specific computing power. A DSP array board [11] is dedicated to voice compression and echo cancellation, which will be described in details in next section. Third, processing is distributed in many tasks, and each is implemented as a separate process. Actually, in psos, a process is just called a task. These tasks are designed to distribute processing and minimize resource access conflict. Real time requirement can be satisfied through balanced task assignment. This also leads to clearer and easier programming. Modules in figure 3, i.e., call control, message handler, OAM, gatekeeper, gateway, and user interface are all implemented as separate tasks [12]. Switching module provides routines employed by call control and gateway. Gatekeeper is further separated into two tasks. One task, GK_POSITIVE, is a positive task to process call requests originated by local PBX to remote phones. The other task, GK_PASSIVE, is a passive task to accept call requests from Internet to local phones. Gateway is implemented in a more distributed manner. As we have described earlier, its function flow is cloned in many gateway objects coded as GW_i. Each is responsible for only one duplex call channel. Thus the function of each object is clearly defined and limited. The processing is equally distributed among gateway objects. Task structure of VoIP module is depicted in figure 6. 4

6 GK_POSITIVE GK_PASSIVE GW_1 GW_2 GW_N Gateway Objects Fig. 6. Task Distribution of VoIP Module Several inter-task communication mechanisms are used to improve performance. These include variable length message queues, fixed length message queues, and global variables. Variable length message queue is a kind of objects psos provides for inter-task communications with the same efficiency with fixed message queue. It provides a good solution for cases in which the number of parameters with a message is variable. For example, variable message queues are used between message handler and call control, and between call control and gateway objects. On the other hand, call control and gatekeeper exchange messages with fixed message queues. Global variable is used as an efficient object for simplex inter-task communication. That is, one task only writes it and the other only read it. The gatekeeper uses this method to notify gateway of some control information. 7 Voice Processing Voice processing techniques in endpoints determine the voice quality to a great degree, though QoS deployment on Internet holds the last word. A number of factors have been optimized to improve the voice quality in our implementation. Particularly, following techniques contribute most: noise suppression, echo cancellation, voice compression, buffer structure, and zero copy [13]. In network channel set-up period, before true voice data are received, what is heard in phone is the noise. It should be suppressed. Also, during whole process of voice communication noise may be heard at intervals between two successive voice data blocks. These intervals should also be muted. Echo cancellation and voice compression are computation intensive. They are implemented using a dedicated DSP array board with eight DSP modules [14]. Each module uses a Motorola processor. One module can process two duplex phone channels simultaneously. So a DSP array board can support 16 phones on a PBX. Multiple DSP array boards can be fitted in the CCP chassis to support more phones. The DSP procedures for incoming voice data from Internet and outgoing voice data to remote phones are given in figure 7 respectively. G.729a [15] and G [16] are implemented for voice compression. Structure of buffers for sending and receiving voice data can affect the voice quality. Longer buffer makes longer delay. Shorter buffer makes more data fragment for transmission and thus increases processing overhead and transmission delay. In addition to optimal buffer length, good configuration of types and number of buffers would also improve the voice quality. These optimizations are all for static structure of buffers. Dynamic control of buffers can also be optimized. One of them is zero copy, which means the same buffer is used in both user program space and system space for a packet, and there is no copying of data between them when the packet is sent or received. Zero copy can save resources and reduce latency. It is especially useful when a larger number of phone calls are processed. 5

7 Initialization Voice Data Input Decompression Log-to-Linear Conversion Echo Cancellation Linear-to-Log Conversion Voice Data Output Initialization Voice Data Input Log-to-Linear Conversion Compression Linear-to-Log Conversion Voice Data Output (a) For incoming data (b) For outgoing data Fig. 7. Functional Flow of Voice Processing on DSP Board 8 On-going Work The communication subsystem has been finished and is running well on Internet environment. When two of such systems are connected to Internet in different points, one can make an IP phone call from one point to another with a good voice quality. Currently the VoIP module implements H.323 in a simplified manner. Details are being appended to complete the protocol standard in full details. Interoperation with IP phones from other providers will then be reached. Integration with voice interface and information agent is at the same time being carried out. Though the two subsystems are generally pre-mature, especially the voice interface, a quick integration and test of the system is important. This will determine to much degree how our effort would be allocated in next phase. 9 Conclusion In this paper we have described our on-going work to develop a cutting-edge VoIP system. It provides a gateway solution for interaction of voice, information, and communication. The design allows remarkable flexibility and performance improvement in servicing voice users. We argue that it will lead to an efficient system to promote voice access business greatly. Overall system architecture and design philosophy are introduced. Implementation of the communication subsystem is described in details. It integrates a PBX and a VoIP module, and embodies the framework of H.323 in a simplified and efficient way. Based on brand-new hardware innovations from Motorola and enhanced with a number of voice processing techniques, the embedded implementation reaches high performance and good realtimeness. References [1] Yangzhao Xiang and Xicheng Liu. An Overview of VoIP Technology and Systems, Chinese Computer World, No. 4, January [2] [3] [4] Xicheng Liu. Research on Some Key Issues for High Performance QoS Router. Technical Report, Institute of Computing Technology, Chinese Academy of Sciences, June [5] ITU-T Recommendation H.323 (1997) - Packet-Based Multimedia Communications Systems. 6

8 [6] Motorola Semiconductor Products Sector. MPC860 PowerQUICC Users Manual, September [7] Xicheng Liu. VoIP System Design, Technical Report JDL-VOIP-DESIGN, Motorola-ICT Joint R&D Laboratory, May [8] ITU-T Recommendation H.225 (1998) - Call Signalling Protocols and Media Stream Packetization for Packet-Based Multimedia Communication Systems. [9] ITU-T Recommendation Q.931 (1998) - ISDN User-Network Interface Layer 3 Specification for Basic Call Control. [10] Integrated System, Inc. ISI psos Programming Guide, [11] Microtek International Development System. Communication Common Platform DSP Array Board User s Manual, Doc. No. 149-M02320, First Edition, February [12] Xicheng Liu and Shimin Liu. VoIP System Software Implementation, Technical Report JDL-VoIP -SW, Motorola-ICT Joint R&D Laboratory, May [13] Xicheng Liu and Fusheng Chen. VoIP System Performance Optimization, Technical Report JDL- VOIP-PERF, Motorola-ICT Joint R&D Laboratory, June [14] Xicheng Liu and Fusheng Chen. DSP Implementation Scheme for VoIP System, Technical Report JDL- VOIP-DSP, Motorola-ICT Joint R&D Laboratory, June [15] ITU-T Recommendation G.729 (1996) - Coding of Speech at 8 Kbit/S Using Conjugate-Structure Algebraic-Code-Excited Linear-Prediction (CS-ACELP), Annex A (1996) - Reduced Complexity 8 Kbit/S CS- ACELP Speech Codec. [16] ITU-T Recommendation G (1996) - Speech Coders: Dual Rate Speech Coder for Multimedia Communications Transmitting At 5.3 and 6.3 Kbit/S. 7